Transmission Systems Design Handbook for Wireless Networks

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Transmission Systems Design Handbook for Wireless Networks

For a listing of recent titles in the Artech House Mobile Communications Series, turn to the back of this book.

Transmission Systems Design Handbook for Wireless Networks Harvey Lehpamer

Artech House Boston London www.artechhouse.com

Library of Congress Cataloging-in-Publication Data Lehpamer, Hrvoj (Harvey). Transmission systems design handbook for wireless networks/Hrvoj Lehpamer. p. cm.(Artech House mobile communications series) Includes bibliographical references and index. ISBN 1-58053-243-8 (alk. paper) 1. Wireless communication systems. 2. Cellular telephone systems. 3. RadioTransmitters and transmission. I. Title. II. Series. TK5103.2.L45 2002 621.3456dc21 2001056655

British Library Cataloguing in Publication Data Lehpamer, Hrvoj (Harvey). Transmission systems design handbook for wireless networks. (Artech House mobile communications series) 1. Wireless communications systemsDesign I. Title 621.38456 ISBN 1-58053-243-8 Cover design by Igor Valdman

© 2002 ARTECH HOUSE, INC. 685 Canton Street Norwood, MA 02062

All rights reserved. Printed and bound in the United States of America. No part of this book may be reproduced or utilized in any form or by any means, electronic or mechanical, including photocopying, recording, or by any information storage and retrieval system, without permission in writing from the publisher. All terms mentioned in this book that are known to be trademarks or service marks have been appropriately capitalized. Artech House cannot attest to the accuracy of this information. Use of a term in this book should not be regarded as affecting the validity of any trademark or service mark. International Standard Book Number: 1-58053-243-8 Library of Congress Catalog Card Number: 2001056655 10 9 8 7 6 5 4 3 2 1

Dedicated to the two most important women in my life, my mother and María.

Contents Acknowledgments

xix

1

Introduction

1

2

Basics of Wireless Networks

5

2.1

Historical Background

5

2.2 2.2.1 2.2.2 2.2.3 2.2.4 2.2.5 2.2.6 2.2.7

Existing Wireless Technologies Mobile Networks and the Wireless Local Loop Analog Cellular Systems FDMA TDMA Personal Digital Cellular GSM and General Packet Radio Services CDMA

6 6 7 9 9 10 10 13

2.3 2.3.1 2.3.2

Evolution of Wireless Technology 1G and 2G Wireless Networks 3G Wireless Networks

15 15 15

vii

viii


2.3.3 2.3.4 2.3.5

CDMA2000 CDMA2000 1xEV-DO Future Directions

19 22 24

2.4 2.4.1 2.4.2

Satellite Networks Fixed Satellite Service Mobile Satellite Systems

26 26 27

2.5 2.5.1 2.5.2

Fixed Microwave Systems Microwave Point-to-Point Systems Microwave Point-to-Multipoint Systems References

28 28 31 44

3

Transmission-Network Principles

47

3.1 3.1.1 3.1.2 3.1.3 3.1.4

Wireline Side of Wireless Networks PSTN Interconnect and Telephony Overview Traffic Engineering SS7 and AIN Telecommunications Act of 1996

47 47 51 54 58

3.2 3.2.1 3.2.2 3.2.3 3.2.4 3.2.5 3.2.6

Digital Transmission Technology About the Transmission Transmission MediaPhysical Layer Transmission (Backhaul) in Wireless Networks DSX-1 Digital Interfaces (North America) North American Digital Hierarchy CEPT Digital Hierarchy

59 59 59 61 62 63 65

3.3

Plesiochronous Versus Synchronous Digital Hierarchy 66

3.4 3.4.1

Multiplexing and Inverse Multiplexing Statistical Multiplexing

70 72

3.4.2 3.4.3

3/1 Multiplexing and Subrate Multiplexing Inverse Multiplexing

73 74

3.5

ATM

79

3.5.1

ATM Basics

79

Contents

ix

3.5.2 3.5.3 3.5.4 3.5.5

Use of ATM Adaptation Layer Inverse-Multiplexing for ATM Protocol QoS in ATM Networks Definition of Availability in ATM

83 86 87 89

3.6 3.6.1 3.6.2 3.6.3 3.6.4

Voice over IP H.323 Network Building Blocks Latency and Jitter Issues Multiprotocol Label Switching IP-Based Wireless Networks

91 91 95 97 99

3.7 3.7.1 3.7.2 3.7.3 3.7.4 3.7.5 3.7.6 3.7.7 3.7.8 3.7.9 3.7.10 3.7.11 3.7.12 3.7.13 3.7.14 3.7.15 3.7.16

Complete T1 Tutorial Signals in a T1 Network Pulse Transmission BERs Overall System Length Single-Cable and Dual-Cable Operation T1 Repeatered Lines Order Wire Lightning Simplex Power Design Data Error Rates Voltage and Temperature Factors T1 Engineering, Installation, and Documentation Troubleshooting and Problem Classification Switch Options, Line Codes, and Framing Fractional T1 T1 (J1) in Japan

102 102 105 105 106 107 107 110 110 112 114 115 116 116 118 118 118

3.8 3.8.1 3.8.2 3.8.3 3.8.4 3.8.5 3.8.6

Complete E1 Tutorial Introduction to E1 Networks Customer Premises Equipment Signal Characteristics Transmission Facilities Pulse Density HDB3

119 119 119 120 120 121 121

x


3.8.7 3.8.8 3.8.9 3.8.10

E1 Framing Synchronization E1 Framing Formats Spare Bits Global Framing Formats References

122 122 124 127 127

4

Wireless-Network Architecture

129

4.1

2G Wireless-Network Architecture

129

4.2 4.2.1 4.2.2 4.2.3 4.2.4 4.2.5 4.2.6 4.2.7

3G Wireless-Network Architecture Directions in 3G Developments Horizontally Layered Network Architecture 3G Core Network Universal Mobile Telephone System 3G in GSM Networks 3G CDMA Network 3G Traffic Classes

131 131 136 140 141 141 142 145

4.3 4.3.1

3G Transmission Networks Replacing TDM with ATM in Transmission Networks Importance of AAL2 QoS Concept ATM Physical Layer Traffic Modeling and Simulation Tools 2G and 3G Coexistence Transmission-Network Architecture References

148 148 154 158 161 165 168 170 179

5

Theory and Principles of Fiber-Optic Transmission

181

5.1

Basics of Fiber-Optic Transmission

181

5.2 5.2.1 5.2.2

Design Principles Bandwidth and Attenuation Optical Power Budgets and Distance Calculations

183 183 185

5.3

Synchronous Digital Hierarchy

189

4.3.2 4.3.3 4.3.4 4.3.5 4.3.6 4.3.7

Contents

xi

5.3.1 5.3.2 5.3.3 5.3.4 5.3.5

Basics of Synchronous Systems Benefits of SONET SONET Architecture SONET Availability Requirements SDH

189 191 191 195 195

5.4 5.4.1 5.4.2 5.4.3 5.4.4 5.4.5

DWDM DWDM Overview DWDM Capacity (Bandwidth) Requirements Network Growth and Flexibility of DWDM Optical Layers Protection in DWDM Networks

196 196 197 198 200 200

5.5

Optical Switching References

201 202

6

Microwave Point-to-Point System Design

203

6.1

Basic Microwave Transmission Theory

203

6.2 6.2.1 6.2.2 6.2.3 6.2.4 6.2.5 6.2.6

Theoretical Aspects of Microwave Link Design Microwave-Radio-Path Calculation Overview Design Fade Margins Diversity Improvement North American and ITU Objectives Reliability and Availability Overview Effects of Rain on Microwave Propagation

206 206 207 210 212 215 217

6.3 6.3.1 6.3.2 6.3.3 6.3.4 6.3.5

Practical Aspects of Microwave-Link Design Design Overview Protected and Nonprotected Microwave Systems Microwave Repeaters Microwave Path Calculations Microwave Interference Analysis and Frequency Coordination Microwave System Design Guidelines Microwave Lookup Table

219 219 221 222 224

6.3.6 6.3.7

225 226 229

xii


6.4

Spread-Spectrum Microwave Systems

230

6.5

Microwave Compatibility and Safety

236

6.6 6.6.1 6.6.2 6.6.3 6.6.4 6.6.5

Coordinate Systems, Datums, and GPS About Datums Geometric Earth Models Reference Ellipsoids and Coordinate Systems GPS Useful Facts to Remember

240 240 241 242 244 247

6.7 6.7.1 6.7.2

Managing the MW Radio Network Introduction Managing a Microwave Network with SNMP References

247 247 249 254

7

Transmission-Network Planning and Design

257

7.1

Overview

257

7.2 7.2.1 7.2.2 7.2.3 7.2.4 7.2.5 7.2.6 7.2.7 7.2.8 7.2.9 7.2.10 7.2.11 7.2.12 7.2.13

General Wireless-Network Planning and Design Principles Identifying the Opportunity and Strategic Planning Customer Requirements Analysis Interconnection Spectrum Auctions Clearing Spectrum and Microwave Relocation RF Design Transmission Media and Topology Planning Mobile Positioning Toll QoS Network Performance Sales and Marketing Regulatory Issues Life Cycle of Wireless Networks

259 259 262 265 266 267 268 270 271 272 273 276 278 281

7.3

Transmission System Design

283

Contents 7.3.1

xiii

7.3.2 7.3.3 7.3.4 7.3.5

Transmission System Design Process and Requirements SDH and PDH Transmission Systems SDH Transmission-Network Protection Ring Protection in the Wireless Network Description of TND Deliverables

283 286 286 288 289

7.4 7.4.1 7.4.2 7.4.3 7.4.4 7.4.5 7.4.6 7.4.7

Leased Lines in Wireless Networks Leased Access and Core Transmission Networks Dedicated Leased Service xDSL Switched Leased Service Higher-Speed Switched and Nonswitched Services Leased Lines Network Build Out Owned Versus Leased Transmission Networks

292 293 294 296 297 299 299 301

7.5 7.5.1 7.5.2 7.5.3

302 302 304

7.5.4 7.5.5 7.5.6 7.5.7 7.5.8

SynchronizationStratum, BITS, and GPS Introduction and Historical Overview Strata General Timing Planning Rules in Transmission Networks Interoffice Distribution Intraoffice Distribution SONET Network Timing Synchronization: Issues in PCS Networks Cell-Site Timing in Wireless Networks

308 310 312 313 314 316

7.6 7.6.1 7.6.2 7.6.3 7.6.4

Transmission-Network Optimization Daisy Chaining and Traffic Grooming Voice Compression Signal Propagation Delay Example of Optimized Network Design

317 317 318 321 323

7.7 7.7.1 7.7.2

Transmission Network: Design Examples Small PDH Microwave Transmission Network Complex Transmission Network

324 324 325

xiv


7.8

Overview of RNC Dimensioning in the 3G Wireless Network Traffic Classes in the UTRAN Network Description of RNC Interfaces User Traffic Modeling Traffic Calculation Guidelines Example of Traffic Calculations

327 328 329 332 334 342

7.9 7.9.1 7.9.2 7.9.3

Alternative Solutions in Transmission Networks Considering Dark Fiber and Dark Copper Partnership with Utilities Optical Laser Communications References

344 344 344 348 350

8

Transmission Equipment

351

8.1 8.1.1 8.1.2 8.1.3 8.1.4 8.1.5 8.1.6 8.1.7

Digital Microwave Radio PDH and SDH Microwave Radios Standard Microwave Radio Configuration Split Microwave Radio Configuration Microwave Antennas Transmission Lines Environmental and Quality Issues Standards and Recommendations

351 351 352 353 358 361 362 363

8.2 8.2.1 8.2.2

Fiber-Optic Equipment SONET and SDH OPGW

363 363 368

8.3 8.3.1 8.3.2 8.3.3

Wireline Equipment Digital-Access Cross Connects CSU/DSU DSL and ADSL

370 370 375 376

8.3.4

Echo Cancellers

379

8.4 8.4.1

Cabling NEC Cable Categorization

384 384

7.8.1 7.8.2 7.8.3 7.8.4 7.8.5

Contents

xv

8.4.2 8.4.3 8.4.4

Digital Cross Connects DS1 Signal Termination Leased Lines and the Network Interface Unit

386 388 392

8.5 8.5.1 8.5.2 8.5.3 8.5.4

Grounding Earth Grounding Basics Ground System Design Fundamentals Types of Grounding Grounding for Wireless Cell Sites

392 392 395 396 398

8.6 8.6.1 8.6.2 8.6.3 8.6.4

Power and Battery Backup ac Power dc Power Batteries Solar Energy

399 399 400 401 403

8.7

GPS Antennas

405

8.8 8.8.1 8.8.2 8.8.3 8.8.4 8.8.5 8.8.6

Quality and Reliability Issues Quality Assurance First Article Inspection Factory Acceptance Testing Equipment Reliability Environmental Specifications Network Equipment Building Standard References

405 405 406 407 407 410 411 412

9

Transmission-Network Deployment

413

9.1 9.1.1 9.1.2 9.1.3 9.1.4 9.1.5

Equipment and Services Ordering Process Planning and Design RFQs, RFIs, and RFPs Negotiating the Statement of Work Negotiating with Telecommunications Providers Negotiating with Equipment and Services Suppliers

413 413 414 421 422 425

9.2

Regulatory Issues

425

xvi


9.3 9.3.1 9.3.2 9.3.3 9.3.4 9.3.5

Services Engineering Services Project Management Outsourcing Services Due Diligence Network Maintenance

430 430 430 432 435 436

9.4 9.4.1 9.4.2 9.4.3 9.4.4 9.4.5 9.4.6 9.4.7

Project Management in Wireless Networks Definitions Project-Management Organizational Issues Project Stages Leased-Lines Tracking Process Change Orders Postinstallation and Optimization Activities Project-Management Tools

436 436 438 440 441 443 443 445

9.5 9.5.1 9.5.2

446 446

9.5.3 9.5.4 9.5.5

Selection of Key Sites Site Acquisition, Zoning Issues, and Colocation Switch and NOC Site Selection and Building Requirements Cell-Site Selection Microwave Repeater Site Selection Colocated Systems and RF Cell-Site Compliance

449 451 452 453

9.6 9.6.1 9.6.2 9.6.3 9.6.4 9.6.5

Microwave Deployment Microwave System Scope of Work Microwave Site Surveys Microwave Path Survey Housing the Infrastructure Microwave Antenna Mounting Structures

454 454 456 461 464 466

9.7 9.7.1 9.7.2 9.7.3 9.7.4 9.7.5

Measurement of Radio-Frequency Fields Health and Safety Issues Measurements and Sources of Emission Near-Field and Far-Field Regions Power Levels and Power Density Radiation Patterns and Polarization

485 485 486 488 489 490

Contents

xvii

9.7.6 9.7.7 9.7.8 9.7.9

Sources of Electromagnetic Field Induced and Contact Currents Instrumentation Measurement Procedure for Microwave Installations

491 492 493 494

9.8 9.8.1 9.8.2 9.8.3 9.8.4 9.8.5 9.8.6 9.8.7 9.8.8

Fiber-Optic Cables and Their Installation Cabling Design Considerations Fiber Protection Fiber-Optic Cable Types Fiber Count Fiber Splicing Connectors Handling Fiber-Optic Cables Fiber-Optic Cable Installation Procedures

495 495 497 498 501 503 504 506 507

9.9 9.9.1 9.9.2 9.9.3 9.9.4

Operations and Maintenance Growth of Multiservice Networks Network Management System Geographic Partitioning Location-Finding Techniques References

509 509 511 516 516 520

10

Transmission-Network Testing and Commissioning 521

10.1

Definitions

521

10.2 10.2.1 10.2.2 10.2.3 10.2.4

BERT T1 Impairments T1/E1 Testing Out-of-Service Testing In-Service Monitoring of Live Data

522 522 523 524 526

10.3 10.3.1 10.3.2 10.3.3 10.3.4

Transmission-Network Testing Procedure Testing Leased Facilities Testing Microwave Systems DS1 and DS3 Performance Objectives DS1 Test Procedure and ATP Form

526 526 527 528 529

xviii


10.4 10.4.1 10.4.2 10.5 10.5.1 10.5.2 10.5.3

Fiber-Optic Cable Testing 530 OTDR Test Procedure for Single-Mode Fiber-Optic Cables 531 Characterizing and Testing Fibers for DWDM Applications 532 Packet-Network Testing Testing the Voice Traffic ATM Network Testing Quality Tests in VoIP Networks References

537 537 538 543 547

Appendix: Units Conversion

549

A.1

About Units of Measurement

549

A.2

The International System of Units

550

A.3

Common Units

552

Glossary

553

About the Author

587

Index

589

Acknowledgments I would like to thank my colleague Alfiya Vali-Durrett, a great engineer, for patiently reading and contributing to a number of the chapters in this book. Many thanks also to Dave Olson for saving the electronic version of the manuscript from disappearing into the twilight zone of my crashing computers hard drive. In addition, I would like to express my appreciation to everyone with whom I was unable to spend time during the past 3 years because I was working on this book. Thank you all.

xix

1 Introduction Public telephone operators and new independent wireless operators throughout the world are deploying wireless access in an effort to drastically reduce delivery costs in the most expensive part of the networkthe local loop. Available radio technology enables both existing and new entrants to access subscribers in a rapid manner and deliver their basic telephony products and broadband-enhanced services. While each operator has his or her own series of business, regulatory, and technical drivers, the following questions are common: • What services should be delivered to the various customer segments? • Which technology strategy should be adopted: TDMA, GSM,

W-CDMA, or CDMA 2000? • How can the network be implemented and service provided in a fast-track manner? • Which programs, processes, and suppliers must be established to realize the business plan? • How can the new wireless provider compete against other wireline and wireless operators, and what are the key factors for success? Issues, such as technology choices, security aspects, network quality, marketing, and customer service, are essential to long-term success, affecting subscriber installation equipment, program management, opportunities in 1

2


emerging nations, license obligations, interconnect strategy, billing systems, and revenue collection. Prior to the license application, it is imperative to formulate the strategic elements of the business plan. The strategy must be developed and agreed upon using a top-down approach, with realistic time scales and budgets, and identified risks. The vital elements of the corporate strategy include personnel, technology, network topology, procurement strategy, regulatory strategy, and project management. Typically, a business plan is based on marketing data and, once the license is granted, implementation. Key issues include geographic rollout, communications links, site requirements, service offerings, and launch strategy. Working closely with the marketing plan, the overall coverage plan and capacity requirements should be established. In line with the chosen technology, the engineering details and specifications need to be generated, agreed upon, and approved. Physical elements, such as structures, antennas, and cabinets, should be professionally specified to ensure expert purchasing. Design activities must be addressed in link planning, transmission networks, switch capacity, and numbering plans. Site surveys have to be performed from both microwave-link and RF-coverage aspects, to ensure exact design. Site acquisition is usually the gating factor in most wireless networks build-out timelines. Once these sites are acquired and planning permission is obtained, the development process (build-out) will accelerate. Some of the key issues here are building services, type of site, zoning issues, access, power, site management, and landlord liaison. Equipment procurement is dependent on the performance of outside agencies (i.e., equipment and systems suppliers). Consequently, there is obvious risk, and a number of key issues will need to be addressed, including competitive tendering, contract management, software development, and change management. Strategic alliances with equipment suppliers and service providers must also be considered. Constructed sites are usually tagged as ready for communications. There is intense activity surrounding installation of all the necessary equipment following the logistical challenge of materials management. At this time, given the pace of the project, it is typical to find numerous sites being developed simultaneously. Some of the key issues are equipment deliveries, rigorous testing procedures, commissioning, optimization, patenting, and integration. Transmission, or transport, design and its integration with the existing networks are important parts of the telecommunications network, as they allow for the transporting of the traffic between various elements of the

Introduction

3

network and provide access to the customer, while enabling connectivity between various networks. The transmission media has been traditionally categorized into three main typescable (copper or fiber-optic), radio (microwave point-to-point, in this case), and satellite. Although the technologies for the backbone network are already well developed, there remain costs and difficulties involved in setting up the local access; the technologies for overcoming these difficulties are continuing to evolve. The question for wireless operators is whether to build and own the transmission network or to lease facilities (lines) from the existing carriers or operators; the answer will be different in different situations. The introduction of third-generation (3G) wireless networks with increased capacity requirements and packet data architecture will also have a great impact on future transmission-network design and deployment. The issues involved in the deployment of various transmission technologies and their impact on the overall wireless-network topology will be presented in this book. Overall strategy and approach to transmission-network planning, design, and deployment will be discussed, but the details of the fiber-optic and microwave designs are outside the scope of this book. Transmissionnetwork design in wireless networks is almost technology independent, although there are differences between the TDMA and CDMA counterparts. This book covers a number of wireless technologies for the purpose of introducing readers to the overall aspects of wireless-network design; it focuses then on the transmission aspects. Many of the recommendations, tables, and descriptions in this book are the result of the authors experience of over 20 years in the engineering field and cannot be traced to any particular standard, book, or article. Regardless of the technology, the transmission network, including its physical layer and issues related to it, is an important part of every wirelessnetwork build-out. In this book, the importance of transmission engineering within wireless networks is given the attention it deserves.

2 Basics of Wireless Networks 2.1 Historical Background The first commercially available radio and telephone system, known as the Improved Mobile Telephone Service (IMTS), was put into service in 1946. This system was quite unsophisticated, as there were no solid-state electronics available at that time. In order to use IMTS, a tall transmitter tower was erected near the center of a metropolitan area. Several assigned channels were then transmitted and received from the antenna atop this tower and any vehicle within range could attempt to seize one of those channels and complete a call. Given these constrictions, the number of channels available did not come close to satisfying the demand. The solution to this problem was cellular radio. Metropolitan areas were divided into cells of no more than a few miles in diameter, each cell operating on a set of frequencies (sending and receiving) that differed from the frequencies of the adjacent cells. Because the power of the transmitter in a particular cell was kept at a level just high enough to serve that cell, these same sets of frequencies could be used at several places within the metropolitan area. Two important characteristics of cellular systems contribute to their usefulness. The first is their controlled handoff. As subscribers are driving out of one cell and into another, their mobile (subscriber) units, in conjunction with sophisticated electronic equipment at the cell sites [also known as base stations(BS)] and the telephone switching offices [also known as mobile switching centers, (MSCs)], are transferred from one frequency set to 5

6


another with no audible pause. Second, systems were designed to locate particular subscribers by paging them in each of the cells. After locating the vehicle in which the paged subscriber was riding, the equipment assigned sets of frequencies to it and conversation could begin. The initial transmission technology used between the vehicle and the cell site was analog in nature and known as Advanced Mobile Phone System (AMPS); the analog scheme used was called frequency-division multiple access (FDMA). Digital transmission was later developed, resulting in timedivision multiple access (TDMA). In Europe, the selected scheme was an adaptation of the TDMA used in the United States, called Groupe Speciale Mobile (GSM), now known as Global System for Mobile Communications. A few years later, a third group of companies (led by Qualcomm, Inc.) developed spread-spectrum technology called code-division multiple access (CDMA). Thus, there are at least four schemes that may be used for communications between a vehicle and the cell site. Communications between the cell site and the switch utilize transmission media such as microwave, copper pairs, and fiber optics. The continuing growth of cellular communications led government and industry in the United States to search for additional ways to satisfy the obvious need not only for ordinary telephone service but also for special services and features, smaller telephones, and cellular phone use. This search led to the development of the personal communications service (PCS) industry. Additional frequency bands were allocated for their use, and rather than assign them to the first comers or by way of a lottery, the Federal Communications Commission (FCC) [1] auctioned them off through a sophisticated bidding contest that brought the U.S. Treasury billions of dollars. Unfortunately, after acquiring the spectrum, many of the new potential operators went bankrupt and never actually had a chance to use it. In Canada, the decision was based on a beauty contest, a comparison of business cases, technology, and future plans, rather than the bidding process.

2.2 Existing Wireless Technologies 2.2.1

Mobile Networks and the Wireless Local Loop

Mobile communication is defined as a system in which the originator of the message, its recipient, or both, is in motion. User movements in mobile networks affect the communications system in many waysfrom the channel behavior in the physical layer to the system performance in the higher layers [2].


7

Wireless networks are not necessarily mobile; they can also be fixed in nature. Sometimes called radio in the loop (RLL) or fixed radio access (FRA), wireless local loop (WLL) is a system that connects subscribers to the Public Switched Telephone Network (PSTN) using radio signals as a substitute for copper for all or part of the connection between the subscriber and the switch. This connection includes cordless access systems, proprietary fixed radio access, and fixed cellular systems. In WLL application, the subscriber units are stationary without roaming capabilities of cellular (800 MHz) or PCS (1,900 MHz) services. Fixed local loop connections utilizing wireless multiple access networks are rapidly emerging as a viable alternative to cable-based networks for providing voice and data connection to subscribers in countries both with and without mature and established telecommunications networks. Much of the growth will occur in emerging economies where half the worlds population lacks plain old telephone services (POTS). Developing nations look to WLL technology as an efficient way to deploy POTS for millions of subscriberswithout the expense of burying tons of copper wire. In the more developed economies, WLL can help unlock competition in the local loop, enabling new operators to bypass existing wireline networks to deliver POTS and data access. The penetration of WLL into developed countries is assumed to be much lower than in emerging countries. The telephone companies are able to keep up with the demand for new POTS lines, thereby removing unserved demand as an issue. The requirement for WLL in developed countries will come from companies that want to bypass the established local phone companies or customers who want the additional services that WLL can provide. There is a significant difference in network design for the mobile wireless networks, fixed wireless networks, or a combination of the two, as equipment, quality requirements, and usage of the network resources can vary greatly. 2.2.2

Analog Cellular Systems

2.2.2.1 Principle of the Frequency Reuse

The first-generation cellular systems in operation were analog FM radio systems that allocated a single carrier for each cell. Each carrier was frequency modulated by the caller and the carriers were typically spaced at 25-KHz intervals. The allocated bandwidth was relatively narrow and only few channels (typically 12) were available [3]. Central to the cellular concept is the concept of frequency reuse. Although there are hundreds of channels available, if each frequency were

8


assigned to only one cell, the total system capacity would be equal to the total number of channels. Adjusting for the Erlang blocking probability results in only a few thousand subscribers per system. By reusing channels in multiple cells, the system can grow without geographical limits. Typical cellular reuse (pre-CDMA, that is) is easily rationalized by considering an idealized system. Assuming that propagation is uniform and that cell boundaries are at the equisignal points, then a planar service area is optimally covered by the classical hexagonal array of cells. Using seven sets of channels, one set in each cell, this unit is then replicated over the service area. While real systems do not ever look like these idealized hexagonal tilings of a plane, the seven-way reuse is typical of that achieved in practice. The capacity of a K-way reuse pattern is simply the total number of available channels divided by K. With K = 7 and 416 channels, there are approximately 57 channels available per cell. At a typical offered load of 0.05 Erlangs (50 mE) per subscriber, each site supports about 1,140 subscribers. Assuming that the cells are using omnidirectional antennas, it might be expected that system capacity is increased by antenna sectorization. Sites are, in fact, sectorized by the operators, usually three ways, with each site equipped with three sets of directional antennas, and their azimuths separated by 120°. Unfortunately, the sectorization does not in practice lead to an increase in capacity. The reason is that the sector-to-sector isolation, often no more than a few decibels, is insufficient to guarantee acceptably low interference. This is due partly to the poor front-to-back ratio of the antennas. The properties of electromagnetic propagation in the real world also conspire to mix signals between sectors. The practical result of sectorization is simply an increase in coverage because of the increased forward gain of the directional antenna. Nothing is gained in reuse. The same seven-way cell reuse pattern of omnidirectional cells applies in sectored cells. Viewed from the standpoint of sectors, in this case the reuse is K = 7 × 3 = 21, not 7. 2.2.2.2 Cellular Digital Packet Data

Cellular digital packet data (CDPD) is designed to provide packet data service on an existing cellular telephone network. First, the basic goal of the CDPD system is to provide data services on a noninterfering basis using existing cellular telephone service with 30-kHz channel spacing. This can be achieved by devoting some of the existing channels to CDPD service. Second, CDPD is designed to make use of cellular channels (those temporarily not being used for voice traffic). Basically, the system is used along with AMPS system, and a possible application is for digital AMPS (D-AMPS). The 30-kHz channels with CDPD support bit rates of up to 19.2 Kbps.


9

The degraded radio channel condition, however, limits actual information payload throughput rates to lower levels such as 5 to 10 Kbps. This will introduce additional time delay due to error detection and transmission protocol in the CDPD radio link physical layer using GMSK modulation at standard cellular frequencies at both forward and reverse links. 2.2.3

FDMA

FDMA is based on frequency division multiplexing (FDM). The digital FDMA systems resemble analog FM, but the carrier is modulated by a digitally encoded speech signal. FDMA allocates a single channel to one user at a time. If the transmission path deteriorates, the controller switches the system to another channel. Although technically simple to implement, FDMA is wasteful of bandwidththe channel is assigned to a single conversation whether or not somebody is speaking. Moreover, it cannot handle alternate forms of data, only voice transmissions. The first generation of analog wireless networks, or cellular networks, were based on this principle. 2.2.4

TDMA

During the late 1980s, the wireless industry began to explore converting the existing analog network to digital as a means of improving capacity. The Cellular Telecommunications Industry Association (CTIA) chose TDMA over Motorolas FDMA (today known as NAMPS) narrowband standard as the technology of choice for existing 800-MHz cellular markets and for emerging 1.9-GHz markets. The TDMA-136 specification, which was defined in the United States in 1988 by the Telecommunications Industry Association (TIA), was developed with the aim of digitizing the analog AMPS. To maintain compatibility with AMPS, the TDMA specification stipulates 30-kHz carrier spacing in a three-slot TDMA solution. TDMA is digital transmission technology that allows a number of users to access a single radio frequency channel without interference by allocating unique timeslots to each user within each channel. The TDMA digital transmission scheme multiplexes three signals over a single channel. The current TDMA standard for cellular divides a single channel into six timeslots, with each signal using two slots, providing a 3-to-1 gain in capacity over AMPS. Each caller is assigned a specific timeslot for transmission. Because of its adoption by the European standard GSM Communications, the Japanese Digital Cellular (JDC), and North American Digital Cellular (NADC), TDMA and its variants are currently the technology of choice

10


throughout the world. Over the last few years, however, a debate has convulsed the wireless community over the respective merits of TDMA and CDMA. The TDMA system is designed for use in a range of environments and situations, from hand portable use in a downtown office to a mobile user traveling at high speed on the freeway. The system also supports a variety of services for the end user, such as voice, data, fax, short message services, and broadcast messages. TDMA offers a flexible air interface, providing high performance with respect to capacity, coverage, and support of mobility and capability to handle different types of user needs. 2.2.5

Personal Digital Cellular

The development of the Personal Digital Cellular (PDC) specification was drafted by the RCR (1990), which later became the Association of Radio Industries and Broadcasting (ARIB). To ensure compatibility with the Japanese analog systems, a carrier spacing of 25 kHz was maintained in a threeslot TDMA solution. There is very little development going on in this technology today, and it is expected that it will be eventually completely phased out. 2.2.6

GSM and General Packet Radio Services

GSM is a set of ETSI standards specifying the infrastructure for a digital wireless service. The new Pan-European digital cellular standard was developed in 1985. The GSM system, which became operational in 1991, has since evolved into the leading global second-generation (2G) standard, in terms of number of subscribers and area of coverage. To ensure interoperability between countries, these standards address much of the network wireless infrastructure, including the radio interface (900 MHz), switching, and intelligent network (IN). An 1,800-MHz version, DCS 1800, has been defined to facilitate implementation in some countries, particularly the United Kingdom. GSM was designed to provide good speech quality with low-cost service and low terminal cost and at the same time provides a range of new services, including international roaming. GSM is an eight-slot TDMA system with 200-kHz carrier spacing. In terms of service, GSM is a mobile integrated services digital network (ISDN), with support for a wide variety of services. IN support in the mobile environment has also been defined for GSM, for example, the virtual home environment, as well as many advanced data services. GSM subscriber data is


11

carried on a subscriber identity module (SIM) card, which is inserted in the mobile phone to make it work. So, the subscriber potentially has an option of either SIM card mobility or terminal mobility across multiple networks. Today, using General Packet Radio Services (GPRS), packet access can also be integrated into GSM. GPRS is a new nonvoice value-added service that allows information to be sent and received across a mobile telephone network. It supplements todays circuit-switched data and short-message service (SMS) and has connectivity to X.25 and Internet Protocol (IP) networks. GPRS involves overlaying a packet-based air interface on the existing circuit-switched GSM network. This gives the user an option to use a packet-based data service. To supplement a circuit-switched network architecture with packet switching is quite a major upgrade; However, the GPRS standard is delivered in a very elegant mannerwith network operators needing only to add a couple of new infrastructure nodes and make a software upgrade to some existing network elements. Services running on mobile IP packet networks can be divided into horizontal and vertical services, where horizontal refers to e-mail, file transfer, information searching, multimedia, and so on, and vertical refers to applications targeting government, police, transport, medical, business, and the like. Theoretical maximum speeds of up to 171.2 Kbps are achievable with GPRS using all eight timeslots at the same time. This is about three times as fast as the data transmission speeds possible over todays fixed telecommunications networks and ten times as fast as current circuit-switched data services on GSM networks. By allowing information to be transmitted more quickly and efficiently across the mobile network, GPRS may well be a relatively less costly mobile data service compared with SMS and circuitswitched data. GPRS facilitates instant connections whereby information can be sent or received immediately as the need arises, subject to radio coverage and when no dial-up modem connection is necessary. This is why GPRS users are sometimes referred to as being always connected. Immediacy (immediate connection) is one of the advantages of GPRS (and SMS) when compared with circuit-switched data. High immediacy is a very important feature for time-critical applications, such as remote credit card authorization, where it would be unacceptable to keep the customer waiting for even 30 extra seconds. GPRS also facilitates several new applications that have not previously been available over GSM networks due to the speed limitations of circuitswitched data (9.6 Kbps) and message length of the SMS of only 160 characters. GPRS will fully enable the Internet applications you are used to on your

12


desktop, from Web browsing to chat, over the mobile network. Other new applications for GPRS include file transfer and home automationthe ability to access and control in-house appliances and machines remotely. Packet switching means that GPRS radio resources are used only when users are actually sending or receiving data. Rather than dedicating a radio channel to a mobile data user for a fixed period of time, the available radio resource can be concurrently shared between several users. This efficient use of scarce radio resources means that large numbers of GPRS users can potentially share the same bandwidth and be served from a single cell. The actual number of users supported depends on the application being used and how much data is being transferred. Because of the spectrum efficiency of GPRS, there is less need to build in idle capacity that is only used in peak hours. GPRS therefore enables network operators to maximize the use of their network resources in a dynamic and flexible way, along with allowing user access to resources and revenues. It should be noted here that the GPRS is not only a service designed to be deployed on mobile networks that are based on the GSM digital mobile phone standard. The IS-136-TDMA standard, popular in North and South America, will also support GPRS. This follows an agreement to follow the same evolution path toward 3G mobile phone networks concluded in early 1999 by the industry associations that support these two network types. It should already be clear that GPRS is an important new enabling mobile data service that offers a major improvement in spectrum efficiency, capability, and functionality compared with todays nonvoice mobile services. It is important to note, however, that there are some limitations with GPRS, which are described here. GPRS does impact a networks existing cell capacity. There are only limited radio resources that can be deployed for different usesuse for one purpose precludes simultaneous use for another. For example, voice and GPRS calls both use the same network resources. The extent of the impact depends upon the number of timeslots, if any, that are reserved for exclusive use of GPRS. However, GPRS does dynamically manage channel allocation and allows a reduction in peak-time-signaling channel loading by sending short messages over GPRS channels instead. Achieving the theoretical maximum GPRS data transmission speed of 172.2 Kbps would require a single user taking over all eight timeslots without any error protection. Clearly, it is unlikely that a network operator will allow all timeslots to be used by a single GPRS user. Additionally, the initial GPRS terminals are expected to be severely limited, supporting only one, two, or three timeslots. The bandwidth available to a GPRS user will therefore be severely limited. As such, the


13

theoretical maximum GPRS speeds should be checked against the reality of constraints in the networks and terminals. The reality is that mobile networks are always likely to have lower data-transmission speeds than fixed networks. GPRS is based on a modulation technique known as Gaussian minimum-shift keying (GMSK). GPRS packets are sent in all different directions to reach the same destination. This opens up the potential for one or some of those packets to be lost or corrupted during the data transmission over the radio link. The GPRS standards recognize this inherent feature of wireless packet technologies and incorporate data integrity and retransmission strategies. However, the result is that potential transit delays can occur. Because of this, applications requiring broadcast-quality video may well be implemented using high-speed circuit-switched data (HSCSD). HSCSD is simply a circuit-switched data call in which a single user can take over up to four separate channels at the same time. Because of its characteristic of end-to-end connection between sender and recipient, transmission delays are less likely. Whereas the store-and-forward engine in the SMS is the heart of the SMS center and a key feature of the SMS service, there is no storage mechanism incorporated into the GPRS standard, apart from the incorporation of interconnection links between SMS and GPRS. When a new service is introduced, there are a number of stages before it becomes established. GPRS service developments will include standardization, infrastructure development, network trials, contracts placed, network rollout, availability of terminals, application development, and so on. 2.2.7

CDMA

CDMA is a spread-spectrum technology that allows multiple frequencies to be used simultaneously. CDMA codes every digital packet it sends with a unique key, and the CDMA receiver responds only to that key (Walsh codes) and can pick out and demodulate the associated signal. One of the characteristics of CDMA is that it uses multiple levels of diverse receptionfrequency, spatial, time, and path diversity. When a signal is received, it is received from a number of paths, some directly and others reflected from buildings, mountains, or other obstacles. The technology combines all these signals to create a clear and reliable signal. In March 1992 the TIA established the TR-45.5 subcommittee with the charter of developing a spread-spectrum digital cellular standard. In July 1993 the TIA gave its approval of the CDMA IS-95 standard. IS-95 (also

14


called cdmaOne) systems divide the radio spectrum into carriers that are 1,250 kHz (1.25 MHz) wide. Unlike GSM and other TDMA-based wireless technologies, CDMA frequency reuse theoretical value is 1. Practical value is close to 1.6 due to interference from users in other cells [4]. CDMA is a spread-spectrum technology, which means that it spreads the information contained in a particular signal of interest over a much greater bandwidth than the original signal [5]. When implemented in a cellular telephone system, CDMA technology offers numerous benefits to the cellular operators and their subscribers: • Capacity increases of 8 to 10 times that of an AMPS analog system

and 4 to 5 times that of a GSM system;

• Improved call quality, with better and more consistent sound as

compared with AMPS systems;

• Simplified system planning through the use of the same frequency in

every sector of every cell;

• Enhanced privacy, meaning low detectability of the transmitted sig-

nal by an intended receiver;

• Improved coverage characteristics allowing for the possibility of

fewer cell sites;

• High tolerance to intentional interference (jamming) or uninten-

tional interference;

• Increased talk time for portables; • Bandwidth on demand.

The narrowband CDMA IS-95 specification stipulates 1.25-MHz carrier spacing for telephony services. Each of the 2G standards essentially defines a mobile telephony system; that is, a system that provides mobile end users with circuit-switched telephony services. Aside from voice services, these systems support supplementary services and some low-bit-rate data services. A number of new flavors of CDMA have been proposed for the next generation of wireless systems (3G); common to all of them is the fact that they are wideband, provide seamless interfrequency handoff, pilot-aided coherent reverse link, and fast closed-loop power control in the forward link [6].


15

2.3 Evolution of Wireless Technology 2.3.1

1G and 2G Wireless Networks

First-generation (1G) wireless cellular systems, analog in nature, introduced the convenience of limited mobility to the consumer. The most widely deployed 1G (analog) mobile phone systems include AMPS, Nordic Mobile Telephone (NMT), and Total Access Communications Systems (TACS). An AMP has major network deployments in North America, the Asia-Pacific region, and Central and Latin America. TACS and NMT first deployments were primarily in Europe (NMT in Scandinavia and TACS in the United Kingdom) and then the Asia-Pacific region. Second-generation systems, digital in nature, consist of GSM, IS-136 or D-AMPS, IS-95 or cdmaOne, and PDC. GSM, D-AMPS, and PDC are TDMA-based systems, while IS-95 is a CDMA system. Although digital in nature, 2G systems were focused on voice and had very little data-transfer capability. The Wireless Application Protocol (WAP) is a hot topic that has been widely hyped in the mobile industry and outside of it. It is also sometimes, like GPRS, referred to as 2.5G. WAP is simply a protocola standardized way that a mobile phone talks to a server installed in the mobile phone network. Mobile information services, a key application for WAP, have not been as successful as many network operators expected, and WAP is seen as a way to rectify this situation. WAP services are expected to be expensive to use because the tendency is to be on-line for a long circuit-switched data (CSD) call. WAP takes a client-server approach. It incorporates a relatively simple microbrowser into the mobile phone, requiring only limited resources on the mobile phone while the intelligence is in the WAP gateways. The WAP is envisaged as a comprehensive and scaleable protocol designed for use with any mobile phone. Examples are those with a one-line display to a smart phone, any existing or planned wireless service such as the SMS, CSD, unstructured supplementary services data (USSD), and GPRS, and any mobile network standard, such as CDMA, GSM, or the Universal Mobile Telephone System (UMTS). 2.3.2

3G Wireless Networks

2.3.2.1 Definition

It is important to clarify what is meant by 3G. The radio component of advanced technologies varies in terms of at least three characteristics:

16


1. The radio frequency (RF) channel width. This can range from 200 kHz for GSM-GPRS to 5 MHz for UMTS. 2. The RF spectrum allocation. This can vary from deployment on currently assigned spectrum at 800 and 1900 MHz for CDMA2000 1x to deployment on newly allocated spectrum at 1900 and 2100 MHz for UMTS. While seldom discussed, this latter spectrum is also suitable for CDMA2000 1x as well. 3. The data rate. Depending on technology, this may range from theoretical rates of 115 Kbps to beyond 2 Mbps. Some advanced technologies are called 2.5G, others are called 3G, and some are, or have been, called both. The International Telecommunication Union (ITU) serves as the arbiter of 3G standards. It does not define 3G in terms of channel width or spectrum allocation, but rather of data rates. By ITU definition, the 3G RF interface can deliver data rates of 144 Kbps or greater. The ITU recognizes Wideband CDMA (UMTS) and CDMA2000 1x as meeting this criterion. Higher data rates will enable end users to experience richer content than is now available and, in conjunction with packet architecture, to gain instant and low-cost access to the Internet. Higher data rates, and especially instant and low-cost access to the Internet, will expand future network traffic. As network traffic expands, operator revenues will increase. However, independent of the capability of technologies, operators must recognize the economic-commercial trade-off of network costs versus data rates. The higher the data rates, the greater the network costs. Eventually, every operator must optimize the data rate it offers to end users in terms of the cost to provide it versus the revenues it generates. While the 2G radio access brought mobile telephony capabilities to the mass market, the 3G radio access is expected to introduce value that extends beyond basic telephony. The widespread growth of the Internet has created a mass-market base for multimedia and information services. The challenge is to merge mobile telephony coverage and the associated user base with the Internet and other multimedia applications. To meet this challenge successfully , 3G radio access must provide the following: • Flexible multimedia management; • Internet access; • Flexible bearer services;


17

• Cost-effective packet access for best-effort services.

Most new multimedia services will be offered via the Internet. Therefore, a characteristic feature of 3G radio access is that it provides mobile Internet. Multimedia requires considerable flexibility; that is, the costeffective ability to support different bearer services with very different requirements, such as different bit rates (constant or variable), real-time or best-effort service, and packet- or circuit-switched service. In addition, 3G radio access must provide full-area coverage (same as 2G voice service), high-peak bit-rate services (384-Kbps full-area coverage, 2-Mbps local coverage) and any kind of service mix. Finally, 3G radio access must use the radio spectrum and network resources in a cost-effective fashion. To succeed, the 3G standards must facilitate efficient migration from 2G radio access. The introduction of multimedia into mobile communication will proceed gradually over time. Thus, a step-by-step migration plan must be developed that begins with the state of present-day 2G systems. Given that there are four separate existing 2G standards (GSM, TDMA, PDC, and CDMA/IS-95), different migration paths must be offered. Throughout the past decade, the ITU Radio Communications Sector (ITU-R) has elaborated a framework for global 3G standards. At the same time, since the early 1990s, the industry has been actively researching 3G radio access. 2.3.2.2 3G Standardization Process

It was expected that the 3G mobile phone system would be available commercially in 2001 or 2002, but it appears likely that it will be delayed for a year or even longer. The idea behind 3G is to unify the disparate standards that todays 2G wireless networks use. Instead of different network types being adopted in the Americas, Europe, and Japan, the plan is for a single network standard to be agreed upon and implemented. In 1998 the ITU called for radio transmission technology (RTT) proposals for IMT-2000, originally called Future Public Land Mobile Telecommunications Systems (FPLMTS), the formal name for the 3G standard. Many proposals were submittedthe DECT and TDMA/Universal Wireless Communications organizations submitted plans for the RTT to be TDMA-based, while all other proposals for non-satellite-based solutions were based on wideband CDMA. The main submissions were called wideband CDMA (WCDMA) and CDMA2000. The ETSI/GSM players including infrastructure vendors such as Nokia and Ericsson backed WCDMA. The North American CDMA

18


community, led by the CDMA Development Group (CDG), including such infrastructure vendors as Qualcomm and Lucent Technologies, backed CDMA2000. After acquiring Qualcomms CDMA infrastructure division in 1999, Ericsson is now supporting CDMA2000 as well. 2.3.2.3 Third Generation Partnership Project

In December 1998 the Third Generation Partnership Project (3GPP) was created following an agreement between six worldwide standards-setting bodies including ETSI, ARIB and TIC of Japan, ANSI of the United States, and the TTA of Korea. This cooperation into standards setting made 3GPP responsible for preparing, approving, and maintaining the technical specifications and reports for a 3G mobile system based on evolved GSM core networks and the frequency division duplex (FDD) and time division duplex (TDD) radio access technology. For example, ETSI SMG2 activities on UMTS have been fully transferred to 3GPP. Interestingly enough, China, an important market for the wireless industry, and the CDG were not original members of the 3GPP. In the first half of 1999, much progress was made in agreeing on a global IMT-2000 standard that met the political and commercial requirements of the various technology protagonists, such as GSM, CDMA, and TDMA. In late March 1999 Ericsson purchased Qualcomms CDMA infrastructure division and Ericsson and Qualcomm licensed each others key intellectual property rights and agreed to the ITUs family-ofnetworks compromise to the various standards proposals. 2.3.2.4 3G Data Rates

The ITU has laid down some indicative minimum requirements for the data speeds that the IMT-2000 standards must support. These requirements are defined according to the degree of mobility involved when the 3G call is being made. As such, the data rate that will be available over 3G will depend upon the environment in which the call is being made. • High Mobility. 144 Kbps for rural outdoor mobile use. This data

rate is available for environments in which the 3G user is traveling more than 120 km/h in outdoor environments. Let us hope that the 3G user is in a train and not driving along and trying to use a 3G terminal at such speeds.

• Full Mobility. 384 Kbps for pedestrian users and those traveling less

than 120 km/h in urban outdoor environments.


19

• Limited Mobility. At least 2 Mbps with low mobility (less than 10

km/h) in stationary indoor and short-range outdoor environments. These kinds of maximum data rates are often talked about when illustrating the potential for 3G technology to only be available in stationary indoor environments.

2.3.3

CDMA2000

CDMA2000 is a wireless standard that supports 3G services as defined by the ITUs IMT-2000 vision. CDMA2000 networks are backward compatible to cdmaOne (IS-95 CDMA), providing simple and cost-effective migration paths to next-generation wireless services and will offer voice quality and voice capacity improvements while delivering high speed and multimedia data services. The CDMA2000 standard was originally divided into two phases, commonly known as 1x and 3x. To realize the IMT-2000 vision in only 1.25 MHz of spectrum, an evolutionary standard under development, known as 1XEV, will enhance the capabilities CDMA2000 can deliver beyond 1x. CDMA operators around the world have recently defined the requirements for this standard through the CDG. CDMA2000 1x is implemented in existing spectrum allocations and delivers approximately twice the voice capacity of cdmaOne and data rates up to 144 Kbps. CDMA2000 1x also offers backward compatibility with cdmaOne networks along with other performance improvements. The TIA has published the CDMA2000 1x standard (IS-2000). The name 1x is derived from the technical term 1xRTT, which refers to CDMA2000 implementation within 1.25 MHz of existing spectrum. The designation 1x means one times 1.25 MHz, and RTT stands for radio transmission technology; 1x can be implemented in existing spectrum or in new spectrum allocations. Building on the 1x standard and CDMA2000s legacy of investment protection and spectral efficiency, 1xEV will improve data throughput, achieving peak rates of 2 to 5 Mbps, without requiring more than 1.25 MHz of bandwidth. Newly defined operator requirements for 1xEV specify two phases. In the first phase, the requirements request data throughput of up to 2.4 Mbps for an efficient, best-effort approach to data delivery. Phase 2 focuses on real-time voice and data capabilities and performance increases for both voice and data efficiency. CDMA2000 3x, part of the original CDMA2000 standard, provides for capacity increases over 1x and data rates up to 2 Mbps using a multicarrier approach. The name 3x is derived from the technical term 3xRTT,

20


which refers to N = 3 (i.e., use of three 1.25-MHz carriers). Thus, the multicarrier approach utilizes three 1.25-MHz carriers to deliver wideband 3G services. The idea of CDMA2000 3x (3xRTT) was recently abandoned in favor of the new and different approach. The first phase of CDMA2000 next-generation technology, called 1xRTT, has a data-only option, called 1xRTT Evolution-Data Only, for example. Called 1xEV-DO for short, this is the recently standardized CDMA technology based on Qualcomms high data rate (HDR) and adopted by top-tier equipment manufacturers at the end of 2000. According to data from the CDG, 1xEV-DO is expected to provide peak data rates of 1.25 Mbps for downloading data and 300 Kbps for uploads when users are mobile and moving at vehicular speeds. Average rates of 600 Kbps for downloads and 144 Kbps for uploads are achieved in a fully loaded system. The 1xEV-DO technology offers most of what 3xRTT does but accomplishes it within the existing 1.25-MHz CDMA carrier. For operators, that is a big step toward cost-effective high-speed data because it avoids the hardware and software upgrade changes required with the 5-MHz carrier of CDMA2000 3x. These speeds are further increased with a second evolutionary phase of 1xEV, called 1xEV-DV, which also supports voice. It is expected that the 1xEV-DV standard will be ready sometime in 2002. WCDMA and CDMA2000 networks are capable of streaming audio and video to large numbers of users, while GSMs 2.5G solution (GPRS) is not and must focus on text, graphics, animation, and electronic music. Figure 2.1 illustrates the next generation of CDMA2000 network architecture. It is obvious that the network is much more complex than the previous generations of wireless networks as a result of the 3G services offered to the customer. The BSC will separate packet data, which will be handled by the PDSN, from voice and circuit-switched data, which will be handled by the BSC or MSC. The MSC, visting location register (VLR), and home location register (HLR) provide the CS-CU functionality. The CDMA2000 packet core network (PCN) consists of three elements: the packet data serving node (PDSN), the home agent (HA) and the authentication, authorization, and accounting (AAA) server. The PDSN has switching and routing funtionality and functions as a connection point for the radio and IP/ATM networks. Point-to-Point Protocol (PPP) links are established, maintained, and terminated here. In addition, the PDSN delivers foreign agent (FA) functionality to register and facilitate services for network visitors (roamers). In conjunction with the PDSN, the HA authenticates Mobile IP registrations from the mobile client and maintains current location information. The HA also performs packet tunneling, a

Internet

Other ATM/IP networks

Supporting AAAF servers AAAH Router

VLR

Router

ATM

Media gateway/PDSN

AC

MSC server

Operators ATM/IP backbone

PSTN

HA

VLR HLR

Media gateway/PDSN

MSC PSTN

AC BSC

BSC

MSC PSTN

Access network

Access network

cdma2000

cdma2000


HA

HLR

HLR

WAP/HDML

Access transmission network Core transmission network

AAA Authentication , authorization, and accounting BSC Base station controller PCN Packet core network Internet protocol IP AC Authentication center IWF Interworking function HA Home agent PDSN Packet data service node ATM Asynchronous transfer mode WGW Media gateway HLR Home location register WAP Wireless application protocol

21

Figure 2.1 3G high-level CDMA2000-based wireless network topology.

22


function that receives packets destined for a mobiles permanent address and routes them to the mobiles new temporary address. The AAA server authenticates and authorizes the mobile client, provides user profile and quality of service (QoS) information to the PDSN and stores accounting data. The AAA server provides true policy management, profile definition, and the ability to offer a range of differentiated services. From premium services like 24-hour, high-bandwidth Internet access to more standard services, such as simple e-mail, the AAA server enables custom service packages with unique characteristics that address target market segments. The AAA servers function in a PCN network is similar to HLRs funtion in a voice network. CDMA2000 networks will support two IP addressing options for packet data: • Simple IP: used when no mobilityis required; • Mobile IP: used when mobility beyond a packet zone is required

based on Moblie IP standards from IETF.

To offer 3G services, operators need to invest in the access and core networks of their systems and will be looking for solutions that are both easy to adopt and provide a wide range of services. Initially, emphasis will be on the introduction of high-speed mobile data services, multimedia services, and services that require a guaranteed QoS. End users will expect to have access to services anywhere and at any time; moreover, they will expect reliable, secure connections during transmissions every time. Third generation is based on a different technology platform (i.e., CDMA, as opposed to the TDMA technology widely used in the 2G world, including GSM). 2.3.4

CDMA2000 1xEV-DO

The TIA recently adopted a specification based on HDR, with the designation TIA/EIA/IS-856, also known as 1xEV. The 1xEV specification was developed by 3GPP2, a partnership consisting of five telecommunications standards bodies: CWTS in China, ARIB and TTC in Japan, TTA in Korea, and TIA in North America. CDMA2000 1xEV-DO supports packetswitched voice and packet-switched high-speed data on separate RF channels. The voice channel facilitates the low latency necessary for transmitting two-way conversation, while the data channel enables the flexible routing and low-cost transmission advantages of a packet network. CDMA2000


23

1xEV-DO provides theoretical data speeds of up to 2.4 Mbps. In theory, using separate channels for voice and data requires more bandwidth than using a combined channel. In practice, the spectrum disadvantage diminishes as data traffic increases. This will be especially true for operators with larger spectrum assignments and large data throughput. Of particular value, in some cases not fully recognized, the migration of cdmaOne to CDMA2000 1x and beyond provides a more flexible use of spectrum compared with the migrations from GSM to UMTS and TDMA/IS-136 through GSM to UMTS. Under present concepts, GSM will not be available for the 1,900 and 2,100 MHz frequencies allocated to UMTS. UMTS will not be available for the 800-, 900-, 1,800-, and 1,900-MHz frequencies allocated to GSM. However, operators can deploy CDMA2000 1xEV-DO (and eventually EV-DV) either on newly available 1,900- and 2,100-MHz spectrum or on currently assigned 800- or 1,900- MHz spectrum. Some operators have deployed CDMA2000 1x on current spectrum. The Japanese operator KDDI intends to deploy CDMA2000 1x on newly available spectrum while most operators will deploy CDMA2000 1x on current spectrum. This flexible use of spectrum is an advantage of CDMA2000 1x. By enabling operators to use their current spectrum, this technique can save them the overt costs of bidding for new 3G spectrum or, in the case of beauty contests, the covert costs of petitioning for it. The latter can be considerable, especially when they include onerous conditions for network construction. Sweden, for example, did not charge for 3G licenses; however, it did require each license recipient to spend what would have been $3 billion or more for constructing full nationwide networks within two years of the license award. The regulator has since eased this burden by allowing the license recipients to share up to 70% of the 3G infrastructure. Operators who deploy CDMA2000 1x on currently assigned spectrum do not gain the added capacity that new spectrum provides. However, this disadvantage is to some extent overcome by the more efficient coding algorithm that CDMA2000 1x deploys. This algorithm doubles the theoretical capacity of cdmaOne, although in practice the capacity gain without voice degradation will be closer to 50%. UMTS will also deploy a more efficient coding algorithm and realize the associated capacity gains. On the other hand, because GPRS is network architecture, not an RF interface, it cannot provide capacity gains. For more information on 1xEV-DO see [7]. The TIA/EIA/IS-856 standard was prepared by Technical Specification Group C of the 3GPP2. This standard is evolved from and is a companion to the CDMA2000

24


standards. This air-interface standard provides high-rate packet data services, and 10 different operating bands have been specified. 2.3.5

Future Directions

The central challenge for the industry lies in creating a stable next-generation network platform that can support operators, service providers, and users efficiently and profitably for the foreseeable future. A significant portion of this challenge involves integrating the existing public-switched network with packet-based technologies. Until recently, the business of telecommunications operators centered around providing basic voice services (i.e., dial tone) to a mass market, and voice and data communications connectivity and services to large and medium businesses. Changes in lifestyles, business practices, technologies, and service possibilities mean that a once simple market is increasingly becoming fragmented. It is now possible to consider the concept of the personalized communications environmenta mix of mobile and fixed voice, data, and media services bundled, tariffed, and marketed on a tailored basis to individual users, with the additional freedom that these users can control and change these services at will. Today, less than 1% of traffic in mobile wireless networks is data. The enormous growth, experienced by todays mobile operators and service providers, has been achieved predominantly by voice services. With the arrival of 3G, initially in the form of packet data capabilities for existing networks, and subsequently by full 3G capabilities over new broadband wireless systems, this is set to change. Looking at the mass market in this way, it is possible to visualize discrete market sectors evolving out of a once undifferentiated whole in the following ways: • Basic voice services for the majority of the population will continue

to be the basic communications requirement, and voice is continuing to grow at a rate of approximately 5% a year. Use of advanced services, such as Caller ID, conferencing, call forwarding, and personal numbering will increase, while penetration of mobile voice and communications into the mass consumer market should continue.

• Internet technologies will have an impact on both the cost and

the functionality of voice services. The ability to both provide highquality voice communications over a packet network using IP and deliver advanced voice services will dramatically lower the cost


25

of voice calls, as well as lead to an increase in interactions between the world of the Web and the public voice network. Users will be able to control their services and access billing information through Web-based interfaces or smart terminals, and call-center services will evolve to mix the two environments for e-commerce applications. • Integrated voice and data services will be building on wideband

and broadband copper and wireless technologies (xDSL, HFC, FWA, CDMA2000, WCDMA, GPRS, and UMTS) and on the shift toward IP and packet technologies that can carry voice, data, and multimedia traffic. Simultaneously, users will be able to switch between various communications applications such as Web browsing, e-commerce, entertainment, e-mail, and voice using similar interfaces on different devices for their control.

The role of wireless networks and telecommunications networks in general in facilitating interoperability and ease of use will become increasingly important to consumers. Content and its presentation will become increasingly important in this scenario, as will QoS, reliability, and security. Most mass-market users expect their applications and services to be extremely dependable and intuitive. In theory, software-based functionality can be placed in end user terminals to enable interoperability, to resolve incompatibilities that would be perceived by customers as application failures, and to make complexity transparent to end users. In reality, this is achieved today by forcing end users to be systems administrators of their complex terminal software, or to engage others to administer their systems for them. Traditionally, the telephone networks have hidden complexity from end users and have resolved incompatibilities among end-user terminals by employing middleware in the networks, and all this should be resolved within the 3G wireless networks. In the planning process, these complexities translate into a fairly standard series of problem areas that along with more speculative considerations must be investigated and resolved as efficiently as possible. The first one is the persistent problem of how to extract more value from existing assets like network infrastructure, base station sites, the terms and conditions of the operating license, or limitations of available spectrum. The second dilemma is strategic in its nature and associated with migrating from 2G to 3G technologieswhat kind of capital investment is required, who are the best suppliers to satisfy technical and economical needs, and so on. Third, there is the

26


question of deciding on the optimum time to deploy a new system, service, or technology, in what market, and in which order.

2.4 Satellite Networks 2.4.1

Fixed Satellite Service

The space age opened many new opportunities for radio communications between widely separated locations. Instead of high-frequency (HF) terrestrial systems with limited bandwidth or a large number of short-range microwave relays, satellites can link distant locations from a point high above the Earth. By the mid-1960s, launch vehicles were delivering communications satellites to locations in the geostationary satellite orbit, about 35,800 km (22,000 mi) above the equator. In this orbit, the satellites circle the Earth at the same rate as the Earth rotates, making them appear nearly stationary from the Earths surface. Communications between two places on Earth can take place by using these satellites; one frequency band is used for the uplink and another for the downlink. Such satellite systems are excellent for the transmission of data, but they leave something to be desired for voice communications. This is a result of the huge distance and the time (delay) it takes for an electrical signal to make an Earth-satellite-Earth round trip, which can amount to more than one-quarter of a second. A reply from the called subscriber takes another quarter of a second or more, and the result of a half a second or more is definitely noticeable. Consequently, voice communications are seldom carried via geosynchronous satellites. Today, geostationary communications satellites continue to play a major role in telecommunications. From the geostationary orbit, satellite antennas can illuminate a small area (using spot beams), a country, or a larger region encompassing many countries. Thus, satellites can theoretically compete with point-to-point microwave and nonradio media (e.g., optical fiber) in providing communications between fixed points. The fixed satellite service primarily involves communications between fixed Earth stations via satellite (i.e., uplinks and downlinks), although the service can also include certain intersatellite links and feeder links. The fixed satellite service can include communications to multiple, specified fixed locations, but does not include broadcasting functions. The fixed satellite service basically involves four frequency bands: 46 GHz, 78 GHz (for military systems), 1114 GHz, and 2030 GHz. Although there are numerous bands above 30 GHz allocated to the fixed satellite service, only one is presently being used. Microwave frequencies and


27

stationary satellites allow the use of high-gain, directional antennas, much like the fixed service. This reduces the power requirements for the satellite transmitters. The fixed satellite service includes international, domestic, and military systems, and although they often carry the same type of traffic, each group has its own set of users. International and domestic systems operate at 46 GHz and 1114 GHz, while military systems use 78 GHz and frequencies near 20 GHz and 45 GHz.

2.4.2

Mobile Satellite Systems

Another of the wireless telecommunications technologies is the low-Earthorbit (LEO) satellite system, made up of satellites that communicate directly with handheld telephones on Earth. Because these satellites orbit at a relatively low altitude (less than 900 miles), they move across the sky quite rapidly [8]. In a LEO system, the communications equipment on a satellite acts in much the same way as a cell site of a cellular systemit catches the call transmitted from Earth and usually passes it to an Earth-based switching system. Because of the speed of the satellite, it is frequently necessary to hand off a particular call to a second satellite just rising over the horizon. This is similar to a cellular system, except that in this case it is the cell site that is moving rather than the subscriber. Due to the high cost of deploying a satellite mobile network, cost of service is very high and today in use only in very special cases and in the remote and unpopulated areas of the world. This is also the main reason for the failure of the Iridium projecta very small number of subscribers. Despite huge debt and failure to meet subscriber acquisition targets, the Globalstar network is unlikely to be abandoned. In the worst case, Globalstar will follow the precedent set by Iridium and Orbcomm (i.e., bankruptcy protection followed by an asset auction). This will allow the highest bidder to obtain a multi-billion-dollar satellite network at a fraction of its original cost. Globalstar offers significant advantages over other satellite networks for both mobile and fixed applications. Due to its high capacity and low latency, Globalstar is ideal for a wide range of voice, Internet access, e-mail, and monitoring and control applications. A broadband mobile satellite network (i.e., a network that employs both satellites on nongeostationary orbits and high-speed switching and routing techniques) could be considered the new generation of satellite networks [9, 10]. Requirements for lower propagation delays and propagation loss, together with the coverage of high latitudinal regions for PCS, have led

28


to the initiation of extensive research in this area and also in the area of Internet applications over the satellite.

2.5 Fixed Microwave Systems Fixed microwave (MW) systems typically use microwave frequencies above 1 GHz. Point-to-point microwave systems (often simply called microwave systems) provide wideband communications over the line-of-sight (LOS) paths. Tropospheric scatter systems provide point-to-point service over paths up to 200 km, using highly directional antennas and high-power transmitters. Point-to-multipoint microwave systems are used over LOS paths, often with an omnidirectional master antenna and directional node antennas. The fixed services are grouped within seven categories: HF services, VHF/UHF services, and five categories of microwave services. The microwave categories include common carrier, private, auxiliary broadcasting, cable relay, and federal government. These five microwave groups represent the licensing categories that are established and defined by the FCC or NTIA. 2.5.1

Microwave Point-to-Point Systems

The main purpose of microwave radio link is to transport data and voice traffic from one place to another. The radio link uses the air as the transport medium to send encoded electromagnetic waves. A typical link consists of two radios and two antennas separated by a distance from a couple of hundred meters up to tens of kilometers. The data and voice traffic are fed into the radio using either an electrical or optical line. In the radio the digital signals are coded into analog signals and converted to microwaves with a typical length of a few centimeters. Microwaves are used because they are able to propagate high bit rates safely through the air. The microwaves are sent using a highly directive parabolic-shaped antenna. At the other end the signals are received and restored to the digital format. This works both ways, simultaneously, of course. Microwaves, which are only centimeters (or inches) in length, are small relative to the surroundings, and hence, do not have this bending property. In order to establish a radio link, it is important to have LOS between the two radio position sites. Figure 2.2 shows basically two types of microwave systems (i.e., for low- to medium-capacity access and high-capacity backbone transmission systems). Terrestrial MW point-to-point systems use frequencies from approximately 1 to 60 GHz with maximum hop lengths of around 200 km (125


Split-configuration MW radio

15, 18, 23 GHz

Short-distance low- and mediumcapacity systems Access radio

Split-configuration MW radio

29

6, 7, 8 GHz

Long-distance Backbone medium- and systems high-capacity systems

Central office switch

Figure 2.2 Microwave transmission networks.

miles). Long hops over the water or flat surfaces (deserts, wheat fields, planes) are usually more difficult and more expensive to build. A typical microwave hop consists of parabolic dish antenna, waveguide or coaxial cable, and terminal (radio) equipment on both ends of the hop. LOS is the main prerequisite (although not the only one) for achieving satisfactory communication performance over the microwave point-to-point link. The greatest growth area for the use of digital microwave radio is currently associated with the emergence of new competitive wireless operating companies as part of a liberalized telecommunications environment. It is also becoming increasingly common for newly licensed wireless operators to be granted the rights to self-provide the transmission infrastructure. It is also fairly standard that the terms of such competitive licenses commit the operators to challenging operational obligations; that is, to provide service throughout a certain percentage of the country within an ambitious time frame. Furthermore, the terms require wireless operators to provide service at the earliest opportunity to realize revenues in line with their business plans. The speed of installation of the microwave radios and flexibility to upgrade in line with network requirements mean that almost all mobile operators that are independent from the Post, Telephone, and Telegraph Company (PTT)

30


organizations have the right to self-provide chosen digital microwave radio as the interconnect solution for base stations. In certain parts of the world, utility and government organizations have long had discretionary rights to build their own networks, and have historically been users of microwave radio. With growing liberalization, many other private users are also recognizing the benefits of digital microwave radio. As the above network applications indicate, there are many reasons that a wireless-network operator, given the right to self-provide (and own) transmission infrastructure, should choose microwave radio as opposed to utilizing leased lines or implementing their own cable-based systems. In summary, the advantages of MW radio systems are as follows: • MW systems are economical compared with fiber or leased lines •

•

• •

• •

over a period of at least 2 to 3 years. An owned transmission network remains under the control and ownership of the end user, which removes sometimes sensitive dependency upon the incumbent telephone company (often a competitor) and provides operational benefits. Modern MW radio architecture has been designed to provide a high degree of flexibility in terms of distance and traffic capacity, enabling links to be designed to fit operator requirements and local conditions precisely. Link capacities can also be field upgraded to cater to a networks growing traffic requirements as subscriber numbers increase. Owned MW networks can be planned to provide a higher QoS than often guaranteed by the leased T1/E1 lines. A MW link can, in the majority of circumstances, be installed and commissioned in a much shorter period of time than cable-based alternatives, because a microwave link does not require the same degree of civil works associated with laying cables. MW radio links can be removed and redeployed to another geographical area, without leaving valuable assets in the ground. MW radio is commercially available, and it can usually be supplied with short lead times.

In todays wireless networks, compact base stations require less time and money when it comes to selecting, acquiring, and preparing a site. The savings can be applied to areas that will generate revenue, such as coverage,


31

capacity, and applications, adding value for both operators and end users. Because of the reduced space requirements, the operator has more choices for site selection. This empowers operators to optimize network design and take control of costs early on by negotiating lower leases. This, of course, means that all the other equipment, including power and transmission equipment, should be as compact and upgradable as possible. One of the solutions is to use split-configuration microwave radio, which has an outdoor RF unit mounted very close (or attached to) the parabolic dish antenna. A small indoor unit (digital circuitry including multiplexer) is placed into the shelter of a small cabinet. Outdoor and indoor units are connected with the regular coaxial cable. Microwave radio is the preferred solution for radio base station (RBS) connectivity in most parts of the world, except in North America, where leased T1 lines are quite common and widely available. Most new wireless service providers (WSPs) in North America, in order to achieve short timeto-market, start with leased facilities (lines) and after a year or so, start redesigning the network using microwave radio. 2.5.2

Microwave Point-to-Multipoint Systems

2.5.2.1 Broadband Wireless Access

Currently, multiple cable TV channels, together with ancillary telephony and low data-rate services, are transmitted to homes and businesses via coaxial cable. However, the capability for interactive services and high-speed data transfer over this medium is limited, and cable TV provision necessitates the installation of underground cables right up to the home or business premises. An alternative technology, local multipoint distribution service (LMDS), which overcomes the inadequacies of cable TV and wireline integrated voice and data communications, is now available. LMDS is a fixed, wireless, point-to-multipoint (PMP) telecommunications platform that facilitates the two-way transmission of voice, data, and video. The system operates in the 2431-GHz bands, which, coupled with a high bandwidth capability in excess of 1 GHz, permits the communication of multimedia services with interactive facilities within a 5-km radius of a central data hub. For PMP architecture, the operator installs base stations around the market area very similar to traditional cellular systems. Those base stations have antennas that transmit and receive on multiple sectors and typical configuration is four sectors using 90º antennas. The subscribers (much like WLL users) use much narrower antennas that are typically installed on their rooftops pointing in the direction of the maximum signal strength from the base station. Similar

32


to cellular systems, frequencies can be reused in neighbor base stations or sectors, as long as the reuse distances are defined in such a way to avoid interference. These systems are used for the bandwidth on demand. Since the bandwidth is shared with other users, bandwidth available per subscriber is reduced with every new subscriber using the system. For subscribers who demand fixed bandwidth availability, point-to-point systems can be offered that will not share bandwidth with other subscribers. Bandwidth on demand is achieved by use of ATM as a transport mechanism. In Canada, the 27.3528.35-GHz (27-GHz) band is designated for LMCS. In the United States, the 27.3527.5-GHz portion of the band is designated for federal government fixed and mobile systems and intersatellite service, and the 27.528.35-GHz portion is designated for LMDS. The 29.129.25-GHz (29-GHz) band is designated for LMDS in the United States and is allocated to fixed and mobile services in Canada. The 31.031.3-GHz (31-GHz) band is designated for LMDS and fixed pointto-point microwave systems in the United States and is allocated to fixed and mobile services in Canada. LMCS is licensed by LMCS service areas and LMDS is licensed by basic trading areas (BTAs). BTAs are defined in [11], which identifies 487 BTAs based on the 50 United States. Further information on U.S. service areas and licensees is available at [1]. Some of the characteristics of PMP systems include the following: • They follow a cellular deployment structure, with multiple cells cov-

ering certain geographic areas.

• Each cell contains a hub with multiple radio nodes equipped with sec-

tor antennas for PMP and directional antennas for point-to-point connections.

• Statistical multiplexing is used. • They can use FDMA, TDMA, spread spectrum (direct sequence

[DS] or frequency hopping [FH]) over-the-air interface.

• They can use FDD or TDD. • LOS between the hub site and all the customer sites is required,

similar to classic point-to-point microwave systems, and therefore standard microwave propagation and prediction methods are used for the system design.

• Frequency coordination with other spectrum owners (and PMP

service providers) can be difficult and must be carefully executed.


33

The services to be provided, aside from regular cable TV, are telephony, video, videoconferencing, video programming (video on demand), data, full-duplex data communications, fast Internet access, and so on. Some other bands are also used for point-to-point systems, like MMDS 2.1 to 2.69 GHz and other unlicensed and newly allocated bands. Air interface standards for new broadband services in the microwave and millimeter-wave frequency range are being developed by the IEEE Working Group 802.16 (Broadband Wireless Access [BWA]). This group, working with the National Institute of Standards (NIST) and National Wireless Electronic Systems Testbed (N-WEST) is the primary focus of industry efforts to develop transmission standards that will support the goal of broadband wireless systems developers and their hardware providers. Originally established to address services in the 1066-GHz range, the 802.16 group has recently added bands below 10 GHz to its portfolio of standards projects. The most economical and practical infrastructure for providing wireless broadband channels to a high concentration of users will probably include a large number of small cells operating at the very high frequencies. It is becoming increasingly apparent that in order to provide wireless communications with bit rates in hundreds of megabits, a large amount of bandwidth in the millimeter-wave (over 30 GHz) must be utilized. A wide, continuous frequency bandwidth is available only in these higher-frequency bands. Many frequency spectrum regulatory agencies around the world, including the FCC, have allocated several large sections of spectrum in the millimeter-wave region [12]. Millimeter-wave characteristics dictate short-range LOS propagation (rain attenuation is predominated at frquencies about 10 GHz) with minimal refraction and reduced interference while providing a bandwidth capacity approaching coaxial or fiber-optic systems. These millimeter-wave characteristics require a cellular network topology to be based on a large number of small cells which facilitate frequency reuse resulting in a large number of traffic channels per service area and, thus, high network traffic capacity. Under FCC Part 15, the 5964-GHz range is available for general use by unlicensed devices based on severe propagation losses that provide protection against interference. In Europe, the 6263-GHz and 6566-GHz bands are allocated for licensed operation and specifically for mobile broadband systems (MBS). In Japan, the 5964-GHz band is regulated for use by MBS as well. The full-scale deployment of fixed wireless technology in the United States was at almost a standstill from 2000 to 2001, because the FCC began studying possible reallocation of the MMDS and Instructional Television

34


Fixed Service (ITFS) spectrum in the 25.026.9-GHz band. They are among several frequencies identified by the World Radiocommunications Conference for possible 3G use. In the United States the use of millimeter-wave broadband systems is already established, while in Europe, some countries (notably Germany) have already licensed the 26-GHz band. In the United Kingdom, the government awarded licenses for networks operating at 28 GHz in the summer of 2000, with 40-GHz licenses on offer later. Figure 2.3 shows a typical PMP system architecture based on the dynamic bandwidth allocation and ATM. During 2000, so-called mesh networks were proposed, wherein the problems of LOS to every subscriber would be avoided. Some of the subscriber sites within the range of the cell site will not have LOS with the cell site. If the subscriber station can act as a repeater and bounce signals to neighboring subscriber stations, the likelihood of having a LOS path to desired points is significantly improved. In addition to improving LOS coverage, mesh networks can improve service range for those locations that

PSTN 2426 GHz or 28 GHz

RBS 1

AT CE

AT

RBS 2

Hub site 1

90 deg

AT

C-QPSK modulation Point-to-point

RN

SDH/SONET

RS

RN RBS 3

MSC

E3/DS3

ATM MUX or ATM switch

AT

ATMbased, wireless broadband access system: F-DCAFast dynamic capacity allocation Frequency: 2431 GHz Radio Capacity: 37.5 Mbps (symmetrical) ATAccess termination RSRadio shelf CECircuit-emulation shelf RNRadio node NMSNetwork management system

Figure 2.3 MW PMP system architecture.

BSC

NMS

RS Hub site 2


35

originally could not be reached because of the distance and RF path losses. Of course, at one point the delay may become an issue. Industry is still debating the merits of TDD versus FDD in PMP networks. While TDD requires a single channel for full duplex communications, FDD systems require a paired channel for communication, one for the downlink (hub to remote) and one for the uplink (remote to hub). In TDD, transmit-receive separation occurs in the time domain, as opposed to FDD, where it occurs in frequency domain [13]. While FDD can handle traffic that has relatively constant bandwidth requirements in both directions, TDD better handles varying uplink-downlink traffic asymmetry by allocating time spent on up- and downlinks. It is obvious from this discussion that bursty traffic (data, Internet) favors TDD. TDD requires a guard time equal to the round-trip propagation delay between hub and remote units and increases with link distance. In FDD, sufficient isolation in frequency between the uplink and downlink is required. In other words, FDD is a simpler, but less efficient, solution for broadband access. 2.5.2.2 Wireless Local Access in U-NII Band

During the past 15 years, high-frequency digital microwave radio has proven to be a cost-effective and quality solution for short-haul local access applications. The high growth of cellular, PHS, PCN, and PCS networks has predominantly fueled the widespread use of wireless solutions from 10 GHz to 38 GHz for last mile solutions. Radio frequency allocation below 10 GHz continues to be used for wideband backbone traffic. The emphasis and requirement for microwave radio equipment operating below 10 GHz have been on new modulation techniques designed to squeeze more data into less bandwidth. Along with fiber networks, these spectral efficient backbone radios feed both wired and wireless local access applications. Although technological advances continue to provide lower-cost highfrequency products and more efficient backbone solutions, these bands are often congested and still too expensive for future wireless applications. As personal communications products and services evolve, faster and less expensive solutions are in high demand. One of the important developments in the telecommunications industry has been the FCC ruling 15.407 in favor of the U-NII frequency spectrum. On January 1997 the FCC allocated 300 MHz of spectrum for U-NII in the 5-GHz band to be used with U-NII products (see Table 2.1). The FCC believes that the creation of the U-NII band will stimulate the development of new unlicensed digital products that will provide efficient and less expensive solutions for local access applications.

36

Transmission Systems Design Handbook for Wireless Networks Table 2.1 FCC U-NII Bands Band 1

Band 2

Band 3

Frequency (GHz) 5.155.25

5.255.35

5.7255.825

Power (W)

0.2 (EIRP)

1.0 (EIRP)

4.0 (EIRP)

Application

Indoors only Campus

Wireless access

The U-NII band is divided into three subbands at 5.15 to 5.25, 5.25 to 5.35, and 5.725 to 5.825 GHz. The first band is strictly allocated for indoor use and is consistent with the European high-performance local area network (HIPERLAN). The second and third bands are intended for high-speed digital local access products for campus and short-haul microwave applications. The FCC rules for products operating in bands 2 and 3 of the U-NII band are best suited for digital microwave applications over distances in excess of 10 miles. FCC spectral efficiency and maximum power requirements for these bands facilitate the deployment of highly reliable microwave links for both data and telephony transmission. Figure 2.4 depicts the relationship between the maximum equivalent isotropic radiated power (EIRP) and occupied bandwidth of the transmitted signal in accordance with the regulations. 4500 4000

EIRP (mW)

3500 3000

5.7

2000

Hz

25 G

5.8 25

2500 1500 1000

5.255.35 GHz

500 0

5

10

Bandwidth

Figure 2.4 Maximum EIRP in the FCC U-NII bands 2 and 3.

15

20


37

The most effective use of the band is by means of robust modulation schemes capable of carrying high-speed Ethernet or multiple T1/E1 digital circuits. Modulation techniques, such as binary phase-shift keying (BPSK), frequency shift keying (FSK), and quadrature phase-shift keying (QPSK) are best suited to provide the most cost-effective and reliable interconnection solution. The U-NII frequency band is an ideal solution for short-haul applications. Unlike high-frequency microwave links above 10 GHz, the U-NII band is not affected by outages due to rain attenuation. Above 10 GHz, rain is a considerable factor in determining the maximum distance of a properly engineered microwave path. High-power amplifiers and larger antennas help; however, these solutions are expensive and often not applicable. Microwave transmission is also less affected by free space loss at 5.25 to 5.825 GHz than high-frequency microwave. Even with FCC limitations on power output and antenna gain in the U-NII bands (5.3 and 5.7 GHz), microwave paths can operate full duplex, using both bands 2 and 3, over 10 miles with 99.995% reliability. The microwave system performance using both bands 5.3 GHz and 5.7 GHz is limited by the FCC transmitter and antenna rules for band 2. The use of dual-band operation, however, does have the benefit of separating the system transmitters and receivers by approximately 480 MHz. This significantly simplifies the equipment transmitter and receiver design, resulting in a lower-cost product. Dual-band operation also promotes frequency reuse, allowing the use of 200 MHz of bandwidth as opposed to 100 MHz in single-band operation. The use of 2- or 4-ft, highly directional parabolic antennas (with gains of approximately 27 and 33 dBi, respectively) in the U-NII band improves the overall performance of the system. As shown in Figure 2.5, a high level of availability is typical for paths in the 8.5-mile range while meeting FCC rules for maximum EIRP and 99.995% of the time availability requirement for RBS-BSC connection (see later chapters on MW design for more details). Antenna gain can exceed 6 dBi as long as the peak power spectral density is reduced proportionately. Parabolic antennas also offer additional isolation from colocated or adjacent microwave signals. One of the concerns with operation in a license-exempt or ISM band is that of interference from other unlicensed band users. With proper use of the spread-spectrum technology and network planning, interference can be almost eliminated. When deploying networks that are PMP in nature in the ISM bands, there is essentially a choice of two spread-spectrum technologies to be considered: DS and FH. It is generally acknowledged that the DS systems can support higher bit rates than FH systems; however, this comes at a

38

Transmission Systems Design Handbook for Wireless Networks 60

2 FT2 FT antenna

Outage (min/yr)

50

2 FT4 FT antenna

40 30

99.995%

20 10 0

99.990% 3

3.5

4

4.5

5

5.5

6 6.5 7 Distance (mi)

7.5

8

8.5

9

9.5 10

Figure 2.5 Typical point-to-point microwave path performance in the U-NII band.

cost of reduced immunity to interference. For resisting narrowband transient or even nontransient interference in PMP networks, FH systems will have an advantage over DS networks. If the FH system experiences a bad hop, or possibly even more, it will merely send the data out on the next clear hop and no data is lost. With a DS system, if the interference is high enough, the link will fail and data transmission will be interrupted. In the U.S. market, wireless access is most successful where the existing infrastructure is weakest. There is little need in downtown New York or Toronto, for instance, for wireless fractional T1 connectivity because the fiber or copper is already installed and easily obtained. Where wireless networks can make the largest contribution and reach the customers are in the tier-two and tier-three markets. These markets are typically less built out and as a result have far fewer ambient 2.4-GHz radio signals floating around. In these environments, coverage of up to 20 miles is easily achievable. There are many system tradeoffs in selecting an optimum backhaul solution for the hub site. For small towns (of less than 100,000 population) with less data traffic, 28 T1 backhaul capacity may be enough. For larger towns with large numbers of subscribers with high data-rate needs, OC3 and higher capacity may be needed. For lower backhaul capacity needs, multiple T1 connections or point-to-point radio link connections may be sufficient. Point-to-point microwave links are appropriate for backhaul when the required capacity is less than a few T1s to DS3 and the cost of fiber near the


39

base station is too high. This is generally the case in the rural and less populated areas. Also, many times the base station hub is located on a ridge that may be inaccessible by fiber. There are unlicensed band (industrial, scientific, medical [ISM] and U-NII bands) products available for backhaul applications from a number of suppliers. The capacity of these products is generally low (16 Mbps) for multichannel multipoint distribution service (MMDS) cell backhaul applications. This type of low capacity is only useful in cells in a rural area with less population. A typical MMDS cell in a larger city needs OC-3 or higher backhaul capacity. In a combined MMDSunlicensed band system, an MMDS PMP system can satisfy the backhaul needs of the unlicensed band cells. In this scenario the upstream link will be the limiting factor in terms of capacity because the MMDS system modulation rates in the upstream are typically QPSK and 16-QAM (quadrature amplitude modulation). Downstream capacity is generally higher due to availability of 64-QAM modulation. Many applications in a building require portable broadband connections to a notebook or a laptop PC. In-building wireless LANs (WLANs) at 2.4, 5.8, and 24 GHz are now available at multimegabit data rates. Integration of MMDS network with the in-building WLAN will be required for many of these applications. Also, as Bluetooth and HomeRF technology is developed, a low-cost in-building network will provide an easier way for mobile computing and communications devices to communicate with one another and connect to the Internet at high speeds without the need for wires or cables. The broadband MMDS network of the future needs to be able to interface seamlessly in these advanced in-building wireless networks. Various architectures are proposed to use license-exempt bands to supplement and enhance the broadband MMDS networks. Specifically, unlicensed band networks can be used in conjunction with MMDS networks to fill holes in coverage due to hills or buildings or extend the coverage of the MMDS network. MMDS and unlicensed bands can also be used for low- to medium-capacity backhaul requirements. As more devices become available that comply with Bluetooth and HomeRF and other in-building networks, it makes sense to integrate these networks seamlessly with broadband MMDS networks. 2.5.2.3 Bluetooth

Bluetooth is a de facto open standard for short-range digital radio. It is designed to operate in the unlicensed ISM applications band, which is generally available in most parts of the world (Table 2.2). The specification includes air-interface protocols to allow several Bluetooth applications to intercommunicate simultaneously, and to overcome external sources of interference, such

40

Transmission Systems Design Handbook for Wireless Networks Table 2.2 Bluetooth Frequency Bands

Area

Frequency Band (GHz)

Number of Channels

United States, 2.40002.4835 Europe, and most other countries

79

Spain

2.44502.475

23

France

2.44652.4835

23

as microwave ovens. The short range referred to above is defined as up to 10m (30 ft) in normal operation, although greater range and penetration can be achieved through higher output powers under some circumstances. Bluetooth is considered to be a PMP system, although it can also be a point-to-point system depending on the application. Bluetooth technology allows for the replacement of the many proprietary cables that connect one device to another with one universal short-range radio link. For instance, Bluetooth radio technology built into both the cellular telephone and the laptop would replace the cumbersome cable used today to connect a laptop to a cellular telephone. Printers, PDAs, desktops, fax machines, keyboards, joysticks, and virtually any other digital device can be part of the Bluetooth system. But beyond replacing the cables, Bluetooth radio technology provides a universal bridge to existing data networks, a peripheral interface, and a mechanism to form small private ad hoc groupings of connected devices away from fixed network infrastructures. Designed to operate in a noisy radio frequency environment, the Bluetooth radio uses a fast acknowledgment and frequency-hopping scheme to make the link robust. Bluetooth radio modules avoid interference from other signals by hopping to a new frequency after transmitting or receiving a packet. Compared with other systems operating in the same frequency band, its radio typically hops faster and uses shorter packets. This makes the Bluetooth radio more robust than other systems. Short packages and fast hopping also limit the impact (interference) of domestic and professional microwave ovens. Use of forward error correction (FEC) limits the impact of random noise on long-distance links. The encoding is optimized for an uncoordinated environment, since Bluetooth radios operate in the unlicensed ISM band at 2.4 GHz. A frequency hop transceiver is applied to combat interference and fading. A shaped, binary FM modulation is applied to minimize transceiver complexity and a time-


41

division duplex scheme is used for full-duplex transmission with the gross data rate of about 1 Mbps. The Bluetooth baseband protocol is a combination of circuit and packet switching. Slots can be reserved for synchronous packets. Each packet is transmitted in a different hop frequency. A packet nominally covers a single slot, but can be extended to cover up to five slots. Bluetooth can support an asynchronous data channel, up to three simultaneous synchronous voice channels, or a channel that simultaneously supports asynchronous data and synchronous voice. Each voice channel supports a 64-Kbps synchronous (voice) link. The asynchronous channel can support an asymmetric link of maximally 721 Kbps in either direction while permitting 57.6 Kbps in the return direction, or a 432.6-Kbps symmetric link. The different functions in the Bluetooth system include the following: • Radio unit; • Link control unit; • Link management; • Software functions.

The Bluetooth system supports both point-to-point and PMP connections. Several piconets can be established and linked together ad hoc, where each piconet is identified by a different FH sequence and all users participating on the same piconet are synchronized to this hopping sequence. The topology can best be described as a multiple piconet structure. Voice channels use the continuous variable-slope delta (CVSD) modulation voice-coding scheme, and never retransmit voice packets. The CVSD method was chosen for its robustness in handling dropped and damaged voice samples. Rising interference levels are experienced as increased background noise: even at bit error rates up to 4%, the CVSD coded voice is quite audible. The Bluetooth air interface is based on a nominal antenna power of 0 dBm. The air interface complies with the FCC rules for the ISM band at power levels up to 0 dBm. Spectrum spreading has been added to facilitate optional operation at power levels up to 100 mW worldwide. Spectrum spreading is accomplished by frequency hopping in 79 hops displaced by 1 MHz, between 2.402 GHz and 2.480 GHz. Due to local regulations, the bandwidth is reduced in Japan, France, and Spain. This is handled by an internal software switch. The maximum FH rate is 1,600 hops per second. The nominal link range is 10 cm to 10m (4 in to 30 ft), but can be extended to more than 100m (300 ft)by increasing the transmit power. The link type

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defines what type of packets can be used on a particular link. The Bluetooth baseband technology supports two link types: • Synchronous connection oriented (SCO) type (used primarily for

voice); • Asynchronous connectionless (ACL) type (used primarily for packet data). Different master-slave pairs of the same piconet can use different link types, and the link type may change arbitrarily during a session. Each link type supports up to 16 different packet types. Four of these are control packets and are common for both SCO and ACL links. Both link types use a TDD scheme for full-duplex transmissions. There are three error-correction schemes defined for Bluetooth baseband controllers: • 1/3-rate FEC code; • 2/3-rate FEC code; • Automatic repeat request (ARQ) scheme for data.

The purpose of the FEC scheme on the data payload is to reduce the number of retransmissions. However, in a reasonably error-free environment, FEC creates unnecessary overhead that reduces the throughput. Therefore, the packet definitions have been kept flexible as to whether or not to use FEC in the payload. The packet header is always protected by a onethird rate FEC; it contains valuable link information and should survive bit errors. An unnumbered ARQ scheme is applied in which data transmitted in one slot is directly acknowledged by the recipient in the next slot. For a data transmission to be acknowledged, both the header error check and the cyclic redundancy check must be okay; otherwise a negative acknowledge is returned. The Bluetooth baseband provides user protection and information privacy mechanisms at the physical layer. Authentication and encryption are implemented in the same way in each Bluetooth device, appropriate for the ad hoc nature of the network. Connections may require a one-way, two-way, or no authentication. Authentication is based on a challenge-response algorithm and represents a key component of any Bluetooth system. It allows the user to develop a domain of trust between a personal Bluetooth device, such as allowing only the owners notebook computer to communicate through


43

the owners cellular telephone. Encryption is used to protect the privacy of the connection. Bluetooth uses a stream cipher well suited for a silicon implementation with secret key lengths of 0, 40, or 64 bits. Key management is left to higher-layer software. The goal of Bluetooths security mechanisms is to provide an appropriate level of protection for Bluetooths short-range nature and enable use in a global environment. Users requiring a higher level of protection are encouraged to use stronger security mechanisms available in network transport protocols and application programs. Public WLANs could be the superior solution for providing nextgeneration wireless services to indoor and campus hot spots. Public WLANs can handle large volumes of data at significantly lower costs, offer a migration path to speeds of 100 Mbps and higher, and deliver additional capacity with pinpoint accuracy compared with leading 3G technologies. That is why 3G wireless-network operators need public wireless LANs to serve the most demanding users in the most demanding locations (hot spots). It also explains why funding, coverage, and roaming are huge challenges confronting independent public WLAN operators (including IEEE 802.11b, Bluetooth, IEEE 802.11a, and HiperLAN/2). Bluetooth can provide access in secondary locations, integrated with pay phones, point-of-sale terminals, and ATMs. IEEE 802.11a offers a migration path to speeds of 54 Mbps and higher. 3G wireless networks, WLANS, and Bluetooth are expected to complement each other and create a total wireless access solution. The following examples demonstrate how 3G wireless systems and Bluetooth could work together, providing local intercommunication as well as wide-area connectivity in a wide range of applications [14]. These are not definitive and by no means exhaustive, but aim to show how complementary standards can work together to provide a greater level of service than either could achieve separately. Vending Machines in Shopping Malls

All the automatic vending machines within a confined area can, through a Bluetooth access system, be connected to a central vending machine administration unit that in turn uses a 3G access system to call for maintenance or supplies. Minor problems can be relayed to the mall technician directly through a Bluetooth communicator. Pricing changes can be sent from central administration and locally broadcast to all Bluetooth vending machines. E-Mail Delivery to the PC

Third-generation terminals will be able to handle several channels simultaneously (e.g., voice, fax, and data each requiring different channel characteristics

44


and speeds). With predictions of terminal penetration being very high (every member of the population above the age of 12 in a few years), the PC itself does not have to be a 3G terminal in order to receive e-mails on the move. A Bluetooth-3G terminal can receive e-mail as a data transmission and forward it, via Bluetooth, to the PC (assuming it is within close proximity). When the reception is complete, the PC can notify the user via Bluetooth and a short message to his mobile terminal that he has e-mail, and if an item is urgent, this fact can also be forwarded. This concept allows the 3G terminal to be the local head-end for a variety of applications that are locally interconnected via Bluetooth. The Underground Train

Underground facilities suffer from poor coverage on cellular systems. Many underground rail operators are overcoming this by installing systems designed to provide driver and station staff with a reliable communication network. Systems such as Terrestrial Trunked Radio (TETRA) in Europe provide sufficient spare capacity to carry some passenger traffic too. Carriages equipped with Bluetooth transceivers would provide a gateway between the train TETRA system and the users 3G Bluetooth terminal, and the TETRA system would provide the gateway to the surface public networks. The Bluetooth Headset

The 3G Bluetooth terminal mentioned in the above example does not in fact need to be in the users hand or pocket during most of the noted transactions. The user will have a Bluetooth headset, allowing him to leave the terminal in his briefcase too. This may provide voice control and recognition functionality, removing most of the need for a keyboard or display on the 3G terminal. These suggestions may raise the question as to where the 3G terminal in fact should reside. Much of the functionality delivered by 3G systems will be directed toward a data terminal device such as a PC or palm-top computer, and it may be logical to build the 3G terminal into it.

References [1]

Federal Commuications Commission, http://www.fcc.gov/wtb/uls.

[2]

Linnartz, J. P., Narrowband Land-Mobile Radio Networks, Norwood, MA: Artech House, 1993.


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[3] Winch, R. G., Telecommunication Transmission Systems, New York: McGraw-Hill, 1993, pp. 6276. [4] Wheatly, C., Trading Coverage for Capacity in Cellular Systems: A System Perspective, Microwave Journal, July 1995. [5] Viterbi, A. J., CDMA: Principles of Spread Spectrum Communication, Boston, MA: Addison Wesley, 1995. [6] Zeng, M., et al., Recent Advances in Cellular Wireless Communications, IEEE Communications Magazine, September 1999. [7] TIA/EIA Interim Standard, CDMA2000 High Rate Packet Data Air Interface Specification, TIA/EIA/IS-856. [8] Gardiner, J., and B. West, Personal Communication Systems and Technologies, Norwood, MA: Artech House, 1995. [9] Chotikapong, Y., et al., Evaluation of TCP and Internet Traffic via Low Earth Orbit Satellites, IEEE Personal Communications, June 2001, pp. 2834. [10] Jamalipour, A., The Role of Satellites in Global IT: Trends and Implications, IEEE Personal Communications, June 2001, pp. 511. [11] Commercial Atlas and Marketing Guide 1992, 123rd Edition, Skokie, IL: Rand McNally, 1992, pp. 3839. [12] Gavrilovich, C. D., Broadband Communication on the Highways of Tomorrow, IEEE Communications Magazine, April 2001, pp. 146154. [13] Bolcskei, H., et al., Fixed Broadband Wireless Access: State of the Art, Challenges, and Future Directions, IEEE Communications Magazine, January 2001, pp. 100107. [14] Intercai Mondiale, Ltd., Bluetooth as a 3G Enabler, White paper, 2000.

3 Transmission-Network Principles 3.1 Wireline Side of Wireless Networks 3.1.1

PSTN Interconnect and Telephony Overview

Based on the statistics, most of the voice calls from wireless phones are still directed to regular wireline telephones (POTS), and therefore every wireless network at some point will have to interconnect with the wireline telephone system. Although not very complicated from the technical point of view, interconnect is full of complex regulatory and legal and political issues. This means that every wireless operator should have a small team of people dedicated to and prepared to deal with regulatory and interconnect issues. Interconnect (MSC-PSTN connection) issues are completely separate from and far more complex than the leased-lines issues dealing with the backhaul (radio-base stationbase-station controller [RBS-BSC] connection). In a wireline network, if each locale were limited to three or four telephones, it would make sense to connect each phone to all other phones and find a simple method of selecting the one desired. However, with as many as three or four thousand phones in a locale, such a method is not feasible. It is more appropriate in this instance to connect each phone to some centrally located office and perform switching there. This switching can be a simple manual operation using plugs and sockets or it can be done with electromechanical devices or with electronics. In any case, this central office solution is the one that has been chosen by the telecommunications industry. 47

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As these thousands of telephones are connected to the central office (CO), we have what is called a star configuration; all lines are particular to one and only one station, and all terminate at the CO. These connections are called the local exchange plant, and the telephone company handling this function is called the local exchange carrier (LEC). The connections themselves are often called the local loop or otherwise referred to as the last mile. In more technical terms, the section closest to the customers premises is called the distribution plant and that section closest to the CO, the feeder plant. But when a particular telephone call is not originated and terminated within the particular COs geographic coverage, the question is how do we get to another city, or another state, or even another country? The answer, of course, is to connect these COs to a higher-level CO. We clarify the distinction by applying numbers to these levels of offices: The local office (or end office) is called a Class 5 office. The office to which it connects is called the Class 4 office, and so on, with the top level, the Class 1 office, appearing in only a few places in the country. It is worth noting that the only office that has people as its subscribers is the Class 5 office, while other offices in this hierarchy have lower-level COs as their subscribers. Lines connecting switching offices to switching offices, rather than to subscribers, are called trunks. This section of the telephone infrastructure (the section leading upward from the Class 5 offices) is handled not by the LECs but by the interexchange carriers (IXCs), the long-distance carriers. The entire structure is called the hierarchy of switching systems. The whole network is usually called the PSTN. When AT&T was the only long-distance carrier, any time a telephone number was dialed using an area code, the LEC knew that it must be handed off to AT&T. Today there are a number of long-distance carriers and it is no longer obvious what an LEC is supposed to do with a particular longdistance call. To whom should it be handed off? In political terms this is called equal access, which means that a requesting long-distance carrier could require that the LEC examine the number and hand off the call to the proper long-distance carrier. This handoff was from the CO of the LEC to the point of presence (PoP) of the IXC. This PoP could be in a building adjacent to the telephone companys CO, or it could be in some convenient site in the suburbs where it could serve several of the telephone companys COs. Clearly, as time went on the very pure hierarchy of switching systems was becoming somewhat corrupted; new hierarchies in the long-distance part of the network were being applied on top of the old one. Although it is not pertinent to the topology of this network, it should be recognized that the interconnections between these various COs can


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be twisted copper-pair carrier systems utilizing copper pairs (e.g., T1/E1), microwave, satellites, and certainly fiber optics. However, this hierarchical network is not the only network in todays telephone system. Other types of networks include the following: • Local-area networks (LANs), are limited-distance networks connect-

ing a defined set of terminals. They could connect workstations in an office, offices in a building, or buildings on a campus.

• Wide-area networks (WANs), link metropolitan or local networks

usually over common carrier facilities.

• The IN is a concept that centralizes a significant amount of intelli-

gence rather than installing this intelligence in individual COs.

• The Synchronous Optical Network (SONET in North America,

synchronous digital hierarchy [SDH] in the rest of the world) is a particular set of standards that allows the interworking of products from different vendors. It usually embodies a fiber-optic ring that will permit transmission in both directions. The Internet is a packet network (rather than a circuit-switched network), but it is an overlay network in its nature.

• The common-channel-signaling (out-of-band signaling) network is

especially important; it works closely with the PSTN. In the original PSTN, signaling (e.g., call setup) and voice utilized the same common trunk from the originating switching system to the terminating switching system. This process seized the trunks in all of the switching systems involved. Hence, if the terminating end was busy, all of the trunks were set up unnecessarily. In the mid-1970s the common-channel-signaling network, separate from the voice network, was established; it utilizes the protocol called Signaling System 7 (SS7). With this system a talking path was not assigned until all signaling had been satisfactorily completed.

The PSTN described so far has a star configuration with local loops (usually one loop per subscriber) terminating in a CO. This CO completes connections from one local loop to another local loop, or from one local loop to a trunk that terminates on some other CO. This CO has gone through a number of fundamental technological changes. The step-by-step system, which is still in operation in many countries, utilized what is known as the Strowger switch [1]. The intelligence in the

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system was located in relays mounted on each switch. In the crossbar system, still electromechanical in nature, the intelligence was separated from the actual switch allowing this common control to be used repeatedly to set up and tear down calls and never sit idle. At the onset of electronics, the electromechanical control of the common control system was replaced with electronics, and the network, or matrix, was usually replaced with tiny glass-encapsulated reed switches. Hence, only a part of the switch was electronic. In the next generation, the stored program operation of a digital computer was applied to the switch, although the network remained a complex of reed switches. In the final generation, called a digital switch, the talking path was no longer an electrically continuous circuit. The speech being carried was digitized. It is important to note that this final generation depicted a significant change from the previous generations in that there was no longer an electrical talking path through the switch. It was, in fact, operating in a digital (rather than analog) domain. However, whether the system was analog or digital, one thing must be recognized: there was an actual talking path, a circuit, from the calling party to the called party. This talking path was established at the beginning of a call and held for the duration of a callit is called circuit switching. This system is not very efficient, because a voice source alternates between talk spurts (active) and silent periods. Silent periods constitute over 50% of the transmission time of voice calls in each direction. There is, however, a different kind of connection, and we see it today in a number of applications called packet switching (PS). In a packet-switching system, the information being transmitted (whether it is data or digitized voice) is not sent in real time over a dedicated circuit; rather, it is stored in a nearby computer until a sufficiently sized packet is on hand. Then a computer seizes a channel heading in the general direction of the destination, and that packet of data is transmitted at very high speeds and after that the channel is released; so, except for some necessary supervisory information (destination, error checking codes, etc.) the channel is 100% efficient. When the distant station gets that message, no more than a few milliseconds later, it responds with the necessary handshaking informationagain, by accumulating a packet of data, seizing a channel, and bursting the information out over that channel. It is again 100% efficient. As mentioned earlier, the packet networks in the world (actually overlay networks to the PSTN) are used extensively for data; only recently are we seeing them used for voice and penetrating wireless networks of the next generation (3G).

Transmission-Network Principles 3.1.2

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Traffic Engineering

Traffic engineering, as it applies to traditional voice networks, is determining the number of trunks necessary to carry a required number of voice calls during a period of time. For designers of a voice over X (VoX) network, the goal is to properly size the number of trunks and provision the appropriate amount of bandwidth necessary to carry the amount of trunks determined. The performance level, often called the grade of service (GoS), is a function of load and the capacity provided. Performance declines at an exponential rate as the load increases on a given amount of capacity [2]. There are two different types of connections to be aware of: lines and trunks. Lines allow telephone sets to be connected to telephone switches, such as private branch exchanges (PBXs) and CO switches. Trunks connect switches together. An example of a trunk is a tie line interconnecting PBXs (ignore the use of line in the tie-line statement; its actually a trunk). Companies use switches to act as concentrators because the number of telephone sets required are usually greater than the number of simultaneous calls that need to be made. For example, a company may have 600 telephone sets connected to a PBX, but may only have 15 trunks connecting the PBX to the CO switch. In telecommunications, an Erlang is a nondimensional unit with a value between 0 and 1 which indicates how busy a telephone facility is over a period of time (usually one hour). Agner Krarup Erlang (18781929) was a Danish mathematician who invented the formula commonly used to forecast telecommunications traffic. Number 1 applied to a particular telephone circuit would indicate busy 100% of the time. Erlang B is a calculation for any one of these three factors, if you know or can predict the other two: • Busy-hour traffic (BHT), or the number of hours of call traffic dur-

ing the busiest hour of operation; • Blocking, or the percentage of calls that are blocked because not enough lines are available; • Lines, or the number of lines in a trunk group. By monitoring daily and weekly variations in traffic intensity, the BHT may be determined. This is a continuous one-hour period during which traffic in a part of the network is at its most intensive. This may occur at different times of the day, depending on which category of subscribers is dominant. The BHT peak is a mean value, measured over several days. The

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aim is to calculate for minimal congestion, while obtaining full utilization of the network. In order to provide a decent level of service, it is necessary to base traffic engineering on a GoS during the peak or busy hour. GoS is a unit of measurement of the chance that a call will be blocked. For example, a GoS of P(0.01) means that one call will be blocked in 100 call attempts, and a GoS of P(0.001) results in one blocked call per 1,000 attempts. It is important to look at call attempts during the days busiest hour. The most accurate method of finding the busiest hour is to take the busiest days in a year, sum the traffic on an hourly basis, find the busiest hour, then derive the average amount of time. In North America it is common to use the 10 busiest days of the year to find the busiest hour. Regardless of which method is used, the intent is to use a number that is sufficiently large in order to provide a GoS for busy conditions and not the average hour traffic. An exchange capacity is not just expressed as the number of possible simultaneous calls. Variations in the holding time of a call (call setup time + conversation time + disconnection time) impact the dimensioning of the exchange, as it may be that the busy hour for switching the calls is not the same as the busy hour for the control equipment. An extended version of Erlang B allows you to determine the number of people who, when blocked, retry their calls immediately. Most of the common models of capacity assume that calls are either served immediately or are blocked and overflow. In some applications, blocked calls can be delayed and served later, which leads us to Erlang C distribution, where capacity estimation is a function of delay criteria instead of blocking. Although the Erlang C traffic model was initially developed for telecommunications applications, it is applicable to any queuing system that meets the following criteria: • Call arrivals are random (Poisson arrivals). • Calls that do not find an idle server enter the queue. • Queue discipline is first come, first served (the queue discipline does

not affect the average call delay, but only the waiting time distribution, meaning the probability of a callers not being served immediately and having to wait).

• The queue length is infinite. • Service channels form a full-availability group with exponentially

distributed service times.


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Other performance parameters developed in conjunction with the Erlang C formula include the probability of a call having to wait longer than a specified period, the average delay on all calls and on delayed calls, and queue size measurements. Erlang C is used for special applications but rarely is used in either wireless or wireline telecommunications network design. The Erlang B and C traffic models make certain assumptions about the nature of the call arrivals. One of them is the assumption that call arrivals are random (Poisson arrivals), and although this is quite reasonable in most applications, it can cause inaccurate results when there is a sudden peak of calls. In many ways, packet-switching theory is comparable to circuitswitching theory, in that the principles of blocking, delay, and alternate routing apply as well as the performance estimation using the same mathematical principles. The major and significant technical difference between packet and circuit switching is the process of assigning the bandwidth. In circuit switching, bandwidth is preassigned per user for the duration of the call, while packet switching allocates total bandwidth per user on a dynamic basis (dynamic bandwidth allocation). The tremendous increase in traffic capacity per circuit through use of packet switching caused a reanalysis of the voice network concepts in combined voice and data networks. New models and simulation methods are currently being developed and tested in order to be able to design and dimension networks capable of carrying voice and data traffic at the same time. For example, in a small wireless network, a simplified calculation for the PSTN traffic (number of required T1 lines between base-station controller [BSC] to PSTN) would look like the following: Assumptions • Small town with a population of 100,000; • 50 milliErlangs (mE) per subscriber; • 5% market penetration first year; • 0.1% GoS; • Private self-contained network80% of traffic is PSTN (no IMTs,

handoffs).

Calculations

First, we determine the number of potential subscribers: • 100,000 [population] × 0.05 [market penetration] = 5,000 subscribers;

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Then we determine the total traffic expressed in Erlangs: • 5,000 × 0.05 [Erlangs per sub] × 0.8 [PSTN traffic] = 200 [Erlangs

traffic];

From Erlang B tables, we then determine the number of T1/E1 circuits: • From Erlang B table with 0.1% GoS = 238 voice channels (DS0s); • 238 voice channels/24 DS0s per T1 = 10 T1s (238/30 = 8 E1s).

This is a coarse estimate of the number of T1s or E1s the wireless operator will have to order from the local telephone company (PSTN) to connect to the BSC. We assume here that 80% of wireless subscribers will call outside of the network and their calls will be routed to the PSTN. Typically only a very small percentage of calls are mobile-to-mobile calls within the same network. Most practical voice modeling at the private or public network level for voice assumes a nonstationary Poisson process dependent on time of day and day of week. As mentioned earlier, Erlang B is a very popular formula for trunk sizing. For certain sizing in COs, Erlang-Engset is used, since the number of subscribers is limited. 3.1.3

SS7 and AIN

In traditional telephony, the basic call control procedure is divided into three phases: call setup, the data-conversation phase, and call teardown. Messages on the signaling link are used to establish and terminate the different phases of a call. Standard in-band supervisory tones and recorded announcements are returned to the caller on appropriate connection types to provide information on call progress. Until 1976, the signaling in Bell Systems was almost entirely on a per-line or per-trunk basis and the signaling information for a particular channel was carried on the same channel as the voice or other message information. This type of signaling had many disadvantages, such as long setup times, inefficient trunk usage, restricted signaling periods, and security problems. Common Channel Interoffice Signaling (CCIS) was introduced by AT&T in 1976 to reroute signaling information over a separate path from the voice path. CCIS was an analog system that provided capabilities to support automated calling card and sophisticated routing services, such as 800 service free to the caller.


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The signaling protocol in all modern circuit-switched networks is SS7, which is an internationally standardized, general-purpose CCS system. This protocol enables the exchange of messages through the network for routing, resource reservation, call admission, address translation, call establishment, call management, and billing. Signaling has a special meaning in the telecommunications world: It refers to the information needed to set up, route, monitor, and terminate a call across either a physical or virtual circuit. SS7 network call handling is normally described as being separate from the actual voice connection. This is only partially accurate. SS7 is a separate network, but the call handling is still related to and tied to a specific call reference in the voice path [3]. SS7 is the global standard for telecommunications but it has the Comite Consulatif International de Telegraphique et Telephonique (CCITT) No. 7 and American National Standards Institute (ANSI) No. 7 version. This standard covers procedures and protocols used by network elements in the PSTN to exchange data packets over a digital network. CCITT began specifying the 2G common-channel signaling system in the mid-1970s. More recently, additions have been issued as individual ITU recommendations. The purpose of exchanging data is to enable wireless and wireline call setup, routing, and control, as well as voice communications. The ITU definition is an umbrella standard that allows for variants such as the ANSI and Telcordia (formerly Bellcore) standards used in North America, and the European Telecommunications Standard Institute (ETSI) standard used in Europe. SS7 performs a number of functions: • The fundamental operations of an ordinary phone call: call setup,

management, and call teardown;

• Toll and toll-free wireline services; • Advanced services such as call forwarding, caller ID, and multiparty

calls;

• Wireless services including wireless roaming; • Subscriber authentication; • Local number portability (LNP).

The introduction of stored program-controlled (computerized) switching systems starting in the 1960s made it possible to go beyond direct dialing

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of calls to offer customized telecommunications services to meet users needs. The earliest customized services to be offered included customercontrolled call forwarding, three-way calling, and network-based speed dialing. Recently these have been supplemented by services that depend upon the calling partys number, such as caller identification, automatic recalling of the last call received, and call blocking. These services depend upon the combination of low-cost memory (storage of information) and processing power that is enabled by state-of-the-art electronic technologies. However, these types of services have traditionally been implemented by making changes and additions to the large software programs that run the switching systems. Making changes to these large mainframe-like systems is very costly and time consuming. Furthermore, since switches are purchased from different suppliers and come in a multiplicity of types, implementing new services has traditionally required the development and deployment of new generic switching software by multiple suppliers for multiple switch types. This costly and time-consuming process is not consistent with the rapid deployment of a wide range of new telecommunications services that are customized to meet users needs. Thus, the LECs have implemented the advanced intelligent network (AIN) as a client-server approach to creating new services. In this approach, the switches act as clients that interface with software-based functionality in server nodes called service control points (SCPs), service nodes, and intelligent peripherals. The switches, service nodes, and intelligent peripherals implement building-block capabilities that can be mixed and matched by the SCPs to create new services for network users. Because all switches implement comparable building-block capabilities, new services can be created and deployed quickly by implementing new functionality in the server nodes. SS7 separates voice channels from signaling. SS7 data packets go over 56Kbps (mostly in the United States) or 64-Kbps (in most other countries) bidirectional channels called signaling links. Signaling occurs on dedicated channels that are different from the voice/data channels. This provides faster call-setup times and more efficient use of voice circuits. Out-of-band signaling is also needed to support IN services. SS7 has three kinds of switches or signaling points: 1. Service switching point (SSP); 2. SCP; 3. Signal transfer point (STP).


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SSPs originate, terminate, or tandem calls and send signaling messages to other SSPs to set up, manage, and release voice circuits. They may also send a query message to an SCP, which is a centralized database, to figure out how to route a call such as a toll-free call. Network traffic between signaling points may be routed via a packet switch called STP. The hardware and software functions of the SS7 protocol stack are divided into levels connected loosely to the Open Systems Interconnect (OSI) seven-layer model defined by the International Standards Organization (ISO). The message transfer part (MTP) of the SS7 protocol stack is divided into three levels. The lowest level, MTP Level 1, is equivalent to the OSI physical layer. MTP Level 2 ensures accurate point-to-point transmission of a message across a signaling link and is equivalent to the OSI data link layer. MTP Level 3 provides message routing between signaling points in the SS7 network and is equivalent to the OSI network layer. The ISDN user part (ISUP) defines the protocol used to set up, manage, and release trunk circuits that carry voice and data between calling parties. ISUP is used for both ISDN and non-ISDN calls. In some parts of the world such as China and Brazil, the telephone user part (TUP) affects basic call setup and teardown. TUP handles analog circuits. The transaction capabilities applications part (TCAP) supports information exchange between signaling points using the signaling connection control part (SCCP) connectionless service for IN. An SSP uses TCAP to query an SCP to determine the routing number(s) associated with a dialed toll-free number. The SCP uses TCAP to return a response containing the routing number(s) back to the SSP. Calling-card calls are also validated using TCAP. When a mobile subscriber roams into an MSC area, the integrated visitor location register requests service profile information from the subscribers home location register (HLR) using mobile application part (MAP) information carried within TCAP messages. SCCP provides connectionless and connection-oriented network services above MTP Level 3. SCCP allows messages to be addressed to specific applications (called subsystems). SCCP is used as the transmission layer for TCAP-based services, such as toll-free calls, calling card, local number portability, wireless roaming, and PCS. While wireless provides the physical access mechanism for the telephone and other appliances, the AIN provides the software-based functionality to people on the move. Home- and visitor-location registers (AIN service control points) keep track of where nomadic users are and provide the information required to direct incoming calls to those users. AIN can screen or block

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incoming calls according to the calling number, time of day, day of week, or other parameters specified by the called party. AIN functionality allows multitier telephones to access cordless base stations, high-power vehicular cellular base stations, low-power pedestrian cellular base stations, and LEO satellite systems, depending on which is most economical and available at any given time. As wireless telephony transitions toward nomadic multimedia computing and communications, the advanced intelligent network will provide access control (security-related) mechanisms, interworking functionality, screening, customized routing, media conversion, and other middleware functionality to support people on the move. 3.1.4

Telecommunications Act of 1996

Prior to the 1996 Telecommunications Act, the service offerings of the Bell operating companies (BOCs) were governed by a modified final judgment (MFJ), or consent decree. The MFJ prohibited the BOCs from entering certain lines of business, including interexchange service. These line-of-business restrictions were based upon the theory that, if the BOCs were allowed to enter the long-distance market, they could use their bottleneck control in the local and exchange access markets to obtain an unfair advantage in the longdistance market. The MFJ also required the BOCs to provide exchange access services, which are necessary to originate or terminate an interexchange service, that are equal in type, quality, and price among the interexchange carriers. Congress overhauled many aspects of federal regulation of communications services with the passage of the Telecommunications Act of 1996. In particular, Congress chose to modify the line-of-business restrictions established under the MFJ. In enacting the Telecommunications Act of 1996, Congress established a pro competitive and deregulatory framework designed to benefit Americans by opening telecommunications markets to competition. As a result, the 1996 Act set the stage for a new competitive regime in which carriers in previously segmented markets would be able to compete in a dynamic and integrated telecommunications market that promises lower prices and more innovative services to customers. Central to the new statutory scheme, and expressly departing from prior jurisprudence developed under the MFJ, are provisions designed to open the local services market to competition. This will lead ultimately to permit all carriers, including those that had previously enjoyed a monopoly or competitive advantage in a particular market, to provide a combined telecommunications offering that includes both local and long-distance services.


59

3.2 Digital Transmission Technology 3.2.1

About the Transmission

The terms transmission and transport are used interchangeably in this text, as both are currently in use, the former being preferred in Europe and the latter in North America. In the telecommunications networks today, digital transmission is being used almost exclusively. A high-pitched voice mostly contains high frequencies while low-pitched voice contains low frequencies. A loud voice contains a high-amplitude signal while soft voice contains a lowamplitude signal. Analog signals can be combined (i.e., multiplexed) by combining them with a carrier frequency. When there is more than one channel, this is called FDM. FDM was used extensively in the past but now has generally been replaced with the digital equivalent called time-division multiplexing (TDM). The most popular TDM system is known as Tier 1 (T1). In a T1 system, an analog voice channel is sampled 8,000 times per second, and each sample is encoded into a 7-bit byte. Twenty-four such channels are mixed on these two copper pairs and transmitted at a bit rate of 1.544 Mbps. T1 in North America (E1 in the rest of the world) remains an important method of transmitting voice and data in the PSTN. E1 has 30 channels with a bit rate of 2.048 Mbps. Such a digital transmission scheme (and certainly there are modifications of it that improve efficiency, capacity, and quality) works very well with the digital-switching schemes discussed previously. Thus a talking path (i.e., a switched circuit) in the PSTN can be either analog or digital or a combination thereof. In fact, a digital signal can be transmitted over a packet-switched network as easily as in a circuit-switched network. Digitized voice is little different from data, and, therefore, if data can be transmitted over a packet network, then so can digitized voice. One of the most common applications is now known as voice over the Internet. The challenge, of course, is to get the transmitted signal to the destination fast enough (delay-related issues) because the conversation is time-sensitive. A second challenge is to get each packet, which is a small piece of a voice conversation, to the destination in the proper order. 3.2.2

Transmission MediaPhysical Layer

There are four types of media that can be used in transmitting information in the telecommunications world: 1. Copper lines (twisted pair and coaxial cables): for low- and mediumcapacity transmission over a short distance;

60


2. Fiber-optic transmission: for medium- and high-capacity transmission over any distance; 3. Wireless: for low- (mobile radio) and medium-capacity (microwave point-to-point) transmission over short and medium distances; 4. Satellite: for low- and medium-capacity transmission over large distances. Years ago, copper wire was the only means of transporting information. Technically known as unshielded twisted pair (UTP), it consists of a large number of pairs of copper wire of varying size in a cable. The cable did not have a shield and therefore the signal (primarily the high-frequency part of the signal) was able to leak out. Also, the twisting on the copper pair was very casual, designed as much to identify which wires belonged to a pair as to handle transmission problems. Even with these limitations, it was quite satisfactory for use in voice communications. Coaxial cable, much larger in dimensions than twisted pair, has been almost exclusively used in broadband systems for video transmission. Coaxial cable consists of a single strand of copper running down the axis of the cable. This strand is separated from the outer shielding by an insulator made of foam or other dielectrics. Covering the cable is a conductive shield. Because of the construction of the cable, obviously coaxial in nature, very high frequencies can be carried without leaking out. In fact, dozens of TV channels, each 6 MHz wide, can be carried on a single cable. Fiber optic is the third transmission medium, and it is unquestionably the transmission medium of choice today. Whereas transmission over copper utilizes frequencies in the megahertz range, transmission over fiber utilizes frequencies a million times higher. This is another way of saying that the predominant difference between electromagnetic waves and light waves is the frequency. Transmission speeds of as high as 10 Gbps have become commonplace in the industry today. At this speed, the entire 15-volume set of the Encyclopedia Britannica could be transmitted in well under one second. Of course, laying fiber, on a per-mile basis, still costs somewhat more than laying copper. However, on a per-circuit basis there is no doubt that fiber is more cost effective. Fiber comes in several forms. The two predominant ones are multimode and single-mode. The total strand diameter for both is about 125 microns (a millionth of a meter). However, the ultrapure glass that forms the core transmission medium is between 50 and 62.5 microns for the multimode fiber and about 8 to 10 microns for the single-mode fiber. It might


61

appear that the multimode fiber would have a greater carrying capacity; however, just the opposite is true. With single-mode fiber, only one ray or mode can travel down the strand, making it easier to regenerate the signal at points along the span. In fact, single-mode fiber makes up the majority of todays long-distance network. The tremendous capacity of fiber certainly makes for more efficient communications; however, placing so much traffic on a single strand, for point-to-point communications, makes for greater vulnerability. Most of the disruptions in the long-distance network are a result of physical interruption of a fiber run. Called backhoe fade, the ring configuration is a solution used most often today. Wireless communication is the final option as a transmission medium. This can take several forms: including microwave (point-to-point or PMP), synchronous satellites, LEO satellites, cellular service, and PCS. In every case, however, a wireless system obviates the need for a complex wired infrastructure. In the case of synchronous satellites, transmission can take place across oceans or deserts. With microwave point-to-point systems there is no need to plant cable, and in mountainous territories (as well as downtown areas) this is a significant advantage. 3.2.3

Transmission (Backhaul) in Wireless Networks

The rapid growth in the number and diversity of WSPs cellsites has a corresponding influence on the size and complexity of their transmission (and backhaul) network. In wireless networks, backhaul is defined as the portion of the network that carries the wireless calls from cell site back to the BSC. It is then routed on to the appropriate service termination points, such as MSC and PSTN (voice) or PCN and the Internet (data). Once considered a second-tier priority in relation to RF and cell-site deployment issues, WSPs today view their backhaul and transmission networks as strategic assets since this portion of a WSPs network can run as high as 20% of an annual expense budget. The following lists some of the technological developments that are driving the transmission-network design and deployment field: • Innovations in optoelectronics leading to a near exponential expan-

sion in the amount of data that can be carried by fiber-optic cables, now reaching terabit per second speeds;

• The development of new digital wireless systems such as GSM,

CDMA, and UMTS, enabling the radio spectrum to carry far more

62


communications traffic at very high quality and deliver mobile Web access and video services; • The widespread adoption of extremely flexible packet-based com-

munications techniques, such as IP, that will eventually replace the large and expensive telecom switches and exchanges currently in use;

• The evolution of technologies and infrastructuresISDN, xDSL,

and CATVthat will give ordinary subscribers access to very high bandwidth services for combined video, voice, and data applications in the home;

• The introduction of intelligence into the network, allowing new

services to be developed for the needs of individual businesses or personal use and the Internet to interact with the public telephone network;

• The increasing familiarity among all types of users with the use of

the Internet and World Wide Web as a ubiquitous communications interface for information searches, carrying out financial transactions and purchases, and accessing entertainment and new media;

• The development of digital media technologies that allow music and

video to be transmitted over relatively low-speed communications links;

• The possibilities of unified messaging applications that can translate

voice, e-mail, and fax messages into the most appropriate medium for receipt by a mobile user.

3.2.4

DSX-1 Digital Interfaces (North America)

DSX-1 is a physical interface of the T1 circuit. Figure 3.1 illustrates some of the most commonly used pieces of equipment in the transmission networks. The transmultiplexer is a device for the conversion between the analog and digital worlds in transmission networks, and while rarely used today was quite common in the early 1980s during the beginning of digitalization in telecommunications. There are two standards for first-order digital transmission systems. The T1 system, developed by Bell Laboratories, is used mainly in the United States, Canada, Taiwan, Jamaica, and few other countries in the world [4]. Most of the countries around the world use the E1 system defined by European Conference of Postal and Telecommunications (CEPT) Administration.


63

Digital switch DS0s 1

Digital MUX*

DS1

Digital MW radio or fiber-optics

DS1 DSX-1 DS1 Cross-connect Two 12-channel Channel 655' (digital MDF) DS1 Trans groups bank MAX (60108 kHz) Analog MUX* (FDM) MUX* (TDM-FDM) 24 PCM channels

24

Voice and 1.544 Mbps data circuits

DS1

T1 office repeater

Span line

TELCO Transmultiplexer converts analog voice channels into digital bit stream without demultiplexing

*Multiplexer

Figure 3.1 DSX-1 digital interfaces (North America).

3.2.5

North American Digital Hierarchy

Details of the T1 systems will be discussed in other chapters of this book. Here we give only a brief description and definition of what comprises T1. • There are 24 DS0 (64 Kbps) timeslots in a T1 line, providing a total

bandwidth of 24 × 64 Kbps = 1,536,000 bps (1,536 Kbps).

• Another 8,000 bps are used as the equivalent of lane markers to

allow traffic to remain in its assigned lanethese overhead bits are known as framing bits.

• Adding up the total bandwidth of the 24 DS0 channels and the 8

Kbps framing bits yields the 1,544-Kbps T1 data rate.

The North American multiplexing scheme is shown in Figure 3.2; the North American T1 digital hierarchy is shown in Figure 3.3. For the hierarchies based on the 1.544-Mbps primary rate, the principle has been established that some bits in the frame should be reserved, in particular to perform quality control of the digital paths when several digital

64

M12Rarely used today M13Most common type of multiplexer today 1 2. ..

PCM channel bank

1.544 Mbps (24 Channels)

24 DS1

DS3 1 2

M12


3 4 DS2

28 1.544-Mbps T1 trunks

1 2. .. 28

M13

1 2 3 4 5 6 7



M23 DS3

1 2 3

The North American hierarchy is standard in the United States, Canada, Taiwan, Korea (except in some new cellular links), and Japan (through DS2). Other countries use the European CEPT 30-channel hierarchy.

Figure 3.2 North American pulse-code modulation hierarchy.

3 DS3s (84 DS1s)

135 Mbps (2016 Channels)


Voice and data circuits


Designation DS0 DS1 DS2 DS3

# DS1 signals 24/DS1 01 04 28

Bit-rate (Kbps) 64 1,544 6,312 44,736

Line code AMI AMI/B8ZS B6ZS B3ZS

65

Voice channel equivalent 1 96 24 672

Line lengths (Ft/m) 655/200 1,000/300 450/140

The AMI, B8ZS, B6ZS, and B3ZS codes are bipolar. Line codingused to ensure that enough Cable types: 100/110 Ω twisted-pair, 75 Ω coax timing information accompanies the digital Ref: ITUT G.703, G.704; Bellcore TRTSY-000499 signal to allow accurate recovery of all the bits

Figure 3.3 North American T1 digital hierarchy.

sections in tandem are involved [5]. A 64-Kbps circuit (DS0) that uses 8 Kbps for signaling is called Switched 56, Digital Data Service (DDS) or Advanced Digital Network (AND). Each carrier used to have its own name for this service. Today, this 56-Kbps service is becoming obsolete. Some important facts to remember about T1 are as follows: • Most T1 circuits today use extended super-frame (ESF) format and

B8ZS line code.

• ESF and B8ZS provide clear channel with only 2 Kbps used for

framing + 2 Kbps facility data link (FDL) + 2 Kbps (CRC-6) = 6 Kbps.

• Older T1s use 8 Kbps for framing and bit robbing for provisioning

and maintenance.

3.2.6

CEPT Digital Hierarchy

E1 is described in more detail in the following chapters. Only a short description and E1 definition are given here; more information is available in [6] and in other chapters of this book: • There are 30 DS0 (64 Kbps) timeslots in an E1 line, providing a

total bandwidth of 30 × 64 Kbps = 1,920,000 bps (1.92 Kbps).

• Sixty-four thousand bps are used as the equivalent of lane markers to

allow traffic to remain in its assigned lane.

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• One 64-Kbps timeslot (TS0) is used as framing bits. • Another 64-Kbps timeslot (TS16) is used for signaling of voice fre-

quency channels.

• Adding up the total bandwidth of the 30 DS0 channels, framing,

and signaling bits yields the 2,048-Kbps E1 data rate.

The CEPT multiplexing scheme is shown in Figure 3.4. In the case of networks using a 2.048-Mbps-based hierarchy (Figure 3.5), there is in principle no basic restriction on the use of full capacity of the digital path. However, it is recognized that compatibility with recommended frame structures at the various levels of the 2-Mbps hierarchy (e.g., the use of the same frame alignment pattern) could be a preferred solution since it offers the following advantages: • Use of the same framing devices for switched and nonswitched

applications;

• End-to-end quality control performed in a unique way by the net-

work when the maintenance entity that terminates the service (e.g., the encoding device) does not belong to the network;

• The possibility of performing additional network management

functions that could be required, depending on the applications.

3.3 Plesiochronous Versus Synchronous Digital Hierarchy Traditionally, transmission systems have been asynchronous, with each terminal in the network running on its own clock. In digital systems, clocking (timing) is one of the most important considerations. Timing means using a series of repetitive pulses to keep the bit rate of the data stream constant and to indicate where the ones and zeros are located in a data stream. Since these clocks are free running and not synchronized, large variations occur in the clock rate and thus in the signal bit rate. Asynchronous multiplexing uses multiple stages; lower-rate signals are multiplexed and extra bits are added (bit stuffing) to account for the variations of each individual stream and combined with other bits (framing bits) to form a higher-level bit rate. Then bit stuffing is used again to produce even higher bit rates. At the higher asynchronous rate, these signals cannot be accessed without a complete demultiplexing procedure.

PCM 1 channel 2. .. bank 30 1st order

2.048 Mbps (30 channels) 1 M2-8 2 2nd order 3 4

E3 8.448 Mbps (120 channels) 1 2 M8-34 3 3rd order 4

E4

E2 E3 2.048 Mbps 2.048 Mbps 2.048 Mbps

1 2. M2-34 .. 16

34 Mbps (480 channels)

1 2 3 4

M34-140 MUX*


E1

34.368 Mbps (480 channels)

Skip (double-step) MUX* The CEPT hierarchy is the international standard everywhere except North America (USA, Canada), Taiwan, Korea, and Japan.

67

Figure 3.4 CEPT pulse-code modulation hierarchy.

*Multiplexer

68


Designation E0 E1 E2 E3 E4

# E1 signals

Bit rate (Kbps)

30/E1 1 4 16 64/63 *

64 2,048 8,448 34,368 139,264

Line code

Voice channel equivalent

AMI HDB3 HDB3 HDB3 CMI

1 30 120 480 1920/1890 *

The AMI, AMI, HDB3, HDB3, & CMI andcodes CMI codes are bipolar. are bipolar. Ω Twisted Cable types: 120 twisted-pair, 75 ΩCoax coax 120W Pair, 75W (Length/type assigned for 6 dB maximum 6-dB maximum loss) loss) Ref: Ref: ITU-T ITU-T G.703, G.703, G.704 G.704

E1 400mtwisted-pair 750mcoax E3 375mcoax

* 63 E1 (1,890 channels) are mapped in SDH systems

Figure 3.5 CEPT E1 digital hierarchy.

In a plesiochronous digital hierarchy (PDH) system, the average frequency of all clocks in the system will be the same (synchronous) or nearly the same (plesiochronous) and every clock can be traced back to a highly stable reference supply. The prefix plesio, which is of Greek origin, means almost equal, but not exactly, meaning that the higher levels in the CCITT (ITU today) hierarchy are not an exact multiple of the lower level. The North American digital hierarchy starts off with a basic digital signal level of 64 Kbps (DS0) (see Table 3.1). Thereafter, all facility types are

Table 3.1 North American Data Rates Name

Rate (Kbps)

DS0

64

DS1

1,544

DS1C

3,152

DS2

6,312

DS3

44,736

DS4

274,176


69

usually referred to as Tx, where x is the digital signal level within the hierarchy (e.g., T1 refers to the DS1 rate of 1.544 Mbps). Up to the DS3 rate, these signals are usually delivered from the provider on twisted-pair or coaxial cables. North American T1 service providers often refer to the signal interfaces between the user and the network as DS1 signals. In the case of user-to-user interfaces, the term DSX-1 is used to describe those DS1 signals at the crossconnect point. The CCITT digital hierarchys basic level is the DS0 rate of 64 Kbps (see Table 3.2). These signals are usually delivered from the provider on twisted-pair or coaxial cables. All E-carriers and T-carriers are based on PDH. In the late 1980s, synchronous network hierarchies were introduced. The term synchronous means occurring at regular intervals, and is usually used to describe communications in which data can be transmitted in a steady stream rather than intermittently. For example, if a telephone conversation was synchronous, each party would be required to wait a specified interval before speaking. The SONET/SDH is a newer technology in the field of digital transmission. The transmission is carried out in a synchronous mode, hence the name synchronous digital hierarchy. The most important advantage in adopting this synchronous technology is to enable the mapping of various user bit rates directly on to the main transmission signals, thus bypassing various stages of multiplexing and demultiplexing as was done in the case of earlier PDH technology. The number of new architectures and topologies made possible as a result of this new technology has rendered it the only viable alternative for adoption today. For instance, add-drop multiplexers (ADMs) have made it possible to use SONET/SDH terminals in a long chain with bit streams Table 3.2 ITU Data Rates Name DS0 E1

Rate (Kbps) 64 2,048

E2

8,448

E3

34,368

E4

139,264

70


added or dropped along the way in an effective manner. By closing the chain at two ends, the ring configuration is possible; this provides enhanced protection features. Other advantages include support of the network management system, easy upgrade to high bit rates, and adaptability to the existing PDH. In synchronous networks, all multiplex functions operate using clocks derived from a common source. The North American SONET system is based upon multiples of a fundamental rate of 51.840 Mbps, called Synchronous Transmission Signal, Level 1 (STS-1). The facility designators are similar, but indicate the facility type, which is usually fiber-optic cable (e.g., OC-1 is an optical carrier supporting an STS-1 signal, while OC-3 supports an STS-3 signal, and so on). Some typical rates are listed in Table 3.3. The international SDH system is based upon a fundamental rate of 155.520 Mbps, three times that of the SONET system. This fundamental signal is called Synchronous Transmission Module, Level 1 (STM-1). The typical transmission medium is defined to be fiber, but the broadband ISDN specification does define a user-network interface (UNI) STM-1 (155.520 Mbps) operating over coaxial cables. Some typical rates within this hierarchy are shown in Table 3.4. The evolution from PDH to SDH worldwide is a phased process. One of the more common problems faced at this intermediate stage is that SDHbased networks must have the flexibility to utilize existing PDH transport media.

3.4 Multiplexing and Inverse Multiplexing Multiplexing is a process in which multiple data channels are combined into a single data or physical channel at the source. Multiplexing can be Table 3.3 STS Data Rates Name

Rate (Kbps)

STS-1

51,840

STS-3

155,520

STS-9

466,560

STS-12

622,080

STS-48

2,488,320


71

Table 3.4 STM Data Rates Name

Rate (Kbps)

STM-1

155,520

STM-3

466,560

STM-4

622,080

STM-16 2,488,320

implemented at any of the OSI layers. Conversely, demultiplexing is the process of separating multiplexed data channels at the destination. One example of multiplexing is when data from multiple applications is multiplexed into a single lower-layer data packet. Another example of multiplexing is when data from multiple devices is combined into a single physical channel (using a device called a multiplexer). Some methods used for multiplexing data are TDM, asynchronous timedivision multiplexing (ATDM), FDM, and statistical multiplexing. FDM involves splitting the frequency band transmitted by the channel into narrower bands. Each of these narrow bands is used to constitute a distinct channel. In FDM, information from each data channel is allocated bandwidth based on the signal frequency of the traffic. Multiple channels are combined onto a single aggregate signal for transmission. The channels are separated in the aggregate by their frequency. There are always some unused frequency spaces between channels, known as guard bands. Guard bands reduce the effects of bleedover between adjacent channels, a condition more commonly referred to as cross-talk. The other method is to allot a common channel to several different information channels one at a time (TDM). FDM was the first multiplexing scheme to be widely used, and such systems are still in use today. However, TDM is the preferred approach today. In TDM, information from each data channel is allocated bandwidth based on preassigned timeslots, regardless of whether there is data to transmit. In ATDM, information from data channels is allocated bandwidth as needed, by using dynamically assigned timeslots. In statistical multiplexing, bandwidth is dynamically allocated to any data channels that have information to transmit. Time-assignment speech-interpolation (TASI) systems represent an example of an analog statistical TDM scheme. These systems enjoyed limited use in the 1980s, and were particularly adept at sharing voice circuits; specifically PBX trunks. A TASI multiplexer is interconnected between the PBX and the trunk facilities. Usually, one analog trunk circuit is used for signaling

72


purposes between TASI units at each end of the link. The remaining voice trunks support analog TASI TDM voice conversations. In normal telephone conversations, the majority of time is spent in a latent (idle) state. TASI trunks will allocate chunks of voice from another channel during this idle time. If individuals were to monitor these TASI trunks, they would hear bits and pieces of various conversations. The signaling channel is used for the signaling conversion between end-point PBX units and also for the allocation of bandwidth once incoming speech energy has been detected. As digital speech processing became more common, TASI systems were created that had analog inputs and digital outputs. This type of multiplexing technique is more commonly known as digital speech interpolation (DSI). There are a few drawbacks in using TASI and DSI systems. First, users can notice a good deal of voice clipping which occurs when a little bit of speech is lost while waiting for the TASI multiplexer to detect valid speech and allocate bandwidth (more about multiplexing can be found in [7]). Clipping also occurs when there is no bandwidth present at the moment. TASI and DSI units are also very susceptible to audio input levels and may have problems with the transport of voiceband data (e.g., VF modem) signals.

3.4.1

Statistical Multiplexing

Any data communications system that has more than one asynchronous line going between common locations can benefit by installing a pair of statistical multiplexers (STATMUX). A STATMUX performs the function of combining several asynchronous data communication channels into one composite synchronous signal that is transmitted between two locations more inexpensively than the cost of individual lines. Individual users connect to asynchronous channels, and the composite communication line between the two locations is called the link. The STATMUX utilizes a different form of TDM. These multiplexers typically use a high-level data link control (HDLC)like frame for aggregate communications between units. As input-output (I/O) traffic arrives at the multiplexer (MUX), it is buffered, then inserted into the I-field of the HDLC frame. The receiving units remove the I/O traffic from the aggregate HDLC frame. Statistical multiplexers are ideally suited for the transport of asynchronous I/O data, as they can take advantage of the inherent latency in asynchronous communications. They are typically faster at transporting I/O data end to end than X.25 systems, but some of these multiplexers can also perform


73

network switching functions between I/O ports. The total I/O bandwidth can (and usually does) exceed the aggregate port bandwidth (multiplexing gain). Later, many of these multiplexers incorporated switching mechanisms that allowed I/O ports to connect themselves intelligently to other destination ports upon user command. While functioning somewhat as an X.25 switch, these statistical multiplexers were usually faster, and provided more transparent I/O data-carrying capacity. Frame relay and X.25 systems are also categorized as statistical TDMs. Both systems utilize aggregate HDLC frame structures. The advantage of frame relay over X.25 is that it can support the same traffic as X.25, while facilitating bandwidth-on-demand requirements for bursty traffic. Frame relay, however, cannot adequately support voice or video traffic because of variable end-to-end delivery times (e.g., variable delay). Voice and video transmissions are of a constant-bit-rate (CBR) nature, and are not happy sitting in a queue waiting for a big LAN packet to finish transmitting.

3.4.2

3/1 Multiplexing and Subrate Multiplexing

An M13 multiplexer is usually a modular, compact unit for multiplexing up to 28 DS1s into a DS3. Compatible with North American standard interfaces, these multiplexers are capable of multiplexing up to seven low-speed DS1 signal groups (each with 4 DS1s) into one DS3 (44.736 Mbps) signal, with each low-speed signal group consisting of four DS1s. Standard features include continuous performance monitoring and extensive local and remote diagnostics, which provide off-premises restoration, thereby avoiding the cost of on-premises maintenance. Remote monitoring and control are the key features, providing remote provisioning, remote inventory, performance monitoring, and remote testing. The FMT150 is a fiber-optic multiplexer that multiplexes and transports one, two, or three DS3 signals, up to 84 DS1s, and a maximum of 2,016 voice circuits. Designed to interface with DS1 or DS3 input signals, the typical FMT150 uses 150-Mbps fiber transport plug-ins (with up to three DS3 inputs). FMT150 usually features a comprehensive maintenance system and provides complete, instantaneous monitoring and troubleshooting capabilities, including fault analysis of the remote site. It can also monitor alarms and controls at local sites. The purpose of subrate multiplexing is to fit more sub-DS0 data circuits into one DS0 (64 kb) channel. The SRDM feeds synchronous or asynchronous data circuits into a single 64-Kbps DS0 signal. For example, the

74


SRDM can multiplex 20 sub-DS0 signals into a DS0 channel at 2.4 Kbps, or 10 at 4.8 Kbps, or 5 at 9.6 Kbps. 3.4.3

Inverse Multiplexing

3.4.3.1 The Need for Bandwidth

The demand for large amounts of bandwidth over extended distances is driving the interest in networking technologies such as ISDN, frame relay, switched multimegabit digital service (SMDS), asynchronous transfer mode (ATM), satellite data communications systems, wireless communications systems, and others. However, making most of these services universally available requires either a new communications network infrastructure, or significant modifications to the existing one. For example, ATM offers high-bandwidth digital connections based upon fixed-size cells that can carry voice, video, and data. But universal ATM also requires the public switched telephone network to replace its time-division multiplexed switching fabric with a new ATM switching fabric and enhanced interoffice trunk facilities. Considering the value of the existing worldwide telephony infrastructure (switches, transmission systems, and embedded wiring plants), this is unlikely to happen any time soon. Therefore, while alternative transmission technologies will certainly be implemented over time to handle the growing demand for high-speed digital bandwidth, full use of the existing digital TDM infrastructure is currently required. While originally conceived as a transmission network for 64-Kbps digitized voice, it is now possible to dial up point-to-point digital connections whose bandwidth ranges from 56 Kbps to 3 Mbps and beyond. Two significant enhancements to TDM networking have made this possible. The first is newly developed software for digital TDM switches that allows dialed connections to exceed the original design channel rate of 56 or 64 Kbps, thus allowing carriers to offer dialed wideband services. The second is the use of specialized equipment that resides at the users premises to allow multiple independent digital connections to be combined to create a single, higherspeed end-to-end connection. This technique is known as inverse multiplexing, and the equipment that performs it is called an inverse multiplexer. NxT1 and NxE1 inverse multiplexing has been around since the early 1990s. Initially, bit-based inverse multiplexers (IMUXes) were developed to aggregate bandwidth from up to eight T1/E1s to provide multimegabit connections for high-speed frame relay, Internet access, videoconferencing, or private network trunks. The delay of each link may be different due to physical path length or buffering in transmission equipment. At the receiving end,


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the IMUXes reconstruct the original bit stream, using buffers to compensate for variations in delay of the individual links. Bit-based IMUXes have several properties that make them easy to use: The order of the bit stream is always preserved, the bundled link can be managed as a single entity, and the data is transparently transported regardless of protocol. However, special proprietary hardware is required at both ends of the link to implement the inverse multiplexing. 3.4.3.2 Basic Features of IMUXes

Up until now, modems have been the basis of the public network data transmission. Their advantage is that they plug directly into the ubiquitous analog lines that access the public telephony network. Their disadvantages include their susceptibility to errors and, more importantly, their limited data rates (typically up to 19.2 Kbps and lately 56 Kbps). While modems are inherently analog transmission devices, the majority of todays public network is digital. Network switches and interswitch trunks have migrated from analog to digital for ease in maintenance, supplemental service offerings, and lower transmission costs. If digital lines are used from the customer premises to the digital network infrastructure, an end-to-end digital call can be made, using the PSDN. The bandwidth of this call far exceeds the bandwidth of a modem call. Since the network is a fabric of 64-Kbps channels, a digital calls bandwidth will be a multiple of 64-Kbps. As mentioned previously, this aggregation of 64-Kbps channels can be performed by either the network itself or by inverse multiplexers located on the customers premises. Inverse multiplexing, which works at the physical layer, spreads a data stream received from any DTE device across the link and reconstructs it at the other end of the link. In other words, inverse multiplexing can be defined as the aggregation of multiple independent information channels across a network to create a single higher-rate information channel. For example, if six different independent 56-Kbps data channels are established between points A and B, inverse multplexing can be used to combine these channels to create a single 336-Kbps (i.e., 6 × 56 = 336) data stream. Likewise, 64-, 384-, and 1,536-Kbps channels can be inverse-multiplexed together. Specifically, the IMUX assures that all the channels are present and accounted for. The IMUX then segments the transmission data stream and sends it out over the individual channels (it could be leased or owned T1/E1 lines). At the receiving end the IMUX accepts the data from these channels, reordering the segments and compensating for variances in channel transit times. Depending upon the inverse multiplexing protocol in use, the IMUX may monitor the integrity of the aggregated connection. Should transmission

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problems occur, the IMUX can take diagnostic action, replacing failed or failing channels with functional channels to maintain the integrity of the connection. The IMUX can also add or remove channels from the aggregated connection without terminating the connection. This allows the total amount of bandwidth between the two sites to vary according to real-time bandwidth requirements or for economies of operation. This feature is sometimes referred to as dynamic bandwidth allocation, or rubber bandwidth. The technology can be used in both dialed and leased-line environments and provides an efficient way to bridge the gap between T1/E1 and T3/E3 and alleviate the bottleneck in the broadband networks. Data-channel reordering and network-delay compensation are important features of inverse multiplexers. Because each of the channels of an inverse multiplexed call is separate and independent, there must be a mechanism within the inverse multiplexer to accept data from each of the channels and then reassemble it in the correct order. In addition, each of the channels of an inverse multiplexed call can and will likely follow a path across the network that has a different propagation time than other channels within the connection (see Figure 3.6). A call from San Francisco to New York may have one channel routed through Dallas, another through Chicago, and yet another through both Atlanta and Washington, D.C. Some channels of an inverse-multiplexed connection might go through a satellite channel, which has nearly a 300 m of propagation delay, while the remaining channels follow much shorter terrestrial paths. The digital switching matrices within each network switch add additional delay to the channel as well. Therefore, the inverse multiplexer must also compensate for the differential delays between channels. The inverse multiplexer compensates for each channels different delay Variable-delay T1/E1 circuits

Data distributed bit-wise to lines

ww

Figure 3.6 IMUX delay issues.

w w w ww w w

ww w

IMUX Outgoing w w Data reassembled in proper order y x

y x

x x x x x x x y y y y y y y

y y

y x

y x

w

x x

Incoming w

y x x x x

IMUX


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characteristics by buffering each channels data as it is received. Each channels data buffer is in essence a first-in-first-out (FIFO) buffer, and these FIFOs are unloaded in such a way that the reassembled data stream is properly reconstructed. The total end-to-end delay of an inverse-multiplexed connection is approximately that of the channel with the longest network transit time (internal inverse-multiplexer processing delays are small compared with network transit times). How is this feature used? In videoconferencing, the quality of video motion increases as the bandwidth between sites increases. A videoconferencing session might use a 384-Kbps inverse-multiplexed connection at the beginning of a course lecture, when the lecturer is on camera and moving about the stage. However, the bandwidth can be reduced when the camera focuses for an extended period on a blackboard, when the relative lack of motion requires less bandwidth and, therefore, less expense. In LAN internetworking, an inverse-multiplexed connection between two remote LANs might be established at 64 or 128 Kbps, adequate for most simple interactive activities. However, if that connection becomes saturated for a predetermined period of time by, for example, a large file transfer, the bandwidth of the connection can be automatically or manually increased to accommodate the increase in traffic, allowing file transfer to take place much faster. At the end of the file transfer, the bandwidth is reduced to the original value. The benefits are increased available bandwidth for all users and, as bandwidth is more efficiently utilized, a reduction in overall bandwidth costs. When IMUXes for the public switched digital network first appeared, standard inverse-multiplexing protocol did not exist. Therefore, each manufacturer developed its own proprietary protocol, requiring both ends of an inverse-multiplexed connection to use the same manufacturers equipment. There was no interoperability between inverse multiplexers manufactured by different vendors. Inverse-multiplexer manufacturers soon realized that, unless an inverse-multiplexer standard was implemented, inverse multiplexing would not reach its true potential as a data communications technology. Therefore, the Bandwidth-on-Demand Interoperability Group (BONDING) consortium was formed by a number of inverse-multiplexer manufacturers to develop an inverse-multiplexing standard. The resulting BONDING specification was published, implemented by vendors, tested, and publicly demonstrated. Today, nearly all inverse-multiplexing equipment implements at least a subset of the BONDING inverse-multiplexing protocol to assure interoperability between different vendors equipment. In wireless networks, bandwidth of more than 1E1 or 1T1 (unchannelized) per cell site may be required. This is usually the case in high-traffic areas

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where RBS will be heavily loaded. In that case, backhaul circuits (RBS-BSC connection) are using IMUXes to divide the bandwidth into smaller, and generally more recognizable (and widely available), E1/T1 circuits. 3.4.3.3 The IMUX Metaframe

Metaframing is part of the inverse-multiplexing process. The IMUX metaframer has three major functions: 1. Creating a metaframe, larger than a DS1 frame, which aligns the received DS1 lines and permits corrections between their various delay times; 2. Providing a metaframe facility data link (MFDL) to the far end for managing logical channels and validating the receiver synchronizer; 3. Providing a metaframe parity indication of the logical channel data for an end-to-end integrity check. The IMUX supports two types of metaframes: the standard metaframe and the extended metaframe. Each is created by robbing the 17th bit in every transmitted frame. The standard metaframe length is 48 milliseconds, allowing the alignment of up to 22 milliseconds of differential delay. Using the extended metaframe (768 milliseconds in length) allows the IMUX to correct up to 45 or 125 milliseconds of differential delay, depending on the model. Again, the metaframe is created by robbing the 17th bit in every transmitted frame. This is the least significant bit in the second DS0 for T1 and the most significant bit in the second DS0 for E1. Because the normal DS1 payload is 1.536 Mbps (T1) or 1.984 Mbps (E1), the use of one bit for the metaframe leaves a user payload of 1.528 Mbps (T1) or 1.976 Mbps (E1). The MFDL, which runs at 4 Kbps within the metaframe, serves as a communication link to the far-end IMUX. The rate is constant, regardless of the number of lines in the logical channel. Each T1 or E1 contains an MFDL, but messages are sent on only one MFDL at a time. The MFDL provides the following: • Access to the far-end status and performance data; • An IP link to the far end that may be used for Telnet, SNMP, and

downloading;

• Management of the logical channel; • Identification of the T1/E1.


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Because the metaframe is carried in the payload, it will show errors when there have been impairments, even when there are intervening multiplexers. Counters like ES, UAS, and ESF are not usually end-to-end in most applications and will not increment when line errors or frame slips occur in a preceding T1 or E1 section (on the other side of a DS0 cross connect). MFE and MSYN errors may be the only indication of line problems (other than application errors). However, the presence of any line impairments that cause line errors to be reported will cause the occurrence of metaframe errors.

3.5 ATM 3.5.1

ATM Basics

ATM is the complement of STM. STM is a circuit-switched networking mechanism whereby a connection is established between two end points before data transfer commences, and torn down when it is completed. In this way, the end points allocate and reserve the connection bandwidth for the entire duration, even when not actually transmitting the data. ATM is a transmission technology that uses fixed-size packets called cells. A cell is a 53-byte packet with 5 bytes of header-descriptor information and 48 bytes of payload, or user traffic (voice, data, video, or a combination of these). Today, telecommunications companies are deploying fiber-optic cross-country and cross-oceanic links with Gbps speeds. They would like to carry, in an integrated way, both real-time traffic such as voice and high-resolution video, which can tolerate some loss but without delay, as well as non-real-time traffic such as computer data and file transfer, which may tolerate some delay, but not loss. The problem with carrying these different characteristics of traffic on the same medium in an integrated fashion is that the requirements of these traffic sources may be quite different. In other words, the data comes in bursts and must be transmitted at the peak rate of the burst, but the average arrival time between bursts may be quite large and randomly distributed. For these connections, it would be a considerable waste of bandwidth to reserve them a bucket at their peak bandwidth rate for all times, when on the average only 1 in 10 buckets may actually carry the data. And thus using STM mode of transfer becomes inefficient as the peak bandwidth of the link, peak transfer rate of the traffic, and overall burstyness of the traffic expressed as a ratio of peak to average, all go up. Terms like fast packet, cell, and bucket are used interchangeably in ATM literature.

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ATM networks are connection-oriented packet-switching networks. Virtual circuits must be established between the end nodes before transmission can start and, as with any packet-switching network, routing of cells is performed at every node for each arriving cell. A virtual path identifier (VPI), an 8- or 12-bit field, together with virtual circuit identifier (VCI), a 16-bit field, contains the routing information of the cell [8]. The main idea rather than identifying a connection by the bucket number each time was just to carry the connection identifier along with the data in any bucket. This would reduce the size of the bucket so that if any one bucket got dropped en route due to congestion, less data would get lost, and in some cases, could easily be recovered. As this resembled packet switching, they called it fast packet switching with short fixed-length packets. The fixed size of the packets arose out of the desire of telecommunications companies to sustain the same transmitted voice quality as in STM networks, even in the event of some lost packets on ATM networks. The two end points in an ATM network are associated with each other via an identifier called the VCI label, instead of by a timeslot number as in an STM network. The VCI is carried in the header portion of the fast packet. The fast packet itself is carried in the same type of bucket used before, but without the label or designation for the bucket. The physical layer specification, while not explicitly a part of the ATM definition, is being considered by the same subcommittees, and SONET was standardized as the preferred physical layer. Therefore, the STS classifications refer to the speeds of the SONET link. STS-3c is 155.5 Mbps, STS-12 is 622 Mbps, and STS-48 is 2.4 Gbps. The SONET physical layer specifications outline a worldwide digital telecommunications network hierarchy internationally known as the SDH. It standardizes transmission around the bit rate of 51.84 Mbps that is also called STS-1, and multiples of this bit rate comprise higher bit rate streams. Thus, STS-3 is 3 times STS-1, STS-12 is 12 times STS-1, and so on. STS-3c is of particular interest as this is the lowest bit rate expected to carry the ATM traffic, and is also referred to as STM-1. The term SONET is the U.S. terminology for SDH. SDH specifies how payload data is framed and transported synchronously across fiber-optic transmission links without requiring all the links and nodes to have the same synchronized clock for data transmission and recovery (i.e., both the clock frequency and phase are allowed to have variations, or be plesiochronous). The intention is to allow products from multiple vendors across geographical and administrative boundaries to be able to plug and play in a standard, with the broadband ISDN network acting as a true international network. All of this rests below the ATM layer, and ATM


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cells are transported across the physical layer as payload, also called the SONET payload or the synchronous payload envelope (SPE). The physical layer is independent of the payload type, and can just as easily carry STM cells as ATM cells. Future telecommunication networks, including wireless networks, must be able to offer todays range of services as well as services with new features (e.g., variable bit rates). The requirements of modern networking involve handling multiple types of traffic (voice, video, and data), all with individual characteristics that make very different (and often opposed) demands of the telecommunication channel. The second requirement is the reliability and flexibility of the communication links. The greatest problem is that transmissions occur at statistically random intervals with variable data rates. A way of solving this problem is to take a service that takes packets on the transport layer from a higher layer and fragments them in small packets of a fixed size. The delays produced by each packet are going to be short and probably fixed, so if voice and video traffic can be assured priority handling, they can be mixed with data without diminishing any reception quality. The services are called the ATM adaptation layer (AAL) and the packets are called cells. ATM networks are connection-oriented, but the service offered is internally implemented using packet switching. In a circuit-switching network, a physical path is established from the source to the destination. In a virtual circuit network, like ATM, when a connection is established, no physical resources are allocated; rather, all the switches in between source and destination make table entries defining the route. All data will follow subsequently. When a packet comes along, the switch looks up its header information and relays it to the next switch indicated in the VCI information. This fulfills the need for variable bandwidth (i.e., if one flow needs more bandwidth, then another ATM can provide it). This is the virtue of packet switching using statistical multiplexing. Note that ATM uses packet switching to emulate circuit mode. Fixed-length packets offer many advantages because it is easier to switch packets when their length is known. There are two different cell formats, depending on where a cell is in the network: UNI and network-network interface (NNI). The UNI format is used when transmitting from host to switch and the NNI format when transmitting between switches, and there is only a minor difference between them. An overview of different ATM interfaces is illustrated in Figure 3.7. The advantage of having two different virtual connections is that it makes routing much easier between intermediate switches. This is accomplished by switching groups of VCIs on VPI level rather than on every single VCI. Only

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Computer Private UNI

Private switch Private NNI

Computer Private Private switch UNI

Public UNI Carriers Public switch

Public NNI

Public switch B-ICI

Router

DXI

DSU

Carrier

Public switch

User-to-network interface (UNI): Public UNI, Private UNI Network-to-node interface (NNI): Private NNI (P-NNI) Public NNI = Interswitching system interface (ISSI) Intra-LATA ISSI (Regional Bell Operating Company) Intra-LATA ISSI (Interexchange carriers) ⇒Broadband Intercarrier Interface (B-ICI) Data exchange interface (DXI) Between routers and ATM digital service units (DSUs)

Figure 3.7 ATM interfaces overview.

the end switch will have to switch on VC level. Note that the virtual path (VP) acts as a fat pipe transporting bundles of thousands of virtual circuits. ATM cells can be transmitted over various carriers such as SDH/SONET, FDDI, or T1/E1. Traffic management is necessary to ensure that both the priorities of the user and the performance criteria of the network are met. The main threat to orderly traffic transmission is congestion and the possibility of having to start discarding cells, due to too many cells arriving at a point in the network. The network must cater to all types of traffic and deal with the different QoS requirements for the various traffic types. CBR connections, for example, are very delay and delay-variation sensitive. They require different parameters to variable bit rate (VBR) connections, such as LAN traffic. LAN traffic, due to its bursty nature, is well able to tolerate variations in end-to-end delay. It also has upper-layer software to protect it against lost or misinserted cells.


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Use of ATM Adaptation Layer

The AAL is the highest layer in ATM, and it is used to adapt traffic into an ATM format. In other words, the AAL maps application data into the ATM 48-byte cell payloads. The AAL function is performed at the edges of an ATM connection, and not within the network. As the AAL is used to adapt traffic to an ATM format, it is needed at the entry point to the ATM network. Once the traffic is adapted to the ATM cell format it travels across the network in ATM cells, which are switched in the ATM layer of switches along the path of the ATM connection. Once the cells reach their destination, there is a need to reassemble the traffic back into the format of the original application. The AAL is thus also used at the exit point of the ATM network. The high-level ATM protocol layer structure is shown in Figure 3.8. The ITU-T I.362 standard provides the functional descriptions for the AALs. Traffic classes are based on the following parameters: • Whether a timing relationship is required between source and

destination;

• Whether the traffic is CBR or VBR; • Whether the traffic is connection oriented or connectionless: •

Class A defines traditional synchronous data, such as that containing T1/E1 voice circuits or uncompressed broadcast video.

•

Class B covers compressed video that requires a timing relationship. End system

End system

ATM adaptation layer

Switch

ATM adaptation layer

ATM layer

ATM layer

ATM layer

Physical layer

Physical layer

Physical layer

Figure 3.8 ATM protocol layers.

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• •

Class C defines bursty data such as frame relay, X.25, or large file transfer. Class D includes broadcast data, such as Service Access Point (SAP) messages in NetWare or an Address Resolution Protocol (ARP1) packet in TCP/IP.

There is also a Class X, which covers undefined or unspecified bit rate (UBR), where the user defines traffic type and timing requirements. Reference may be found, in ANSI documentation, to Class Y. In the ITU and ATM Forum documentation this is known as available bit rate (ABR). AAL5 is the ATM Forums response to the ITU-Ts excessively complex AAL3/4. AAL5 is sometimes known as simple and efficient adaptation layer (SEAL). The AAL is used at the entry and exit point of the ATM network, and all ATM switches require an AAL function. For example, signaling traffic needs to be interpreted by all switches. To interpret incoming signaling traffic, the ATM switch needs firstly to reassemble the signaling traffic. This requires an AAL function. The switch knows that traffic coming in from a given connection is signaling traffic, as a special reserved VPI/VCI value is used. Similarly, all ATM switches need to interpret operation and maintenance traffic and management traffic. The AAL provides services to application programs, taking packets from higher layers and fragmenting them so that they will fit in the cell payload. Five different types of AALs have been defined to serve the demands of the different services: • AAL1 is for real-time services with a constant bit rate. • AAL2 is used for real-time services with variable bit rates. By far the

most important benefit of AAL2 is the ability to substantially reduce bandwidth requirements for supporting voice on ATM networks, and the inherent flexibility to add features via the service-specific convergent sublayer (SSCS) structure. • AAL3 and AAL4 are for connection-oriented and connectionless transmission of non-real-time services. As there was no need for distinguishing AAL3 and 4, the two AALs merged to become AAL3/4. • AAL5 is a simplified version of AAL3. 1. A low-level protocol within the TCP/IP suite that maps IP addresses to the corresponding Ethernet addresses, ARP is used to obtain the physical address when only the logical address is known.


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In AAL0 no AAL functionality is used. The data is directly written in the payload field of the cells. AAL0 is no real AAL, as the functionality of adaptation is not needed. AAL1 is used for data streams with a constant bit rate. Clock information is transferred with the data. This layer is further split up into two sublayers: the segmentation and reassembly (SAR) sublayer and the convergence sublayer (CS). The 48-byte payload is split up into 47 bytes of data and 1 byte of header. The header contains a sequence number to detect lost packages or wrongly transmitted packages. AAL2 is responsible for the efficient transmission of delay-sensitive narrowband applications with variable bandwidth (see Figure 3.9). This means that for each transmission the QoS and the maximum cell delay or the maximum cell-loss rate must be guaranteed and the necessary bandwidth must be available. Typical applications are telephony and MPEG compressed video. In 3G wireless networks, AAL2 will be used, among other functions, to carry mixed voice and data traffic between the RBS and BSC (backhaul). AAL2 is new, and the data that would be supported by it was traditionally transmitted in AAL5 instead. A problem with the utilization of AAL5, however, is the lack of delay parameters. By contrast, AAL2 is inherently designed for the support of VBR traffic, for which timely delivery is an issue. A feature of AAL2 is the ability to accept several streams of traffic and multiplex them together. The manner of multiplexing is to accept samples and to append a small header to each sample. The primary function here is to add a channel number to identify the higher-layer stream. Once so labeled, blocks are then transferred to the ATM cell payloads. Part of this transfer is to add yet another header. The principal use of this header is to identify the start of Ideal for low-bit-rate voice Variable and constant rate voice Multiple users per VC Compression and silence suppression Idle channel suppression Payload 1

Payload 2

Payload 3

Pkt Payload 1 Pkt Payload 2 Pkt Payload 3 Hdr Hdr Hdr Pkt Payload 1 Pkt Payload 2 Pkt Payload 3 Cell Hdr Hdr header Hdr

Figure 3.9 AAL2 structure.

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a flow after a short period of inactivity (one of the streams may show a blank screen for a couple of seconds and the stream may produce no output). AAL3 and AAL4 specify the connection-oriented and non-connectionoriented transmission of data packets in ATM networks. Connections can be point-to-point or PMP. Like AAL1, AAL3 and AAL4 are divided into an SAR and a CS sublayer; however, in the CS we distinguish a common part convergence sublayer (CPCS) and an application-specific part, the SSCS. AAL5 is a simplified version of AAL3 and AAL4. There is no possibility of multiplexing cells. AAL5 has significantly lower overheads than AAL3 and AAL4 and is, therefore, very widely adopted. In practice, AAL3 and AAL4 are seen as overly complex and cumbersome; only AAL1 and AAL5 are widely used. AAL1 is used for CBR traffic and AAL5 for all other: VBR, UBR, and ABR. SAAL is used for signaling protocols. SAAL guarantees the higher signaling layer a transmission of their messages with the Service-Specific Oriented Protocol (SSCOP). 3.5.3

Inverse-Multiplexing for ATM Protocol

Until recently, ATM services were limited to the core of the public network, or to enterprises with data volumes large enough to justify the T3/E3 and higher-rate facilities on which ATM was transported. Businesses with less substantial volumes of data were left to choose between dedicated T1/E1 lines and frame-relay services topping out at T1/E1 rates, and data pipes in the bandwidth gap between T1/E1 and T3/E3 were achievable only through inverse multiplexing. Recognizing the need for ATM at speeds lower than E3/T3 and STM-1/OC-3c, the ATM Forum has developed the inverse multiplexing for ATM (IMA) specification. As the name suggests, this specification offers a standard for ATM transmission at nxE1 (or nxT1) rates over inverse-multiplexed E1 (or T1) lines. Formally approved in 1997, IMA has received widespread acceptance, with numerous products implementing IMA currently on the market or in development. The IMA protocol defines a method for taking an ATM cell stream and breaking it down for transmission over multiple E1/T1 links. Where previous E1/T1 inverse multiplexing had been bit-based, with no regard for the content or structure of the original payload, IMA recognizes ATM cells and distributes them individually across the multiple links. The separate E1 or T1 links across which the ATM stream is transported are called an IMA group. IMA is very robust in its recovery features, including physical and logical layer monitoring, error detection, and correction. Physical link monitoring is based on loss of


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signal and bit error rates. For logical event monitoring, a link state machine function determines which links are stable and are maintaining data rates that meet the IMA group timing requirements. Links that are identified as part of the group are continuously monitored and can be taken in and out of the group dynamically. To properly order and account for the cells, a special IMA control protocol (ICP) cell is used to track timing information and frame sequence numbers, as well as other management information. The timing information is of particular importance, because it allows for some variance in clocking among the physical E1/T1s used in the IMA group. E1/T1 signals are delayed approximately eight milliseconds per 1,000 miles. Signals taking routes of different lengths have what is referred to as differential delay. Because the IMA protocol includes the ability to handle relatively large differential delays, an IMA group can include diversely routed E1/T1 facilities, either from a single carrier offering diverse routing or from multiple carriers. Even if the E1/T1 lines on one route are all brought down by a catastrophic failure, the lines on the other route will still generally be available, and the service can continue to run at a lower data rate until the affected lines are restored. 3.5.4

QoS in ATM Networks

ATM is a high-speed networking technique capable of supporting many different classes of traffic. CBR service is ideal for any data, text, or image transfer application that requires a fully reserved channel. VBR service is suitable for any application that can benefit from statistical multiplexing or that can tolerate or recover from potential and random packet loss. Real-time VBR service can be used by native ATM voice with bandwidth compression and silence suppression. Non-real-time VBR can be used for data transfer and frame-relay interworking. ABR provides economical support to applications that have vague requirements for throughput and delay and require a low cell-loss ratio. UBR offers a solution for less demanding applications that take advantage of any spare bandwidth and profit from it. UBR is a besteffort service with no guarantees. QoS metrics in ATM networks can be categorized into two different classes: the first one includes the call control parameters associated with connection-oriented networks and the second class is the set of information transfer parameters defined in packet networks. 3.5.4.1 Call Control Parameters

Three of the most important components of interest in connection-oriented networks are setup delay, connection release delay, and connection acceptance

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probability. Connection setup delay is the time interval between the call setup message transfer to the call setup acknowledge message transfer, excluding the called users response time. This is not an ATM-specific parameter and is mainly determined by the processing delays at various signal transfer points in the network. Connection release delay is the time interval between the callrelease message transfer and receipt of the call-release acknowledge message. This is also an ATM-independent parameter. Connection acceptance probability is the proportion of the accepted calls over a long period of time, which is also referred to as the blocking probability in telephony networks. This probably is one of the most important performance metrics used in network design and allocation of network resources (such as network topology, switch capacities, link capacities, and so on). Parameters, such as call attempt rate, average call holding time, and busy-hour call attempts are used for the initial network planning. 3.5.4.2 Information Transfer Parameters

Information transfer parameters in ATM networks include cell information field bit error ratio (BER), cell-loss ratio (CLR), cell-insertion ratio (CIR), end-to-end transfer delay, cell delay variation (CDV), and skew. BER is defined as the ratio of the bit errors in the information field to the total number of bits transmitted in the information field. Assuming that errors occur randomly, the probability that there is no bit error in the 48byte (384 bit) ATM cell payload on a link is

(1 − BER ) 384 Assuming BER = 10−6, this probability is 0.99617 and increases to 0.999996 with BER = 10−9. CLR is the ratio of the number of lost cells to the total number of cells sent by a user within a specified time interval. This metric is ATM-specific and has an important impact on the quality of service provided to users. All AALs, except type 5, include cell sequence numbers to detect lost cells at the receiver. The effect of cell losses and actions taken on lost cells in ATM networks is different for different types of services. It is possible that an error occurring at the header (CIR) may not be detected by header error checking. In this case, if an undetected change in the bit pattern of a cell header corresponds to the address of another connection, then the cell is misrouted to a wrong destination.


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Cell-transfer delay (CTD) is caused by different factors like coding delay, packetization delay, propagation delay, transmission delay (5 microsec/km in fiber-optic cable), switching delay, queuing delay, and reassambly delay. CDV (jitter) consists of the constant delay component (CTD) and the random delay component that arises out of buffering within the network. The only way of controlling jitter at the receiver is at the expense of large buffers and delaying cells. Skew is defined as the difference in the presentation times of two related objects (like video stream and audio stream) and is important in multimedia applications, where it can cause delay between an image and accompanying voice or between lip motion and voice. A key consideration in the ATM standard is given to the concept of VPs. These are groups of VCs bundled together to reduce setup, switching, and control costs. Although VPs have not yet been extensively deployed, their use in networks that process a large number of switched VCs, or calls, is fundamental [9]. Many of the QoS concepts and notions developed for ATM networks explicitly depend on the network providers ability to integrate different services with different service characteristics over the same physical transport trunks via VPs. Performance management entails the periodic evaluation of ATM equipment and software. The idea is to assess the ATM system in a systematic way in order to determine how the network is performing and if error conditions are acceptable. Performance management consists of forward monitoring, backward monitoring, and reporting [10]. Forward monitoring means generating cells from one network element to a receiving network element. Backward monitoring is checking the cells at the receiving network element and reporting back to the generating network element. Reporting usually means storing the results of the monitoring activities based on the filtering of selected parameters and thresholds. An excellent source of information regarding ATM QoS, required connection availabilities, and performance management is ITU-T Recommendation I.356, B-ISDN ATM Layer Cell Transfer Performance (03/2000) and ITU-T Recommendation I.357, B-ISDN Semi-permanent Connection Availability (11/2000). 3.5.5

Definition of Availability in ATM

From a dependability point of view, a portion of an ATM semipermanent connection should have the following properties:

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• The fraction of time during which it is in a down state (i.e., unable

to support a transaction) should be as low as possible. • Once a transaction has been established, it should have a low probability of being either terminated (because of insufficient data transfer performance) or prematurely released (due to the failure of a network component) before the intended end of a transaction. Availability of an ATM semipermanent connection portion is defined as the fraction of time during which the portion is able to support a transaction. Conversely, unavailability of a portion is the fraction of time during which the portion is unable to support a transaction (i.e., it is in the down state). A common availability model is used which applies to any semipermanent connection type. The model uses two states corresponding to the ability or inability of the network to sustain a connection in the available state. Transitions between the states of the model are governed by the occurrence of patterns of severely errored seconds in the ATM layer (SESATM). In order to define the availability of an ATM semipermanent connection portion, a criterion is defined for entry into the unavailable state. This criterion is applicable to any ATM semipermanent connection portion, whether the user continuously transmits cells or not. This is achieved by defining a cell transfer outcome, the SESATM. A given second is considered to be an SESATM in the following cases: • User information cells are presented during this period of time

to the connection portion and either the CLR > 1/1,024 or the severely errored cell block ratio (SECBR) > 1/32, where CLR and SECBR are computed over the considered period of time. The above CLR threshold is intended to support QoS classes in which the CLR objective is ≤ 10−5. • User information cells are not presented during this period of time to the connection portion, but the ATM connection is considered to be unable to provide acceptable cell transfer performance, because an interruption has occurred within the connection portion. This interruption prevents cells from being transmitted on the connection portion during the considered one-second period of time, should the user attempt to transmit cells. An interruption corresponds to a failure occurring within the connection portion, either of the physical layer or of the ATM layer.


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The methods for estimating the occurrence of an SESATM are taken from the set of cell-transfer parameters defined in ITU-T I.356 and the OAM facilities defined in ITU-T I.610. The onset of unavailability begins with the occurrence of 10 consecutive SESATM. These 10 seconds are part of unavailable time. A period of unavailability ends with the occurrence of 10 consecutive seconds, none of which SESATM. These 10 seconds are part of available time. The 10-second criteria are supported using a sliding window with one-second granularity [11]. A portion of a bidirectional B-ISDN connection is available if and only if both directions are available. It is recognized that in-service measurement of availability as defined above may not be practicable in many cases. Performance objectives are defined for two availability performance parameters: availability ratio (AR) and mean time between outages (MTBO). AR applies to ATM semipermanent connection portions. The AR is defined as the proportion of scheduled service time that the connection portion is in the available state. The AR is calculated by dividing the total service available time by the duration of the scheduled service time. During the scheduled service time the user may or may not transmit cells. MTBO applies to ATM semipermanent connection portions. The MTBO is defined as the average duration of continuous periods of available time, but where scheduled service time is not contiguous they are concatenated in calculating MTBO.

3.6 Voice over IP 3.6.1

H.323 Network Building Blocks

The transfer of voice traffic over packet networks, and especially voice over IP (VoIP), is rapidly gaining acceptance. Many industry analysts estimate that the overall VoIP market will become a multi-billion-dollar business within the next few years. While many corporations have long been using voice over frame relay (VoFR) to save money by utilizing excess FR capacity, the dominance of IP has shifted most attention from VoFR to VoIP. These high-speed backbones take advantage of the convergence of Internet and voice traffic to form a single managed network. This network convergence also opens the door to novel applications. Interactive shopping (Web pages incorporating a click-to-talk button) is just one example, while streaming audio, electronic white-boarding, and CD-quality conference calls in stereo are other exciting applications. But along with the initial excitement, potential users are also worried over possible degradation in voice quality when voice is carried over these

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packet networks. Whether these concerns are based on experience with the early Internet telephony applications, or whether they are based on understanding the nature of packet networks, voice quality is a critical parameter in acceptance of VoIP services. As such, it is crucial to understand the factors affecting voice over packet transmission, as well as obtain the tools to measure and optimize them. VoIP services need to be able to connect to traditional circuit-switched voice networks, and the ITU-T has addressed this goal by defining H.323, a set of standards for packet-based multimedia networks. The basic elements of the H.323 network are shown in the network diagram (Figure 3.10) where H.323 terminals such as PC-based phones (left side of drawing) connect to existing ISDN, PSTN, and wireless devices (right side). The H.323 components include the following: • H.323 terminals that are end points on a LAN. • Gateways that interface between the LAN and switched-circuit

network.

• A gatekeeper that performs admission-control functions and other

duties.

• A multipoint control unit (MCU) that offers conferences between

three or more end points.

H.323 terminals are LAN-based end points for voice transmission and they support real-time, two-way communications with other H.323 entities. H.323 terminals implement voice-transmission functions and specifically include at least one voice compressor/decompressor (CODEC) that sends and receives packetized voice. Common CODECs are ITU-T G.711 (pulse code modulation [PCM]), G.723 (MP-MLQ), and G.729A (CA-ACELP). CODECs differ in their central processing unit (CPU) requirements, in the resultant voice quality, and in their inherent processing delay. Terminals also need to support signaling functions that are used for call setup, teardown, and so forth. The applicable standards here are H.225.0 signaling, which is a subset of ISDNs Q.931 signaling; H.245, which is used to exchange capabilities such as compression standards between H.323 entities; and registration, admission, status (RAS), which connects a terminal to a gatekeeper. Terminals may also implement video and data communication capabilities. The gateway serves as the interface between the H.323 and non-H.323 network. On one side, it connects to the traditional voice world, and on

Gateway H.323 terminal

H.323 terminal

Gatekeeper

Wireless

Router

PSTN

Router MCU


ISDN Router

Enterprise network Gateway 93

Figure 3.10 VoIP principle diagram.

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another side to packet-based devices. As the interface, the gateway needs to translate signaling messages between the two sides as well as compress and decompress the voice. A prime example of a gateway is the PSTN-IP gateway, connecting an H.323 terminal with the circuit-switched network. There are many types of gateways in existence today, ranging from support of a dozen or so analog ports to high-end gateways with simultaneous support for thousands of lines. The gatekeeper is not a mandatory entity in an H.323 network. However, if a gatekeeper is present, it must perform a set of functions. Gatekeepers manage H.323 zones, a logical collection of devices (for example, all H.323 devices within an IP subnet). Multiple gatekeepers may be present for load balancing or hot-swap backup capabilities. The philosophy behind defining the gatekeeper entity is to allow H.323 designers to separate the raw processing power of the gateway from intelligent network-control functions that can be performed in the gatekeeper. A typical gatekeeper is implemented on a PC, whereas gateways are often based on proprietary hardware platforms. Gatekeepers provide address translation (routing) for devices in their zone. This could be, for instance, the translation between internal and external numbering systems. Another important function for gatekeepers is providing admission control, specifying what devices can call what numbers. Among the optional control functions for gatekeepers are providing SNMP management information, offering directory and bandwidth management services. The MCU allows for conferencing functions between three or more terminals. Logically, an MCU contains two parts: 1. A multipoint controller (MC) that handles the signaling and control messages necessary to set up and manage conferences; 2. A multipoint processor (MP) that accepts streams from end points, replicates them, and forwards them to the correct participating end points. An MCU can implement both MC and MP functions, in which case it is referred to as a centralized MCU. Alternatively, a decentralized MCU handles only the MC functions, leaving the multipoint processor function to the end points. It is important to note that the definition of all the H.323 network entities is purely logical. No specification has been made on the physical division of the units. MCUs, for instance, can be stand-alone devices, or they can be integrated into a terminal, a gateway, or a gatekeeper.


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Latency and Jitter Issues

In contrast to broadcast-type media transmission, a two-way phone conversation is quite sensitive to latency. Most callers notice round-trip delays when they exceed 250 ms, so the one-way latency budget would typically be 150 ms, also specified in the ITU-T G.114 recommendation as the maximum desired one-way latency to achieve high-quality voice. Beyond that roundtrip latency, callers start feeling uneasy about holding a two-way conversation and usually end up talking over each other. At 500-ms round-trip delays and beyond, phone calls are impractical. For reference, the typical delay when speaking through a geostationary satellite is 150 to 500 ms. Data networks are not affected by such delay. An additional delay of 200 ms on an e-mail or Web page goes mostly unnoticed. But when sharing the same network, voice callers will notice the delay. When considering the one-way delay of voice traffic, one must take into account the delay added by the different segments and processes in the network. Some components in the delay budget need to be broken into fixed and variable delay. For example, for the backbone transmission there is a fixed transmission delay, which is dictated by the distance, plus a variable delay, which is the result of changing network conditions. The most important components of this latency include the following: • Backbone (network) latency. This is the delay incurred when travers-

ing the VoIP backbone. In general, to minimize this delay, it is necessary to minimize the number of router hops between end points. Some service providers are capable of providing an end-to-end delay limit over their managed backbones. Alternatively, it is possible to negotiate or specify a higher priority for voice traffic than for delayinsensitive data.

• CODEC latency. Each compression algorithm has certain built-in

delay. For example, G.723 adds a fixed 30-ms delay. When this additional gateway overhead is added in, it is possible to end up paying 32 to 35 ms for passing through the gateway. Choosing different CODECs may reduce the latency, but it may also reduce quality or result in more bandwidth being used.

• Jitter buffer depth. To compensate for the fluctuating network condi-

tions, many vendors implement a jitter buffer in their voice gateways. This is a packet buffer that holds incoming packets for a specified amount of time before forwarding them to decompression. This has the effect of smoothing the packet flow, increasing the

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resiliency of the CODEC to packet loss, delayed packets, and other transmission effects. The downside of the jitter buffer, however, is that it can add significant delay. The jitter buffer size is configurable and can be optimized for given network conditions. The jitter buffer size is usually set to be an integral multiple of the expected packet interarrival time in order to buffer an integral number of packets. It is not uncommon to see jitter buffer settings approaching 80 ms for each direction. While network latency affects how much time a voice packet spends in the network, jitter controls the regularity in which voice packets arrive. Typical voice sources generate voice packets at a constant rate, so the matching voice decompression algorithm also expects incoming voice packets to arrive at a constant rate. However, the packet-by-packet delay inflicted by the network may be different for each packet. The result is that packets that are sent in equal spacing from the left gateway arrive with irregular spacing at the right gateway. There are three interesting configurations for measuring latency: 1. Measuring the latency of a device; 2. Measuring round-trip delay; 3. Measuring one-way delay. Measuring the latency of a device is important to understand how the delay budget gets spent over the network. In particular, it is interesting to measure the latency of data going through a gateway, since several user-configurable parameters, such as jitter-buffer size, affect the latency. Thus, after configuring such parameters, it is important to be able to verify that the gateway actually behaves as expected. Some products allow measuring the latency by generating controlled data through an ingress port and capturing it off an egress port. Some protocol analyzers can operate two technologies at the same time, with a synchronized timestamp that allows interport or intertechnology latency measurement. The analyzer can measure the latency through a network using a similar method. When the two end points are geographically distant, it is often less convenient to perform one-way latency measurements, because such an operation requires synchronizing the control and timestamp of two separate analyzers. Instead, many users measure the round-trip time and assume it is twice the one-way time for each direction. Round-trip measurements can be


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done using a protocol analyzer, or as a first approximation using the ping utility, generating Internet Control Message Protocol (ICMP) echo requests through the network. Since the receiving decompression algorithm requires fixed spacing between the packets, the typical solution is to implement a jitter buffer within the gateway. The jitter buffer deliberately delays incoming packets in order to present them to the decompression algorithm at fixed spacing. The jitter buffer will also fix any out-of-order errors by looking at the sequence number in the RTP frames. While the voice decompression engine receives packets directly on time, the individual packets are delayed further in transit, increasing the overall latency. Jitter is calculated based on the interarrival time of successive packets. Frequently, two numbers are given: the average interarrival time and the standard deviation. On a good network, the average interarrival time will be the interarrival time of the emitted packets, and the standard deviation will be lowpointing at a consistent interarrival time. When correct jitter measurements are desired for audio streams, it is important to take into account three phenomena: silence suppression, packet loss, and out-of-sequence errors. CODECs take advantage of periods of silence in the conversation to reduce the number of packets being sent. Typically, up to 50% bandwidth savings can be realized in this way. The RTP packet immediately after a period of silence is marked with the silence-suppression bit. Jitter calculations look at the silence-suppression bit and disregard the long gap between the packet right before the silence and the packet right after the silence period. In the event of packet loss, the interarrival time between two successive packets will also appear excessive. For instance, if three packets were sent at a time of 0, 20, and 40 ms, and the second packet was lost in transit, the interarrival time would appear to be 40 ms even if the network induced no jitter. Correct jitter measurements would discover these cases by looking at the packet sequence number and compensate for packet loss in the jitter calculation. 3.6.3

Multiprotocol Label Switching

Multiprotocol label switching (MPLS) is a layer-3 switching technology aimed at greatly improving the packet-forwarding performance of the backbone routers in the Internet or other large networks. The basic idea is to forward the packets based on a short, fixed-length identifier termed a label,

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instead of the network-layer address with variable length match. The labels are assigned to the packets at the ingress node of an MPLS domain. Inside the MPLS domain, the labels attached to packets are used to make forwarding decisions. Thus, MPLS uses indexing instead of a longest address match as in conventional IP routing. The labels are finally popped out from the packets when they leave the MPLS domain at the egress nodes. By doing this, the efficiency of packet forwarding is greatly improved. Routers that support MPLS are known as label switching routers (LSRs). Although the original idea behind the development of MPLS was to facilitate fast packet switching, currently its main goal is to support traffic engineering and provide QoS [12] within IP networks. The goal of traffic engineering is to facilitate efficient and reliable network operations, and at the same time optimize the utilization of network resources. Most current network routing protocols are based on the shortest path algorithm, which implies that there is only one path between a given source and destination end system. In contrast, MPLS supports explicit routing, which can be used to optimize the utilization of network resources and enhance trafficoriented performance characteristics. For example, multiple paths can be used simultaneously to improve performance from a given source to a destination. MPLS provides explicit routing without requiring each IP packet to carry the explicit route, which makes traffic engineering easier. Another advantage is that using label switching, packets of different flows can be labeled differently and, thus, receive different forwarding (hence, different QoS). The differentiated-services (diffserv) model allows (theoretically) definition of 63 different classes of service, each of which can be mapped to the applications being used by the customer. A label-switched path (LSP) is referred to as a path from the ingress node to the egress node of an MPLS domain followed by packets with the same label. A traffic trunk is an aggregation of traffic flows of the same class, which are placed inside an LSP. Therefore, all packets on a traffic trunk have the same label and the same three-bit class-of-service field in the MPLS header. Traffic trunks are routable objects like virtual circuits in ATM and frame-relay networks. These trunks can be established either statically or dynamically (on demand) between any two nodes in an MPLS domain. A trunk can carry any aggregate of microflows, where each microflow consists of packets belonging to a single TCP or UDP flow. In general, trunks are expected to carry several such microflows of different transport types.


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IP-Based Wireless Networks

The definition and specification of the all-IP network is now taking place in 3GPP2, and some equipment suppliers have proposed a phased, evolutionary stepwise migration toward the all-IP network. This evolutionary approach makes it possible for operators to choose a migration pace that fills their unique needs. Packet Core Network (PCN) will serve as the cornerstone in an operators strategy to move from todays circuit-switched environment toward the Internet-enabled world of tomorrow. IP-based transport protocols are introduced in the transport and signaling legacy network. The anticipation is that the signaling will still be access specific while carried over an IP protocol. Payload protocols will most likely be evolved from current IETF protocols and initiatives, taking into account wireless requirements. IP-based transport protocols are introduced in the legacy core network. More specifically, it is proposed to carry VoIP instead of PCM-structured trunks. The protocols used in the CS network should be to the largest possible extent based on open IETF-based protocols. In the current packet data architecture there is an inherent tight connection between the CS domain and the PS domain. The current architecture requires the MSC for setting up and maintaining packet data services. Legacy voice services should evolve toward IP transport, and PS and CS services will evolve independently toward IP. Furthermore, the packet data services should evolve from the current architecture. The tasks that in GPRS are performed by the SGSN (e.g., packet session control and mobility management) are handled by the MSC in a CDMA2000 system. As a consequence, it has been shown that it is difficult to make significant changes to the session control and mobility management of packet data, because it has an immediate effect on the A interface, the MSC, and on ANSI-41. To replace this MSC-centric model so that it can serve all the foreseeable and unforeseeable features needed for multimedia services, a new entity called access control server (ACS) has been proposed. The ACS can be seen as an entity that manages the control part of the MSC for packet data without being involved in the actual switching or routing of user packet data. By introducing the ACS, the PS domain will be free to evolve independently of the CS domain, thereby enabling a smooth migration to all-IP. The multimedia call models are provided in the packet core network. The network is not only a bearer network, but also an IP multimedia service provider network able to provide multimedia services to users. Signaling gateways and media gateways are introduced in order to provide seamless

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interworking between VoIP and the legacy ISUP-based networks, such as the PSTN. The shortcoming of so-called best-effort packet data is that it has limited ability to distinguish between or prioritize different types of payloads. This is sufficient for many types of traffic, such as Web browsing, e-mail download, or file transfers. On the other hand, other types of traffic, such as voice or videos have their own special requirements with respect to delay, jitter, and bit errors. For this reason it is necessary to introduce two controls: • QoS control differentiates among different types of payloads, ensur-

ing that each is handled according to its own QoS profile. • Admission control monitors the available free bandwidth and does not admit more traffic than can be handled. In order to gain from the service flexibility that IP as a service mechanism offers, the transparency of the IP protocols must be maintained. This costs spectrum in terms of packet overhead. Using the IP protocol family implies a high ratio of overhead, especially when considering narrowband services like voice. Voice CODECs generate frames of typically 15 to 30 bytes every 20 to 30 ms, whereas the Realtime Transport Protocol (RTP), IP, and Transmission Control Protocol (TCP) overhead (IPv4) is 40 bytes. Hence, we have more than 50% overhead in the VoIP. The IP header is approximately equal to 20 octets, and a speech frame is 35 octets (i.e., the IP header gives a significant overhead). In narrowband systems where transmission efficiency is important, header compression must be used. This will reduce the header to 3 or 4 octets. There are proposals for an IP header compression standard, but no standard has been accepted yet. The header compression is used on a per-link basis, which means that it does not have to be used on all links. Companies have been actively contributing to Internet Engineering Task Force (IETF) in order to standardize robust IP header compression algorithms that address these problems, and soon after IP header compression for wireless applications is fully standardized, it will be easy to implement in the base station and mobile terminal. Every host [RBS, processor in BSC, operations support systems (OSS), and so on] will have an IP address. The BSC processors have the responsibility for call control in the Base Station Subsystem (BSS). But the BSC will have no responsibility for the Operations and Maintenance (O&M) for the RBS. The OSS and the RBS will handle all O&M functionality. An IP network example is given in Figure 3.11. The IP network clouds shown could


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Internet

PSTN MSC CPS

GGSN SGSN

ATM IP

BS RBS

PDH/ SDH

TDM pipes for ATM

IP

IP RBS

BSC BS

ATM

ATM

RNC

Figure 3.11 IP-based wireless system.

be existing BSS transmission networks or they could be new Intranets built for the wireless operators data-communications usage. The basic idea is that the BSC consists of a number of processors that communicate with RBS using IP. Real-time traffic must have priority over TCP traffic, in order not to get too long delays. Some routers have RSVP and priority already implemented. Within the IETF there is a lot of activity within the field called differential services. The idea is to define something fairly simple and straightforward so that the present routers can be software upgraded. In a packet network the delay is never constant, even if the packets are delivered with priority. Therefore, the end points must have buffers to be able to cope with the delay variation. The question is how big is the variation and how big should the buffers be. If the absolute time is known, the delay variation can be kept at a minimum. A long packet can delay the sending of shorter speech packets, if the long packet started to be transmitted when the speech packet queue was empty. On a 2-Mbps link, 1,600 octets will take 6.4 ms to be transmitted and accepted. On a 384-Kbps link, 1,600 octets

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will take approximately 33 ms to transmit, which is too much delay for the speech packets, so the long packets must be fragmented to a suitable size depending on the link rate. Certain requirements of the radio access network set a challenge to IP deployment. A traditional packet network cannot carry synchronization signals, and special means must be provided to base stations. Current strict delay requirements and low-bandwidth transmission links between the RBS and BSC need careful network planning to enable smooth evolution toward IP.

3.7 Complete T1 Tutorial 3.7.1

Signals in a T1 Network

The telecommunications user company must design its own facilities, purchase the correct products from various vendors, install and test the products properly, and document what it has for future reference. This section addresses engineering planning and design of a T1 line so that service can be delivered on a more predictable schedule. Details of the T1 systems, used mainly in the United States, Canada, and Japan (but also in Jamaica and some other countries around the world), are presented in the material that follows. The 1.544-Mbps bipolar PCM line signals of T1-type terminal equipment (such as channel banks, data terminals, and repeatered lines) are designated DS1, meaning digital signal, level one. At the standard cross-connect point for DS1 signals, DSX-1, the voltage level of pulses is about +3 volts and −3 volts. DS1 means that the signal meets the interface specification for DS1 signals (1.544 Mbps, bipolar +3 and 3 volt pulses, 50% duty cycle, and so on). The data pattern means that the 192 bits of payload data contain live traffic (thats 24 timeslots and 8 bits per timeslot) or some test pattern. After each 192 bits, there is a framing bit for bit number 193. That bit is either 1 or 0, depending on whether it must follow the special pattern for superframe format (SF, also referred to as D4) or ESF [13]. For details see Figure 3.12. This 193-bit frame is repeated 8,000 times per second. This is also called a frame rate of 8 kHz. Each of 24 channels is sampled and transported 8,000 times per second. If a terminal receiver cannot determine the start and finish of a frame, then receivers will always be out of frame sync with respect to transmitters, and very poor performance is noted. There is one more qualifier on the signal, and that is line code. The two line codes are alternate mark inversion (AMI) and bipolar eight zero substitution (B8ZS). This is very important to network elements that carry live traffic.


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One DS-1 frame 192 payload bits (typically 24 samples of 8 bits per sample)

193rd position for frame bit

Figure 3.12 Pulse-code modulation.

In the data communications world, all digital data is either 1 or 0, nicknamed either mark or space. If the data has a pattern of all 1s, then the first 1 is transmitted on the AMI transmission link as +3 volt pulse, the next 1 as −3 volt pulse, and the next 1 as +3 volt pulse. Internally to the electronics of equipment, the 1.544-Mbps signal might be unipolar, but everywhere accessible to the user, the signal is bipolar. DS1 signals may fall into one of three categories: SF, ESF, and possibly unframed. Sometimes test equipment will use an unframed DS1 test signal to check a transmission facility, but most terminal equipment must be selected for SF or ESF only. If terminal equipment expects to see SF and it receives ESF instead, it will probably not be able to lock its framer circuit onto the input signal and everything will stay in alarm. A T1 line provides a physical four-wire transmission path for cable carrier systems that transmit bipolar pulse streams at bit rates of 1.544 Mbps. At each CO, cable pairs connect to an office repeater. Between offices, line repeaters are located at nominal spacing of 32 dB at 772 kHz. Automatic line buildout (ALBO) equalizers in each repeater can compensate for a range of losses in the preceding cable section. Note that the overview diagram in Figure 3.13 is simplified by the use of one line in each direction to symbolize one twisted pair of wires. Other diagrams use one line to indicate both directions (two twisted pairs), and still other diagrams use four lines to indicate both pairs. The nature of the diagram dictates which symbol convention is used. The triangular symbol indicates a digital regenerator, which is somewhat related to an amplifier. Note that office repeaters use a regenerator in only one direction, and line repeaters use regenerators in two directions. A T1 channel service unit (CSU) interfaces a typical piece of customer equipment (such as a channel bank) with a public T1 transmission facility. There are several specific types of CSUs and the most common type has a DS1 signal interface on each side of the unit (the data terminal side and the

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End office Line repeaters

DSX-1

End office Line repeaters

DSX-1

Mux or Channel bank

Mux or Channel bank

DSX-1

Intermediate office

Office repeaters

Office Office repeaters repeaters

Office repeaters

Figure 3.13 Transmission paths of a PCM system.

network side). In some respects, this CSU resembles an office repeater; however, additional diagnostic features are present on most CSU products. One of the most common functions is that of ones density enforcement (also referred to as ones stuffing). Customer DS1 signals containing very long strings of 0s are not allowed onto the public network (since there may be network elements that require occasional 1s for timing purposes). As a result, most CSU devices change a 0 to 1 to suppress the 16th consecutive 0, depending on specific requirements for a public facility. If the traffic has simply voice circuits, an occasional forced error of this sort is a negligible problem. If, on the other hand, the traffic is high-priority unrestricted data, then an occasional error is not acceptable. If this is the case, then the line code must be selected as B8ZS and not standard AMI. B8ZS uses selected bipolar violations (BPVs) as a means of signaling the far end that strings of 0s are in the customer data. Therefore, no errors are introduced and the system works correctly. If B8ZS line code is used, then each and every network element in the path must be provisioned for B8ZS instead of AMI (or else the element must completely ignore line codes). Note that some T1 office repeaters are aware of B8ZS and others are not aware. Note that one line code or the other will not disrupt these repeaters, but many have BPV monitors that will detect the unintentional BPVs but ignore the intentional BPVs that are part of B8ZS coding.


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Pulse Transmission

Pulses generated by the terminal equipment (e.g., channel bank) and repeaters are subject to distortion by attenuation and phase characteristics of the cable. In the line and office repeater units, just preceding the actual regenerator is an ALBO equalizer, which restores adequate pulse shape for detection and regeneration. Pulses generated in the terminal equipment must reach the office repeater in a predictable fashion, even if it is in the same room. In many office repeaters, the line build out (LBO) setting tells it through how many feet of cable the signal has traveled since the DSX-1 cross connect. The office repeater adapts to that attenuated signal. Most LBO settings assume the use of 22-gauge twisted-pair cable (with around 1416 pF/ft of capacitance). Note that 24-gauge cable has approximately 25% more loss than 22 gauge. By default, many office repeaters are shipped set for 0 to 133 ft of cable (assuming 22 gauge). For transmission calculations, the cable attenuation at 772 kHz is used. This is half of the 1.544-Mbps clock rate, but this is valid because the power spectrum of the pulse stream is maximum at approximately 772 kHz. 3.7.3

BERs

Pulses sent along the repeatered line are regenerated at each repeater point. The repeater looks at each timeslot and decides whether or not a pulse is present. If the logic circuit determines that there is a pulse, the repeater outputs a new pulse that is free of noise, distortion, or interference. As a result of many possible factors, a few pulses may be incorrectly regenerated. A 0 is sent instead of a 1, or vice versa. The ratio of error pulses to the total number of timeslots is called the error rate or the BER. One error in 1,000 is referred to as BER 10−3; one error in a million is BER 10−6. −6 −7 For strictly voice circuit application to T1, a BER of 10 or 10 might −8 −9 be considered acceptable performance, although a BER of 10 or 10 might be required for some data purposes. In the most common systems, once live traffic errors have entered into the transmission stream, they cannot be sorted out or corrected. Error rates tend to accumulate through an end-to-end system, although the rates tend to be low for digital systems as compared with analog systems. Voice is more forgiving of transmission errors or high BER than is data (also called nonvoice content). The reason for this forgiveness comes from the ability of the human brain to reconstruct the missing parts of conversations. If a syllable or even a word is dropped, the brain assumes the missing information from the context of the conversation and reconstructs it. This

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enables people to communicate through the noise or the breakup of a marginally understandable mobile radio call. The transmission of data does not provide this reconstruction advantage. If nonvoice content is lost, it can only be recovered, if at all, through sophisticated error-correction algorithms. Such error corrections add overhead that slows the true data rate, or transmission rate, of the desired content. The greater the number of errors, the more error correction required and the slower the true data rate. 3.7.4

Overall System Length

Each repeater in the series adds a small amount of jitter to every pulse of the bit stream. A conservative limit of 200 tandem repeaters in a system ensures that the accumulated jitter will not exceed the synchronization capability of terminal equipment or higher-order multiplexers. Based on the accumulation of error rates in tandem repeater sections, and particularly in end sections (the repeater section next to the CO), the system should not include more than ten tandem span lines (nine intermediate offices). Long-range design of the span line is necessary to plan the expected cross section of the span. The selection of one-cable or two-cable operation, locations for the line repeaters, and repeater section length will depend on the future requirements of the route. Also, the cable plant must be carefully studied in terms of the number, type, and age of cables; freedom from bridge taps and branches; splicing integrity; suitability for line repeater locations; and minimum exposure to electrical and mechanical hazards. Major factors that control the design of the span include the following: • Ultimate number of systems within the cable; • Cable-pair attenuation at 772 kHz; • Cross-talk coupling loss between cable pairs; • CO noise; • Ambient temperature range.

If a maximum length section is engineered, cable loss should be measured before the final repeater location is established. This allows for changes and small errors. Note that aerial cable can be exposed to higher ambient temperatures compared with buried cable and ducted cable. Higher temperature equates to higher copper attenuation and dc resistance. For design purposes,


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the losses for aerial cable are estimated at about 5% higher than for buried cable. 3.7.5

Single-Cable and Dual-Cable Operation

In normal one-cable operation, low-level repeater inputs and high-level repeater outputs appear at the same point of the cable. As a result, near-end cross-talk (NEXT) is the limiting factor in repeatered line design. The number of systems that can be installed in a single cable is mainly controlled by the physical separation of the pairs in the two directions of transmission. Greater separation increases the coupling loss, resulting in decreased interference. A general rule is that if transmit and receive pairs are in the same cable binder group, the maximum section loss should be reduced to 15 dB to prevent cross talk. In two-cable operation, NEXT does not limit the number of systems for one cable. The choice of one-cable or two-cable operation is based on cable route, circuit requirements, availability of suitable cables, and economics. A span line includes one or more repeater sections. Typical maximum section loss is 32 dB (normal section) measured at 772 kHz. In the end section next to the CO, maximum loss is limited to 50% to 70% of the normal loss limit, or about 23 dB. In some early line repeaters, the ALBO equalizer has a range of 6 to 31 dB. In many newer line repeaters, the ALBO has a range of 0 to 35 dB, meaning the signal can be attenuated through up to 35 dB of cable loss and still become regenerated properly. In practice, repeater sections are designed with a safety factor of several dB, so spacing of 28 to 32 dB is quite common. The minimum section loss is frequently set at 9 dB because of repeater design and to attenuate reflections. In other words, loss is generally added via an XMT span pad or via a 7.5-dB equalizer in the office repeater for the end section. Equalizers help control NEXT by simulating cable attenuation characteristics, whereas the pad displays flat loss. 3.7.6

T1 Repeatered Lines

In the case of a CO-to-CO T1 repeatered line, there is normally one originating office repeater and one terminating office repeater. The convention is that the originating end powers the line with simplex readings of +V and −V, where the −V is typically 10 volts more in magnitude with respect to the +V. If both ends originate power, then there may be a simplex power loop strap set at one of the line repeaters out in the middle of the span. Note that the

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voltage will appear at the span line interface even if that is open loop. Current will not flow into an open loop, however, so the voltage sensed across the 10-ohm current sensing resistor is the best indication of the validity of the simplex current. In one-cable operation, errors can be caused by NEXT between cable pairs in opposite directions of transmission. Physically separating the different groups of pairs as much as possible is preferred. Refer to Figure 3.14. Shortening the repeater spacing will reduce the level differential and the sensitivity to NEXT. One single T1 system in a cable has only itself with which to interfere (transmit signal with respect to receive signal). Twenty systems in a cable makes many more cases for NEXT. Not only does one single system have itself for interference, but 19 other systems are also producing NEXT. If a particular cable use is expected to grow over time to its maximum capacity, then the correct safety margin must be calculated into the design. As stated earlier, in the end section next to the CO, maximum loss is limited to 50% to 70% of the normal loss limit (32 dB), or about 23 dB. The 23-dB limit includes the loss in the tip cable and in the office wiring to the line-terminating shelf for office repeaters. If exchange service is mixed in the same cable with T1, then an extra safety factor of 8 dB should be allowed to tolerate impulse noise. This brings down the end section to a maximum of 15 dB. An end section of 15 dB represents approximately 3,700 feet of cable, depending on the exact type. If this cable distance is too short to be

System 1 West-East

Transmit West-East System 2 West-East System 2 East-West Transmit East-West System 1 East-West Twisted copper pairs D shield Cable sheath

Figure 3.14 Typical cable layout.


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economically tolerable, then consideration should be given to leaving out the exchange service pairs to gain back the 8-dB safety margin. Office wiring typically goes from the cable vault directly to the office repeater shelf, from the office repeater shelf to the automatic protection switch (if equipped), and from there to the channel bank, radio MUX, or other terminal equipment. Along the way, there may be one or more DSX-1 cross-connect panels. Western Electric ABAM cable is a traditional selection to connect all of these elements. ABAM is a designation for 22-gauge, 110-ohm, insulated, twisted-pair cable. Use a loss figure of approximately 0.4 dB per 100 feet of office cabling, assuming 22 American wire gauge (AWG), remembering that 14 to 16 pF/ft is the normal cable capacitance. Cable with higher capacitance will give problems on the longer cable runs. CAT 5 LAN cable was not intended for T1, but it has low capacitance, so it makes a good substitute. In some rural service areas, lightning strikes aerial cable so frequently that one extra measure is applied at the cable vault. The outside cable might be 22 gauge and the tip wiring to the line terminating shelf might be 22 gauge, but one short section of higher gauge (smaller diameter) cable is added at the cable vault. This 2550 ft length of 24- or 26-gauge cable is called fuse cable. As its name implies, it acts as a fuse element that will open when huge currents from lightning appear. This technique is effective in keeping lightning surges out of the CO, but it has side effects. The 24- or 26-gauge insert may be short, but it interjects one extra attenuation factor in cable loss estimation. Furthermore, when the fuse cable opens up, that pair must be abandoned. If route junctions are present along the cable, then it is advisable to adjust repeater section spacings to place repeater housings at the junctions. Otherwise, there is greater possibility of signal-level differences from adjoining cable branches, which would contribute to cross talk. If a repeaterless junction is necessary, T1 span pads may be used to equalize levels between branches. Manufacturers of repeater housings typically offer several configurations depending on such factors as the number of line repeater units in the nest, whether there are attached stub cables, the length of such stubs, and the presence or absence of lightning protectors. It often seems as though each repeater-housing vendor has chosen to describe T1 pairs by different methods. It is important to be aware that in a single direction, the transmit pair from one repeater becomes the receive pair 6,000 ft away. The east-west transmit is separate from the west-east transmit.

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3.7.7


Order Wire

The order wire is simply a loaded voice frequency pair that runs through the T1 cable and appears inside each line repeater housing. When one technician is troubleshooting at the remote housing and one technician is at the CO, it is very convenient to have a place to plug in a technicians telephone (butt-in phone). The order wire is typically terminated to an automatic ringdown line at the office repeater area of the CO. If the copper order wire is not implemented (as it is usually the case with the new wireless network) then the technician at the remote cell site must have a two-way radio or cellular telephone with plenty of battery and talk time. 3.7.8

Lightning

Lightning is an electrical discharge pulse in the atmosphere, which averages 20 kA or more in current. Commonly, these discharges are viewed as a flash from cloud to ground, although it can also work in reverse. Once ionization of the atmosphere occurs, this becomes luminescent, conductive plasma (a lightning bolt) sometimes reaching 60,000°F. Lightning can deliver a tremendous discharge of energy at any grounded object. Lightning strikes are somewhat predictable over a geographical region. The isokeraunic level is the number of thunderstorm days per year. This isokeraunic number varies from over 100 along the Gulf Coast of Florida to less than 5 in the Pacific Northwest. Nevertheless, virtually all areas in North America are subject to lightning strikes to some degree. Exterior equipment, rooftop equipment, and equipment connected to aerial and buried copper cables are subject to possible damage. In low-lightning areas, protected-type line repeaters are used along aerial cable and unprotected-type line repeaters are used along buried cable. In contrast, in areas of heavy lightning activity, it is quite common to use protected-type line repeaters regardless of whether the cable is aerial or buried. Aerial cables are especially susceptible to lightning strikes. The huge energy pulse from lightning momentarily raises the potential of ground at that strike point and then travels along the copper pairs to the CO where it finds a lower ground potential. At the CO, primary protection consists of three-element gas tube protectors, typically installed at the well-grounded protection frame or near the cable vault. In some COs, gas tubes are installed at the top of the relay rack with office repeaters. Primary protectors must be present and grounded properly. Many items of transmission equipment, such as office repeaters, have secondary protection in the form of solid-state surge limiters, but they are not effective if primary


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protection is bad. When the lightning voltage causes the primary gas tube to conduct, it represents a short circuit to the span-powering regulator. In a typical T1 transmission line, span power is fed only from the CO. In some cases, additional power comes from the far end unit. During normal operation, the 60-mA simplex current flows normally, the gas tubes sit idly, waiting for a lightning strike, and the DS1 traffic moves along. In the instant of a lightning strike (somewhere mid-span), the lightning acts as a huge current pulse that raises the ground potential at that strike location. If the cable is not properly grounded everywhere, the lightning can enter the copper pairs and flow toward the best ground point, which might be toward the nearest CO or, in the other direction, toward the remote end. The lightning might be in the form of a metallic voltage appearing from tip to ring on a pair. It might be in the form of a metallic voltage appearing from tip to ground or from ring to ground or it might be in the form of a longitudinal current surge. If a metallic voltage appears at a three-element gas tube protector (called the primary protector), the tube will fire either tip to ground, ring to ground, or tip to ring. This assumes that the gas tube is both working and grounded correctly. Some companies, however, fail to periodically test their gas tubes with a gas tube checker. If measured with a simple meter, gas tubes appear to be an open circuit whether they are working or not working, making it unreliable. If the gas tube has an improper voltage rating, it will not work correctly. If the voltage rating is too high, lightning voltage can seep in before it fires (thereby stressing equipment). If the voltage rating is too low, the normal dc voltage applied, at the simplex power feed end of the span, is enough to set it off prematurely, or at least to hold the gas tube in glow mode after the strike (thereby forcing a failure situation after the lightning strike). When the gas tube fires on schedule, it is effectively producing a short to ground. If there is some span power feed repeater nearby, this acts as a dead short on its current loop, which causes a big current surge. Various T1 products have built-in secondary surge protectors to withstand this secondary surge, called the current surge. But if the primary surge protector fails, it will most likely burn out the equipment or anything around it. In some cases, this surge protector is in the form of extra series resistance to limit current peaks. Otherwise, this protector is in the form of a fuse that will open up at a high current point. In yet other cases, the protector is a combination resistor and fuse. As a general rule, however, two-element protectors are not recommended for T1 circuits. Two-element protectors are suitable for POTS on a two-wire circuit. Due to the nature of T1 and its simplex current

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loop, equipment may be damaged through the use of anything other than three-element gas tubes. Some telephone operating companies have the policy that T1 is only placed on new cables, dedicated for T1. This is because the headaches of rehabilitating old exchange cable can be severe. The job of eliminating every last bridge tap is difficult on some older cables. Impulse noise from ordinary analog subscriber loops can become a problem when mixed side-by-side with T1. 3.7.9

Simplex Power Design

Office repeaters are located at the CO where noise-free 48-VDC power is available. Line repeaters seldom have any local source of ac or dc power. Instead, the power for line repeater electronics is fed down the copper pairs from the office repeater. Inside a line repeater, there are two digital regenerators designated side 1 and side 2. Line repeaters are powered by dc current flow through a loop formed from the simplexes of the two cable pairs associated with side 1 and side 2. For proper operation, the line repeater must have current flowing through it in the right polarity. In this case, the repeater represents an equivalent resistance of 100 to 120 ohms on a common 60-mA simplex loop. Note that in most cases the loop must be completed at the office repeaters for this to be a valid loop, and office repeaters must have some type of switch or jumper specifically for this purpose. In a few cases, however, the simplex loop is not made this way and the power is fed all the way into some CPE terminal equipment where the loop is made there. However, these cases are not common. After this simplex power loop is engineered and installed correctly, a voltage drop can be measured across side 1 of a line repeater (about 7 VDC). Similarly measure from the span-receive side of the office repeater to the span-transmit side (about 712 VDC). In many far-end office repeaters, T1 channel service units, smart jacks and network interface units (NIUs) that receive power from the span line, this simplex arrangement presents a voltage drop of 11 to 12 VDC. This is one method of verifying that the current has the correct polarity. If it is wrong, then the voltage measurement is only 0 to 1 VDC. The positive current flows in the same direction as the PCM signal direction. In the classic T1 repeatered line, the simplex power planning must account for the equivalent dc resistances of office repeaters, line repeaters, attenuation pads and equalizers, and the copper conductors themselves (Figure 3.15). The longer the span line, the more repeaters must be in series; hence, more voltage must be applied at one end to feed the current loop. In

Transmission-Network Principles Network interface unit (NIU)

Office repeater

C.O. switch

Current + regulator −

Frm Gnd 48VDC Gnd

113

Optional CPE power

Simplex x loop selected

CPE

Frm Gnd Gnd 48VDC Optional

Figure 3.15 Shortspan power feed.

a short-length repeatered line, this voltage might be only 20 to 30 VDC, but as the length is stretched out to 10 to 15 mi, this might become 130 VDC. As it gets extremely long, simplex current might be fed from both ends, with the simplex looping back both ways in the middle. If these dc calculations are made and there are no further engineering guidelines, it is assumed that 22-gauge cable has an equivalent dc resistance of about 18 ohms per 1,000 ft. Of course, 24- and 26-gauge cable have much higher resistance. Assume that each line repeater is 120 ohms and one unpowered office repeater is 170 ohms. There are small additional resistances for LBO networks and other pads in the circuit, and these must be added into the calculation. If you know this total equivalent resistance in the loop and you know the current is 60 mA, then applying Ohms law will result in the minimum necessary voltage. Note that many of the most modern automatic span powering repeaters simply need local 48 VDC and they will develop the necessary voltage to regulate 60 mA into the simplex loop up to a maximum loop of 4,000 to 4,200 ohms. However, in many high-rise building installations, the T1 span line is rather short, perhaps from the basement equipment room to the tenth floor, and therefore no line repeaters are needed. Calling it the repeaterless T1 span line, we might see an office repeater at the near end and a smart jack or NIU at the far end, only 3,000 cable feet away. In this short-span-line case, the

114


voltage necessary to drive the simplex current loop is only 20 VDC, but the current loop must still be connected at the ends for the current flow to be correct. Transmission people will often refer to simplex power, loop current, and simplex voltage, using these terms interchangeably to mean about the same thing. Obviously, the voltage applied to a loop resistance yields a loop current. It is important to keep in mind that equivalent resistance through cable is based on one twisted pair acting as parallel resistance (resistance of one wire divided by 2). However, the simplex loop current must pass out the cable length and then pass back the same length (× 2). This effectively makes equivalent resistance the same as one wire for the one-way length. 3.7.10 Data Error Rates

Pulses sent along the repeatered line are regenerated at each repeater point. The repeater looks at each timeslot and decides whether or not a pulse is present. If the repeater logic determines that there is a pulse, the repeater puts out a new pulse free of noise, distortion, or interference incurred in the preceding repeater section. Owing to degradation factors, a small number of pulses may be incorrectly regenerated; that is, a pulse will be transmitted where it was not present or vice versa. The BER is the ratio of transmitted time periods (pulse or no pulse) that are received incorrectly at the end point to the total number of time periods. The total error rate is the arithmetic sum of the error rates of the individual repeater sections. Because of the effect of impulse noise from CO equipment, end sections (the section nearest the CO) are the principal source of errors and, therefore, are shortened to increase the signal-to-noise ratio at the office repeater. Between the terminal ends, a maximum error rate of 1 in 106 (BER = 10−6) will result in good voice communications. Pulse errors cause transients in the individual voice channels, but at this rate they are not noticeable to the average listener. On the other hand, while a higher error rate of 1 in 105 (BER = 10−5) will result in acceptable voice transmission, audible clicks are noticeable. Many data systems are less tolerant of bit errors, and facilities are engineered to meet error rates of 1 in 108, 109, or 1010 wherever possible. Any bit error on the span line results in a BPV, which can show up in a T1 span line. A BPV occurs as a result of bad wiring connections anywhere. A normal DS1 signal uses alternate mark inversion as the line code. In other words, if the terminal data stream is 11111111, then the line code is transmitted as marks (pulses) of alternating polarity, so we would see +1, −1, +1, −1, +1, −1, +1, −1. If a BPV has been created, we would see two consecutive pulses of the same polarity. In this case, we might see +1, −1, +1, +1, −1,


115

+1, −1, +1. B8ZS is a different line code that intentionally sends and receives BPVs in a specific pattern to carry a special meaning related to 64-Kbps clear channel unrestricted data. Most basic transmission line elements such as line repeaters are completely transparent to any type of framing (SF, ESF, or otherwise). In contrast, most pieces of DS1 terminal equipment, such as channel banks, are very sensitive to proper framing format. Between these two types of elements exist intermediate elements, such as higher-order multiplexers and automatic protection switches. These elements may or may not be sensitive to framing format and some even convert from one framing format to another. A few also have the ability to autoconfigure depending on the signal format that is first received. 3.7.11 Voltage and Temperature Factors

Normal simplex loop current is 60 mA for modern line repeaters. Some older repeaters use more current (100, 120, or 150 mA). Occasionally, problems with ac power induction can be overcome by increasing the loop current within the tolerance of the repeaters. However, in the absence of 60-Hz induction, most simplex loops are set between 55 mA and 65 mA. Copper cable tends to have more resistance at higher ambient temperatures (it will take more voltage to drive the constant current through the loop), so knowing the expected cable temperature extremes for a locality helps set a strategy for optimizing loop current for best performance. Aerial cable is exposed to much higher temperatures than buried cable, and this must be calculated into the design. The interesting situation is presented when there are no line repeaters. If the total length of the facility is less than 3,000 to 4,000 ft, then probably no mid-span T1 line repeaters are necessary. In that case, the dc polarity might accidentally be applied backward at the CO and there is nothing on the copper line to fail until it gets to the NIU. So, if the NIU does not work on day one, it must first be established that dc is going correctly (this takes a dc voltmeter across the span side, and tests for a 68-VDC voltage drop), then move to tracing the high-frequency signal. A DS1 signal at 1.544 MHz is pretty unique and can be traced from point to point with the right kind of full-featured DS1 test set, not a receive-only DS1 monitor. It is important to remember that simplex current is only seen from the office repeater or NIU to the outside. Once DS1 signals are inside the CO, inside from the office repeater, then simplex is no longer present. Inside, the DS1 signals are +3 volt and −3 volt pulses with no simplex.

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3.7.12 T1 Engineering, Installation, and Documentation

Once the installation has been completed, the installation crew must supply as-built drawings for files. These may be the original untouched engineering drawings or they may contain notations from the crew for changes made during installation. Examples include setting option switches differently to achieve better performance, using different cable pairs from the plan, and installing plug-in cards into specific slots. In some operating companies, this is done with paper drawings, while in others it is handled exclusively with electronic files. Every installation team and maintenance technician must have a good T1 bit error rate test (BERT) instrument. In many cases, a single BERT is adequate, but in others having one BERT at each end of a facility is an advantage. The best BERTs feature is to have two receive ports instead of one. The second port becomes handy for troubleshooting synchronization problems. In most cases of troubleshooting, all that is needed is a good BERT. A really complex case of finger pointing over the exact DS1 waveform can only be solved by using a good high-frequency dual-trace oscilloscope. With an oscilloscope, use two calibrated high-impedance probes set for A-B, or differential mode. Tie the two probe grounds together at the logic ground of the device under test. That way it can be checked if there is a dc offset voltage present. The new generation of BERTS can also show the DS1 waveform on the small LCD screen and perform detailed analysis based on applicable standards. The transmission acceptance test procedure (ATP) on T1 circuits will be described in later chapters of this book. 3.7.13 Troubleshooting and Problem Classification

Within the telephone industry there exists a standard terminology problem. In some places, the abbreviation T&R stands for tip and ring, which identifies one wire from the other wire in a two-wire pair; elsewhere T&R stands for transmit and receive. Frequently, these are not single wires. In DS1, these are a transmit pair and a receive pair. Of the transmit pair, there is a tip wire and a ring wire. Sometimes on a schematic they are named TT and TR (for transmit tip and transmit ring) and then RT and RR (for receive tip and receive ring). On still other equipment these are named T and R on the transmit pair and T1 and R1 on the receive pair. Again, recall that the transmit signal at the near end becomes the receive signal at the far end. In some places transmit is abbreviated as XMT or Tx, and receive is abbreviated as RCV or Rx. Unfortunately, there is no consistency.


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Another such set of terms is east and west. In the early days of longdistance circuits, AT&T Long Lines developed standard terminology to explain which long-haul equipment at one town was connected to other equipment located in the next town. All transmission directions were arbitrarily called either east-west or west-east. Therefore, it is quite common to be tracking a signal from one office repeater shelf marked east, meaning this is facing the span to the east. It might then go to another office repeater shelf marked west, meaning this is facing the span to the west. It is often more foolproof to simply draw lots of arrows on a circuit diagram, indicating the direction of all signals. Backbone microwave systems connecting cities use the same nomenclature. Exact symptoms are extremely important, since an intermittent problem is much harder to fix than a constant one. If the problem is intermittent, then its pattern of occurrence must be made reproducible. In other words, if it fails once per day every day, then that is significant. If it fails only on the hottest day of the year, something else is indicated. It is more difficult if it is an intermittent noise problem on a DS1 circuit, and in general, with an intermittent signal, it is important to keep testing it until it fails consistently. To a large extent, on DS1 signal it either works or it doesnt work. Rarely is there a marginal noisy problem except with grounding problems. If there is a clean BER with one legal DS1 test pattern, but a dirty BER with another legal pattern, often this is a clue to a bridge tap on the copper pair. As a general rule, BPVs are generated from two sources: a poorly performing T1 line repeater or bad copper T1 line pairs, or it is a problem within the last few feet of DS1 jumper wires, such as a single broken wire (not a broken pair) at a wire-wrap post. BPVs can also be caused by a shiner, a wire where the plastic insulation has been sliced off to reveal the shiny tinned copper conductor. Shiners can easily make an intermittent short to a grounded shelf or anything else. Thermal intermittent problems are hard to reproduce without a lab temperature chamber. It may help to have a can of freeze spray, but this is not usually a solution. Fortunately, with modern low-power components, thermals are not now common. An easy test is to tap an intermittent shelf with the handle of a screwdriver to see if some intermittent component will produce a bit error. Many systems and products include complete documentation, covering acceptance testing, while individual circuit packs may have their own segmented card-level tests. Once everything is hooked up end to end, a real-world test is necessary to verify that the entire system is fully ready to support live traffic.

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3.7.14 Switch Options, Line Codes, and Framing

With DS1 signals inside the CO, it is necessary to verify where the framing format is SF (also erroneously called D4) or ESF. In some cases, it may not matter (they automatically detect and set SF/ESF), but it is helpful to know (line repeaters and office repeaters generally dont care). It is important to verify whether the line coding is AMI or B8ZS. 3.7.15 Fractional T1

In the early 1990s, fractional T1 (F-T1) became very popular and many operators began offering this service. In this way, customers could lease a portion of the T1s 1.544-Mbps bandwidth at a fraction of the cost of a full T1. Compared with F-T1 service, full T1 links were still more cost-effective if the bandwidth was almost fully used or if the links were fairly short. For most applications today, F-T1 local loops are rarely used, as they are neither economical nor practical. 3.7.16 T1 (J1) in Japan

The basic format for T1 transmission facilities in Japan is similar to North American ANSI standard and called J1. However, the CMI line coding used in Japan is different from the one used in North America [14]. Rather than using 50% duty cycle (half width) mark pulses, continuous marks are transmitted, inverting the voltage after each mark. The spaces also have a transition, but in the middle of the pulse period. This ensures that there are always sufficient transitions to keep synchronization, regardless of whether the signal is all 0s or all 1s. Japanese digital hierarchy (J1) is shown in Table 3.5. Table 3.5 Japanese Digital Hierarchy

Designation

Bit Rate (Mbps)

Number of Voice Channels

DS1

1.544

24

DS2

6.132

96

DS3

32.064

480

DS4

97.728

1,440

DS5

400.352

5,760


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3.8 Complete E1 Tutorial 3.8.1

Introduction to E1 Networks

The E1 (CEPT) digital transmission system is the lowest level of the European digital communication hierarchy. An E1 facility provides full-duplex transmission at 2.048 Mbps. Bandwidth is divided into 32 multiplexed (byte-interleaved) 64-Kbps channels. Depending on the framing format, one or two of the 64-Kbps channels are used for framing and other overhead functions. The remaining bandwidth (1.920 or 1.984 Mbps) can be used for either voice or data information. For digitized voice applications, the information bandwidth typically consists of 32 multiplexed 64-Kbps channels. For the transmission of data, the information bandwidth may be channelized as for voice, or it may carry from one to as many as several hundred multiplexed signals on an unchannelized basis. 3.8.2

Customer Premises Equipment

Various types of equipment (CPE) are employed at the customers location. Data terminal equipment (DTE) provides the source for the transmitted signal and the destination for the received signal. DTE includes such equipment as the following: • Multiplexers; • PBXs; • Channel banks; • Front-end processors; • Computers.

Other equipment interfaces the DTE to the public E1 network and provides various termination and interface functions, including the following: • Electrical interface; • Surge and lightning protection; • Signal regeneration and pulse-density assurance; • Keep-alive and yellow signal; • Loopback to the line controlled from the network.

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Specifications for E1 interfaces are defined in International Telephone and Telegraph Consultative Committee (CCITT, an international standards group) recommendations. For instance, E1 signal levels are defined in CCITT Recommendation G.703. Note that CCITT is now called ITU, but the recommendation designations have not changed. 3.8.3

Signal Characteristics

The E1 signals are bipolar pulse trains and the data is assembled using a TDM scheme. When voice is transmitted over E1 networks, it is digitized, using PCM. An E1 signal is divided into timeslots of 488 nanoseconds each, or 2,048,000 timeslots per second. The presence or absence of a pulse in each timeslot encodes data (or digitized voice). A pulse, if it exists, will have onehalf the duration of the timeslot and amplitude of three volts. When a pulse is present, the timeslot data is a 1; when no pulse is present, the timeslot contains a 0. E1 signals employ alternate mark inversion (AMI) line coding, in which consecutive pulses are expected to be of opposite polarity. Consecutive pulses of the same polarity, called a bipolar violation or code violation, indicate a transmission error, except in special density-preserving codes described under HDB3. An example of AMI line coding is illustrated in Figure 3.16. 3.8.4

Transmission Facilities

E1 signals are transmitted primarily over standard twisted-pair copper wire. Signal loss on the wire is approximately 5 or 6 dB per 1,000 feet. Repeaters are typically employed every 6,000 feet along the transmission facility to compensate for the signal losses and to ensure an adequate signal level at the network interface (the termination of the E1 line at the customer premises). Figure 3.17 illustrates a typical E1 network interface. E1 signals may also be transmitted via satellites, digital microwave radios, fiber-optic systems, and coaxial cable modems. In the carrier networks, signals may be multiplexed into even higher-speed signals. Bits

0

0

Signal

Figure 3.16 AMI line coding.

0

1

1

0

1

1

Transmission-Network Principles Customer premises equipment

DTE

DCE

121

E1 line

3000 ft. Repeater

6000 ft. Repeater

Network interface (NI)

Figure 3.17 One end of a typical E1 link.

3.8.5

Pulse Density

In order to interpret and regenerate an E1 signal, repeaters and other network equipment must be able to determine timeslots based on the pulses in the received signal. Since pulses occur only when 1s are transmitted, signals with too many consecutive 0s can cause timing problems. There is no specific pulse-density requirement for the E1 environment. Ones density is automatically maintained by high-density binary 3 (HDB3) coding, as described in the next section. Because there are different national standards for connecting CPE equipment to public E1 networks, often there is a need for two different line interfaces. One is designed for direct connection to an E1 line; the other is designed for connection to an E1 line through an approved networktermination unit. 3.8.6

HDB3

HDB3, which is used in E1 networks, replaces strings of four 0s with special sequence (i.e., fixed codes containing intentional bipolar violations). In other words, HDB3 is a coding method that does not allow more than three consecutive 0s. This code is recommended by the ITU-T G.703 for the 2-, 8-, and 34-Mbps systems [14]. Equipment receiving data containing the special sequences automatically translates them back into strings of four 0s. The special sequences in HDB3 coding are 000V and 100V. In these sequences, V represents a bipolar violation. The choice to use 100V or 000V is made so that the pulses violating the bipolar rule take on alternate +1 and −1 levels. Sequence 100V is used when there have been an even number of 1s since the last special sequence; 000V is used when there have been an odd number of 1s since the last special sequence. Special sequences

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follow each other if the string of 0s continues. The choice of the first special sequence is arbitrary. An example of HDB3 coding is illustrated in Figure 3.18. 3.8.7

E1 Framing Synchronization

In E1, data is grouped into frames of 256 bits. Each frame consists of 32 8-bit timeslots, and 8,000 frames are transmitted each second. This provides for the following E1 transmission rate: 8,000 × 256 = 2,048,000, i.e., 2.048 Mbps 8,000 × 8 = 64,000 (i.e., 64 Kbps) and 32 × 64,000 = 2.048 Mbps A 64-Kbps timeslot has the same data rate as a 64-Kbps DS0 in the T1 North American digital hierarchy. Both evolved from the restrictions imposed by digitized voice. Framing information is carried in timeslot 0 (TS0), while signaling information, if used, is carried in timeslot 16 (TS16). The remaining 30 timeslots carry user information. A group of 16 frames constitutes a multiframe. Figure 3.19 illustrates the basic E1 frame structure. 3.8.8

E1 Framing Formats

E1 defines two primary framing formats: • TS0 framing formats; • TS16 framing formats. 4 consecutive zeros Input bit stream

1 0 1 1 0 0 0 0 0 1 0 0 0 0 0 0 0 0

HDB3-coded bit stream

1 0 1 1 1 0 0 V 0 1 0 0 0 V 1 0 0 V

HDB3-coded levels

− 0 + − + 0 0 + 0 − 0 0 0 − + 0 0 +

Figure 3.18 HDB3 line coding.

Transmission-Network Principles 1

2

TS0

FR 0

3

TS1

4

5

6

8

8-bit timeslot 3.91 µs

TS16

TS2

FR 1

7

123

FR 2

TS31

FR 15

256-bit frame 125 µs

16-frame multiframe 2 ms

Figure 3.19 E1 framing structure

The TS16 multiframe format was developed to provide signaling information to a public switched E1 network. In a switched network, individual 64Kbps timeslots can be routed independently through the network. When TS16 is not used for signaling, that timeslot carries data. As the name implies, TS0 framing formats use timeslot 0 (TS0) to provide the framing pattern that allows equipment receiving the E1 signal to synchronize on the pattern and correctly interpret data. There are two basic TS0 framing formats: • TS0 framing format (without CRC-4); • TS0 multiframe format (with CRC-4).

In the TS0 framing format (without CRC-4), frame synchronization is maintained via the frame-alignment signal (FAS), a bit pattern (X0011011) present in TS0 (bits 2 through 8) in every other frame. Frames that contain the FAS pattern are designated as word frames; frames not containing the FAS pattern are designated as not-word frames. Frame synchronization does not require all 8 bits of TS0 in every frame. Therefore, the TS0 bits have been defined by CCITT for other uses. To use the TS0 bits effectively, CCITT defined a multiframe format. A multiframe consists of 16 consecutive frames numbered 0 to 15. While the FAS distinguishes word frames from not-word frames, the multiframe is found by looking for the 001011 pattern, known as the multiframe-alignment signal (MAS), in bit position one of the not-word frames 1, 3, 5, 7, 9, and 11. This pattern is interleaved with the CRC-4 bits and the E bits.

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As a further refinement, each multiframe is divided into submultiframes, which consist of 8 consecutive frames: frames 0 to 7 and frames 8 to 15. The submultiframe construction, sometimes referred to as the CRC multiframe, is used when CRC-4 error checking is employed. It is the network (or receiving) equipments job to synchronize on the FAS and MAS patterns to identify frames and multiframes properly. An out-of-frame (OOF) event is declared whenever there are three consecutive framing errors. As described in CCITT Recommendation G.732, a framing error is defined as an incorrect bit in one of the seven framing bits in theTS0 word, or an error in bit 2 of TS0 not-word. Three consecutive words or notwords containing errors results in OOF. When CRC-4 is not used, bit 1 of TS0 carries the international spare bits instead. Without CRC-4 coding, only two types of frames can be identifiedframes containing the FAS and frames not containing the FAS. Since the TS0 Multiframe formats are defined in CCITT Recommendations G.704 and G.706, the E1 NSM describes its framing options as G.704 and G.706. G.704 refers to a framing format without CRC-4 capabilities, and G.706 refers to a multiframe format with CRC-4 capabilities. 3.8.9

Spare Bits

The CCITT has set aside a number of spare bits in TS0 that can be used to transmit additional information between end points. These bits include the following: • International spare bits (Si); • Remote-frame alarm bits (Yf ); • National bits (Sa4Sa8).

The following sections provide a brief description of the location and application of these spare bits. 3.8.9.1 International Spare Bits

The international spare bits (Si) are transmitted in bit 1 of TS0. The last two Si bits in each CRC multiframe have been redefined per CCITT Recommendation G.704. When CRC-4 is enabled, these two bits represent CRC-4 error-indication bits (E bits). These bits are transmitted and received in two consecutive frames that do not contain the FAS pattern. In the receive


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direction, a 0 in the E-bit location indicates a far- (or remote-) end block error (FEBE or REBE). In the transmit direction, E-bit locations identify received errored submultiframes by setting the binary state of one E bit from 1 to 0 for each errored submultiframe. The delay between the detection of an errored submultiframe and the setting of the E-bit indicating the error state is less than 1 sec. If CRC-4 is not enabled, the Si bits are differentiated as either odd or even, depending on their frame. The odd Si bits can be used as one 4-Kbps data link, and the even Si bits can be used as a different 4-Kbps data link. 3.8.9.2 Remote-Frame Alarm Bits

Bit 3 of each not-word frame has been defined as a remote-frame alarm bit (also known as a remote or distant FAS). This bit is set to 1 to indicate loss of frame alignment with the received signal. 3.8.9.3 National Bits and Performance Monitoring

The strategic importance of communications continues to grow at a rapid pace. In todays marketplace, corporations look to their high-speed digital networks to give them a competitive advantage. With so much depending on their networks, managers are becoming more and more quality conscious, and when it comes time to implement a new E1 circuit, the quality of the service is just as important as the cost. Because quality is of such concern, users need better ways to track the performance of E1 circuits offered by a service provider. A corporation that manages its services closely is more likely to get the most out of its network and is also likely to get a quicker and more congenial response from a carrier when problems do arise. A user who relies on the carrier to monitor circuit quality is literally putting the business in the carriers hands. In the United States, AT&T originally tariffed T1 in a format known as D4, which did not provide for in-service monitoring of the T1 circuits. As the requirement to meet contracted performance levels for T1 became a more significant factor, AT&T developed the ESF to incorporate performancemonitoring-management functions in the overhead bandwidth (used solely for frame synchronization in the D4 format). In ESF, 2 Kbps of the overhead is dedicated to logic-error monitoring, and 4 Kbps is reserved as a two-way communication channel. This maintenance channel is called, as discussed in previous chapters, the facility data link, or FDL. For companies using E1 circuits, CCITT has defined certain national spare bits in the frame-alignment timeslot (TS0). The performance monitoring employs one of these bits to create a

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4-Kbps data link (the Embedded Operations Channel) which is analogous to the ESF FDL. Five national bits (Sa4Sa8) are transmitted and received in bit positions 4 through 8 of TS0 of the FAS not-word. Each national bit can be a 4-Kbps data link to the far end. These data links are sometimes used to transmit system control and status information. The E1 NSM uses one of the national spare bits (Sa4Sa8) to transmit performance data and to communicate with the far-end unit. This 4-Kbps data link serves the same purpose as the FDL in T1. The E1 NSM allows the user to determine which national bit position will serve as the data link. Traditionally, customers have relied on their multiplexers for E1 performance monitoring. The typical performance-monitoring method used by multiplexers involves dedicating a channel (864 Kbps) to network management. This management channel is used for various transmissions, including configuration parameters, routing information, and bandwidth-use statistics. In most cases, error-rate monitoring is performed only on this management channel and is extrapolated to give an approximate error rate for the entire E1. Assuming a full 64-Kbps management channel, error-rate monitoring is performed only on about 3% of the bandwidth, leaving 97% essentially unmonitored. With an 8-Kbps maintenance channel, error monitoring occurs on less than 0.4% of the bandwidth. For users interested in maintaining or improving service quality to make sure they get maximum efficiency from their networks, the installation of performance-monitoring units (PMUs) is recommended and offers numerous advantages: • The PMU monitors the full E1 bandwidth full time, without dis-

rupting service or decreasing usable bandwidth. • The same performance data can be accessed by both carriers and customers. • The location of PMUs in the network is ideal for gathering and reporting performance data. Because PMUs are installed at the customer premises, at the interface between the customer and carrier portions of an E1 circuit, they provide a critical point of demarcation. Looking in one direction, the PMU monitors carrier performance from the users point of view. Looking in the other direction, the PMU monitors the customers input into the carriers network. Because the entire bandwidth is monitored for a range of format and logic


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errors, virtually all performance failures and degradations are detected and isolated to the customers or carriers portion of the network. And because both the customer and carrier can read the same performance data, the nature of the problem is equally clear to both parties. 3.8.10 Global Framing Formats

As a general rule, outside of the United States and Canada (T1), most countries use E1 as a primary rate service. Japan has a variation of the T1, and Germany and France impose a slight variation of the E1 and, thus, have unique formats different from the rest of the market. Only one element remains constant and universal, and that is DS0. However, the form of PCM encoding differs between T1 (mu-law companding) and E1 (A-law companding). The term companding is a contraction of the words compressing and expanding. Companding is the process of compressing the amplitude ranges of a singal for economical transmission and then expanding it back to its original form at the receiving end. The ITU-T companding standard is based on A-law used in the conversion between analog and digital signals in PCM systms. Mu-law is used in Japan and North America. Therefore, sometimes required E1/T1 conversion involves both the compression law and the signaling format.

References [1]

Clark, M. P., Networks and Telecommunications: Design and Operations, Second Edition, New York: Wiley, 1997.

[2]

Gunn, H. J., Principles of Traffic and Network Designs, Geneva, IL; abc TeleTraining, Inc., 1986.

[3]

Bellcore, CCS/SS7 Computer Based Training, Reference Guide, NJ, 1993.

[4]

G. van Bosse, J., Signaling in Telecommunication Networks, New York: Wiley, 1998.

[5]

Winch, R. G., Telecommunications Transmission Systems, New York: McGraw Hill, 1993.

[6]

ITU-T Recommendation G. 702, Digital Hierarchy Bit Rates.

[7]

Auki, P., et al., RATES: A Server for MPLS Traffic Engineering, IEEE Network, March/April 2000

[8]

Onvural, R. O., Asynchronous Transfer Mode Networks: Performance Issues, Norwood, MA: Artech House, 1994.

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[9] Doverspike, R., and I. Saniee, eds., Heuristic Approaches for Telecommunications Network Management, Planning and Expansion, Norwell, MA: Kluwer Academic, 2000. [10] Black, U., QoS in Wide Area Networks, Upper Saddle River, NJ: Prentice Hall, 2000. [11] ITU-T I.357, B-ISDN Semipermanent Connection Availability, November 2000. [12] Aukia, P., et al, RATES: A Server for MPLS Traffic Engineering, IEEE Network, March/April 2000. [13] Larus Corporation, Transmission Engineering Tutorials: T1 Repeatered Lines, 1996. [14] Flanagan, W. A., The Guide to T1 Networking, Fourth Edition, New York: Telecom Library, 1990.

4 Wireless-Network Architecture 4.1 2G Wireless-Network Architecture A simplified diagram of todays 2G wireless-network architecture is shown in Figure 4.1. Shown here are three RBSs, the BSC, and the MSC and connection with the PSTN. In addition to that, there are a number of other nodes not shown here, like the voice mail system (VMS), SMS, HLR/VLR, and so on. Regardless of whether the access transmission network (connection between BSC and RBS or backhaul) is leased or owned by the operator, RBS can be connected to the BSC via microwave links, fiber-optic, or wireline (usually copper) systems. The BSC provides the connectivity between the MSC/PSTN and the radio network. It performs the radio call management functions and radio network management functions and its capacity is usually given in Erlangs. BSC blocking probability is defined to be the probability that a new request for service is rejected at the BSC due to lack of resources. Resources are understood to be card processing, transmission link capacity, countable resources such as channel cards, and so on. BSC blocking probability does not include blocking due to limitations of the air interface or call admission control, and the BSC blocking probability is usually specified to be 0.5%. BSC and MSC are usually colocated and, therefore, shown as one node. It is important to note that the physical layer of the transmission network, copper, microwave, or fiber optic will not change from 2G to 3G wireless networks. The only change will be more stringent requirements on its 129

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MW

RBS 3

MW radio MW radio T1/E1 leased lines

RBS 2

PSTN T1/E1 leased lines

BSC/MSC T1/E1 leased lines RBS 1

Figure 4.1 Example of 2G wireless network.

quality, availability, and reliability as well as higher capacities required to carry the traffic. There are today several vertically oriented, single-service networks capable of delivering similar services. We have wireline and wireless networks as well as pure data/IP networks as well as cable TV/CATV networks. These are basically separate networks that build on different principles and practices to ensure the reliability of a single service with different approaches to network management, guaranteed service levels, and so on. For over 100 years, classic telephony networks have been optimized to carry real-time voice traffic between fixed points in the network. The classic telephony network supports an integrated service concept, involving the following: • One service typevoice; • One subscription; • One user terminal.

This leads to a vertical industry orientation, where the operator offers everything from subscriber access to service creation and service delivery


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across a wholly owned network infrastructure optimized for a particular service category. Each vertically integrated network incorporates its own protocols, nodes, and end-user equipment and terminals. This means that the telephony (voice) and data service domains are still more or less kept separate. The rapid convergence between telecommunications and datacommunications will lead to a convergence of these purpose-built networks into ATM/IP-based multiservice, or next-generation, networks that can provide reliable and real-time communications. This network convergence raises some fundamental issues of network characteristics and how to bridge the inherent value of reliable circuit-switching technology with more besteffort-oriented packet-switching technology.

4.2 3G Wireless-Network Architecture 4.2.1

Directions in 3G Developments

Multiservice switching is the foundation of next-generation networks. The need for wireline- and wireless-network infrastructure to support future data-communication and IP services is obvious. However, not only is the type of traffic changing, the quantity of traffic is also growing rapidly. Both of these new factors are of critical importance in the development of future networks. Another significant fact is that the major source of revenue for most operators around the world is still voice services. And it is likely to remain so for the coming yearswhile the main source of growth will be data-based services. This means that networks must be optimized to carry packet-oriented traffic at the same time that they are delivering a reliable and high-quality voice service; that is, the technologies are converging. So far the networks have been shaped by the concept of one network, one service, one subscription, and one user terminal. This changed with the introduction of the Internet. The future networks will be multiservice networks, able to carry a full range of services, from voice communications and simple file transfers to high-speed Internet and real-time, broadband multimedia services. These services will be accessible via different access networks and a number of different terminals. Using packet technology as the infrastructure for telephony services implies that packet networks must meet stringent requirements; QoS is a key issue. IP is today not a mature bearer technology for efficient large-scale telephony QoS solutions as regards delays and latency in voice streams. The performance available from todays IP environment cannot provide carrierclass telephony services, but that may change very soon.

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The only technology available today that can meet those requirements is ATM. ATM seems to be the only appropriate technology for use in core multiservice networks, to carry both IP/data services and voice. ATM is able to act as a connectivity layer for both traditional voice and IP/data services. ATM was developed and standardized for both telephony and data, and the large incumbent telecom operators were involved in the standardization process. Therefore, most operators see ATM as the only safe migration path to new generation networks. In other words ATM has now found its role as a network infrastructure technology capable of carrying all the different transport needs of the future networks, including IP. The fundamental structure of the next generation multiservice network is based on a shared bearer network providing multiple services based on ATM. Figure 4.2 shows the layered backbone network model. The bearer network is accessed through so-called media gateways (MGWs) or hybrid switches. The MGWs control the traffic in and out of the bearer network and handle interconnections to other networks, different access networks or to traditional switches and routers. In this new network infrastructure, telephone calls will be converted to ATM switched virtual circuits while separate telephony servers control the setup of the calls through the network. The telephony servers thus provide the equivalent intelligence of todays telephony exchanges, but are not involved in the actual connections as such; the calls are set up end to end

MSC server

HLR

Wireless

MGW

Internet, intranets

MGW

ISDN/PSTN wireless

MGW

Connectivity backbone network ATM, IP Servers handling control Media gateway handling connection

Figure 4.2 Layered backbone network model.


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across the ATM network using a switched virtual circuit between the edge devices. This separation of the switching and connectivity functions in the network is a key to the evolution path toward a single bearer network based on ATM. Telephony subscribers may be connected to the ATM bearer network via the MGW from traditional switches, standard access nodes, or if it is an enterprise customer, PBX. The telephony server does not only control phone calls, it also contains all the required telephony intelligence for all MGWs in the domain. The data access to the bearer network may range from LANs and ATM switched leased lines, to cellular networks and data transmitted from or through these. All access, voice as well as data, terminates in an MGW, which ensures that traffic gets onto the ATM bearer network. A single telephony server with several stand-alone MGWs (multiservice switches) would satisfy regulatory requirements on geographically spread points of presence. The operators would not have to deploy a traditional countrywide circuit-switched network, but would be able to simultaneously offer the full range of telephony and data services from day one. As network management is very expensive and complicated, another main benefit is the fact that only one network has to be managed instead of several different networks. Today there are major developments in virtually all areas of network access and infrastructure. In access, new digital subscriber line (DSL) and wireless technologies are opening up bandwidth and giving users more and more network capacity, instant access, and on-line services. In the core network (CN), new switching and transport technologies such as ATM and WDM are expanding the capacity and the flexibility of the networks. While the focus for initial wireless 3G deployment is voice services, it is expected that wireless 3G will also become a key Internet access technology. Although expected 3G data rates of up to 2 Mbps will never compete with wireline services, the trigger that will fuel exceptional 3G growth is the promise of true mobility, Internet-connected data, and information services anywhere, any time, from a variety of handheld platforms. Global standards for 3G systems are still evolving, but high-level architectures and overall network characteristics are beginning to solidify, accelerating the need for new testing strategies and capabilities for a key component of 3G, the terrestrial radio access network (RAN). RAN infrastructure will support 3G functions including access, roaming, transparent connection to the PSTN and the Internet, and QoS management for data and Web connections. The concept of the 3G wireless network is shown in Figure 4.3. The high-QoS characteristics of classic telephony networks must now migrate onto horizontally oriented next-generation networks that can support

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MSC server MGW

PSTN

MGW

Internet

Backbone network

MGW Radio network Radio base station

MGW

RN server RBS (IP/ATM)

Figure 4.3 Concept of 3G wireless network.

multiple services based on ATM and the IP protocol. Similar developments have been under way in the enterprise market and took off around 1995 and 1996 with the widespread adoption of the TCP/IP protocol for intranet technology. For public network operators and service providers, the heavy increase in data traffic is leading to a bottleneck in narrowband networkswireline or wireless or both. Operators have to offer circuit-switched and packet-switched services, and must expand even further to offer multiservice and multimedia networks (Figure 4.4). It is also essential for the operator to optimize the network resourcesfor example, to use one transmission network that will be suitable for all services. Operators need to ensure that their investments will address their transmission requirements well into the future. The future lies with packet-based transmission technologies. Almost all networks of the past and today are vertically integrated. Vertically integrated networks are single-service networks where the operator offers anything from subscriber access to service creation and delivery. Services are offered across a wholly owned network infrastructure, optimized for a particular service category. Each vertically integrated network incorporates its own protocols, nodes, and end-user equipment and terminals. They operate different principles to ensure the reliability of a single service. Vertically


Wireless voice

Differentiation + usage

Wireless Internet

135

Wireless + multimedia 3G Real-time Internet Mobile video

Convergence Mobile office Ubiquitous (mobile e-mail) Personal coverage Mobile WWW Good voice mobile Wireline quality Bluetooth Coverage multimedia Capacity Mobile image services Cost Mobile e-com VAS WLAN ~384 (512) wide area Tariffing/prepaid ~115/384 Kbps (2 Mbps) locally ~64 Kbps ~10 Kbps Time 02/03+ 01/02 99/00 Figure 4.4 Wireless evolution.

integrated networks have different approaches to network management, guaranteed service levels, etc. Vertical integration means that telephony and data service domains are kept more or less separate. The rapid convergence between telecommunications and datacommunications will lead to a convergence of these purpose-built networks into ATM/IP-based multiservice, or next-generation, networks. This network convergence raises some fundamental issues with respect to network characteristics and how to bridge the gap between the inherent value of reliable circuit-switching technology and best-effort packet-switching technology. In order to survive in a converged communications marketplace, operators need to draw on truly open systems that invite competition in horizontal layers. The QoS characteristics of classic telephony networks must now migrate to horizontally oriented next-generation networks that can support multiple services based on ATM and IP, since the 3G network will ultimately be an open IP platform supporting a wide range of new global services [1]. IP telephony (IPT) has quickly emerged as a serious alternative to traditional circuit-switched telephony. Not just because it offers a more cost-effective service, but because the underlying technology offers a wealth of new business opportunities. The evolution of IPT will not only depend on its integration into successful business operations, but

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also on the evolution of the underlying technology. One of the most limiting barriers for this type of service offering is, without a doubt, lack of interoperability. If different vendor offerings and different carrier networks cannot interoperate, this will limit the possibility for end-user connectivity. A number of all-IP wireless-network architectures have been proposed and planned for 2004 and beyond and it is too early to say exactly how the evolution of the wireless network will go from there. In 3G networks and beyond, bandwidth flexibility is a key issue and involves a flexible decentralized provision of bandwidth to a single user as the need varies, but also cost-effective bandwidth provision to a large number of users with different bandwidth requirements in the same network. 4.2.2

Horizontally Layered Network Architecture

In the horizontally layered network architecture (Figure 4.5), functionality and nodes are arranged in layers according to their specific areas of use. The layered concept of the network architecture of the 3GPP specifications comprises three distinct layers: 1. Application layer; 2. Network control layer; 3. Connectivity layer. 4.2.2.1 The Application Layer

The application layer is where the end-user applications reside. In modern networks, applications are implemented in mobile terminals and in dedicated application servers in the network. The application servers are often complemented with content servers, which host service-related databases or libraries, such as video-clip libraries or news history databases. Concepts such as the virtual home environment (VHE) and open service architecture (OSA) were developed in the 3GPP to allow operators to provide unique services. Operators benefit from being able to differentiate themselves from one another by providing unique services, thus securing for themselves a higher position in the value chain. They also have the option of developing these services themselves or of obtaining them from third-party software houses and they can even get external service providers to run them. This flexibility allows the operator to choose from a huge portfolio of services that it can offer its subscribers. The application layer interfaces with the

Application servers

External servers

Content servers

Application layer Control servers

Network control layer Access network

GCP Connectivity layer Media gateways

ISDN, PSTN, Internet

137

Figure 4.5 Horizontally layered network architecture.


APIs

Control servers

138


network control layer via a defined set of open application program interfaces (API). By using open APIs, application developers can make use of the features of standardized service capabilities to design new services and applications. 4.2.2.2 The Network Control Layer

The network control layer incorporates all the functionality needed to provide seamless, high-quality services across different types of networks. The different networks can be seen as a set of domains, each of which houses control servers that are specific to a given network. Generally speaking, the network control layer houses several different kinds of network servers. The servers are responsible for controlling mobility management, the setup and release of calls and sessions requested by end users, circuit-mode supplementary services, security, and similar functions. These domains can be owned by various individual operators or by a single operator. 4.2.2.3 The Connectivity Layer

The connectivity layer is a pure transport mechanism that is capable of transporting any type of information via voice, data, and multimedia streams (Figure 4.6). Its backbone architecture incorporates core and edge equipment. The core equipment transports aggregated traffic streams between the different nodes at the edges of the backbone. As a rule, core equipment is a backbone router or backbone switch that handles traffic streams either according to very simple classification principles or to routes that the network operator has predefined by means of traffic engineering. Edge equipment collects customer-specific data and statistics for accounting and billing purposes and provides the single bit-pipes that guarantee an appropriate QoS. The edge equipment is usually an MGW, which operates under the full control of the nodes in the network control layer. In addition, an MGW allows the bit streams to be processed, thus providing coding and decoding of speech streams, canceling echo, bridging multiple party calls, and converting transport protocols. The nodes in the network control layer also control these manipulations. This exertion of control down to the bit-stream level allows the variety of services and applications implemented by the different network control domains to be achieved via a common connectivity layer. At the same time, the services and application are independent of the transport technology applied, which may be mixed or

GCP

ATM ATM

MGW

GCP

IP router ATM

ATM switches

GCP ATM Connectivity layer

MGW ATM

ATM

MGW

139

Figure 4.6 The connectivity layer.

ATM

ATM

MGW


GCP

140


vary over time as the network evolves. Connectivity-layer solutions can be based on either ATM transmission or IP transmission. It may also use a QoS-enabled IP-backbone network running IP over SDH, carrying packetand circuit-mode communications. 4.2.3

3G Core Network

The 3G CN supports circuit- and packet-switched services. It contains the hardware and the software needed to provide end users with multimedia applications. The CN spans both the control and connectivity logical layers. One of the new nodes required in the 3G wireless network is the MGW. The MGW performs functions such as speech coding and decoding, echo cancellation, conference-call bridging, tone and announcement generation, setup and release of user data bearers, and QoS IP routing and switching, including QoS handling and packet retunneling. The MGW will also contain interface functions for different transport standards, for example between an IP- or ATM-based CN and an external STM network. For volume-based charging support, the MGW keeps track of the volumes sent and received (for packet-based services), and it performs some security functions (for example, for packet mode services). Most MGW resources are shared between packet- and circuit-communication service, or can easily be reconfigured from one communication mode to the other. The MSC server handles control-layer functions related to circuitmode communication services at the WCDMA or CDMA2000 RAN and PSTN/ISDN borders, and performs MGW control, ISDN services control, mobility management, authentication, data collection and output, services switching function (SSF), Internet dial-in services (RAS), and element management. In addition to these functions, the MSC server houses the interworking and gateway functionality necessary to act as an SMS-IWMSC and SMS-GMSC for the Short Message Service. The HLR is a network database for mobile telecommunications in general. The HLR holds all mobile-specific subscriber data and contains a number of functions for managing this data, controlling services and enabling subscribers to access and receive their services when roaming within and outside their home PLMNs. The HLR communicates with GSNs, MSCs, and other network elements via the MAP protocol. The HLR is a real-time mobile telecommunications node of GSM, CDMA2000, and UMTS systems. The HLR is vital for the operation of a network, as it holds all subscriber data, and it is equally important for the call setup in the network, as well as for the control of the roaming subscribers.

Wireless-Network Architecture 4.2.4

141

Universal Mobile Telephone System

UMTS is the accepted 3G standard for GSM operators. UMTS requires paired 5-MHz RF channels, four times as wide as the paired 1.25-MHz channels required for CDMA2000. For this reason, UMTS is sometimes referred to as WCDMA. By migrating to UMTS, operators will gain access to additional spectrum as well as the greater capacity and expanded functionality of the new technology. UMTS incorporates a more efficient variable vocoder (CODEC). In common with CDMA2000 1x, this vocoder will increase the voice capacity of a given amount of spectrum. Outside of the Americas, UMTS is being deployed on the 1,900-MHz (uplink) and 2,100MHz (downlink) frequencies. Because of this, some operators, primarily those in the Americas who now use the 1,900-MHz frequencies for PCS, would be unable to migrate to UMTS. Allocation of other frequencies for UMTS may or may not be possible. The well-publicized failures of U.S. operators to acquire frequencies at 700 MHz (occupied by TV broadcasters), 1,700 MHz (occupied by the military), or 2,5002,600 MHz (occupied by educational broadcasters) provide examples. 4.2.5

3G in GSM Networks

New 3G GSM networks will require new radio and CN elements as well as a new air interface. This will require new BSSs, which will include radio network controller (RNC) and Node B. The RNC will include support for connection to legacy systems and provide efficient packet connection with the CN packet devices (SSGN or equivalent). The RNC performs radio network control functions that include call establishment and release, handoff, radio resource management, power control, diversity combining, and soft handoff (handoff ). A Node B is equivalent to a base station in the 2G network, but also incorporates support for the 3G air interfaces. New cell-planning methods will be needed to support the new frequency allocations for 3G and the radio interface changes. More 3G base stations will be needed than are necessary in a comparable 2G coverage area. This gives an advantage to GSM 1800 and 1900 network operators whose cells already cover a smaller coverage area than those for GSM 900 networks. GSM 900 network operators will need to fill in coverage in between existing cell sites. The 3G CN will be an evolution from GPRS or equivalent 2.5G CN systems. Upgrades to the mobile and transit switching systems to deliver packets will also be needed. A new piece of network infrastructure for 3G is also the MGW, which resides at the boundary between different networks to

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process end-user data such as voice coding and decoding, convert protocols, and map quality of service. The connectivity layer also provides access to backbone switches and nonmobile networks such as cable television. In some vendor solutions, MGWs are controlled remotely by the MSC and GSN servers by means of the Gateway Control Protocol (GCP). The ITU is working to ensure that the GCP is an open standard protocol. Existing network operators can then upgrade their MSC and GSNs to implement 3G or alternatively to implement a new stand-alone MGW that is controlled from the server part of an upgraded 2G node. 4.2.6

3G CDMA Network

The ITU manages the 3G umbrella standard known as IMT-2000. This standard endorses five different modes of RF interface and three major types of terrestrial infrastructure known as RAN. The intention is for any of the RF modes to work with any of the RAN types. The two major types of RAN are UMTS WCDMA and IS-2000 (also known as CDMA2000). UMTS originated in Europe and WCDMA in Japan, but these are now almost identical, having been converged into a single specification under the 3GPP. IS-2000 is predominantly North American and is defined by the 3GPP2 organization. More recently, a third RAN concept has been added, providing direct access to and from IP-based networks. Third-generation systems are ultimately expected to migrate toward IP as part of the global trend toward carrying all traffic types over packet networks. However, the current UMTS WCDMA specification explicitly defines ATM as the transport layer in the RAN. While some IS-2000 RANs also use ATM, the 3GPP2 specifications allow the manufacturer to choose the underlying transport layer. Given the planned migration paths of various 2G systems to 3G, 70% of 2G subscribers worldwide are expected to eventually migrate to some type of the CDMA version of 3G. The main network elements and interfaces referred to in the 3GPP specifications (see Figure 4.7) include the terrestrial components of the 3G system, referred to collectively as the RAN: • User equipment, also called mobile station, subscriber unit, or sim-

ply handset, including mobile cellular telephones, handheld PDAs, and cellular modems connected to PCs;

• Node B, usually called the RBS, providing gateway services between

the handset/RF interface and the RNC, via Iub interface;


Uu

(Handset) User equipment

Iub Node B

Radio network controller

ATM Radio access network

Iu

ATM

143

Mobile switching center

PSTN

Core network interface Internet core IP, WDM, PoS, ATM.

Figure 4.7 Main interfaces in 3G wireless network.

• RNC (or BSC), connecting to and coordinating as many as 150 base

stations (the RNC manages activities such as handing over active calls between base stations);

• CN interface, referring to other terrestrial CN infrastructure con-

nected to the RAN through Iu interface, such as the Internet and PSTN (the gateway device for this activity is usually called a mobile switching center or mobile multimedia switch).

An extensive set of protocols for communication within the RAN, to and from the user equipment, and between other networks has been developed by 3GPP. These protocols sit above AAL2 and AAL5. Together, they implement control-plane functions (e.g., signaling required to establish a call) and user-plane functions (e.g., voice or packet data). Wireless access in 3G network design constitutes one of the most important and driving requirements for the application of ATM extended with AAL2 switching (more information on ATM and AAL can be found in [2]). The following description is the high-level data flow for voice and data call in CDMA2000 BSC and toward other parts of the RAN showing practical application of AAL2 and AAL5. New terminology used here aside from the well-known backhaul includes sidehaul and fronthaul. Fronthaul is a connection between BSC and MSC, and sidehaul is a connection between two BSCs.

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Figure 4.8 shows the data flow for a land-to-mobile voice call in the 3G wireless system architecture. Voice traffic arrives at the BSC from the MSC (1). The fronthaul module receives PCM-encoded DS0s from the MSC and converts this traffic into AAL1 for transmission to the echo cancellation and vocoding module (2, 3). It also performs the required functions on the traffic channel, and converts the resulting data stream to the modified AAL2 format for transfer to the SEP module (4, 5). Modified AAL2 is a proprietary format used within the BSC. It is very similar to AAL2, with the exception that the channel identifier (CID) bit of the common part sublayer (CPS) packet is not used. This results in a simpler internal format, and therefore processing complexity and delays are both reduced. It should be noted that modified AAL2 is only used internally within the BSC, while all external AAL2 interfaces use standard AAL2. The selector element processing (SEP) module performs the selector element processing functions and sends the resulting modified AAL2 traffic channel to either the backhaul module for transmission to the appropriate RBS or to the sidehaul interface module for inter-BSC handoff (6, 7). The appropriate interface module converts the modified AAL2 data stream to the standard AAL2 format for transmission to the RBS or BSC (8). In the case of backhaul transmission to the RBS, the interface module also provides the necessary formatting. In the case of sidehaul transmission, the interface module converts the modified AAL2 format into AAL2. T1/E1 MSC

1

Radio-packet

Fronthaul

PDSN

AAL1 2 3 AAL1

Echo cancellation Modified vocoding AAL2 4 Modified AAL2 7

ATM

Sidehaul interface module AAL2 packet 8 over T1/E1 Second BSC

Figure 4.8 Voice call in 3G wireless network.

6 Modified AAL2 multicast AAL2U

SEP

5 7

Modified AAL2 Backhaul inteface module AAL2 packet 8 over T1/E1 RBSs


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Figure 4.9 shows the data flow for a land-to-mobile packet call in the BSC architecture. Packet traffic arrives at the BSC from the PDSN (1). The interface module receives the IP/AAL5-formatted data stream over an OC-3 link. The interface module converts this traffic into AAL5 packets for transmission to the SEP module (2, 3). The SEP module performs the required selector element processing. This converts the output to the AAL5 format and sends the resulting AAL5 traffic channel to either the backhaul module for transmission to the appropriate RBS, or to the sidehaul module for inter-BSC handoff (4, 5). The appropriate backhaul module converts the AAL5 data stream to the standard AAL5 format for transmission to the RBS or BSC (6). In the case of transmission to the RBS, the backhaul module also provides the necessary formatting. In the case of sidehaul transmission, the interface module converts the modified AAL2 format to standard AAL2. 4.2.7

3G Traffic Classes

When defining the UMTS QoS classes, also referred to as traffic classes, the restrictions and limitations of the air interface have to be taken into account. It is not reasonable to define complex mechanisms as they have been defined in fixed networks, due to different error characteristics of the air interface [3]. T1/E1 Radio-packet

Fronthaul

MSC

IP/AAL5 over OC-3

PDSN

1

2 AAL5 Modified AAL2/5 4 multicast

Echo cancellation vocoding Modified AAL2/5

ATM

5

Sidehaul interface module 6

AAL2/5 packet over T1/E1

Second BSC

Figure 4.9 Packet call in 3G wireless network.

SEP

AAL5 3 5

Modified AAL2/5 Backhaul interface module 6

AAL2/5 packet over T1/E1

RBSs

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The QoS mechanisms provided in the wireless network must be robust and capable of providing reasonable QoS resolution. There are four separate QoS classes: 1. Conversational class; 2. Streaming class; 3. Interactive class; 4. Background class. The main distinguishing factor between these QoS classes is how delay sensitive the traffic is. Conversational class is meant for traffic that is very delay sensitive, while background class is the most delay-insensitive traffic class. Conversational and streaming classes are mainly intended to be used to carry real-time traffic flows. Conversational real-time services, like video telephony, are the most delay-sensitive applications, and those data streams should be carried in conversational class. The interactive background and classes are mainly meant to be used by traditional Internet applications like the Web, e-mail, Telnet, FTP, and news. Due to looser delay requirements as compared with conversational and streaming classes, both provide better error rate by means of channel coding and retransmission. The main difference between interactive and background class is that interactive class is mainly used by interactive applications (e.g., interactive e-mail or Web browsing), while background class is meant for background traffic (e.g., background download of e-mails or background file downloading). Responsiveness of the interactive applications is ensured by separating interactive and background applications. Traffic in the interactive class has higher priority in scheduling than background-class traffic, so background applications use transmission resources only when interactive applications do not need them. This is very important in wireless environments, where the bandwidth is low as compared with fixed networks. 4.2.7.1 Conversational Class

The most well-known use of this scheme is telephony voice. But with Internet and multimedia, a number of new applications will require this scheme, for example voice over IP and videoconferencing tools. Real-time conversation is always performed between peers (or groups) of live (human) end users. This is the only scheme where the required characteristics are strictly given by human perception.


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A real-time conversation scheme is characterized by the transfer time being low because of the conversational nature of the scheme, and at the same time, the time relation (variation) between information entities of the stream must be preserved in the same way as for real-time streams. The maximum transfer delay is given by the human perception of video and audio conversation. Therefore, the limit for acceptable transfer delay is very strict, as failure to provide low enough transfer delay will result in unacceptable lack of quality. The transfer delay requirement is therefore both significantly lower and more stringent than the round-trip delay of the interactive traffic case. Real-time-conversation fundamental characteristics for QoS are as follows: • Preserved time relation (variation) between information entities of

the stream; • Conversational pattern (stringent and low delay). 4.2.7.2 Streaming Class

When the user is looking at (listening to) real-time video (audio), the scheme of real-time streams applies. The real-time data flow is always aiming at a live (human) destination. It is a one-way transport. This scheme is one of the newcomers in data communication, raising a number of new requirements in both telecommunication and data communication systems. It is characterized by the fact that the time relations (variation) between information entities (i.e., samples, packets) within a flow must be preserved, although it does not have any requirements on low transfer delay. The delay variation of the end-to-end flow must be limited to preserve the time relation (variation) between information entities of the stream. But as the stream normally is time aligned at the receiving end (in the user equipment), the highest acceptable delay variation over the transmission media is given by the capability of the time-alignment function of the application. Acceptable delay variation is thus much greater than the delay variation given by the limits of human perception. Real-time streamings fundamental characteristic for QoS is the preservation of time relation (variation) between information entities of the stream. 4.2.7.3 Interactive Class

When the end user (either a machine or a human) is on-line requesting data from remote equipment (e.g., a server), this scheme applies. Examples of human interaction with the remote equipment are Web browsing, database

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retrieval, and server access. Examples of machines interaction with remote equipment are polling for measurement records and automatic database inquiry. Interactive traffic is the other classic data-communication scheme that on an overall level is characterized by the request response pattern of the end user. At the message destination there is an entity expecting the message (response) within a certain time. Round-trip delay time is therefore one of the key attributes. Another characteristic is that the content of the packets will be transparently transferred (with low BER). Interactive traffic fundamental characteristics for QoS include the following: • Request response pattern; • Preserve payload content. 4.2.7.4 Background Class

When the end user, typically a computer, sends and receives data files in the background, this scheme applies. Examples are background delivery of e-mails, SMS, download of databases, and reception of measurement records. Background traffic is one of the classic data-communication schemes that on an overall level is characterized by the fact that the destination is not expecting the data within a certain time. The scheme is thus more or less delivery-time insensitive. Another characteristic is that the content of the packets is transparently transferred (with low BER). Background traffic fundamental characteristics for QoS include the following: • Destination does not expect data within a certain time; • Preserved payload content.

4.3 3G Transmission Networks 4.3.1

Replacing TDM with ATM in Transmission Networks

Wireless networks are leading the evolution of the information and communications society toward the mobile information society (MIS). This means that subscriber numbers are continuing to increase as mobile penetration reaches new heights. Also, multimedia communications and other packet-based traffic will gradually increase their role and finally predominate in mobile networks. This development has already started with modest data volumes over current


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mobile (and wireless in general) networks; a rapid increase in data applications and traffic is expected soon. New technologies and technical solutions enable higher data volumes right now in existing networks. In GSM networks, HSCSD and GPRS greatly expand these networks capabilities to handle data traffic; they also enable new and user-friendlier applications thanks to the higher bit rates available. This development will continue with still higher bit rates over the air interface in the new 3G WCDMA- and CDMA2000-based networks and 1xEV-DO (also called HDR). These increasing data traffic volumes mean that the share of the packet-based traffic in the total traffic mix in the mobile network is increasing, that the same time as total traffic volumes are also rising rapidly. Evolution of the circuit-switched networks into packet-based networks will take some time, and should be done in well-planned and managed steps, so that the efficiency of the mobile network is preserved during the change over phase. In many cases, basic mobile voice services are also growing quickly due to growth in the number of subscribers, which also contributes to the overall traffic increase and continues to require economic solutions for this type of traffic. Therefore, the well-planned steps are vital to manage mobile operators cash flows and to make full use of existing investments. It is in the interest of a mobile-network operator to direct his future transmissionnetwork strategy toward this expected increase in the penetration of advanced data services. Transmission is an important element in any wireless network, affecting both the services and service quality offered, as well as the costs of the wireless operator. Optimization of transmission solutions is thus certainly worthwhile from the operators business point of view. In current mobile networks, transmission has been optimized for the narrowband circuitswitched traffic and this type of traffic will continue to dominate for some years. However, as stated above, packet-based information over the mobile network will show rapid growth and any reasonable network development plan must take this into account and plan for a smooth and economic transition and evolution path for the transmission network. So, in broad terms, the transmission network must continue to provide well-engineered and economically optimized solutions for the growing volumes of circuit-based traffic, while at the same time develop the strategy and the readiness to cope with the even faster growing data traffic of the future. This type of transmission solution is needed in all parts of the mobile network, both in access networks with many points and low-capacity links, as well as in CNs with high traffic volumes. This means, for example, that in a wireless network, a transmission solution is needed that provides for efficient transport of large number of

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voice channels and that can evolve to also carry packet-based traffic, either ATM or IP or both. The solutions might be similar or different in different parts of the network and even the role and share of the different traffic types (TDM, ATM, and IP) might be different, but the transmission network must support them all in a planned and managed way. In any mobile network, there are different transmission needs, typically divided into two main application areas with their own characteristics: 1. The access transmission network, which connects the base stations to the closest network control or network hub point, and called here the access transmission network or backhaul. It would include the radio base station, BSC (or RNC), and BSC/MSC. 2. The core transmission network, which connects the control (or hub) points to the mobile network switching centers, and called here the core transmission network. The radio network will be connected to the CN by a backbone network (access and core transmission network), allowing wideband access and interconnection of subscribers. The 3G backbone network can use any transport technology, but is certain to be based on packet technologies, such as ATM and IP. The backbone network is built as a mesh of IP routing or ATM switching nodes interconnected by point-to-point links. Technologies such as IP over ATM may be used that uses ATM switching to multiplex IP traffic. This IP over ATM architecture supports voice traffic alongside IP. Many vendors prefer a pure end-to-end IP approach, whereas others prefer an ATM/IP hybrid to guarantee quality of service. Alternatively, IP over SONET/SDH could be a different backbone network solution that would eliminate the ATM layer by establishing point-to-point links between IP routers directly over SONET/SDH rings that run over a dense-wavelengthdivision multiplexing (DWDM) layer. This enables terabits per second (Tbps) of aggregate network bandwidth. When transporting voice over a packet-based network, overhead is added to each voice packet. The amount of overhead depends very much on what the protocol stacks for the user data look like. Figure 4.10 sketches the different possible protocol stacks. So if speech is transported over an AAL2/ATM connection, 5 octets are needed for the ATM cell header and 3 octets for the AAL2 minicell header. Assuming the AMR CODEC is used (WCDMA systems), the speech frame size can vary between 10 and 40


Voice RTP UDP IP LAC-U GTP UDP IP AAL5 ATM SDH Voice over GPRS

Header size 12 8 20 13? >=97 20 8 20 >=8

Voice RTP UDP IP PPP AAL5 ATM SDH Voice over IP on ATM AAL5

Header size 12 8 20

>=49

12 >=8

Voice RTP UDP IP PPP SDH Voice over IP

Header size 12 8 20 >=41 12

151

Voice AAL2 ATM SDH

Header size 3 5

Voice over ATM AAL2

Figure 4.10 User-plane protocol stack alternatives.

octets, depending on the quality of the radio link. If for an average connection the speech frame size is about 21 octets, two speech frames will fit into an ATM cell. So the overhead is 5.5 octets per speech frame in average. Bandwidth increase factor is 1.26. If IP is used directly on SDH, the overhead is calculated as about 41 octets per speech frame. By applying IP header compression, the overhead can be reduced to about 13 octets per speech frame. Bandwidth increase factor is 2.95 (1.62 for header compression). If IP is transported over ATM, the overhead is about 49 octets per speech frame plus the ATM cell header. Assuming again AMR CODEC, the IP datagram varies between 59 (49 + 10) and 89 (49 + 40) octets in size. This means that two ATM cells always have to be used, leaving empty space in the second ATM cell. So 106 (2 × 53) octets are always sent per speech frame. Bandwidth increase factor is 5.05. If the speech is sent via GPRS access, the calculation is even worse. When the transmission is packet based, silence suppression can be applied. This means that typically only 40% of the time does data need to be transferred. The rest is silence, and no bandwidth is used. This is common for ATM and IP transmission. IP is a best-effort transport, so IP as such does not guarantee any QoS. Of course, platforms and systems such as the value-added service centers, gateways, billing systems, customer service elements, IN systems, and the

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like will also need to be upgraded. High-bandwidth over-the-air applications (data, video, etc.) also require high-capacity transmission systems, much more so than 1G and 2G networks. Increasing subscriber numbers and share of data traffic creates significant growth in the transmission capacity needed for long-term network evolution. The implication of this is an increasing role of fiber and high-capacity microwave systems (SDH/SONET) in future transmission networks and their physical-layer implementation. There are more and more data traffic and applications that create bursty traffic. Thus, in order to provide cost-effective transmission solutions for data traffic, packet-switching solutions introducing dynamics to traffic handling must be provided. Also, investments made in fiber- and microwaveradio-based transmission links are very important as new evolution phases can be built on top of existing infrastructure. Wireless networks of the next generation will also require new, more advanced solutions for the core and access transmission networks. Data traffic is inherently variable, and transporting it over the TDM network is inefficient. In a TDM-based approach, timeslots are dedicated for connections regardless of whether information is actually being sent. In a multiservice network, the underlying network can be physically subdivided into multiple networks, one for each service (i.e., voice, data, private lines, etc.). Using an ATM-based infrastructure, much more efficient use of transmission network is possible, since ATM allocates bandwidth on demand based on immediate user needs. In 2G wireless networks, deterministic multiplexing is applied, when each connection is characterized by a constant bandwidth (e.g., one timeslot). The minimum needed bandwidth over the physical link is then simply the sum of the constant bandwidths of the connections. Since the traffic characterization is not probabilistic, statistical gain is not available. Third-generation wireless networks use packet-switched (ATM) systems and statistical multiplexing. When several connections from VBR sources are multiplexed together, a statistical multiplexing gain is obtained (Figure 4.11), because there is a certain probability that traffic bursts on different connections do not appear at the same time. It is possible to maintain the same blocking probability with less bandwidth if statistical multiplexing is used instead of deterministic multiplexing. The price for it is that the QoS (packet delay and loss) will not be ensured in a deterministic, but in a probabilistic, fashion. Statistical multiplexing of data traffic can occur side by side with the transmission of the delay and loss-sensitive traffic such as voice and video [4]. Like voice telephony, ATM is fundamentally a connection-oriented tele-


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X Mpbs 1

Y Mbps

X Mpbs 3

2

Z Mbps

Y Mbps

Due to the multiplexing gain, here: Y < X + X; Z < Y + Y;

Figure 4.11 Multiplexing gain.

communications system. That means that a connection must be established between two points before data can be transferred between them. An ATM connection specifies the transmission path, allowing ATM cells to self-route through an ATM network. Being connection-oriented also allows ATM to specify a guaranteed QoS for each connection. Since voice telephony is a real-time application, delay, among other quality measurements, is the most important factor that affects the quality of voice. According to ITU-T Recommendation G114, an end-to-end delay of 0 to 150 ms is acceptable for most applications. A delay of 150 to 400 ms is acceptable assuming that the administrators are aware of the transmission time impact of the transmission quality of user application, but any delay over 400 ms is unacceptable for general network planning purpose [5]. AAL2 has been designed and used to reduce the packing delay for narrowband services. The idea is to multiplex voice packets from several sources onto one ATM cell so that the time to fill a cell can be reduced significantly. IP over ATM and use of IP routers is a basis for 3G wireless network transmissions. An IP router is a packet-switching device used to connect several different networks to form one common network based on IP

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networking technology. Based on its understanding of the network of which it is a part, the router decides how each packet is going to be forwarded, but it also must be able to differentiate between high-priority packets and lowpriority packets and make the right decision and avoid internal congestion at the same time. IP routers for wireless networks must efficiently be able to handle small packets of data, low-speed links, delay-sensitive traffic, synchronization, a large number of nodes, and continuous on-line connections. These demands come from the nature of wireless traffic, where low-priority packets cannot block the way for high-priority voice packets. Routers in wireless networks must also be able to provide radio base stations with a high-quality synchronization signal that is distributed via transmission links between routers and base stations, assuming that a global positioning system (GPS) is not used. Routers in 3G wireless networks are an integral part of products, such as RNC for WCDMA, BSC for CDMA2000, and MGWs for packet data services. 4.3.2

Importance of AAL2

Third-generation wireless networks will integrate multimedia services, 2G voice services, and TCP/IP networks [6]. As mentioned earlier, data traffic is inherently variable, and transporting this type of traffic over an underlying TDM network is inefficient and thus expensive. An ATM-based approach can take advantage of the statistical nature of data traffic in addition to the constant rate of voice in order to provide a more bandwidth-efficient solution. Of course, ATM over the fiber-optic media has to have the same traditional transmission-network functionality like multiplexing, grooming, add-drop multiplexing, and protection while offering additional services like guaranteed performance, virtual private networks, prioritized rerouting, adjustable statistical multiplexing levels, security screening, customer network management, and so on. With an ATM-based transmission infrastructure, provisioning circuits, changing their bandwidth, and monitoring them is possible through the use of a standard network management system. In deterministic multiplexing, each connection is allocated its peak bandwidth. In ATM, statistical multiplexing is used where the amount of bandwidth allocated in the network to the VBR source is less than its peak, but greater than the average bit rate. So, the sum of the peak rates of connections multiplexed can be greater than the link bandwidth as long as the sum of their statistical bandwidths is less than or equal to the provisioned link bandwidth. The bandwidth efficiency due to statistical multiplexing increases as the statistical bandwidths of connections get closer to their


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average bit rates and decreases as they approach their peak rates. In general, though, statistical multiplexing allows more connections to be multiplexed in the network than deterministic multiplexing, therefore allowing better utilization of network resources. Generally speaking, efficiency gain due to statistical multiplexing is a factor of a number of different connection characteristics and network parameters. For example, depending how bursty the sources are and the length of those bursts, the efficiency gain due to statistical multiplexing may or may not be significant. In 3G wireless networks, core and access transmission networks are ATM based, so the calculations of the required transmission links will be a lot different from 1G and 2G circuit-switched wireless networks. The AAL performs functions required by the user, control and management planes, and supports the mapping between the ATM layer and the next higher layer. The functions performed in the AAL depend upon the higher-layer requirements. In short, the AAL supports all of the functions required to map information between the ATM network and the non-ATM application that may be using it. The transport of voice traffic over ATM networks has been a fundamental principle of its design from the start. However, its deployment has been problematic, and telephony over ATM is still an area that is being developed. Normally we regard voice over ATM as the transport of voice over emulated circuits (a replacement for a PRI or T1 carrying voice calls). ATM has always been able to carry voice and data over the same wires, since that is what it was designed to do. In an uncompressed format (standard 64Kbps PCM), the traffic is CBR and is presented to the network over a circuit using AAL1. When compression is deployed it is possible to use AAL5, or the AAL2, which is specifically designed to work with compression hardware [7]. But until recently, voice transport via ATM relied primarily on AAL1 circuit emulation, which adds 12% to 15% overhead to every voice circuit. Moreover, AAL1 lacks bandwidth-saving features like voice compression and silence suppression. With AAL1, an emulated T1 (1.544-Mbps) circuit requires 1.74 Mbps of ATM bandwidth and is not really applicable for costsensitive real-world applications. The new AAL2, on the other hand, was designed specifically for costeffective voice transport. AAL2 is used in 3G wireless networks as a backhaul connection between RBSs and the BSC. A new adaptation layer was required to provide the flexibility for network operators to control delay on voice services and to overcome the excessive bandwidth needed by using structured circuit emulation. AAL1 simply cannot be extended to meet these new ATM

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networking requirements. AAL2, as specified in ITU-T Recommendations I.363.2 (1997), I.366.1 (1998), and I.366.2 (1999), carries the specific mandate to provide efficient voice-over-ATM services. Developed by the ITU and adapted by the ATM Forum in February 1999, AAL2 as defined in ITU-T Recommendation I.363.2 includes the following capabilities: • VBR-real time (VBR-rt) support. While AAL1 supports only CBR

transmission, AAL2 supports both CBR and a traffic class called VBR-rt, which is a better fit for voice calls and other applications that send information at a variable rate.

• Statistical multiplexing. Unlike AAL1 circuit emulation, which

reserves a fixed amount of bandwidth for each circuit, AAL2 can allocate unused bandwidth to other traffic on demand.

• Cell sharing. AAL2 can pack several short packets from different

sources into one ATM cell, letting multiple connections share the same bandwidth with less overhead.

• Variable packet fill delay. To let service providers balance delay

against efficiency, AAL2 supports variable settings for packet fill delay, the time allotted to stuff packets into cells before putting them on the wire.

• Voice optimization. AAL2 includes specific bandwidth-saving fea-

tures like voice compression, silence detection and suppression, and idle voice channel suppression.

Voice communication by nature is half duplex; one person is silent while the other speaks. There are also pauses between sentences and words with no speech in either direction. By taking advantage of these two characteristics, it is possible to save bandwidth by halting the transmission of packets during these silent periods. This is known as silence suppression or digital speech interpolation (DSI). The extra bandwidth saved from the silent period of one voice channel can be used by other connections if using AAL5or AAL2-based connections. With AAL1 (circuit emulation-based services), these savings can only effectively be used by ABR or UBR services since the connection admission control mechanism will have allocated the bandwidth required for CBR QoS. This technique can improve bandwidth utilization by as much as 40%. In order to create a natural-sounding conversation, background noise can be generated at the far end to recreate a realistic environment.


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AAL2s VBR service handles voice-over-ATM far more efficiently than the CBR service of AAL1, with its inefficiently utilized, permanently allocated bandwidth. Before AAL2, users wanting to implement voice-overATM had to live with AAL1s limitations or adopt a proprietary solution (increasing network efficiency but negating interoperability). The new standard means that ATM switches can extend to voice the benefits of ATMs statistical gain. Access connections using this scheme can transport voice circuits over the same facilities as data circuits, minimizing the use of precious bandwidth. AAL2 is designed to make use of the VBR ATM traffic classes (with higher multiplexing gains), providing bandwidth-efficient transmission of low-rate, short and variable packets for delay-sensitive applications. AAL2s structure lets network administrators take traffic variations into account in the design of an ATM network optimized to match traffic conditions. AAL2 also enables multiple user channels on a single ATM virtual circuit and varying traffic conditions for each individual user or channel. Its structure also provides for the packing of short-length packets into one (or more) ATM cell and the mechanisms to recover from transmission errors. Compared to AAL1 and its fixed payload, AAL2 handles a variable payload within cells and across cells. This provides a dramatic improvement in bandwidth efficiency over either structured or unstructured circuit emulation using AAL1. It has been proven that AAL2 provides good mechanisms for fine adjustment of packetization delay and for achieving high bandwidth efficiency for low bit rate and VBR application. The cost is some overhead in the form of headers. This overhead is not important at all when compared with that for techniques, such as partial filling of cells (AAL2 behaves at least as well as partial filling), but may be significant when users have a choice among several AALs. For example, for transmitting long packets (longer that 45 octets), the user may chose either AAL2 (through it segmentation SSCS) or AAL5 (explicitly specified for long packet data). It is known that for small packets (up to a few hundred octets) AAL2 is more efficient than AAL5, while for long packets AAL5 is definitely recommended. In this case the user should select the AAL based on its data-generation patterns. A very realistic application for AAL2 switching has been identified for 3G of wireless networks, in the frame of IMT-2000 standardization, to support the functionality known as soft handoff. The ATM using AAL2 for narrowband services described in this specification fulfills an urgent market need for an efficient transport mechanism to carry voice, voice-band data, circuit-mode data, frame-mode data, and fax traffic. Voice transport will include support for compressed voice and

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noncompressed voice together with silence removal. Reference [8] describes the procedures and signaling required to support the efficient transport of narrowband services across an ATM network between two interworking functions (IWFs) to interconnect pairs of non-ATM trunks. It specifies the use of ATM virtual circuits with AAL2 to transport bearer information and ATM virtual circuits with AAL2 or AAL5 to transport CCS. The virtual circuits used may be PVCs, SPVCs, or SVCs. The specification supports the transport of common channel signaling (CCS) information as well as channel-associated signaling (CAS) information. ATM trunking using AAL2 provides both switched and nonswitched services to the narrowband network. 4.3.3

QoS Concept

From an engineering planning prospective, the initial choice of network topology can seriously impact initial investment and future flexibility. Although much has been learned in terms of network topology (system design), LOS and path-loss (coverage) issues, and QoS optimization issues, much is still in the process of being developed, tested, deployed, and tested again in the field. It is already proven that the delivery of raw bandwidth over wireless media without acceptable QoS will not result in market acceptance. Without clearly understanding QoS in the context of a wireless broadband access system, it is unlikely that the underlying architecture and the resulting hardware and software design will result in a successful system. The issues of quality delivery are somewhat more complex for wireless broadband access systems than for wireline systems. Data-delivery problems include slow peripheral access, data errors, dropouts, unnecessary retransmissions, traffic congestion, out-of-sequence data packets, latency, and jitter. In addition, wireless access introduces high inherent BER, limited bandwidth, user contention, radio interference, and TCP traffic-rate management. QoS mechanisms must address all of these concerns. In data networking, quality usually implies the process of delivering data in a reliable and timely manner. The definition of reliable and timely is dependent upon the nature of the traffic being addressed. These terms may include references to limitations in data loss, data retransmission, and packet order inversions, as well as data accuracy expectations and latency variations (jitter). QoS is a complex concept, requiring a complex mechanism for implementation. A casual user doing occasional Web browsing, but no FTP file downloads or real-time multimedia sessions, may have a different definition of QoS than a power user of large databases or financial files, frequent


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H.323 video conferencing and IP telephony. For example, in a wireless system, QoS mechanisms must cope not only with considerations particular to the wireless environment, but with wireline-networking considerations as well. In ATM networks, traffic descriptors are usually used only as a rough guide, and many service providers systematically practice overprovisioning and allocating a lot more bandwidth than necessary [9]. Although not a very efficient system, the perceived QoS is satisfactory in most cases. The migration from circuit-switched to ATM and a packet-switched network has also affected QoS mechanisms. An IP-centric wireless system for packet-switched network traffic requires a new approach to provide optimal QoS performance. The use of QoS as the underlying guide to system architecture and design constitutes the fundamental differentiation between wireless broadband access systems designed with traditional circuit-centric or ATM-centric approaches and IP-centric wireless broadband access systems. Queuing is the commonly accepted tool for manipulating data communications flows. Data packets must be queued for packet headers to be examined or modified, for routing decisions to be made, or for data flows to be output on appropriate ports. However, queuing introduces a delay in traffic streams that can be detrimental and can totally defeat the intent of queuing. Excessive queuing can delay time-sensitive packets beyond their useful time frames or increase the round trip time, producing unacceptable jitter or causing the data transport mechanisms to time-out. Therefore, queuing must be used intelligently and sparingly, without introducing undue delay in delaysensitive traffic. To achieve high-quality (often referred to as toll-quality in circuitswitched networks) voice transmission, the absolute amount of transmission delay, as well as the variation in that delay, called jitter, must be kept low. In a wireless environment where TDMA, FEC, and other such techniques are necessary, queuing must be used only to enable packet and radio-frame processing. However, in the case of real-time flows, the overall added delay in real-time traffic should be held to below about 20 to 25 ms. This amount of delay is not perceptible to the human ear. Several additional factors contribute to delay in data networks, including cell-packing delay, coding and compression delay, queuing delay, and freeze-out delay. Cell-packing delay, or packetization delay, is simply the amount of time it takes for the sending device to fill packets before they can be sent. Coding and compression delay represents the amount of computational time that is needed for the sampling, quantization, and compression of the signal. This delay can become significant if a large amount of compression is being done to the signal. Queuing delay occurs in networks experiencing congestion and represents

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how long packets must wait in queue at a bottleneck node in the network. The use of queue management as the primary QoS mechanism in providing QoS-based differentiated services is a simple and straightforward method for wireless broadband systems. Wireless systems are usually more bandwidthconstrained, however and are therefore more sensitive to delay than their wireline counterparts. So QoS-based differentiated services must be provided with mechanisms that go beyond simple queuing. In a network not experiencing congestion, the most significant type of delay should be freeze-out, or serialization delay. This delay is due to the fact that packets take a finite amount of time to transmit, and during this time, the channel is not available to transmit any other information. For example, when two voice channels are packetized and sent over the same data link, each has to wait to send its packets while the link is busy transmitting the other channels packets. Freeze-out delay can be roughly quantified as the size of packets divided by the speed of the channel (in bps). Freeze-out delay by itself provides a minimum estimate of one-way delay for packetized voice traffic; in reality, actual delay would be higher due to the additional delays mentioned before. TCP controls transmission rates by sensing when packet loss occurs. Because TCP/IP was created primarily for the wireline environment with an extremely low inherent BER (today it is on the order of −9 1 × 10 or better for fiber optics), TCP assumes any packet loss is due to network congestion, not error. Therefore, TCP assumes that the transmission rate exceeds the capacity of the network and slows the rate of transmission; however, packet loss in the wireless link segment is due primarily to the high inherent BER, not congestion. With a range of data flows, each having different bandwidth, latency, and jitter requirements, the IP-centric wireless system must be able to manage QoS mechanism parameters over a wide range in real time. The QoS mechanism must be able to alter system behavior to the extent that one or more data flows corresponding to specific applications be transparently switched on and off from the appropriate end users. This approach is in contrast to other QoS mechanisms that seek to achieve high QoS by establishing circuit-centric connections from end to end without regard for the underlying applications actual QoS requirements. By providing an applicationspecific QoS mechanism, scarce wireless bandwidth can be conserved and dynamically allocated where needed by the QoS mechanisms associated with each application type. Mobile quality of service (M-QoS) has been defined in order to augment traditional QoS requirements found in wired ATM networks [10]. This wireless QoS refers to the QoS parameters associated with the wireless


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links, such as link delay, bit error rate, and channel reservation, as well as the performance parameters associated with the handoff blocking probability and cell loss. 4.3.4

ATM Physical Layer

Bandwidth is the key resource in both circuit- and packet-based networks. In circuit-based networks, a fixed amount of bandwidth is dedicated to a call in progress. Since 55% to 60% of the call consists of silence, circuit-based networks do not make optimum use of bandwidth. The principal advantage of packet-based networks is that they use bandwidth much more efficiently. Unlike circuit-based networks, packets from many different sources share a circuit, allowing for efficient use of fixed capacity. ATM seems to be the technology of choice for many different 3G wireless networks. The physical layer is at the lowest level of the ATM stack. It takes the full cells from the mid-layer and transmits them over the physical medium. The ITU-T originally defined only two speeds that should be supported by ATM (i.e., 155.52 Mbps and 622.08 Mbps); however, over time a number of additional speeds and interfaces have evolved, going as low as E1/T1 and as high as the Gbps range. The physical layer itself is subdivided into two sublayers: the transmission convergence (TC) sublayer and the physical medium dependent (PMD) sublayer. These two sublayers work together to ensure that the optical or copper interfaces receive and transmit the cells efficiently, with the appropriate timing structure in place. ATM, being an international transmission technology, has to be able to work with a variety of formats, speeds, transmission media, and distances that may vary from country to country. The standardization of the physical layer interfaces enabled just such connectivity. Single-mode fiber, multimode fiber, coaxial pairs, and shielded and unshielded twisted pairs are today all standardized for use in the ATM environment. The TC sublayer takes care of header error check (HEC) generation and verification, cell scrambling and descrambling, cell delineation, and decoupling. The HEC is a one-byte field in the ATM cell header, which protects the header from errors. The PMD sublayer covers bit timing, line coding, the physical connectors, and signal characteristics. The UNI documents detail the physical media types allowed at the user interface and the details differ for the public and private UNIs. For example, Category 5 twisted pair is permitted at the private UNI, but not at the public UNI. The original objective for the physical layer was operation over the SDH and SONET only. When the ATM Forum V3.0 and V3.1 specifications were

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ratified, however, other interfaces were included. These interfaces were the DS3 and a 100-Mbps interface based on the transparent asynchronous transmitter/receiver interface (TAXI) fiber distributed data interface (FDDI) standard. DS1 operates at 1.544 Mbps and is approved for ATM over twisted pair at a distance of up to 3,000 feet. DS3 operates at 44.736 Mbps on coaxial cable up to 900 feet. STS-1 (51.84 Mbps), STS-3c (155.52 Mbps), and STS-12 (622.08 Mbps) operate over single-mode fiber up to 15 km. E1 (2.048 Mbps) and E3 (34.368 Mbps), together with the Japanese standard J2 (6.312 Mbps), are standardized for coaxial cable with no distance specified. Recently, a definition for a 2.5-Gbps physical interface (an SDH interface) was completed. This definition describes how cells are mapped to this higher-speed transport. 4.3.4.1 ATM in SONET/SDH Fiber-Optic Networks

The integration of SONET/SDH and ATM involves more than offering interfaces that allow connectivity between the technology networks. The high-speed transmission attributes of SONET/SDH and the switching and bandwidth-management capabilities of ATM complement each other to form the foundation of broadband networking. SONET/SDH provides ATM with access to a high-speed infrastructure; conversely, ATM offers the high-speed traffic that takes full advantages of a SONET/SDH infrastructure. Key to integration of ATM and SONET/SDH is the ability to monitor and react to failures, ensuring survivability. SONET/SDH rings provide protection against the failures within the core transmission network, but should also extend to protect broadband services against link and node failures. The integrated ring provides facility-layer reliability (link) complementing ATM reroute, which provides ATM-layer reliability (node) in addition to link reliability. The ATM service platform should be able to intelligently monitor the SONET/SDH payload overhead. Upon the detection of a failure, the ATM services should be able to provide SONET/SDH 1+1 automatic protection switching as defined by ITU and Bellcore specifications. The ATM switch needs to provide one-for-one redundancy on the ports, and the complete switchover must take place within the specified 50-ms QoS parameter defined by service providers. If SONET/SDH switchover is unable to overcome the network failure, ATM reroute will reestablish the failed connection over a new route. These inherent SONET/SDH switchover-protection capabilities and ATM reroute capabilities are critical to new broadband services. Both SONET/SDH and ATM have operations and maintenance information in their headers, but use the information in complementary ways, with


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SONET/SDH checking for errors on a span-by-span basis, while ATM considers performance on an end-to-end basis between different switches. ATM will send 50 cells per SONET/ATM frame, each with its own operations, administration, and maintenance functionality, making it possible for the ATM layer to detect a fiber cut much sooner than the SONET/SDH layer. This way, ATM can enhance the speed with which SONET/SDH detects problems. As per ITU-T I.630, ATM Protection Switching, the individual VP/VC protection-switching concept was developed to apply primarily to the situations where server-layer protection-switching does not exist. It is useful to protect only a part of VPs/VCs that need high reliability. The rest of the VPs/VCs remain unprotected. This helps to reduce the necessary bandwidth for protection and can be used for protection against ATM-layer defects as well as physical-layer defects. The ATM protection-switching architecture can be a 1+1 type or an m :n type. In the 1+1 architecture type, a protection entity is dedicated to each working entity with the working entity bridged onto the protection entity at the source of the protected domain. The traffic on working and protection entities is transmitted simultaneously to the sink of the protected domain, where a selection between the working and protection entity is made based on some predetermined criteria, such as server defect indication. In the m :n architecture type, m dedicated protection entities are shared by n working entities, where m ≤ n typically. The bandwidth of each protection entity should be allocated in such a way that it may be possible to protect any of the n working entities in case at least one of the m protection entities is available. When a working entity is determined to be impaired, it first must be assigned to an available protection entity followed by transition from the working to protection entity at both the source and sink of the protected domain. It is noted that when more than m working entities are impaired, only m working entities can be protected. In general, if lower-layer (e.g., SDH or optical) protection mechanisms are being utilized in conjunction with ATM-layer protection mechanisms, then the lower layers should have a chance to restore working traffic before the ATM layer initiates protection actions. The objective here is to avoid unnecessary protection actions and any issues of contention. 4.3.4.2 ATM in Microwave Radio Networks

ATM was originally developed for use in high-reliability fiber-optic SONET/SDH networks and not for more difficult media like radio. Over

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the last few years more and more transmission systems, especially those in wireless networks, are using ATM over the microwave networks. These radio systems carrying packetized traffic such as ATM (or frame-relay) have to be designed in a way that takes into account behavior of this kind of traffic. Because ATM is primarily designed for an essentially error-free environment, in the wireless arena the sources of errors and their consequences on ATM traffic and its QoS are being studied today [11]. Although important in any network, error bursts are expected to be very significant sources of degradation in the microwave network. ATM is designed for low-BER links, and the radio links with just a moderate BER can cause unacceptably high cell-loss and misinsertion rates [12]. By definition, a misinserted cell is a received cell that has no corresponding transmitted cell on the considered connection. Cell misinsertion on a particular connection is caused by defects on the physical layer affecting any cells not previously associated with this connection. Since the mechanisms that cause misinserted cells have nothing to do with the number of cells transmitted on the observed connection, this performance parameter cannot be expressed as a ratio, only as a rate. In a process of dimensioning microwave point-to-point systems for ATM traffic, there are a number of issues to be considered. Since bit errors in the microwave system typically appear in multiples and spread less than the ATM header length, the singlebit-header correction feature may not improve cell-loss rate as much as predicted and intended. The latest research shows that the BER is degraded approximately one decade from the microwave radio system to the ATM CBR virtual circuit due to the cell loss. In wireless networks, broadband terminals as well as smaller and denser cells will increase the total capacity needs enormously. Many of todays 1E1/T1 links will be increased to STM-1/OC-3 and higher capacities and will require high-capacity SDH/SONET microwave radios even at spur links to the last cell site in the network. Aside from the capacity, these microwave radios need a very sophisticated error-correction technique to satisfy ATM transport layer requirements. −9 Normally, in a fiber-optic system, BER should be 10 measured at the −10 ATM CBR virtual circuit. The same quality corresponds to BER = 10 in −6 the microwave radio system. Systems with BERs worse than 10 are considered unavailable. Although still under research, the following facts should be kept in mind: −9

• To achieve 1 × 10 user BER from the ATM network, 1 × 10

required on the SDH radio link.

−10

is


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−6

• When BER is above 1 × 10

(low BER) end users quality is very low (1 cell lost per second). AIS generation at low BER should be considered both for the purpose of rerouting and disconnecting. −5 • For ATM NMS operation, 8 × 10 is required on the SDH microwave radio link. 4.3.5

Traffic Modeling and Simulation Tools

The transmission network consists of the elements required for the transport of call and signaling information between the major nodes in the service network. The core transmission network covers elements required for the transport of calls and signaling between nodes in the CN. Typically, the major nodes are POIs, MSCs, BSCs, TSCs, STPs, HLRs, and ESNs. The access transmission network consists of the elements that are required for the transport of call and signaling information between the RBSs and BSCs in the service network. Due to recent and ongoing efforts of both regional and global standardization and research processes, there is a wide consensus regarding the basic architectural aspects of 3G mobile communications systems, including applying ATM as switching and multiplexing technology [11]. In order to utilize transmission facilities efficiently, traffic simulation models and tools will be used in the process of wireless-network planning. All three aspects of network planning (RF, core, and access) will be based on statistical methods and traffic-simulation tools. Multiplexing of different traffic streams has consequences on the dimensioning, since there will in many cases be a nonnegligible gain compared to just adding the capacity demands of individual traffic streams. The reason is that the variation of the individual streams is usually smoothed out in the aggregated traffic stream. It is often convenient to split up this socalled multiplexing gain into traffic levels: • Multiplexing gain on a call level for all services (only call arrivals are

considered, often modeled as a Poisson process); • Multiplexing on a transaction level (for services with discontinuous transmission, such as interactive services, voice with DTX); • Multiplexing on a packet level (as with VBR packet services). For constant bit rate services with continuous transmission, multiplexing gain can be obtained on a call level only. In order to include multiplexing

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gain on subcall levels (for example, with VBR services) when dimensioning, it must be considered that arrivals of transactions or packets (depending on the level considered) often exhibit a pattern that deviates significantly from the Poisson process. Inverse multiplexing for ATM is a method that makes it possible for several physical links to carry a single ATM stream. The main advantage is increased robustness. The traffic is distributed on all physical links and in case of a failure on one physical link, the traffic is distributed over the remaining physical links. No traffic will be lost if the remaining capacity is sufficient. Another important factor is that larger links result in an increased potential for statistical multiplexing gain. In order to perform the calculation, the following data is needed: • Service characteristic description; • Traffic-related data; • GoS requirement.

The service description contains parameters representing the basic behavior of the service. The traffic mix is responsible for setting how intensive the given services are used in the RBS. This data is most likely based on the outcome of the cell-planning procedure. The GoS requirement reflects the percentage of the users that will not get service because of the limited resource on the transmission and switching network. There are two types of services that can be defined and characterized: circuit- and packet-type services. Circuit-type services are for services with definite bandwidth demand and holding time. These types of services are delay sensitive and demonstrate CBR or VBR with well-defined characteristics (like voice). The parameters that describe the circuit-type services are equivalent bandwidth and overhead. The equivalent bandwidth for CBR is the bit rate itself. For VBR it is somewhere between the peak and the average bit rate and it is very dependent on the traffic characteristics. The overhead of the service takes into account the amount of extra data volume that must be transferred on the transport network excluding the ATM overhead, which is included later on, when the logical links are mapped onto the physical ones. The overhead can consist of protocol overheads like frame protocol, MAC, RLC, and so on. Calculation of the overhead should take into consideration what kinds of overheads are already added in the equivalent bandwidth.


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Packet-type services can be used to describe best-effort services that are not delay-sensitive services. For best-effort services, well-defined resources are not guaranteed within a certain time, and only an average throughput over a longer period of time (at least an hour) can be guaranteed. Mean BHT, burstiness factor, and overhead characterize this type of service. The average throughput is given by the mean BHT in bytes per hour. The burstiness factor describes how bursty the traffic is; that is, what portion of a given bandwidth can be utilized by the traffic (in an average level) to be able to handle the deviations from the average bit rate. Its value is between 0 and 1, where burstiness equal to 1 means nonbursty traffic (a typical value is 0.6 or 0.7 in the case of traffic generated by IP applications). The smaller the burstiness factor, the more bursty the traffic, requiring higher bandwidth. The overhead parameter has the same role as in the circuit-service type. The other group of information is traffic data and GoS requirement specific to an RBS. The traffic data is given by the traffic mix parameter, which is a list of services offered by the RBS and their level of usage. In case of circuit service, the latter one is an Erlang value describing the traffic volume. For packet service it is the average number of simultaneously attached users (as defined in WCDMA). In both cases they are meant to be a busyhour value. The GoS requirement is responsible for adjusting the percentage of the calls that cannot be served because of transmission resource shortage. The corresponding parameter name is blocking. It can be set for each RBS and affects only the circuit-type services. Trunk calculation for RBS-BSC backhaul interface is based on RBS capacity and can be defined in terms of the number of sectors and the number of simultaneous calls it can support. Let us assume that under full configuration, the CDMA2000 RBS will support 4 RF carriers, equating to a maximum of 12 RF sectors. This requires that 8 E1 (or 2 E1s per RF carrier) backhaul connections between the RBS and BSC. The actual number of E1/T1 spans, however, depends on traffic demand. Due to the high efficiency of ATM-based backhaul, the maximum number of 8K traffic channel per E1 span is 180 while the number of 13K traffic channels per E1 span is 125. These are typical CDMA2000 numbers that may differ from supplier to supplier. The RBS traffic load can be acquired from RF planning in terms of Erlangs. Then, the traffic is converted to the number of channels by Erlang B formula. Based on above maximum number of channels per E1 span, the number of E1 spans of RBS-BSC can be derived accordingly. For example, under mobility environment, 1 carrier and 3-sector RBS may require 1T1 or 1E1 backhaul to the BSC.

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The interface of BSC-BSC is to support inter-BSC soft handoff which carries packet-based voice traffic channels. The inter-BSC handoff traffic Erlangs can be determined by the formula below: Handoff traffic (in Erlangs) = a × l × T where l = the handoff arrival rate at border cell, which is the function of mo-

bile speed, border cell radius, and the number of calls at border cell (air capacity); T = the mean usage duration, which is the average handoff call duration;

a = the ratio of the rate of system border crossings and the rate of cell

border crossing.

Based on the above equation, we are able to calculate handoff traffic in Erlangs. Again, the number of E1/T1 spans is calculated in the same way as the backhaul interface (RBS-BSC). 1 The interface of BSC-MSC is based on the IOS standard to support telephony service. The trunk group size depends on the voice and circuitswitched data traffic load on the BSC. Basically, the BSC traffic Erlang can be calculated from Erlang/sub and the amount of subscribers within the BSC. Then, it is converted to the number of trunks and E1s/T1s by Erlang B formula. Usually, trunk size is dimensioned under the load of 70% to protect the system from overloading. 4.3.6

2G and 3G Coexistence

The ATM Forum specification for circuit emulation service (CES) defines the means for ATM-based networks to employ AAL1 to emulate, or simulate, synchronous TDM circuits over the asynchronous infrastructure of ATM networks [2]. The circuit emulation (CE) function enables existing TDM circuits to be mapped over ATM. CE thus enables operators to migrate an existing TDM network to ATM while preserving the investment in TDM equipment. The reader should note that CE products are available for all major circuits, for example the American T1 standard and the 1. International Organization for Standarization.


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European E1 circuit standard. CE is broken into two versions, structured and unstructured, but both versions of CE use AAL1 CBR connections. In structured CE, the ATM network recognizes the internal structure of the circuit and is able to recover this structure at the receiving end. Circuit structure refers to the timeslots. Structured T1/E1 supports N × 64 Kbps (fractional T1/E1). This means that particular timeslots may be mapped to different virtual circuits and hence to different destinations. Several timeslots from a source circuit may be mapped to one virtual circuit. E1 contains 32 timeslots per frame, and the first timeslot, timeslot 0, is used for framing. As framing is irrelevant within the ATM network this timeslot (timeslot 0) is often terminated within the first ATM switch and regenerated at the destination ATM switch, which produces an E1 output. Additionally, some means of recovering the original signal clocking must be available. Which method is to be used at the destination is communicated across the network in a CE call setup request, or is set by the administrator. All timeslot 1s from each frame are mapped to an ATM cell and, hence, to an ATM VC. In mapping the source information into cells in this manner, an important issue is encountered, that of latency. An AAL1 1-byte header will be included in each cell payload; thus, to use the ATM bandwidth as efficiently as possible we should wait for 47 timeslot 1s. In waiting this long to fill the ATM cell we may reach a point where the QoS of the application is compromised. To get around the latency issue we may choose to pad out the cell, which is to send a cell only partially filled with voice samples. The level of padding will depend on several factors, including how far the call has to go overall. Different strategies may be employed for different types of calls. For example, a call that is recognized as a long-distance or international call may be heavily padded by the CE function in order to minimize latency. That call will inherently have a long end-to-end delay. Similarly, it may be acceptable to map local calls to ATM cells without any padding to maximize bandwidth usage. Structured CE also specifies support CAS, commonly used by PBXs to indicate off-hook and on-hook conditions. Since the structured mapping of the individual timeslots does not convey TDM framing information end-toend, the CAS information is encoded and transported separately, requiring additional overhead. In unstructured CE, the network does not attempt to recognize the internal circuit structure. Rather, it simply transmits the entire circuit across the network. This unstructured service emulates a 1.5- or 2-Mbps data leased line. On E1 circuit, a 376-bit (or 47-byte) chunk of the source signal is taken. These 47 bytes have the AAL1 one-byte header added to make up the full 48-byte payload. An ATM cell header is added, making a full ATM cell.

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These cells are then sent on a CBR connection. CE IWF provides timing to the TDM equipment in a synchronous mode, or accepts timing in an asynchronous mode. Timing transfer is critical for many legacy TDM networks, especially T1/E1 multiplexer networks. CE is most likely the method that will be used to combine existing TDMA networks with the new ATM-based 3G networks. On the other hand, existing transmission networks are based on PDH as defined in Recommendation G.702. ATM is considered the suitable technique to support B-ISDN. The SDH will form the basis of transport of the ATM cells, but during the transition period, there is the need to transport ATM cells using existing PDH transmission networks. Recommendation ITU-T G.804 provides the mapping to be used for this transport of ATM cells on the different PDH bit rates for both 1.544- and 2.048-Mbps hierarchies [13]. These mappings cover both the 1.544-Mbps and 2.048-Mbpsbased hierarchies and are used in conjunction with the frame structures defined in Recommendation G.832. The detailed requirements on how to map ATM on a fractional physical link will be in accordance with ATM Forum Document af-phy0130.000, ATM on Fractional E1/T1 (September 1999). 4.3.7

Transmission-Network Architecture

4.3.7.1 Transmission-Network Objectives

The main objectives of the transmission network are to connect all the points of interest, satisfy the capacity demands and provide reliable service using microwave, copper, fiber-optics or satellites. During the wireless-network build-out, it is important to establish a transmission plan that will include all the present requirements as well as future expansion (number of cell sites, RBS type, and future capacity requirements). Transmission network design typically involves a trade-off between network reliability and speed of deployment and price. An example of the small, three cell-site, RF network where mixed backhaul (transmission) media are usedleased T1/E1 lines and microwaveis shown in Figure 4.1. In wireless networks, the term backhaul (and sometimes access transmission or access transport network) is used to describe RBS-BCS connectivity exclusively. The terms core transport and core transmission are usually used to describe network connectivity between other network nodes. The core transmission network is the connection between both MSCs and BSCs and MSCs and the PSTN. BSC-MSC connectivity refers to the ability of the BSC to support the reliable transfer of signaling messages with the MSC for call (e.g., voice, fax,


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and packet data) setup and teardown, mobility management, radio resource management, and transmission facilities (terrestrial circuit) management. BSC-BSC connectivity involves the ability of the source BSC to support signaling messages with the target BSC for direct RBS to RBS soft or softer handoff, access handoff, access probe handoff, and channel assignment into soft or softer handoff in CDMA networks. The supported signaling protocol will allow for efficient resource allocation and deallocation (inter-RBS connection setup and teardown) and call connection control between a source and a target RBS during soft handoff. Various types of transmission-network topologies are shown in Figure 4.12. Star and tree formats are examples of linear transmission-network architecture and used for small- to medium-size wireless networks. The size of the network is assessed based on the number of cell sites and the required backhaul capacity. The small wireless network shown in Figure 4.1 is an example of this type of transmission-network architecture. There is no network protection in this case and a problem on any of the E1/T1 links will affect one or more (in case of daisy-chain sites) cell sites. Added service reliability can be achieved with automatic rerouting. Many successful mobile operators protect transmission by using automatic traffic rerouting, assuring additional reliability in normal situations, such as when access microwave radio links suffer cutoff due to poor weather conditions or possible fiber-

Star Ring

Switch/NOC location BSC MSC

Tree Hub site Spur (end) site

Figure 4.12 Transmission network topology.

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optic cable cuts or any other human error. With a flexible rerouting transmission system such as T1/E1 trunk rerouting, backup capacity can pass via physically separate routes, as the problem is not likely to interrupt both routes simultaneously. The base station trunk is the entire physical transmission link between two base stations or sites or between a base station and its base station controller, typically T1/E1 or nxT1/E1 links. In case of traffic failure, trunk rerouting switches all traffic in the main trunk simultaneously to the backup trunk. Large base stations comprising a number of circuits are switched simultaneously for minimum service downtime. Rerouting can be arranged for all sites or only critical sites, such as base stations that are labeled as higher priorityfor example, hub sites. Hub sites are those sites that collect traffic from more than one other site (typically three to four other sites) and carry that traffic toward the BCS location or fiber-optic ring hub site. For a larger transmission-network it is recommended that a ring configuration be used as a high-capacity backbone carrying traffic to the switch location. Additional fibers in the fiber-optic network or cross-polarization in the case of a microwave network can be used to further increase (double) the capacity of the ring. The ring configuration shown in Figure 4.13 has the BSC and MSC not colocated but interconnected through two SDH highcapacity networks. It is usually recommended that the BSC and MSC be colocated and placed close to the point of presence of the PSTN to simplify interconnect. Ring architecture is considered a reliable communication facility since it provides automatic protection from the following: • Site hardware (batteries, towers, antenna system) failures; • Radio or MUX equipment failures; • Propagation failures in the microwave network; • Cable cuts in the fiber-optic network.

A ring topology also provides basic user features such as simple operation, fault location, and maintenance, and it provides alternate routing of E1/T1 traffic automatically and no loss of E1/T1 traffic due to signal failure. Each E1/T1 circuit must be dedicated completely around the ring, and reuse of the same E1/T1 in the opposite direction is not possible. For ultimate reliability, both directions can be 1+1 hardware protected. In microwave systems

Wireless-Network Architecture RBS

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RBS RBS N × 2 Mpbs

RT RT RT

ECC STM-1 RT

RT

RT

N × 2 Mpbs

N × 2 Mpbs

ECC

BSC

RBS RBS

RT

RBS

WS

Q3

RT

RT

STM-0

RBS

RT STM-1 ring

STM-1 ECC

TMN OS SDH transport network

Mobile MSC exchange

RBS RT

RT

RBS BSCBase station controller RBSRadio base station BRTRadio terminal

Figure 4.13 Microwave ring topology.

additional protection (e.g., space or frequency diversity) at lower frequencies may be required against short-term multipath outages. All the sites that belong to the ring are considered hub sites and have to be planned so that during the deployment stage they are completed first in order to provide connectivity and protection for the rest of the network (spur links). In PDH networks, additional hardware with built-in intelligence to assess the T1/E1 quality and switch circuits, if needed, will be required. This hardware has to be added at every site and it is useful for small networks. SONET/SDH have incorporated several protection and switching techniques from their inception. These include linear APS, path-switched rings, line-switched rings, and virtual rings. These techniques provide the ability for a network to detect the problem (under 10 ms) and heal itself automatically in the case of failure with a restoration time under 50 ms. Self-healing schemes use fully duplicated transmission systems and capacity for alternate routing of todays TDM or STM circuit facilities. The restoration capacity and the associated transmission systems are essentially unused, except in the rare occasions of network failure.

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Although expensive and relatively complex to implement, the dualhomed ring architecture is the choice for high-capacity digital service providers. This architecture uses a drop-and-continue feature that ensures that traffic is available to pass between adjacent rings at two separate nodes or offices. If an entire office is lost, the receiving ring equipment will select traffic from the other office or node. Although this architecture looks expensive, due to network survivability it offers a high potential for cost reduction in the long run. ATM features dynamic bandwidth allocation and ATM switches which together with SONET/SDH transmission will provide support for the emerging broadband multimedia services and existing legacy low-bandwidth telephony and data services. ATM can provide fault detection and traffic rerouting much faster than the existing 50-ms switching time requirement in the SONET/SDH networks. ATM-based capacity management and dynamic reconfiguration have the potential to significantly reduce the transmission facilities required for network survivability, providing economic justification for ATM deployments in large networks. 4.3.7.2 Cluster Topology

The cluster topology is a time-saving, cost-efficient, flexible, reliable, and future-proof way of building and implementing the transmission network for wireless systems. This topology is applicable to large wireless networks. In this context, the transmission network refers to the access network connections from the BSC to the RBSs. The basic idea is to group the RBSs into several independent subnetworks, or clusters. Each cluster has a separate connection to the BSC or to an intercity transfer point. Grouping the RBSs into several clusters creates the initial cluster topology network. The number of clusters and the number of RBSs in each cluster are dependent upon the total number of RBSs and their configuration. Some of the more important factors governing the design are listed below: • The total number of timeslots generated by the RBSs in a cluster

must not exceed the capacity of the link connecting the cluster to the BSC.

• The cluster size must support a flexible, uncomplicated, and effi-

cient network topology inside the cluster. A too large cluster will result in a large and inflexible topology.


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• In a microwave link network, the number of clusters connected to

one BSC must not be too high, as this will result in high concentration of microwave equipment at the BSC site, which may cause interference problems as well as tower stability problems (too many antennas).

• In general, a cluster size of 10 to 25 RBSs is recommended.

The RBSs in each cluster can be connected to a cross-connect node placed at a central hub-site in each cluster. The cross-connect node consolidates the traffic from the RBSs on the link to the BSC in order to minimize the required link capacity toward the BSC. Figure 4.14 gives an example of a cluster. The topology inside the clusters can be of any type: cascade star, tree, ring, or a combination of these. Each cluster is then directly connected to a hub site in the central cluster, which is either the BSC site or an intercity (backbone) transfer point connecting to the BSC cluster in another city, as shown in Figure 4.15. The RBSs in this central cluster can be connected directly to the BSC or the intercity transfer point. They can also be connected through a cross-connect node to the BSC or the intercity transfer = RBS site = RBS site with = cross-connect = node

To BSC or intercity transfer point

Figure 4.14 RBS cluster.

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Transmission Systems Design Handbook for Wireless Networks City A (no BSC)

Intercity connection

City B (BSC)

= RBS site = RBS site with = cross-connect = node = BSC

Figure 4.15 Intercity connection.

point. In a microwave link scenario, the links from the clusters toward the central cluster should be hardware protected as 1+1 to guarantee high availability. If required, considering the hop length the links should operate in the lower frequency bands to reduce fading due to rain. Operating in the bands below 10 GHz completely eliminates fading problems caused by rain. In the cluster topology, it is both easy and cost efficient to create a high level of redundancy. Redundancy is achieved by connecting several clusters in ring structures (Figure 4.16). A ring is built simply by adding a link between the cross-connect nodes in two of the outer clusters. All links in a microwave ring should be nonprotected 1+0 links. However, in the initial network the links from the outer clusters to the central cluster were 1+1 protected. Therefore the standby radios from the 1+1 links can be used for the new connections between the outer clusters required to create the ring, but also for expansion in other parts of the network, thus minimizing investment cost for new equipment. As shown in Figure 4.16, there are two possible paths from each cluster. Should the primary path be down due to a link failure, there is always a secondary path to the BSC. In a ring structure, all links in the ring must be able


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= RBS site = RBS site with = cross-connect = node = BSC Figure 4.16 Redundancy configuration.

to transport traffic from the other cluster(s) in the ring. Therefore, the existing links may have to be upgraded to higher capacity, unless the links were dimensioned for a future ring structure already in the design of the initial network. More complex redundancy schemes can be constructed by connecting rings to each other to form a meshed network. These connections should, as before, be made between the cross-connect nodes in the clusters. The cross-connect nodes in each cluster are also excellent points of connection to a fiber-optic network. The cluster topology is an ideal topology when introducing one or more remote BSCs. A remote BSC is a small and compact stand-alone node that can be introduced to add BSC capacity to high-density traffic areas. A remote BSC should, however, not be introduced initially but at a later stage when the traffic distribution becomes clear and when the need for such a node can be analyzed. The idea is to invest as the network grows. In the cluster topology, a remote BSC is introduced simply by replacing one of the cross-connect nodes with a remote BSC. Furthermore, by introducing a remote BSC in a cluster, the required capacity on the link from the cluster in question to the central cluster (MSC) is substantially reduced. The reduction

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is a consequence of the trunking efficiency of the remote BSC. Thus, there is no need for large rearrangement of the transmission network when a remote BSC is introduced. 4.3.7.3 Quality, Performance, and Availability

In todays wireless networks, with converged voice and data, performance degradation may be as dangerous and costly as hardware failures. A degraded transmission network can result in unacceptable signal transmission quality, loss of information, and dropped connections. High availability does not mean just preventing catastrophic failures but also preventing quality and performance degradation. BER, errored seconds (ESs), and one-way delay are usually parameters of interest that will define the quality of the transmission network. Some wireless technologies are more sensitive to delays in the T1/E1 links than others. For example, cdmaOne and CDMA2000 due to the soft handoff have much more stringent requirements on the network delays and synchronization than TDMA-based wireless technologies. High availability of the wireless network is an end-to-end network goalthe network management system (NMS) can help identify critical resources, traffic patterns, and performance levels. NMS can also be used to configure error thresholds, set corporate policies, and provide reports showing end-to-end results. Transmission-network survivability is usually measured by its long-term availability or average network uptime. Most operators expect their network to be continuously available (or at least as little downtime as possible) to minimize potential loss of revenue. The survivable wireless network has an infrastructure of transmission facilities and reliable network elements that are used to manage them. High network availability at the transport level may be achieved using millisecond restoration schemes provided by self-healing network configurations such as SONET/SDH rings or fast facility protection (FFP). DACS in combination with SONET/SDH ring configurations will ensure network availability and survivability. An FFP network comprises two physically diverse routes with identical transmission systems (route diversity). Each route carries half of the working traffic and half of the restoration traffic. The restoration traffic on each route is the duplicate of working traffic on the other route. If the media on these routes are different (i.e., one is fiber optic and the other is, for example, microwave), we talk about the media diversity. These highly reliable solutions dont come cheap, and in many cases a compromise must be made between the cost of the network and its deployment time and network reliability.


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Regardless of the transmission-network medium and topology, hardware redundancy is an option when designing the transmission network. Protection types usually used are 1+1, where one card or module serves as a protection for another one, or 1+N, where one card or module protects N other units. Linear 1+1 protection switching means that identical payloads will be transmitted on the working and protect fibers or working and protect frequencies in case of the microwave system. Liner 1+N protection switching assumes that there is one protect fiber or frequency for N working fibers or frequencies. A rule of thumb is that if all the hardware is protected with 1+1 and 1+N configuration, fewer spare parts are needed. In case of hardware failure, protection will kick in and the operator will have sufficient time to order replacement parts from the supplier. A ring configuration could provide protection against hardware failures as well, so additional hardware protection might not be required. This is something that transmission engineers must decide, and it is a decision that will be based not only on technical but also budgetary requirements.

References [1]

Bos, L. and S. Leroy, Toward an All-IP-Based UMTS System Architecture, IEEE Network, January/February 2001, pp. 3645.

[2]

McDysan D., and D. Spohn, ATM Theory and Applications, New York: McGraw Hill, 1998.

[3]

ETSI, UMTS-QoS Concept and Architecture, ETSI TS 123 107 V4.0.0, December 2000.

[4]

Malis, A.G., Reconstructing Transmission Networks Using ATM and DWDM, IEEE Communications Magazine, Vol 37, No. 6, June 1999, pp. 140145.

[5]

Liu, C. et al, Packing Density of Voice Trunking Using AAL2, Globecom 99 General Conference, 1999.

[6]

Eneroth, G., et al, Applying ATM/AAL2 as a Switching Technology in 3G Mobile Access Networks, IEEE Communications Magazine, June 1999, Vol 37, No. 6, pp. 112122.

[7]

McDaysan, D., and D. Spohn, ATM Theory and Applications, New York: McGrawHill, 1998.

[8]

ATM Forum, ATM Trunking AF-VTOA-0113.00, February 2000.

[9]

Roberts, Jim W., Traffic Theory and the Internet, IEEE Communications Magazine, January 2001, pp. 9499.

Using

AAL2

for

Narrowband

Services,

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[10] Hac, A., Multimedia Applications Support for Wireless ATM Networks, Upper Saddle River, NJ: Prentice Hall, 2000. [11] Zorzi, M., R.R. Rao, On the Impact of Burst Errors on Wireless ATM, IEEEE Personal Communications, Vol.6, No. 4, August 1999, pp. 6576. [12] Lankl, B., and M. Salerno, ATM Traffic and Its Impact on Radio System Design, Sixth European Conference on Fixed Radio Systems and Networks, Bergen, Norway, June 1998. [13] ITU-T G.804, ATM Cell Mapping into Plesiochronous Digital Hierarchy (PDH), February 1998.

5 Theory and Principles of Fiber-Optic Transmission 5.1 Basics of Fiber-Optic Transmission The three primary components of a fiber-optic link are the optical transmitter, the optical receiver, and the fiber-optic cable. In the transmitter, the input signal modulates the light output from the semiconductor laser diode, which is then focused into the fiber-optic cable. This fiber carries the modulated optical signal to the receiver, which then reconverts the optical signal back into the original electrical (analog or digital) signal. Fiber-optic communications offers several advantages over metallic systems. One of them is that any form of outside electronic, magnetic, or radio frequency interference does not distort the transmitted signals. Therefore, optical cables are completely immune to lightning or high-voltage interference. Furthermore, optical fibers will emit no radiation, which suits them for todays tougher standards for interference, also known as electromagnetic compatibility (EMC). Because optical signals do not require grounding connections, the transmitter and receiver are electrically isolated and free from ground loop problems. With no chance of terminal-to-terminal ground potential shifts, plus safety from sparking and shock, fiber optics is increasingly the choice for many processing applications where safe operations in hazardous or flammable environments is a requirement. Digital computing, telephone, and video broadcast systems require new avenues for improved transmission. The high signal bandwidth of optical 181

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fibers means increased channel capacity. Also, longer cable runs require fewer repeaters, because fiber-optic cables have extremely low attenuation rates. This suits them for broadcast and telecommunications use. Compared with conventional coaxial cables with the same signal carrying ability, the smaller diameter and lighter weight of fiber-optic cables mean relatively easier installation, especially in crowded duct areas. A singleconductor fiber-optic cable weighs about 6 lbs per 1,000 ft. A comparable coaxial cable weighs 80 lbs per 1,000 ftabout 13 times more. Electronic bugging depends on electromagnetic monitoring; fiberoptic systems are immune to this technique. They have to be physically tapped to extract data, which decreases signal levels and increases error ratesboth of which are readily detected. Some of the features of fiber-optic systems include the following: • All dielectric: low signal radiation, secure transmission, EFI and EMI

immunity, lightning immunity; • Optical signal: no ground loops, spark hazard, suitability for operation in flammable areas; • Low attenuation: greater distance and fewer repeaters; • Small size: less duct space, fewer additional ducts installed; • High bandwidth: future signal capacity expansion (especially today

with DWDM).

The cross section of an optical fiber is shown in Figure 5.1. Core refers to the light transmission area of the fiber, either glass or plastic. The larger the core, the more light that will be transmitted into the fiber. The function of the

Cladding Core Coating

Figure 5.1 Cross section of optical fiber.


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cladding is to provide lower refractive indexes at the core interface in order to cause reflection within the core so that light waves are transmitted through the fiber. Coating is a multilayer of plastic applied to preserve fiber strength, absorb shock, and provide extra fiber protection. These buffer coatings are available from 250 µm to 900 µm. Fiber size is commonly referred to by the outer diameter of its core, cladding, and coating. For example, 50/125/250 indicates a fiber with a core of 50 µm, cladding of 125 µm, and a coating of 250 µm. The coating is always removed when joining or connecting fibers. A micron (µm) is equal to one-millionth of a meter, and 25 µm are equal to 1/1,000 of an inch (a sheet of paper is approximately 25 µm thick). Fiber types can be identified by the types of paths (or modes) that the light rays travel within the fiber core. There are two basic types of fiber, multimode and single-mode. Multimode-fiber cores may be either step index or graded index. Step-index multimode fiber derives its name from the sharp steplike difference in the refractive index of the core and cladding. In the more common graded-index multimode fiber, the light rays are also guided down the fiber in multiple pathways. The effect of this grading is that the light rays are speeded up in the outer layers, to match those rays going the shorter pathway directly down the axis. The result is that a graded index fiber equalizes the propagation times of the various modes so that data can be sent over a much longer distance and at higher rates before light pulses start to overlap and become less distinguishable at the receiver end. Graded index fibers are commercially available with core diameters of 50, 62.5, and 100 µm. The core of the single-mode fiber is extremely small, approximately 5 to 10 µm. The single-mode fiber allows only a single light ray or modes to be transmitted down the core, virtually eliminating any distortion due to the light pulses overlapping. The single mode has a higher capacity and capability than either of the two multimode types.

5.2 Design Principles 5.2.1

Bandwidth and Attenuation

When selecting components for a fiber-optic system, there are two optical fiber factors that affect transmission performancebandwidth and attenuation. Bandwidth is the measure of the data-carrying capacity of the fiber. The greater the bandwidth, the greater the information capacity. Bandwidth is expressed in a frequency-distance form (MHz-km) and can be limited by

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transmitter, receiver, or dispersion of the optical fiber. Fiber can affect bandwidth due to the fact that different wavelengths travel at different speeds along the fiber. To avoid this chromatic dispersion, lasers with narrow optical bandwidths are used with fiber that has low dispersion. In addition to physical changes to the light pulse that result from frequency or bandwidth limitations, there are also reductions in the level of optical power as the light pulse travels to and through the fiber. This optical power loss or attenuation is expressed in decibels per kilometer at a specified wavelength. Light is an electromagnetic wave, and short wavelengths are in the ultraviolet spectrum, while microwaves, radar, television, and radio operate in the longest wavelength areas. In between the ultraviolet and the microwave spectrums, there are fiber-optic wavelengths, which are in the infrared spectrum. Just as the speed of light slows when traveling in transparent materials, each infrared wavelength is transmitted differently within the fiber; therefore, attenuation, or optical power loss, must be measured in specific wavelengths for each fiber type. Transmission loss or attenuation varies with wavelength ( l). Wavelengths are measured in nanometers (nm)billionths of a meterwhich represent the distance between two cycles of the same wave and are calculated as l=1 f where f is the frequency of the signal. Losses of optical power at the different wavelengths occur in the fiber due to absorption, reflection, and scattering. These occur over distance depending on the specific fiber, its size, purity, and refraction indexes. The amount of optical power loss due to absorption and scattering of optical radiation at a specified wavelength is expressed as an attenuation rate in decibels of optical power per kilometers. Fibers are optimized for operation at certain wavelengths. For example, less than 1 dB/km loss is attainable in 50/125 µm multimode fiber operating at 1,300 nm, and less than 3 dB/km (50% loss) is attainable for the same fiber operating at 850 nm. Two wavelength regions, 850 or 1,300 nm, are the areas most often specified for fiberoptic transmission but lately optical fibers have also been optimized at the 1,550 nm region for single-mode transmission systems. Without protection, an optical fiber is subject to losses of optical power caused by microbending; they are minute fiber deviations caused by lateral forces. Different types of protection for the fiber are available to minimize


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microbending. Step index fibers are relatively more resistant to microbending losses than graded index. When its freezing cold outside, fiber-optic cable enclosed in innerduct systems that are not buried beneath the frost line (for example, innerduct strung under a bridge) could be at risk. Innerduct is a smaller, usually PVC duct used for the protection of the fiber-optic cables within the larger cable ducts housing a large number of different types of cables. Exposed innerduct could collect water which if ice can exert crushing pressure on the fiber-optic cable inside the conduit. That pressure can cause microbends in the fiber and degrade the signal, especially at higher speeds, or it can even break the fiber. This freezing phenomenon was discovered in the early 1990s by a large telephone company, which noticed that its fiber-optic network experienced problems on cold nights in certain parts of the network. As it warmed up, the signal would be restored to its usual quality.

5.2.2

Optical Power Budgets and Distance Calculations

The key to network distance is the optical power budgetthe amount of light available to make a fiber-optic connection. This chapter will explain how to determine the maximum fiber-optic distances attainable using media converters in various network environments. A simple calculation is used to determine how much fiber-optic light, measured in dBm, is available. It tells us how many decibels over 1 mW of electrical, or in this case optical, power are available. The first step in calculating the optical power budget is determining how much light is available for the electronic devices themselves. Two measurements are needed from the manufacturer of the equipmentminimum transmit power and minimum receive sensitivity. Minimum transmit power represents the worst-case transmit power for a device guaranteed to provide at least that much power. It is important to note that some vendors will list an average minimum transmit power. Average minimum transmit power does not guarantee that a product will perform at that products minimum transmit power. The second piece of information required is the minimum receive sensitivity. This figure represents the minimum amount of light required by the receiver to operate correctly. Again, the actual minimum should be used, not an average of minimums. With minimum transmit power and minimum receive sensitivity data, it is possible now to calculate the available light as a worst-case scenario. In reality, the available light will probably be higher than this calculated value:

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available light = minimum transmit power − minimum receive sensitivity Minimum receive sensitivity is usually a negative number, such as −33 dBm. Subtracting a negative number is the same as adding its absolute value. For example, if a device has a minimum transmit power of −10 dBm and a minimum receive sensitivity of −33 dBm, the available power will be

(−10 dB) − (−33 dB) = 23 dB − 10 dB + 33 dB = 23 dB Also, when connecting devices from different vendors, or different models of products from the same vendor, the available power calculation needs to be computed in both directions with the smaller of the two used for the rest of the calculations. For example, assume connecting two devices labeled Device 1 and Device 2. Device 1 has a minimum transmit power of −3 dBm and a minimum receive sensitivity of −32 dBm, and Device 2 has a minimum transmit power of −1dBm and a receive sensitivity of −31 dBm. The available power going from Device 1 to Device 2 would be calculated by −3 dBm − (−31 dBm) or 28 dB. The available power going from Device 2 to Device 1 would be calculated by −1 dBm − (−32 dBm) or 29 dB. There is less light available in the Device-1-to-Device-2 direction, so we will use that figure for our calculations; if that half of the link works, then so will the other (worst-case scenario). See Table 5.1.

Table 5.1 Fiber-Optic Link Budget Device 1

Device 2

Min Tx Power

−3 dBm

−1 dBm

Min Rx Sensitivity

−32 dBm

−31 dBm

Budget 1

Budget 2

Device 1 Tx

−3 dBm

−1 dBm

Device 2 Rx

−31 dBm

−32 dBm

Available Power

28 dB

29 dB


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From available power we must subtract out all of the losses including cable attenuation, connector losses, and splice losses. Cable attenuation is often the most significant loss factor and should be determined by getting the exact number off of the cable being installed or using the manufacturers worst-case number of the type of fiber you planned to install. This number will range from 0.22 to 0.5 dB/km. Multiply this factor by the number of kilometers in the installation. A fiber with 0.4 dB/km of loss will lose 16 dB over a 40-km distance. Also, it is important to remember that fiber does not come in 40-km spools; therefore a 40-km installation will have several splices. Each splice will typically introduce 0.1db of additional loss, and the fiber installers should be able to provide a guaranteed worst-case number. The number of splices should be multiplied by this number. Connectors are another source of light loss. A typical long-haul installation will have six connectors in the installation. The first connects the fiber to electronics. This connector is usually on indoor fiber. This fiber connects the equipment room to the building entrance for the outdoor (buried or aerial) fiber. There is another connector on this end of the indoor cable and one on the outdoor cable. This is repeated at the other end of the network for a total of six connectors. Individual networks can vary, however, and the exact number must be determined. The connector manufacturer or the installer provide connector loss. Multiply the number of connectors by the loss for each connector to get total connector loss. Each of these lossescable attenuation, connector loss, and splice lossis then subtracted from the available power. If this number is negative, there is no need to continue, as there is not enough power to drive the network. If this number is positive, there are two more things to considerfirst, what happens if the fiber gets cut and has to be spliced back together? A proper design will count on this happening and account for it in the power budget; also, an estimation of the number of anticipated repairs over the life of the fiber needs to be made. These repairs will add splice loss, so we must multiply the number of anticipated splices by the loss of each splice (same number we used above), and subtract this from the remaining power. The number should still be positive. Finally, we must account for temperature extremes, as well as any other unforeseen factors. Determining a safety factor (margin) typically does this. This number will be different for every organization, depending on how much risk they want to assume in their network. Typically a value around 3 dB is used. To guarantee error-free operation, a value no less then 1.7 dB should be used. This safety factor is subtracted from the remaining power

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from above. If the number is still positive after all of this, we can be assured that the fiber network will deliver the required performance over the life of the installation. Table 5.2 contains some typical numbers that can be used to approximate optical-link budget calculations. If at all possible, real numbers from the network in question should be used. The simple worksheet presented in Table 5.3a can help in these calculations. Table 5.3b is an example that plugs the numbers from the previous discussion into the worksheet presented in Table 5.3a. Table 5.2 Fiber-Optic Losses TIA standard for connector loss

0.75 dB

Typical cable attenuation at 850 nm (MM)

3.0 dB

Typical cable attenuation at 1,310 nm (MM)

1.0 dB

Typical cable attenuation at 1,310 nm (SM)

0.4 dB

Typical cable attenuation at 1,550 nm (SM)

0.2 dB

Typical distance between splices

6.0 km

Typical safety margin

3.0 dB

Typical splice attenuation (usually < 0.05 dB for SM and < 0.1 for MM)

0.1 dB

Table 5.3a Optical Budget Calculator Minimum Transmit Power

________

Minimum Receive Sensitivity

________

Available Power

________

________ km of cable

×

_______ dB/km

________

________ connectors

×

_______ dB/con

________

________ splices

×

_______ dB/splice

________

Link Margin _______ repair splices ×

________

_______ dB/splice

________

Safety Margin

________

Excess Power

________


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Table 5.3b Optical Budget CalculatorExample Minimum Transmit Power

10 dBm

Minimum Receive Sensitivity

33 dBm

Available Power

23 dB

×

____0.5 dB/km

_____10.0

_______6 connectors

×

____0.75 dB/con

_____x4.5

_______4 splices

×

____0.1 dB/splice

______0.4

_____ 20 km of cable

Link Margin ______ 5 repair splices ×

______8.1

_____ 0.1 dB/splice

_____ 0.5

Safety Margin

______3

Excess Power

______4.6

5.3 Synchronous Digital Hierarchy 5.3.1

Basics of Synchronous Systems

The fiber-optic media have proven to be the best high-speed transmission media where electrical and physical environmental factors are more considerable for enterprise network infrastructure design. It may primarily resolve some of the networking issues, but still durability and survivability of linear fiber-optic implementation raise more questions regarding cable cuts, disaster recovery, and so on. Since initial fiber installation followed a linear route between endpoints and the back-up fiber was installed with same right of way (meaning that the main and back-up fiber are probably placed very close to each other and possibly even in the same cable or same cable duct), definitely creates serious restoration problem in case of a natural or man made disaster. Usually, the prediction and prevention of such an undesired condition are unavoidable. A fiber-optic cut may involve the primary and the back-up cable in a linear route. In addition to the actual cut itself, glass fibers may experience stress fractures several hundred feet in either direction from the cut. Restoration under such circumstances may take a few days or weeks to complete. The above factors were not alone sufficient to drive SONET as a significant

190


transmission technology. The clients increasing demand for bandwidth, error-free delivery, and related requirements were too much for the older PDH solution, so the decision was made to use SONET for transmission means. Above all, PDH had the following limitations: • Lack of performance; • Incompatibility of different vendors equipment; • Optical interfaces that were not universally defined; • No self-checking; • No standard for high-bandwidth links; • Not synchronous above DS1.

To find a solution for PDH problems, we need an error-free, synchronous, and high-speed technology. In 1984 Bell Communications introduced SONET, a high-speed fiber-optic system that provides an interface and mechanism for optical transmission of digital information with nearly errorfree conditions. Later, SONET was quickly accepted by ANSI. In 1988 the Comite Consultatif Internationale de Telegraphique et Telephonique (CCITT) published a similar set of standards called SDH, currently used in Europe and many other parts of the world. SONET/SDH are true implementations of fiber-optic media, and because they are typically WAN standards, SONET/SDH use a point-to-point connection type. They use time-division multiplexing over mesh and ring physical topology. One of the principal purposes of SONET is to improve the durability and survivability of fiber-optic networks. SONET/SDH are physical transmission mechanisms for various high-end implementations, such as FDDI, ATM, and SMDS. SONET is not a network, but is used as an infrastructure transmission means for the foundation of a high-quality network. It is a set of standards that define the rates and formats for optical networks as specified in ANSI T1.105, ANSI T1.106, and ANSI T1.117. Both SONET and SDH are now the industry standards for voice, video, and data optical transmission. Three key requirements have driven the development of SONET. The first was to push the multiplexing standards beyond 44.736 Mbps that is DS-3 level. A number of vendors have introduced their own proprietary schemes for combining multiple DS-3s into an optical signal. The second requirement is to provide economic access to small amounts of traffic within the bulk payload


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of a SONET signal. The third is to provide bandwidth on demand and virtual private networking with a sophisticated optical error-free mechanism. Almost all new fiber-optic transmission systems, as well as highcapacity microwave systems, now being installed in public or private networks use SDH or SONET. Bit rates in long-haul systems are expected to rise to over 40 Gbps over the next few years and at the same time, systems of 155 Mbps and below penetrate more deeply into access transmission networks. New 3G wireless networks will definitely have a big impact on the urgency of adding more and more capacity into the existing transmission networks. 5.3.2

Benefits of SONET

Some of the benefits of using SONET systems in the transmission-network design include the following: • Performance. SONET provides excellent performance with almost •

• •

•

•

5.3.3

100% error-free transmission through the optical medium. Integration. The SONET services combine bandwidth and multiplexing capabilities to let users fully integrate voice, video, and data over a single means of transmission facility. Security. SONET maximizes security and eliminates contention by containing the traffic on the dedicated private lines. Disaster recovery. SONET is designed to provide survivability and reliability to self-heal within 50 ms in case of any physical medium failure. High-speed transmission. The SONET/SDH supports up to 9.9 Gbps transfer rate. In the DS-level hierarchy, SONETs basic bandwidth is a DS-3 (44.736 Mbps) channel plus overhead. Stability and robustness. Todays SONET architecture offers more stable and robust communication that also eliminates multiple mapping and demapping multiplexers in the long distribution circuit.

SONET Architecture

5.3.3.1 Transmission Layers

At the interface, SONET converts digital signals from electrical form to optical form and back to electrical form at the destination. It supports a transmission range from 51.84 Mbps to multiples of DS3s, such as OC-248 (9.9

192


Gbps) and the multiplexing of lower-capacity channels, down to the 64Kbps DS0 channel. The SONET standard defines four layers to deal with the task involved in getting transmission from one end point to another. It can adjust timing and framing during operation with additional support to drop and insert capabilities that make it easier to identify and remove channels going to different destinations. SONET can be used as carrier service for ATM, SMDS, FDDI, and other broadband ISDN networks. It uses an 810-byte frame as its basic transmission and transmits 80,000 frames/s. The four layers of SONET are photonic, section, line, and path (see Figure 5.2). Photonic is the physical layer of SONET that deals with cable, signal, and component specification, such as the optical fibers minimum power requirements and the dispersion characteristics of transmitting lasers. Signals are converted between electrical and optical form in this layer. In the section layer, basic frames are created and scrambled whenever appropriate. It converts electrical signals to photonic ones and monitors for errors. The line layer is responsible for getting frames from one end of a line to the other. The synchronization and multiplexing of data onto the SONET frames are done in this layer, so timing adjustments and adding and dropping are done at this level. This layer is also responsible for protection, maintenance functions, and switching. The path layer completes the transmission (e.g., getting from one end to another with overall path); this layer provides transport of data at the appropriate signaling speed. At the end-point terminal, a signal is converted to synchronous transmission signal (STS), which is the building block of the SONET optical interface, with a rate of 51.84 Mbps. An STS signal travels through various SONET networks in STS format until it terminates. The terminating Path layer Line layer Section layer Frame Light Photonic layer

Terminal

Envelope Section Photonic

Line Section Photonic

Path layer Line layer Section layer Photonic layer

Regenerator

STS mulitplexer

Terminal

STS-N block

Figure 5.2 SONET transport layers.


193

equipment converts the STS to the user format. The STS consists of two parts: the payload and the overhead. The payload of STS carries the information portion of the signal, and the overhead carries the signaling as well as protocol information. These allow communication between intelligent nodes in the network, permitting operation, administration, maintenance, and provisioning (OAM&P) of a network from a central location. The end points are the source and destination for the DS-3 or smaller channel that makes up the SONET transmission. 5.3.3.2 Basic Frame Format

The functionality of SONET is achieved by defining the basic STS-1 signal and an associated byte-interleaved multiplex structure that creates a group of standard rates at N times the STS-1 rate (see Table 5.4). The frame structure must be defined describing how bits are assembled into standard units fit for transmission. N can be any integer value from 1 to 255. Today, the following values for N are available and defined as N = 1, 3, 9, 12, 18, 24, 36, 48, 96, 192, and 255. The transmission over fiber-optic facilities also describes an optical counterpart of the STS-1 signal called optical carrier level 1 (OC-1). Three STS-1 channels can be combined into one 155.52-Mbps (STS-3) channel by merging the frames, producing frames that are three times larger. Similarly, the STS-48 is 48 times the basic SONET rate of 2,488.32 Mbps. Table 5.4 High Data Rates STS Level

OC Level

STM Level Bit Rate (Mbps) DS3

001

001

51.84

001

003

003

01

155.52

003

006

006

311.04

006

009

009

466.56

009

012

012

04

622.08

012

018

018

933.12

018

024

024

1,244.16

024

036

036

1,866.24

036

048

048

16

2,488.32

048

096

096

4,976.64

096

192

192

9,953.28

192

255

255

13,219.20

255

194


The OC-1 signal forms the basic SONET transmission building block from which higher-level signals, such as OC-3, OC-48, and so on, are derived. For example, OC-48 systems carry 48 DS-3s (1,344 T1s). In the international arena, STS-3 equates with STM-1; thus, STS-48 equates with STM-16. It is possible to examine the SONET rate by applying the frame-format attributes to see how an STS-1 signal is formed. So, the STS-N will be multiples of STS-1 replaced by the value of N. A SONET frame is 810-bytes long and transmitted every 125 ms at 8,000 frames/s. Each frame is a coordination of 9 rows by 90 columns. In other words, the 810-byte frames are grouped into nine 90-byte portions that are then transmitted one after another. Therefore, the STS-1 line rate can be derived as follows: 9 rows × 90 columns × 8,000 frames/s × 8 bits/byte = 51.84 Mbps Three bytes, or the first three columns in each row, are the transmission overhead; the remaining 90 − 3 = 87 bytes are data or payload. The overhead in the first three rows is allocated for the monitor section; in the remaining six rows, it is for the line overhead. So, the STS-1 frame format can be summarized as follows: • STS-1 frame is the basic building block of SONET. • It consists of 810 bytes, 9 rows, and 90 columns. • There are 27 bytes of overhead formed from the first three octets of

each row, 9 used for section overhead, 18 for line overhead.

• Payload is 87 × 9 = 783 bytes. One column of the payload is the

path overhead, positioned by a pointer in the line overhead.

• Transmission is top to bottom, row by row, from left to right. • STS-1 frame is transmitted every 125 ms.

Column 4 to column 90 contain the SPE. One column in the SPE is used for path overhead. Both the section and line overhead include channels for communicating. These channels are used to send alarms and other administrative information. The line overhead includes several bytes for pointer. These allow the SPE to be moved. A good source of information on SONET and SDH vendors and analysis of key products available on the market can be found in [1].

Theory and Principles of Fiber-Optic Transmission 5.3.4

195

SONET Availability Requirements

In the SONET network design, the end-to-end availability objective for a DS1 channel transported along the 250-mile short-haul interoffice route is 99.98%. This is the equivalent of 0.02% unavailability or 105 minutes per year. This 0.02% unavailability along the route is due to unavailability of all facilities, including the terminal and repeater equipment, fiber-optic cable, connectors, splices, software failures, and procedural errors. This objective is linearly prorated with distance [2, 3]. The allocation to hardware reliability failures is 15 min/yr for a 250-km route and prorated for shorter distances. 5.3.5

SDH

ETSI defined the SDH for Europe. It is now used everywhere outside North America and Japan. SDH has provided transmission networks with a vendor-independent signal structure that has resulted in new network applications, the deployment of new equipment in new network topologies, and management by operations systems of much greater power than previously seen in transmission networks. The development of optical-fiber transmission systems made more complex standards possible. It was widely accepted that the new multiplexing method should be synchronous and based on byte interleaving and not on bit interleaving, as was the PDH. SONET is an ANSI standard; it can carry as payloads the North American PDH hierarchy of bit rates1.5/6/45 Mbps, plus 2 Mbps. SDH embraces most of SONET and is an international standard, but it is often regarded as a European standard because its suppliers, with few exceptions, carry only the ETSI-defined European PDH bit rates of 2/34/140 Mbps (8 Mbps is omitted from SDH). The flexibility of SDH can be used to best advantage by introducing new network topologies. Traditional networks make use of mesh and hub arrangements, but SDH, with the help of DACSs and hub multiplexers, allows these to be used in a much more comprehensive way. SDH enables these topologies to be combined with rings and chains of ADMs to improve flexibility and reliability across the core and access parts of the network. SDH defines traffic interfaces that are independent of vendors. At 155 Mbps they are defined for both optical and copper interfaces and at higher rates for optical ones only. These higher rates are defined as integer multiples of 155.52 Mbps in an N × 4 sequence, giving, for example, 622.08 Mbps (622 Mbps) and 2,488.32 Mbps (2.5 Gbps). To support network growth and the demand for broadband services, multiplexing to even higher rates is being

196


developed with upper limits set by technology rather than by lack of standards, as was the case with PDH. Each interface rate contains overheads to support a range of facilities and a payload capacity for traffic, and both of them can be fully or partially filled. Rates below 155 Mbps can be supported by using a 155-Mbps interface with only a partially filled payload area. An example of this is a microwave radio system whose spectrum allocation limits it to a capacity less than the full SDH payload, but whose terminal traffic ports are to be connected to 155-Mbps ports on a cross connect. Interfaces are sometimes available at a lower synchronous rate for access applications. North America has for some time used 51.84-Mbps SONET, and ETSI has defined a 34-Mbps SDH interface whose data rate is identical to that of a 34-Mbps PDH system. To support a range of operations, SDH also includes a management layer whose communications are transported within dedicated datacommunications-channel (DCC) timeslots inside the interface rate. They have a standard profile for the structure on network management messages, irrespective of vendor or operator. However, there has been very little agreement on the definition of the message sets to be carried, so there is very little interworking of management channels between equipment vendors at the SDH interface. More information on the management functions of the SDH can be found in [4].

5.4 DWDM 5.4.1

DWDM Overview

DWDM is a fiber-optic transmission technique that employs light wavelengths to transmit data parallel-by-bit or serial-by-character and can, at present, increase the bandwidth of the existing fiber-optic facilities up to 32 times. There is a huge importance of scalable DWDM systems in enabling service providers to accommodate consumer demand for ever increasing amounts of bandwidth. DWDM is becoming a crucial component of optical networks for the transmission of e-mail, video, multimedia, data, and voicecarried in IP, ATM, and SONET/SDH, respectively, over the optical layer. In order to squeeze more bandwidth out of their fiber networks, longhaul carriers are deploying dense-wavelength-division multiplexers to build backbones that might have dozens of channels riding on a single strand of fiber, with each channel operating at multigigabit speeds. Now the DWDM equipment vendors are moving their technology away from the WAN and positioning it in the metropolitan area network (MAN). The target customers


197

for metro DWDM technology are the incumbent local exchange carriers (ILECs) and competitive local exchange carriers (CLECs), cable companies, wireless-network operators, and even enterprise customers with dark fiber. 5.4.2

DWDM Capacity (Bandwidth) Requirements

To understand the importance of DWDM and optical networking, these capabilities must be discussed in the context of the challenges faced by the telecommunications industry, and in particular, service providers. Most U.S. networks were built using estimates that calculated bandwidth use by employing concentration ratios derived from classical engineering formulas, such as Poisson and Reeling. Consequently, forecasts of the amount of bandwidth capacity needed for networks were calculated on the presumption that a given individual would only use network bandwidth for six minutes out of each hour. These formulas did not factor in the amount of traffic generated by Internet access (300% growth per year), faxes, multiple phone lines, modems, teleconferencing, Internet, and data and video transmission. Had these factors been included, a far different estimate would have emerged. In fact, today many people use the bandwidth equivalent of 180 minutes or more each hour. No one could have predicted the network growth necessary to meet the demand. For example, one study estimated that from 1994 to 1998 the demand on the U.S. IXCs network would increase sevenfold, and for the U.S. LEC network, the demand would increase fourfold. In actuality, one company indicated that its network growth was 32 times that of the previous year, while another companys rate of growth in 1997 alone equaled the same size of its entire network in 1991. Another operator has said that the size of its network doubled every six months in that 4-year period. In addition to this explosion in consumer demand for bandwidth, many service providers are dealing with fiber exhaust in their networks. Today, many carriers are nearing 100% capacity utilization across significant portions of their networks. Another problem for carriers is the challenge of deploying and integrating diverse technologies in one physical infrastructure. Customer demands and competitive pressures mandate that carriers offer diverse services economically and deploy them over the embedded network. DWDM provides service providers with an answer to that demand (Figure 5.3). Use of DWDM allows providers to offer such services as e-mail, video, and multimedia carried as IP data over ATM and voice carried over SONET/SDH. Despite the fact that these formats (IP, ATM, and

198


DWDM/point-to-point optical transport

Fixed DWDM/multipoint network

l1 l2 lN

l1 l2 lN

...

... l1 l1

Optical XC reconfigurable DWDM/multipoint network

Capacity expansion

Bandwidth management

lk lk

oxc

Figure 5.3 Fiber-optic capacity expansion and bandwidth management.

SONET/SDH) provide unique bandwidth-management capabilities, all three can be transported over the optical layer using DWDM. This unifying capability allows the service provider the flexibility to respond to customer demands over one network. DWDM can be deployed in unidirectional and bidirectional applications [5]. Unidirectional applications use two fibers to send and receive traffic separately in the opposite direction. Bidirectional applications, on the other hand, use a single fiber to carry traffic in both directions. Bidirectional transmission, which typically supports fewer wavelengths than unidirectional transmission, is a good solution only if there is no need for upgrading to large capacity of hundreds of gigabytes per second in the network. 5.4.3

Network Growth and Flexibility of DWDM

Faced with the challenges of increased service needs, fiber exhaust, and layered bandwidth management, service providers need options to provide an economical solution. One way to alleviate fiber exhaust is to lay more fiber, and for those networks where the cost of laying new fiber is minimal, this will prove the most economical solution. However, laying new fiber will not necessarily enable the service provider to provide new services or utilize the bandwidth-management capability of a unifying optical layer.


199

A second choice is to increase the bit rate using TDM, where TDM increases the capacity of a fiber by slicing time into smaller intervals so that more bits (data) can be transmitted per second. Traditionally, this has been the industry method of choice (DS1, DS2, DS3, and so on). However, when service providers use this approach exclusively, they must make the leap to the higher bit rate in one step, having purchased more capacity than they initially need. Based on the SONET hierarchy, the next incremental step from 10-Gbps TDM is 40 Gbps, which is a huge leap that many believe will not be possible for TDM technology in the near future. This method has also been used with transmission networks that are based on either the SONET standard for North America or the SDH standard for international networks. The third choice for service providers is DWDM, which increases the capacity of embedded fiber by first assigning incoming optical signals to specific frequencies (wavelength λ) within a designated frequency band and then multiplexing the resulting signals out onto one fiber. Because incoming signals are never terminated in the optical layer, the interface can be bit-rateand format-independent, allowing the service provider to integrate the DWDM technology easily with existing equipment in the network while gaining access to the untapped capacity in the embedded fiber. DWDM combines multiple optical signals so that they can be amplified as a group and transported over a single fiber to increase capacity. Each signal carried can be at a different rate (OC-3/12/24, etc.) and in a different format (SONET, ATM, data, etc.). For example, a DWDM network, with a mix of SONET signals operating at OC-48 (2.5 Gbps) and OC-192 (10 Gbps) over a DWDM infrastructure, can achieve capacities of over 40 Gbps. A system with DWDM can achieve all this while maintaining the same degree of system performance, reliability, and robustness as current transmission systemsor even surpassing them. Future DWDM terminals will carry up to 80 wavelengths of OC-48, a total of 200 Gbps, or up to 40 wavelengths of OC-192, a total of 400 Gbps. By beginning with DWDM, service providers can establish a grow-asyou-go infrastructure, which allows them to add current and next-generation TDM systems for virtually endless capacity expansion. Carriers can address specific problem areas that are congested because of high-capacity demands. This is especially helpful where multiple rings intersect between two nodes, resulting in fiber exhaust. Service providers searching for new and creative ways to generate revenue while fully meeting the varying needs of their customers can benefit from a DWDM infrastructure as well. By partitioning

200


and maintaining different dedicated wavelengths for different customers, for example, service providers can lease individual wavelengths. Furthermore, DWDM allows service providers to increase capacity on a broad range of wavelengths in the 1.55-µm region. For example, with a DWDM system multiplexing up to 16 wavelengths on a single fiber, carriers can decrease the number of amplifiers by a factor of 16 at each regenerator site. Using fewer regenerators in long-distance networks results in fewer interruptions and improved efficiency. 5.4.4

Optical Layers

Aside from the enormous capacity gained through optical networking, the optical layer provides the only means for carriers to integrate the diverse technologies of their existing networks into one physical infrastructure. DWDM systems are bit-rate- and format-independent and can accept any combination of interface rates (e.g., synchronous, asynchronous, OC-3, -12, -48, or -192) on the same fiber at the same time. If a carrier operates both ATM and SONET networks, the ATM signal does not have to be multiplexed up to the SONET rate to be carried on the DWDM network. Since the optical layer carries signals as-is without any additional multiplexing, carriers can quickly introduce ATM or IP without deploying an overlay network. But DWDM is just the first step on the road to full optical networking and the realization of the optical layer. The concept of an all-optical network implies that the service provider will have optical access to traffic at various nodes in the network, much like the SONET layer for SONET traffic. Optical wavelength add-drop (OWAD) offers that capability, where wavelengths are added or dropped to or from a fiber, without requiring a SONET terminal. But ultimate bandwidth management flexibility will come with a crossconnect capability on the optical layer. Combined with OWAD and DWDM, the optical cross connect (OXC) will offer service providers the ability to create a flexible, high-capacity, efficient optical network with full optical bandwidth management. These technologies are todays reality: DWDM has been utilized in the long-distance network since 1995; OWAD products became available in 1998; and the first all-OXC was showcased at industry conventions in 1997. 5.4.5

Protection in DWDM Networks

The basic types of network failures generally considered are link and node failures. Link failure usually occurs because of cable cuts, while node failure is


201

due to equipment failure at network nodes. Besides node and link failures, which are common failure situations in any communication network, channel failure is also possible in DWDM optical networks [6]. Offering protection on a per-channel basis has become an important factor as equipment providers roll out their metro DWDM systems. A number of equipment suppliers already offer wavelength protection on a per-channel basis as part of their metro platforms. Per-channel protection is becoming increasingly important because carriers want to cap their investments in legacy SONET/SDH equipment but still protect higher layers of traffic. Because SONET/SDH already offers protection, the ability to turn DWDM protection on and off depending on the traffic type is essential. The reason behind per-channel protection is dualif one channel suffers a failure and if you dont have protection on a per-channel basis, all channels suffer a 50-ms switch. If half of the channels already have SONET/SDH protection, it is not necessary to provide duplicate protection.

5.5 Optical Switching Optical networking provides the backbone to support existing and emerging technologies with almost limitless amounts of bandwidth capacity. Most current networks employ electronic processing and use the optical fiber only as a transmission medium [7]. All-optical networking (not just point-to-point transmission) enabled by optical cross connects, optical programmable adddrop multiplexers, wavelength routers, and optical switches provides a unified infrastructure capable of meeting the telecommunications demands of today and tomorrow. DWDM is a key technology in long-haul networks, because it gives carriers the ability to multiply the capacity of fibers by factors of 16, 32, and even higher. Service providers are moving from leasing dark fiber (fiber without terminal equipment), the physical medium, to leasing light paths, an optical end-to-end path through the network. Carriers are starting to offer individual wavelengths providing clear channel OC-48c capacity to their wholesale customers. The service provides a transparent interface with a customers network for completion of SONET rings, construction of new or diverse routes, and transmission of ATM or IP traffic. The digital cross connects are able to switch a large number of circuits, but they have to convert optical signals to electrical, switch them, and then convert them back into wavelengths of light and send them out. New optical cross connects do the switching on the optical level without converting optical signals into the electrical signals. Miniature mirrors with

202


surfaces no bigger than the diameter of hair are bouncing waves off the surface whose angles are changed by the small changes in temperature. Optical switches aim for the ability to provision bandwidth intelligently and switch and route wavelengths based on demand, the way standard Class 5 switches do today, but in the optical layeron a wavelength level rather than an individual voice channel. DWDM in combination with optical switching will provide a huge increase in the capacity of the new and existing transmission systems. It is probably the only available solution for OC-768/STM-256 (40 Gbps), OC-1920/STM-640 (100 Gbps), and OC-3840/STM-1234 (200 Gbps) systems over one fiber pair. Also, a wavelength-routing device can route signals arriving at different input fibers (ports) of the device to the different output fibers (ports) based on the wavelengths of the signals. A new optical internetworking initiative, the Optical Domain Service Interconnect (ODSI), is building interoperability between electrical and optical systems and among optical vendors, as well. ODSI is attempting to develop an interface that will allow electrical network elements, such as routers, ATM switches, and cross connects, to provision services, such as setting up and tearing down, on optical networks, on demand.

References [1]

Goralski, W., SONET, Second Edition, New York: McGraw-Hill, 2000.

[2]

Lewin, B., SONET Equipment Availability Requirements, IEEE, 1989.

[3]

Bellcore TR-TSY-000418, Generic Reliability Assurance Requirements for Fiber-Optic Transport Systems, Issue 2, September 1989.

[4]

ITU-T Recommendation G.784, Synchronous Digital Hierarchy (SDH) Management.

[5]

Fujitsu, Wave Division Multiplexing, white paper, April 1998.

[6]

Zhou, D., and S. Subramanim, Survivability in Optical Networks, IEEE Network, November/December 2000, pp 1623.

[7]

Sivalingam, K. M., and S. Subramaniam, eds., Optical WDM Networks: Principles and Practice, Norwell, MA: Kluwer Academic Publishers, 2000.

6 Microwave Point-to-Point System Design 6.1 Basic Microwave Transmission Theory RF signals can be transmitted over a wide range of frequencies, expressed in cycles per second, or hertz (Hz). Furthermore, 1 kilohertz (kHz) equals 1 thousand Hz, 1 megahertz (MHz) equals 1 million Hz, and 1 gigahertz (GHz) equals 1 billion Hz. Amplitude modulation (AM) radio signals are at the lower end of the RF spectrum, while other radio services, such as analog and digital TV (DTV), cellular and PCS telephony, and point-to-point microwave (MW) services are much higher in frequency. Free-space loss is defined as the loss between two isotropic antennas in free space in the absence of ground or atmospheric influences or obstructions; in other words, where fades due to refraction, reflection, and diffraction fade activity and atmospheric absorption do not occur. Radio energy is lost in free space because of the triangular spreading of energy in the wavefront as it travels through space in accordance with the inverse-square law (wavefront area quadruples for each doubling of distance for a 10 log 1/4 = 6 dB increase in path loss). The derivation of the free-space loss formula is presented in other references and will not be repeated here. The formula itself, however, is as follows: A = 96.6 + 20 log f + 20 log D 203

204


where A = free-space attenuation between antennas (dB); f = frequency (GHz); D = path length (miles); or A = 92.4 + 20 log f + 20 log D where A = free-space attenuation between antennas (dB); f = frequency (GHz); D = path length (km). The unfaded receive-signal level (RSL) is computed by introducing the net path loss (NPL) in decibels between a radio transmitter output and the far-end radio receivers input. The microwave radio transmit power is reduced by the NPL as determined by free space and atmospheric absorption losses, antenna gains, and waveguide feeder and network losses at both ends of the single radio link (or hop, as it is commonly referred to). Unfaded RSL at the distant receivers RF input is RSL = Pt − NPL = Pt + G t − L al + G r − L f − L mis where RSL = unfaded RSL (dBm); Pt = transmit power (dBm); Gt = transmit antenna gain (dB); Lfs = free-space loss (dB); Lal = absorption loss (dB); Gr = receive-antenna gain (dB); Lf = coax or waveguide feeder loss (dB); Lmis = network and miscellaneous losses (dB). The propagation of radio waves is generally affected by several factors, irrespective of the radio communication service or the specified purpose of telecommunication. These factors are described below.


205

Frequency Effects

The relative importance of the factors influencing the propagation of radio waves mainly depends on the frequency band. In the low-frequency (LF) and medium-frequency (MF) bands the propagation is strongly influenced by the electrical characteristics of the ground and by the ionospheric conditions. In the VHF and UHF bands the effects from the ionospheric conditions decrease to such an extent that the terrain features and, to some degree, the meteorological characteristics of the troposphere (the lower parts of Earths atmosphere) begin to dominate. At microwave frequencies, the importance of the terrain features and the meteorological characteristics of the troposphere are still predominant. However, above about 6 GHz the effects of gas absorption and precipitation must also be taken into account. At frequencies close to 10 GHz the effects of precipitation begin to dominate. Gas absorption starts influencing at about 22 GHz, where the water vapor shows a characteristic peak. Radio waves theoretically travel in a straight line in a vacuum. However, due to nonhomogeneous pockets of warm and cold air, in actuality waves can and do bend much the same as a light wave propagates through a prism. Long waves will bounce from the ionosphere above the Earth and travel back to Earth, thereby allowing an electromagnetic signal to be transmitted over a very long distance to a remote receiver. Microwaves and other very short waves (for example, 15, 23, and 38 GHz) do not bounce off the ionosphere and suffer much higher attenuation as they travel through the air. In addition, millimeter waves are subjected to rain losses (above 10 GHz), which can impact availability of the radio link if there is no sufficient fade margin to handle the rain-fading effect. Terrain Effects

When radio waves propagate near the surface of the Earth, their characteristics are dominated by the electrical characteristics of the Earth and by the topography of the terrain, including the vegetation and man-made structures. Tropospheric Effects

The gaseous constituents of the atmosphere influence the propagation of radio waves both by absorbing energy and by variations in the refractive index. Variations in the refractive index of the atmosphere cause radio waves to reflect, to refract, and to scatter. The magnitude of these effects depends, of course, on the frequency.

206


Multipath Effects

The term multipath effects applies to those cases in which the effective received signal is made up of several components arriving at the receiving antenna over different paths. The components may have different phases and different amplitudes, and their mutual relationship may also vary continuously with time. Multipath effects result from reflections from buildings, from the surface of the Earth, or from horizontal interfaces between different layers in the atmosphere. Multipath effects caused by reflections are responsible for the fast fading observed on microwave radio links. They can seriously degrade the quality of a service. On frequencies below 10 GHz, multipath fading is the main concern of microwave engineers. Multipath fades result when there are two or more transmission paths for a communication link and there are many active transmission paths for the signal to propagate on, and all contribute to the signal at the receiver. Signals that arrive delayed have been phase shifted as a function of frequency with respect to those signals that travel a shorter path. At some frequency the delayed signal phase change will be 180° out of phase with the nondelayed signal (frequency-selective fading), and at this frequency the received frequency spectrum will contain a notch, since the two received signals will add destructively. The depth of the notch will depend on the strength of the delayed and nondelayed signals and will introduce dramatic distortions, which must be corrected in some way; the use of antenna space diversity or equalizers is a potential solution. To test in the lab (two radios in back-to-back configuration) the receivers ability to cope with dynamic effects of multipath fades, it is necessary to be able to accurately simulate these degradations in the lab environment, and by measuring M-curves (or signature) determine the equalizer performance. The M-curve must always include the BER threshold value and the delay. The M-curve is the most common multipath measurement, and the smaller the M-curve, the better the radio can cope with the multipath.

6.2 Theoretical Aspects of Microwave Link Design 6.2.1

Microwave-Radio-Path Calculation Overview

This is an overview of the basic calculations associated with the digital microwave radio path using widely recognized performance and availability objectives. More information can be found in [13]. Calculations of the


207

short-term (due to multipath fading) outage time are based on Vigants widely used, field-verified model as defined in [4] for North American links and [5] for international (ITU-R) links. Newer ITU-R Rec. PN.530-7 multipath-outage-prediction methods (with Hosoyas space-diversityimprovement model) can produce results widely different from Vigants model, especially for flatland paths using space diversity in difficult geoclimatic regions. Long-term (due to rain fading) outage-time calculations are based upon Cranes model, which user-selects either Cranes or ITU-Rs point rain-rate tables for both North American and international (ITU-R) radiorelay links. Cranes point rain rates [6] are in close agreement with ITU-R point rain-rate data [7] in all regions worldwide. The actual rain regions and their letter designations in Crane and ITU-R are, however, different. While the Crane data may be used for international rain calculations (and vice versa), Crane regions are usually user-selected for North America and ITU-R regions are usually selected internationally. The ITU-R PN.530-7 rain model, which typically computes lower rain attenuation and, thus, less rain outage on shorter high-frequency links, will not be discussed here. 6.2.2

Design Fade Margins

The thermal (or flat) fade margin is the difference between the unfaded receive signal level and the receivers static or dynamic threshold, as measured with back-to-back radios, at a given BER. In digital microwave links, MW equipment specifications provide the ability to select either the 10−3 BER dynamic threshold for outage computations or the 10−6 BER static threshold for other performance and availability computations. It is important to understand that an internationally defined outage in a digital radio link corresponds to a 10−3 BER severely errored second (SES) event at the receivers dynamic threshold, very near the initiation of a T1 or E1 alarm (AIS) condition in the connected PABX trunk, channel bank, transmultiplexer, and so forth. The 10−6 digital-radio-static threshold (operating point) is used for factory and in-service manual measurements (with attenuators) and as a measure of circuit quality, not outage, although it is sometimes assigned as the outage threshold by some users. This method could lead to overdimensioning of the MW system, making it more expensive than necessary. A few digital microwave radio links with adverse geometry are susceptible to dispersive fading (spectrum distorting), as well as interference and flat fading, discussed above [8, 9]. Dispersive fading may cause the loss of the

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digital receiver synchronization due to amplitude or group delay slope and notch effects even in the absence of significant flat fade activity. In addition to antenna heights, sizes, and vertical alignments optimized to the path geometry for minimum dispersive fading, effective countermeasures include space diversity (SD), adaptive time domain equalization (ATDE), and adaptive IF slope amplitude equalization (ASAE). As the RF bandwidth or number of signaling states increases, the effect of dispersive fading intensifies, requiring these more sophisticated countermeasures typical of todays digital microwave receivers. Fade margins in digital radios are far more complex than in analog microwave systems. In addition to the thermal (flat) fade margin, dispersive and interference fade margins may impact upon the performance of some digital links and, thus, have to be considered. The formula for the composite fade margin (CFM), accommodating all of these influences upon digital link performance, is CFM = TFM + DFM + IFM The dispersive fade margin (DFM) provided by the manufacturer is included, where appropriate, in the CFM computation. It is derived from the digital radios 6.3-ns multipath delay signature curve for preliminary link design, where the actual delay (and therefore the actual link DFM) from path geometry calculations has not yet been made. Final design (beyond the scope of this tutorial) computes the link DFM (with antenna discriminations and actual multipath delay) from path geometry computations. The term interference fade margin (IFM) (usually TFM + 6 dB, a TIA standard) defines the digital links vulnerability to cochannel and adjacent channel interference and is provided by the frequency search company based on the manufacturers threshold-to-interference (T/I) curves and the interference ambiance. IFM is often ignored in the preliminary link design. Based on the standard TIA-TSB-10F, a 1-dB or other threshold (fade margin) degradation due to interference may be entered as a miscellaneous loss, if desired. The combining of different decibel power levels requires more than simply arithmetic addition or subtraction: CFM = −10 log (10 − TFM /10 + 10 − IFM /10 − 10 − DFM /10 ) where TFM = thermal (flat) fade margin (dB);


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IFM = interference fade margin (dB); DFM = dispersive fade margin (dB). Once the composite or thermal (flat) fade margin has been determined, it is used to compute outage times and expected one-way path reliability or two-way availability. Path reliability (from the annual one-way short-term outage time) is influenced by the effects of the following: • Multipath fading, defined by geographical propagation characteris-

tics (climate and terrain) for given locales; • Average annual temperature for the same geographical region.

CFM is also input for computing the expected annual two-way path availability in high-frequency links that accommodates all long-term (10 CSES traffic disconnect) rain outage events. Annual rain outage and path availability are computed from the following: • Rain attenuation tables, which assign coefficients based upon fre-

quency and polarization; • Rain-rate tables based upon thunderstorm and similar high rain-rate activity in all worldwide regions.

Rain outage is considered a self-healing unavailability event, and unlike other such events, rain outage is statistically predictable over a long-term (more than one year) period. Equipment, antenna system, and infrastructure failures and manual intervention (e.g., switching), which could also cause long-term outage, are not a part of these two-way availability calculations. The fade margin of a digital or analog microwave radio path provides a safety margin to protect the microwave signal from the adverse effects (carrier-to-noise degradations) of multipath fading, interference, and rain attenuation. Digital-microwave-link fade margins are typically smaller than for analog radio links whose fade margins are often increased to provide baseband quieting (low thermal noise) even in short and other nonfading paths. The provision of an adequate fade margin and path clearance to protect against long-term CFM degradations and outage due to surface ducting and earth blocking, plus optimum diversity on longer or otherwise vulnerable

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paths, assure per-hop propagation reliabilities in a Rayleigh-distributed multipath fade environment exceeding 99.999%. Path reliability and quality [residual bit error rate (RBER), %EFS, etc.], define digital-radio-link performance during traffic availability periods. Path availability is that percentage of time that a given microwave link is operational (traffic is not dropped or disconnected and performance measurements are possible) over a specified period, typically a year. With the exception of high-frequency (10 GHz) microwave links in rain areas, the availability objective of most microwave links is 100%. Predicted (measured) only during available (traffic-connected) periods, path reliability is a measure of annual short-term (CSES) one-way multipath fade outage occurring over a 24.5-month fade season. 6.2.3

Diversity Improvement

The probability of outage for a diversity system is U d = U nd /I d where Ud = one-way probability of outage for a diversity path; Id = diversity-improvement factor. The diversity-improvement factor differs for each type of diversityspace, frequency, hybrid, and so on. Space-diversity (SD) systems provide more efficient use of the spectrum, as well as extremely good diversity protection. In a typical diversity system, the radio at each terminal may contain an unprotected transmitter or Monitored Hot Stand By (MHSB) transmitters and redundant receivers (Figure 6.1). The transmitter is energized and operates on the same RF channel as

Data in

Tx A Tx B Rx A

Data out

Switch Rx B

Switch Antenna Data in −1 dB Power splitter −7 dB

Tx A Tx B Rx A

Data out

Figure 6.1 Hot stand by and space diversity.

Switch

Switch Rx B

−1 dB

Main antenna

330 m. 1090 ft. Vertical separation

Diversity antenna


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the diversity receivers at the far end. It uses the same antenna as one of the diversity receivers at the same site. Both receivers operate on the same RF channel (as the far-end transmitter); however, each receiver uses a separate antenna. When equipment failure or path fading affects the main path to the point where the signal is degraded, a decision is made (electronically) at the receiver end of the path as to which RF path is placed on-line. The SD arrangement has two receivers connected to vertically spaced main and diversity dishes. Hitless or errorless switches are used for switching the signal between the main and diversity branch. In a frequency-diversity (FD) system, the radio at each terminal contains redundant transmitter-receiver pairs. Each transmitter operates on a different RF channel and both transmitters are energized, and similarly, each receiver operates on a different RF channel, but is identical to the corresponding transmitter at the far end. When equipment failure or path fading affects an RF channel to the point where the signal is degraded, a decision is made at the receiver end of the link as to which RF channel is placed on-line. There are a number of other diversity methods for improving the particular microwave hop performance. Angle diversity (AD) has been used in LOS digital microwave links since the mid-1980s and in troposcatter links since the 1950s. The calculations use Vigants model, requiring vertically spaced dishes, and assume flat (analog) rather than dispersive (spectrumdistorting digital) fade activity on a path. This method used to be popular when digital radios were less robust, but today AD antennas are assigned mostly where installation constrictions (space, aesthetics, tower loading, and so on) prohibit SD and, thus, justify these less effective, more costly dishes. The AD antenna is a single dish with two feeds vertically offset by about 1° (the smaller the better). AD is most effective when path outages are dominated by dispersive fade activity (dispersive fade outage approaches or exceeds flat fade outage). As reported in the 1980s in AD studies [10], links with several of the above characteristics showed improvement factors ten or more times greater than space diversity (500 versus 50, for example). Lowercapacity radio links and those with new digital radios with more robust modulation and adaptive countermeasure schemes realized less improvement from angle diversity. Depending on path geometry and climatic conditions, AD improvements of perhaps 20 or even much more can be achieved. Optimum AD improvements are only obtained through an antennaalignment procedure that matches the antenna size and alignment to the path and its climatic characteristics. AD dishes require a more exacting, long-term alignment procedure than that for space and nondiversity antennas.

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Hybrid diversity (HD) is an enhancement of space diversity, using FD, when permitted (SD + FD). Hybrid diversity is the most effective of all of the diversity arrangements and is preferred in difficult propagation areas and in space-limited mountaintop, urban areas, and other sites restricted to single antennas. It is possible to compute frequency- and hybrid-diversity improvements for links in regions where regulatory rules or waivers so permit. The hybrid-diversity improvement factor, Ihd, is derived either from the SD or FD improvement factor Ifd described above. Above 3 GHz, Isd is nearly always higher (better) than Ifd and is therefore selected unless the diversity spacing exceeds about 5% (300 MHz/0.3 GHz in the 6-GHz band, for example). Below 3 GHz, Ifd is usually larger than Isd and therefore selected. The higher T&R frequencies must always be assigned to the upper antenna at the SD (usually the lower elevation) end of each HD link for optimum performance. 6.2.4

North American and ITU Objectives

The recommended short-term one-way outage objective for a T1/E1 trunk or FDM circuit, regardless of system length, is 1,600 outage seconds per year (SES/yr) for a 99.995% end-to-end propagation reliability. This is as equally true of a 5-hop short-haul system as it is of a 150-hop long-haul system, except that in a very long-haul system (perhaps > 50 hops), only half are considered as fading hops. A 99.999% per-hop reliability (320 SES/yr outage) objective (floor) is often assigned in spur links and on short systems of less than about 5 tandem hops. Although computed over the fade season, this is considered an annual outage compared, perhaps, to Bell short-haul (< 400 km/250 mi), Bell longhaul (6,400 km/4,000 mi), or other end-user objective. The long-term rain outage in a high-frequency (10 GHz) microwave link is often engineered to this same 99.995% objectivenow an annual two-way rain availability, not a one-way multipath reliability, objective. This corresponds to the ITU-R per-hop availability objective, since the 60-hop 2,500-km ITU-R reference circuit end-to-end two-way availability objective of 99.7% [11] scales to a 99.995% availability objective for a typical 40-km link. The 1,600 SES/yr outage objective for a system (longest one-way T1 trunk or FDM message circuit) may be over links making up a part or the whole of a long-haul system, a short-haul system, or other distance. While the following computations are for link(s) making up a full short- or long-


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haul system, the 1,600 end-to-end SES/yr outage allocation may be proportioned only to the actual number of tandem hops in an actual system. That means the assignment of a 1,600/20 = 80 SES/yr (99.99975% path reliability) per-hop objective for a 20-hop digital microwave system. The ITU publishes recommendations for the telecommunication and radio communication areas. The recommendations for the telecommunication part are published by ITU-T, whereas the radio communication part is published by ITU-R. The ITU also provides maps and data, proven adequate and reliable, which allow identification of the applicable rain-rate statistics in any part of the world. For this purpose, the ITU has divided the world into 24 regions, each identified by similar rain-rate statistics. For example, in Europe, the maximum rain rate (i.e., rain rate not exceeded for more than 0.01% of the time) varies from about 20 mm/hr in southern Europe to about 40 mm/hr in most parts of Mediterranean Europe. It peaks at 60 mm/hr in small areas of southern Europe. In total, the ITU identified six regions in Europe and five regions in the United States, each with the same maximum rain rate. Regions in the United States range from 19 mm/hr on the west coast to 63 mm/hr in the southernmost states (based on 99.99% availability). The recommendations G.801, G.821, and G.826 define error performance and availability objectives. The objectives for digital links are divided into separate grades. These are high, medium, and local grade. The medium grade has four quality classifications. The following grades are usually used in wireless networks: • Medium-grade Class 3 for the access network; • High grade for the backbone network.

The ITU Recommendation G.703 defines the physical and electrical characteristics of digital interfaces, while Recommendation G.704 defines synchronous frame structures. These two recommendations are followed for all links in the transmission network. An RBS site becomes unavailable when the transmission link connecting it to the next site or BSC becomes unavailable. ITU Recommendation G.821 provides guidance on allowable unavailable time (AUT) for networks. For wireless networks, transmission link performance is typically based on meeting ITU medium-grade Class 2 or Class 3 objectives. They are based on a hypothetical reference connection, as depicted in Figure 6.2. Specific details concerning the application of performance and availability objectives to telecommunications connections can be found in ITU-T

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Transmission Systems Design Handbook for Wireless Networks 1,250 km

LE Local grade

25,000 km

IG

1,250 km

LE

IG High grade

Medium grade

Medium grade

Local grade

IG = International gateway LE = Local exchange

Figure 6.2 ITU-T G.821 hypothetical reference connection.

Recommendation G.821, and the application of these objectives to digital radio-relay systems (microwave links) is detailed in ITU-R Recommendation 696-2. Table 6.1 shows the Rec 696-2 medium-grade performance and availability objectives that are typically applied to the design of microwave links operating at fundamental bit rates below 2 Mbps (this covers PDH links typically used in the access network). These ITU objectives are typically applied to microwave links in wireless networks. If we take the medium-grade Class 3 objective and consider the unavailable time (U) objective: U = 0.05% per year The intention of the ITU is that this objective should be applied to an entire connection; for example, if there is a chain of 10 links connecting RBSs to Table 6.1 Medium-Grade Performance and Availability Objectives Percentage of Any Month Performance Parameter −3

Percentage of Any Month

Class 2 (280 km) Class 3 (50 km)

SES (BER10 )

0.0075

0.002

ES

0.16

0.16

Unavailability

0.05

0.05


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the BSC, then the unavailability objective for each link should be 0.05%/10 for each link, or 0.005%. In a real network there may be fewer than 10 links connecting a series of RBSs to a BSC; however, the network operator may still choose to apply the figure of 0.005% to each link, to build in a performance margin. 6.2.5

Reliability and Availability Overview

In considering the assignment of a realistic short- or long-term outage objective, several things need to be kept in mind. A single overall design objective of not more than x hours, minutes, or seconds outage over some period such as a year, is an oversimplification. The character of the particular kind of outage and its effect on the system should be taken into account, and perhaps there should even be different objectives for different types of outages. For example, propagation outages due to multipath fading are usually short. A cumulative outage of an hour per year due to multipath might represent thousands of individual outages, each averaging one second or less (one SES) on a properly engineered path. On the other hand, propagation outages totaling an hour per hop due to rain attenuation might consist of only four or five individual outages averaging 10 to 15 minutes each. The effects of long-term and short-term system outages on trunks are very different. The many short-term unreliability outage events do not disconnect circuits or reduce (in most circuits) data throughput. The few longterm unavailability events cause both traffic disconnect and loss-of-data throughput. A distinction should be made between those circuits for which an outage of a few seconds or a few minutes is just an inconveniencefor example, spur cell site in the wireless networkand those circuits for which such an outage might result in a danger to life, great economic loss (important or hub sites in a wireless network), or other catastrophic consequences. The suitability or unsuitability of a higher frequency band such as 18- or 38GHz, in a high-rain-rate region could differ widely for these two situations. Even if the maximum possible reliability and availability objectives are established and a path or a system is engineered to the full limit of the state of the art, the probability of outage can never be eliminated, but only reduced to a very low value. Thus, it is imperative to make any very important services as fail-safe as possible against a loss of the communications channel. The system should be engineered with appropriate protection schemes and diversity arrangements so that short- or long-term outages are tolerated, or at least kept within acceptable bounds. Ring (route diversity) protection is often used to eliminate long-term rain outage, for example.

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It seems that in most cases, a more relaxed attitude might be taken toward rain-induced outages than toward multipath outages or even equipment outages. In several respects, rain outage is somewhat benign in nature. If the fade margins are kept high and the paths are not stretched out too much, even in less advantageous areas, the number of outages per year should not be very large. The length of individual rain outages on a hop should only rarely exceed 5 to, perhaps, 10 minutes. Short (fewer than 2 seconds) microwave outages, common on a typical longer diversity or shorter nondiversity digital microwave link with adequate fade margin, will not drop telephone or data lines. Such outages quickly clear with all circuits remaining connected and little note taken of these transient events. Longer outages associated with low fade margins, rain, and so on, disconnect all subscribers and may block access to a digital link for at least 10 seconds after each long-term outage event. Such traffic disconnects are unacceptable to most users; thus, these more vulnerable links clearly require appropriate diversity or ring protection. For high-reliability links (usually in long-haul systems with many hops in tandem), the per-hop objective may approach or exceed 99.9999%, allowing only 20 to 30 seconds of per-hop outage per year. Short-haul systems up to about 10 hops are often assigned a per-hop design objective of about 99.9995% for 160 SES/yr outage. In a wireless network, this would apply to the BSC-MSC and MSC-PSTN connections. Spur legs or short systems with 1 to 5 hops may be assigned a relaxed 99.999% perhop path-reliability objective equating to 320 SES (5.3 minutes) outage per year. In wireless networks, this would apply to hub or other important sites. For other services, even dramatically lower path reliabilities may be acceptable, perhaps approaching 99.99% or about 1 hour of outage per year. This would apply to the end cell sites in the wireless network. It is important to note that all the path reliability formulas and models in this text represent short-term one-way outage. To calculate two-way multipath outage time (rarely used to characterize link performance), double the calculated multipath outage. Unavailability outage times due to rain fading are not doubled, because they occur simultaneously in both directions of transmission and are always two way. It is important also to understand that per-hop performance and availability objectives are just that, objectives, not per-hop requirements. Seasonal, terrain, and geoclimatic changes average out on longer systems, thus permitting end-to-end guaranteed compliance to a users performance requirement. But on a per-hop basis, many links will perform much better, a few somewhat worse, than performance and availability objectives computed here.

Microwave Point-to-Point System Design 6.2.6

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Effects of Rain on Microwave Propagation

The principal gaseous absorption is by oxygen and water vapor. The attenuation due to oxygen is relatively constant in the 214-GHz frequency range. Water-vapor absorption, on the other hand, is highly dependent on the frequency, as well as the density of the water vapor (absolute humidity, gm/m3) [10]. Heavy rainfall, usually in cells accompanying thunderstorm activity and weather fronts, has a great impact on microwave-path availability above 10 GHz. This long-term (515 minutes) outage time usually causes traffic disconnects, and such long-term outage is never added to short-term multipath outage (previously discussed). Rain outage increases dramatically with frequency, and then with path length. Duration fades of 10 to 15 minutes to over 50 dB have been recorded on an 18-GHz, 5 km/3 mile path in Houston, for example, and increased outage at 23 GHz can require a 2-to-1 reduction in path length compared to 18 GHz for a given availability. The predicted annual outage may not occur for years, and then accumulate over a single rainy season for a long-term average [12]. Early studies, both theoretical and experimental, resulted from the recognition of the importance of rain in designing microwave paths with an availability objective in excess of 99.9%. In recent years the emphasis has been on establishing predictive techniques for the statistical estimation of the attenuation-probability distribution for a particular path. R. K. Crane has developed a model for determining the attenuation due to rain based on several factors, including path length, frequency, and point rain rates. Coefficients given by Crane [13] and ITU-R Rec. 838 are identical. The point rain-rate tables in Crane and ITU-R [2] are, however, dissimilar only as worldwide rain regions are assigned different letters, although the rain rates are very nearly the same. We may select either Crane or ITU-R point rain letters worldwide for all regions. However, Crane rain regions are usually selected from maps for North America, and ITU rain regions are usually selected from maps for international links. The more optimistic ITU-R PN.530xx rain model typically computes lower rain attenuation and, thus, less rain outage on shorter high-frequency links. Rain attenuation at the higher microwave frequencies (>10 GHz) has been under study for more than 40 years. Much is known about the qualitative aspects, but the problems faced by the microwave transmission engineerwho makes quantitative estimates of the probability distribution of the rainfall attenuation for a given frequency band, polarization, path length, and geographic (rain distribution rate) areastill remain difficult. In order to estimate this probability distribution, instantaneous rainfall data is needed. Unfortunately, the

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available rainfall data is usually in the form of a statistical description of the amount of rain that falls at a given measurement point over various time periodsgenerally at least an hour in length. The rain-induced attenuation along a given path at a given instant in time is a function of the integrated effect of the rainfall existing at all points along the path. It is affected not only by the total amount of water in the path at that instant, but also by its distribution along the path in volume and drop size (intensity). For heavy rain rates, the instantaneous distribution of volume and drop size along the path is highly variable and is difficult to predict with the accuracy expected of the rainfall data generally available. Increasing the fade margins, shortening path lengths, and increasing antenna sizes are the most readily available tools for reducing the per-hop annual rain outage in a given area. Route diversity (ring protection) or a lower, less vulnerable frequency band (perhaps 10 GHz, in digital systems) is often considered to reduce or essentially eliminate the impact of rain outage on system availability. The total annual rainfall in an area has little relation to the rain attenuation for the area. Within the United States, for example, the northwestern states have the greatest annual rainfall (in excess of 100 in/2,500 mm per year) produced, however, by long periods of steady rain of relatively low intensity at any given time. Other areas of the country with lower annual rates experience thunderstorms and frontal squalls, which produce short-duration rain rates of extreme intensity. It is the incidence of rainstorms of this type that determines the rain rates for an area and, thus, the high-frequency microwave links long-term path outage time and unavailability. Even the rain statistics for a day or an hour have little relationship to rain attenuation. A day with only a fraction of an inch or centimeter of total rainfall may have a path outage due to a short period of concentrated, extremely high-intensity rain. Another day with several inches or centimeters of total rainfall may experience little or no path attenuation because the rain is spread over a long time period or area. The most common reason for a preference for a lower frequency is the susceptibility of bands above 10 GHz to rainfall attenuation. Although the effect is present to some degree at lower frequencies, it increases rapidly with frequency. For example, a rain-cell intensity causing only a few decibels of attenuation at lower frequencies could be sufficient to cause a path outage at 18 GHz. Although fades caused by rain cells are occasionally observed at lower frequencies (1020-dB fades at 6 GHz have been recorded even in North America), this type of fade generally causes outages only on paths above


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10 GHz. The outages are usually caused by blockage of the path by the passage of rain cells (e.g., thunderstorms), perhaps 4 to 8 km, or 2.5 to 5 mi, in diameter and 5 to 15 minutes in duration on the path. Such fading exhibits fairly slow, erratic level changes, with rapid path failure as the rain cell intercepts the path. The fades are nonselective in that all main and diversity paths in both directions are affected simultaneously. Vertical polarization is far less susceptible to rainfall attenuation than horizontal polarized frequencies. Increased fade margin is of some help in rainfall attenuation fading; margins as high as 45 to 60 dB, some with automatic transmitter power control (ATPC), have been used in some highly vulnerable links for increased availability. When permitted, seldom used crossband diversity is totally effectivethe lower-frequency path is stable (affected only by multipath fading) during periods when the upper-frequency path is obstructed by rain cells. Route diversity (ring-protected paths separated by more than about 8 km, or 5 mi) is also used successfully. In summary, these are the things to bear in mind in connection with rain attenuation fades: • Multipath fading is at its minimum during periods of heavy rainfall

with well-aligned dishes, so the entire path fade margin is available to combat the rain attenuation (wet-radome loss effects are minimized with shrouded antennas). • Neither SD nor in-band FD, provides improvement against rain attenuation fade outage. • Vertically polarized high-frequency-link rain outage is 40% to 60%

less than those links horizontally polarized. • Use of mircowave systems below 10 GHz will eliminate problems caused by rain.

6.3 Practical Aspects of Microwave-Link Design 6.3.1

Design Overview

The first step in the microwave system design for the wireless network involves determining the actual system requirements and site locations, which should typically be defined by the RF group in coordination with the transmission planning group, and with the help of the real estate group as well as path surveyors. Shown here is a general list of the areas of information that must be gone over with the customer during the proposal and initial

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planning stage; more detailed questions will have to be answered during the design and deployment stage. • Information on the existing PSTN facilities in the areaswitching and

transmission, ILECs, CLECs, and so on. If possible, a map should be created showing all of the below that exist in the area. • Microwave backhaul questionsfrequency bands, regulatory issues, preferred suppliers, existing systems in the area, and so on. • Fiber-optic backhaul questionsexisting fiber-optic facilities, leasing of dark fiber, types of cables and equipment, and so on. • Operator preferences regarding owned versus leased facilities, equipment supplier, engineering and installation services, points of interconnect (POI) questions, and so on. • Miscellaneous (optional) questionsorganizational and logistical information. Based on the information provided in answering these questions, transmission engineers will work with microwave engineers to determine the best approach and strategy in formulating an overall transmission plan. The transmission plan could be completely microwave or it could be a combination of microwave systems and leased lines. Routing design and preliminary path analysis should be done prior to any field trips. Computer programs are available to plot an initial system map, which shows the sites in geographical relationship to each other. Approximate site coordinates (avoid GPS unless the measurement is done in differential mode) and the feasibility of various paths are determined from topographical maps, digital terrain databases (minimum 1:50,000), or old survey information. This map is used together with traffic requirements to define the paths between sites and the type of radio for each hop. For very short microwave hops (sites very close to each other, less than 3 km), a detailed path survey will not be required. Construction managers could be trained to perform this quick analysis and, using binoculars, see the remote site of the MW hop. This is generally sufficient to confirm LOS. Any written observations, comments, and photographs will later be useful to the microwave designer. Field verification of the LOS (path survey) between proposed wireless sites should be done before any further negotiations with the potential landlord take place. Interference analysis and frequency coordination play an important part in proposed route design. Governments usually require users of the radio spectrum to frequency-coordinate their planned and existing microwave radio


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systems with other users of the radio frequency spectrum. Such coordination is a prerequisite in any microwave radio license application by a microwave radio system operator. The radio license applicant must determine if the planned radio system will experience any interference from the existing environment and vice versa. Potential interference can be calculated for three different cases: interference between microwave point-to-point radio stations, between microwave radio relay stations and earth stations [14], and between radio relay stations and geostationary satellite orbit. The results of these calculations will indicate whether or not there is potential interference and whether redesign or relocation of the planned MW system is required. The microwave radio site survey report (checklist) is a document that will include site details, address, directions to the site, access, restrictions, general information on the site, tower or rooftop leasehold, available space and equipment at the site, dc and ac power availability, and zoning restrictions [13]. The document is usually issued and signed by the construction manager or project manager. At this point, if all the results of the previous analysis are positive and the sites have been accepted by all groups, a detailed microwave path design (based on required availability and reliability of the system) and equipment selection can be completed. The typical output of the MW system engineering design is as follows: • MW path calculations; • System layout; • T1/E1 plan; • System block and level diagrams; • Rack profiles; • Tower loading specifications and antenna placement. 6.3.2

Protected and Nonprotected Microwave Systems

The terms protection and diversity are often used interchangeably when applied to microwave links. This is not correct, since protection commonly improves long-term traffic interruptions [10 consecutive severely errored seconds (CSES) or more], while diversity arrangements greatly reduce the number and duration of short-term outages (less than 10 CSES). The radio path has two transmitters and two receivers always on-line (hot). There is a switch that keeps one radio transmitting or receiving until a failure occurs,

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and at that moment, the signal is switched to the standby radio. The MSHB configuration (1+1) protects against equipment failures only, not path propagation problems. The hot-standby-protection system uses a common frequency channel, with only one of the two transmitters at any end of the microwave link active at any particular time. The transmitter in standby mode will be fully operational except that the transmitter will be muted. The hot-standby (protected) system configuration provides hitless receiver changeover on each side of the radio relay link in case of receiver equipment failure or sudden propagation path fading on one of the four microwave paths. If a transmitter fails, there will be a short break in transmission until the standby transmitter is activated. In contrast to receiver changeover, transmitter changeover will therefore not be hitless. 6.3.3

Microwave Repeaters

Sometimes it is impossible to connect to Point B from Point A in one microwave hop, although from a propagation or quality perspective it is feasible. There could be an obstacle (a tree, building) between points A and B obstructing LOS. If there is no other solution and it is imperative to connect these two points (cell sites, cell site to the switch office), a microwave repeater must be used [15]. An active MW repeater site contains two complete microwave radio terminals, together with the antennas, waveguides, or coax cables, and the like, connected back to back. It is a much more costly solution than the passive repeater. It requires enclosure for the equipment, a power plant, an antennamounting structure of some kind, and so on. In other words, it requires the complete microwave site without the benefits of a revenue-generating cell site providing the RF coverage to that area. This is a very expensive solution to be avoided if possible. The best way to avoid use of microwave repeater sites is to plan and execute the RF coverage carefully and to strategically place cell sites in such a way that they all have an LOS with at least one other cell site. This requires careful coordination between RF and transmission or microwave planners from the very beginning of the project. Where a direct microwave path cannot be established (no LOS) between two points, it is sometimes possible to establish a path by using a passive MW repeater. The function of such a repeater is to redirect the microwave beam in order to pass the beam around or over the obstacle (buildings, hills, etc.). The main requirement is that there is a LOS between the passive repeater and both sides of the microwave link. There are two types of passive repeaters in use: One requires two parabolic antennas connected back to


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back through a short piece of transmission line; the other, more commonly used, is a flat billboard-type metal reflector, which acts as a microwave mirror. The first type is rarely used due to its inefficiency. The effectiveness of a passive repeater is an inverse function of the product of the lengths of the two paths, rather than the sum of their lengths, as one might suppose [15]. Thus, it is highly desirable to keep one of the paths very short (a few hundred meters if possible). Billboard passives are used for frequencies of 6 GHz and above and fall into two basic configurations, depending on the geometric relationship. If the site of the passive repeater is off to one side, or behind the terminal, so that the angle between two paths at the reflector is less than 130° (the smaller the angle the better), a single billboard can be used. This is the most common application. However, if the only available location happens to be more or less in line with the path, a double billboard may be required, consisting of two reflectors usually fairly close together and geometrically arranged to reflect the beam at the proper angles. For single billboards, things are simple if the billboard is in the far field of both antennas. In that case, the antenna gains and the billboard gains are independent and do not interact with each other. Then we simply calculate a total path loss, which is the sum of the two antenna gains and the two-way gain of the reflector, in order to arrive at the end-to-end path loss through the reflector. Net path loss = A 1 + A 2 − G1 − G 2 − G3 where A1 = Short-leg attenuation (dB) A2 = Long-leg attenuation (dB) G1, G2 = Antenna gains (dB) G3 = Free-space, two-way gain of a single passive billboard (dB) Passive-repeater-gain calculations could be complicated in cases where the billboard is in the near field of one of the parabolic antennas. A rule-ofthumb formula to determine the near field boundary is D = 2 × f × B2 where D = Near-field zone distance (ft)

224


F = Frequency (GHz) B = Antenna diameter (ft) In that case, Net path loss = A 1 + A 2 − G1 − G 2 − (G3 − K ) where K = Correction factor (dB) Correction factor K can be calculated using empirical formulas and graphs, and its value will depend on the parabolic dish diameter, frequency, and distance between the parabolic antenna and the passive repeater. Its value is usually between 0.2 and 1.6 dB. 6.3.4

Microwave Path Calculations

Microwave path calculations are performed as part of a detailed microwave system design, and all the detailed hardware requirements are defined based on this information. There are a number of activities involved in preparation for path engineering: • Perform site and path surveys and become familiar with the market

and the terrain;

• Consider linear or ring architecture and MW repeater facilities; • Define availability, outage, and quality standards for the individual

links and the overall system;

• Specify capacity requirements and available frequency-band choices

(higher-frequency bands for shorter distances and lower-frequency bands for longer distances);

• Determine (equipment or route) protection requirements; • Determine propagation diversity arrangements; • Use a microwave design tool (e.g., Ericssons LinkPlanner or

vendor-independant PathLoss 4.0) for detailed path engineering and interference analysis;

• Generate a bill of material (BOM) for the microwave system.


225

Transmission-network requirements for reliability of the wireless networks in North America are shown here (equipment not included). This is a basic rule of thumb and a fairly simplified practical approach, but can serve as a guideline. • For high-reliability links (usually in long-haul systems with many

hops in tandem or backbone), the per-hop objective may approach or exceed 99.9999%, allowing only 20 to 30 seconds of per-hop outage per year. • Short-haul systems of up to about 10 hops are often assigned a perhop design objective of about 99.9995% for 160 SES per year outage; this objective should be used to design BSC-PSTN MW links. • Spur legs or short systems with 1 to 5 hops in tandem assigned a relaxed 99.999% per-hop path-reliability objective equating to 320 SES (5.3 minutes) outage per year; the same criteria should be used for the links connecting important hub sites. • For single MW hops (or maybe up to 2 tandem hops), the required objective should be 99.995% per hop; this applies to RBS-BSC and RBS-RBS microwave hops. Sometimes these numbers are different and could be proposed by the customer; it is important to keep in mind that by definition, higher objective numbers lead to more expensive MW networks. Further information about the microwave system theory and practical microwave design can be found in [1618]. 6.3.5

Microwave Interference Analysis and Frequency Coordination

Radio frequency coordination is the term given to procedures followed by users of a common band of radio frequencies to minimize and control potential interference between systems. The key aspect of the procedure involves cooperative (or at least informed) radio-frequency planning. Radio systems should be designed in such a manner that they do not cause or suffer objectionable interference with other existing or planned systems using the same frequency band. This coordination is facilitated by sharing coordination data among users, so that accurate and up-to-date information is available with which estimates of potential interference can be made during the system design stage. Radio-frequency-interference studies and frequency coordination are necessary not only when designing a new system, but also when determining

226


the potential interference effects of other users radio construction proposals on existing and planned systems. Thus, coordination is involved when one party initiates construction plans as well as when reacting to other parties plans. The frequency spectrum is a valuable resource and is generally subject to appropriate planning and management to prevent misuse and interference between the many and varied applications. National administrations will allocate some or all of these bands for fixed microwave radio use in line with local requirements. Before network planning commences, an operator must determine available frequency bands and channel plans specific to that country. Often, and preferably, an operator, in this case a wireless operator, is able to obtain a number of frequency allocations as a block, enabling him to perform his own network planning in advance without risk of interference from other users. Most regulatory authorities also operate a local-link-length policy, where the length of a particular path will determine what frequency bands are available for the operator to choose from. Typically, the shorter the path, the higher the frequency required. The first step is to perform intrasystem frequency coordination (within the given network); then, if results are satisfactory, perform intersystem frequency coordination. As stated earlier, the radio license applicant must determine if the planned radio system will experience any interference from the existing environment and vice versa. Potential interference can be calculated for three different cases: interference between microwave stations, between microwave stations and earth stations, and between microwave stations and geostationary satellite orbit. The results of these calculations will indicate whether or not there is potential interference and whether redesign or relocation of the planned MW system is required. In many cases, the most reliable information on the potential interference cannot be obtained by calculations, since there is little or no information on the existing terrestrial or satellite systems in the area. The best way to determine potential interference is to sweep the entire spectrum using test equipment at the future microwave system antenna location. 6.3.6

Microwave System Design Guidelines

The transmission network must be designed to meet service demands, but always with the most economical routing in mind. Two scenarios are most common in wireless-network deploymentleased facilities or the microwave network. For larger networks, it is usually some combination of both, and even the leased-lines (facilities) network requires careful transmission-


227

network planning. Survivability and reliability of the network are achieved by means of transmission loopsthe ring configuration or a combination of star and ring configurations. For a larger wireless network (and transmission network), the ring configuration is recommended as a high-capacity backbone carrying traffic to the switch location (Figure 6.3). Not all the cell sites are shown in this network diagram. As a backbone microwave system, 7-GHz, SDH microwave radios with the STM-1 (63 E1s) capacity are used. Cross-polarization can be used to further increase (double) the capacity of the ring. Higher-capacity microwave radios are used to bring traffic from the cell sites and hub sites into the ring and to the BSC. The following criteria can be used for determining whether to consider using a ring configuration: • More than 20 to 30 cell sites in the network; • More than 40 to 50 T1/E1s required; • More than 4 to 5 MW links required at the switch location;

Cell site

MW ring hub site/cell site

7 GHz/63 E1 BSC/MSC

13 GHz/16 E1

13 GHz/8 E1 PDH microwave link (13 GHz) SDH microwave link (7 GHz) SDH microwave link (XPIC)

Figure 6.3 Microwave system design.

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• High network reliability required; • Customer requirement.

The ring configuration provides a reliable communication facility, since it offers automatic protection from site hardware (batteries, towers, antenna systems) failures, radio or MUX equipment failures, propagation failures in the microwave network, and cable cuts in the fiber-optic network. Since additional sites for microwave use only (MW repeater sites) are very expensive, they should be avoided if possible, so it is usual to assume one microwave hop per cell site. These are some of the basic steps for the microwave design as part of the overall wireless-network design: • Use an RF network design tool or existing terrain database to check • • • •

• • • • •

for the basic LOS; If possible, perform preliminary site and path surveys and become familiar with the market and the terrain; Consider linear, star, and ring architecture and MW repeater facilities; Define availability, outage, and quality standards; Specify capacity requirements and available frequency band choices (higher-frequency bands for shorter distances and lower-frequency bands for longer distances); Determine equipment (infrastructure, route) protection requirements; Determine propagation diversity arrangements; Choose the MW equipment vendor; Choose the MW installation services contractor (radio and antenna installation could be, and often is, provided by different contractors); Prepare the ATP.

In microwave systems, additional protection (e.g., space or frequency diversity) at lower frequencies may be required against short-term multipath outages. Diversity systems as a means of increasing reliability of the MW system should be used only for very high-capacity systems or systems of great strategic importance. All the sites that belong to the ring are considered hub sites and must be completed and built first to provide protection.


229

Microwave Lookup Table

To determine roughly the number of MW hops required for the wireless network in a specific market, Table 6.2 can be used. Calculations are done using the PathLoss 4.0 microwave software tool [18]. Rain regions of the world are described using a range of letters, with A, B, and C referring to the areas with very little or no rain, and M, N, and P referring to the rain forest, which experiences heavy downpours. The statistical variations of rainfall intensity, specific attenuation, and attenuation along the path depend in a very complex way on the number, type, and intensity of rainstorms that traverse the

Table 6.2 Microwave Lookup Table Availability

Frequency

99.995%

99.999%

ITU Rain Region

A

B

C

D

E

F

G

H

J

K

L

M

N

P

2

43

43

43

43

43

43

43

43

43

43

43

43

43

43

6

28

28

28

28

28

28

28

28

28

28

28

28

28

28

7

27

27

27

27

27

27

27

27

27

27

27

27

27

27

8

26

26

26

26

26

26

26

26

26

26

26

26

26

26

10

23

23

23

23

23

23

23

23

23

23

22

21

20

15

11

23

23

23

23

23

23

23

23

23

23

22

21

20

15

13

22

22

22

22

22

22

22

22

21

21

19

18

17

17

15

22

22

22

22

22

21

21

21

20

19

17

16

14

13

18

14

14

14

14

14

13

13

13

12

11

9

9

7

6

23

12

12

12

12

11

10

10

9

9

8

6

6

4

3

38

11

9

7

7

6

5

5

4

4

3

3

3

2

1.5

8

29

29

29

29

29

29

29

29

29

29

29

29

29

29

6

19

19

19

19

19

19

19

19

19

19

19

19

19

19

7

18

18

18

18

18

18

18

18

18

18

18

18

18

18

8

17

17

17

17

17

17

17

17

17

17

17

17

17

17

10

16

16

16

16

16

16

16

16

16

16

15

14

13

9

11

16

16

16

16

16

16

16

16

16

16

15

14

13

9

13

15

15

15

15

15

15

15

15

14

14

12

11

9

8

15

15

15

15

15

15

14

14

13

13

12

9

7

6

5

18

10

10

10

10

10

9

9

8

8

7

5

5

4

3

23

9

9

9

8

8

6

6

6

5

5

3

3

2

2

38

9

6

5

4

3

3

3

3

2.5

2.5

2.2

2.2

1.5

1

Average temperature 10°C 211 GHz: 6-ft. dish 1315 GHz: 4-ft. dish 18 GHz and above: 2-ft. dish

Vigants/Crane model used (more pessimistic) Vertical polarization assumed Standard Tx power Average terrain

230


path each year. No theoretical basis exists for the calculation of the desired rainfall statistics, and they must be obtained empirically. Cranes rain-fading model [6] recommends using a rain-rate distribution tailored to each climatic area. The rain heavily affects frequencies over 10 GHz, while frequencies below 10 GHz are only affected by multipath. Frequencies in the 10-GHz neighborhood experience the effects of both. For example, if the distance between cell sites in the South American market (rain region M or N) is 5 km and we are using a 23-GHz microwave system, there is a chance that we might need MW repeaters between the sites. Since that would be an expensive solution, another option is to use a 15-GHz or even lower frequency, but we may be limited by local regulations or the customers license for microwave spectrum. It is obvious from the table that frequencies around 10 GHz are affected only by heavy rain, while even the smallest amount of rain could affect 38-GHz microwave systems. Cranes model is used mainly in North America, while the ITU method is used in the rest of the world. These two methods for calculating rain fading result in slightly different values. Cell-site backhaul is usually designed to satisfy 99.995% reliability (path propagation only; this does not include equipment), with important sites and higher-capacity hub sites designed for 99.999% reliability. There are a number of excellent microwave design tools on the market. Some microwave equipment manufacturers insist on using their own software tools while some operators and consultants prefer tools available on the open market. One of those vendor-independent tools is PathLoss 4.0. This is probably one of the best tools for complex microwave design including North American and ITU standards, different diversity schemes, diffraction and reflection (multipath) analysis, rain effects, and so on. PathLoss 4.0 is widely accepted by microwave system design engineers around the world. Figure 6.4 shows an example of a PathLoss 4.0 microwave path design worksheet.

6.4 Spread-Spectrum Microwave Systems Spread spectrum is a digital-coding technique originally developed for military purposes; coding protects the data by increasing the transmitted bandwidth and reducing the power density [16]. This technique resists jamming, interference, selective fading, and interception, as well as minimizing interference to and from other users. Spread-spectrum systems can coexist with other radio


231

Microwave Worksheet027c049d.pl4 Elevation (m) Latitude Longitude True azimuth (°) Antenna model Antenna height (m) Antenna gain (dBi) Vertical angle (°) Connector loss (dB) Frequency (MHz) Polarization Path length (km) Free space loss (dB) Atmospheric absorption loss (dB) Net path loss (dB) Radio model TX power (watts) TX power (dBm) EIRP (dBm) Emission designator RX threshold criteria RX threshold level (dBm) Maximum receive signal (dBm) RX signal (dBm) Thermal fade margin (dB) Geoclimatic factor Path inclination (mr) Average annual temperature Worst month multipath outage (%) (sec) Annual multipath outage (%) (sec) (%-sec) Rain region Rain rate (mm/hr) Rain attenuation (dB) Annual rain outage (%-sec) Annual multipath + rain (%-sec)

027C 1536.67 20 40 52.00 N 103 18 07.60 W 84.17 VHP2-142 24.00 36.50 −0.32 0.00 14,825.00 Vertical 5.67 130.95 0.15 58.11 9415-UX (2E-1) 0.16 22.00 58.50 3.5MOD7W BER = 10-6 −89.00 −20.00 −36.11 52.89 4.45E-05 5.26 21.00 100.00000 2.61E-05 100.00000 1.09E-04 100.00000-0.00 D2-96 Temp. Continent 234.09 52.89 99.99996-12.55 99.99996-12.55

049D 1513.87 20 41 10.70 N 103 14 52.80 W 264.19 VHP2-142 17.00 36.50 0.28 0.00

58.11 9415-UX (2E-1) 0.16 22.00 58.50 3.5MOD7W BER = 10-6 -89.00 -20.00 -36.11 52.89

100.00000 2.61E-05 100.00000 1.09E-04

Figure 6.4 PathLoss 4.0 microwave worksheet.

systems, without being disturbed by their presence and without disturbing their activity. The immediate effect of this elegant behavior is that spread-

232


spectrum systems may be operated without a license, making spread-spectrum modulation the chosen technology for rapid-deployment systems. There are two basic types of spread-spectrum systemsdirect-sequence spread spectrum (DSSS) and frequency-hopping spread spectrum (FHSS). Fundamentally, DSSS is accomplished by chopping up the signal into data packets using a predefined sequence called the spreading code. Only the coded data, not the signal, is transmitted and only receivers with the matching encoding scheme can reconstruct the signal. A system with a particular encoding scheme will see any foreign encoding pattern or frequency as background noise. With FHSS, hopping the signal randomly to a different frequency several times per second spreads the signal. Both DSSS and FHSS use a pseudorandom sequence. While for DSSS the signal is mixed with the pseudorandom sequence, for FHSS the signal changes frequencies based on the pseudorandom sequence. The bandwidth of the transmitted signal is much greater than the bandwidth of the original message, and the bandwidth of the transmitted signal is determined by the message to be transmitted and by an additional signal known as the spreading code. Low power density relates to the fact that the transmitted energy is spread over a wide band, and therefore the amount of energy per specific frequency is very low. The effect of the low power density of the transmitted signal is that such a signal will not disturb (interfere with) the activity of other systems receivers in the same area. Redundancy relates to the fact that the message is (or may be) present on different frequencies from where it may be recovered in case of errors. The effect of redundancy is that spread-spectrum systems present a high resistance to noises and interference, being able to recover their messages even if noises are present on the medium. Spread-spectrum modulation techniques are composed of two consecutive modulation processes: one executed by the message to be transmitted and the other executed by the spreading code (spreading process). It is this spreading process that generates the wide bandwidth of the transmitted signal. In FHSS systems, the two modulation processes are as follows: • Process 1. The original message modulates the carrier, thus generat-

ing a narrowband signal. • Process 2. The frequency of the carrier is periodically modified (hopped) following a specific spreading code. In FHSS systems, the


233

spreading code is a list of frequencies to be used for the carrier signal. The amount of time spent on each hop is known as dwell time. In DSSS systems, the two modulation processes are as follows: • Process 1. The original message is modulated by the spreading code.

In DSSS systems, the spreading code is a sequence of bits (known as chips), and the first modulation step is an exclusive-OR (XOR) operation executed between the message and the spreading code (in a process known as chipping). The result of the first modulation step is that a 0 bit of message is converted into a chip sequence representing the 0 bit, and the 1 bit of message is converted into another chip sequence, representing the 1 bit. Instead of transmitting the original message bit, a chip sequence representing the bit will be transmitted. • Process 2. The sequences representing the message bits modulate the carrier signal. Dwell time in FHSS is represented as a 3x data bit duration. Spreading sequence in DSSS is represented as being five chips long. In DSSS systems, colocation could be based on the use of different spreading codes (sequences) for each active system. For FHSS systems, IEEE 802.11 defines 79 different hops for the carrier frequency. Using these 79 frequencies, IEEE 802.11 defines 78 hopping sequences (each with 79 hops) grouped in three sets of 26 sequences each. Sequences from the same set encounter minimum collisions and they may be allocated to colocated systems. Theoretically, 26 FHSS systems may be colocated. However, as synchronization among independent systems is forbidden (synchronization would eliminate collisions), the actual number of systems that can be colocated is around 15. Figure 6.5 shows an example of the spread-spectrum microwave system application in the wireless network where the hub site is connected to the BSC/MSC site, as well as to other cell sites, by means of this kind of microwave point-to-point system. The greatest interest in spread-spectrum products has resulted from the creation of rules permitting spread-spectrum transmitters to operate on an uncoordinated basis without requiring individual user licenses. The 2.4 and 5.8 GHz bands are unlicensed bands (in North America) that can be used without any interference analysis or frequency coordination with the existing

234


Road coverage

Base station Urban/suburban coverage 2.4 or 5.8 GHz Hub site

Base station

MSC/end office switch

Figure 6.5 Spread-spectrum MW system.

microwave systems. Spread-spectrum MW systems should be used only as a temporary solution when rapid deployment is required or in rural areas only. The reason is that the 2.4 GHz band is also used with a number of other applications within the unlicensed ISM band, such as garage-door openers, microwave ovens, and Bluetooth systems. Special rules to take advantage of land-based spread-spectrum technology have been issued by the United States, Canada, Argentina, and the United Kingdom. The United States permits unlicensed operation with high power (1W) in three bands under Part 15.247 of the FCC Rules: 902928 MHz, 2,4002,483.5 MHz, and 5,7255,850 MHz. Argentina has followed with rules identical to current U.S. rules. Those in Canada are similar, and based on Industry Canada, Spectrum Management, RSS-210, Low Power Licence-Exempt Radiocommunication Devices. (More information about


235

Industry Canada can be found on its Web page: http://strategis.ic.gc.ca/ sc_mrksv/spectrum/engdoc/spect1.html.) Most of the regulations permitting unlicensed spread spectrum penalize the use of directional antennas. Manufacturers using directional antennas are forced to reduce the transmitter output power by the gain of the antenna to comply with an effective radiated power (ERP) limit. Manufacturers using omnidirectional antennas can use the full power, usually 1W. The use of digital radio communication has increased tremendously in recent years, and the FCC and other agencies have made the ISM band available for wireless digital communication in order to satisfy the increasing demand. It is well known that microwave ovens are the most significant man-made noise source within the ISM band because of their widespread popularity. They have become a major source of interference for digital communication systems. Microwave ovens are designed for heating and cooking food through the use of electromagnetic energy at frequencies in the ISM bands, ranging from 890 to 6,000 MHz. In particular, the frequency of 2,450.50 MHz is applied to drive most ovens used for heating and cooking food in the household or commercial establishment. A primary source for oven emission comes from the leakage of microwave radiation from the door. Certainly, designs and research studies should be initiated to mitigate the effect of this man-made noise and to enhance the performance of wireless communication systems under this man-made-noise environment. Nonionizing radiation is the energy associated with electromagnetic radiation and may be classified in terms of the quantum energy required to eject or promote electrons from biological materials exposed to electromagnetic radiation. Microwave radiation has low quantum energy; under ordinary circumstances, it is too low to affect ionization, excitation, ejection, or promotion of electrons. Consequently, microwave radiation is referred to as nonionizing radiation. A new and more realistic performance standard serves to better control the emission of nonionizing radiation from microwave ovens. It will minimize human exposure and will add to the protection of the publics health and safety. Concurrently, it will provide relief to the spectral congestion in a frequency band allocated to digital radio communication. It also will help to improve the performance of many digital radio communication systems using the ISM band. From the regulatory perspective, these unlicensed bands come with two major constraints, a transmit-power limitation of essentially 1W and a minimum processing gain of 10 dB for either a frequency-hopping or directsequence system. This implies that the desired data capacity per bandwidth (in other words, bandwidth efficiency) may have to be sacrificed to achieve

236


the processing gain, and the total transmit power is not high enough to support multilevel quadrature-amplitude-modulation (QAM) techniques to increase the data rate. The challenge is to overcome these difficulties and still achieve sufficiently high data capacity.

6.5 Microwave Compatibility and Safety The Radiation Control for Health and Safety Act (PL90-602) was adopted by the U.S. Congress in October 1968, to protect the public from unnecessary exposure to potentially harmful radiation, which includes microwaves emitted by electronic products. The act prescribes different and individual performance standards, to the extent appropriate and feasible, for different electronic products so as to recognize their different operating characteristics and uses. The ANSI guidelines have been updated and refined several times since 2 the first frequency-independent limit of 10 mW/cm was issued. In 1986, a report from the National Council for Radiation Protection and Measurements (NCRP) provided a comprehensive evaluation of the scientific literature on the biological effects of radio-frequency electromagnetic fields (RFEMs). On the basis of this evaluation, the report made recommendations for exposure criteria (Biological Effects and Exposure Criteria for RadioFrequency Electromagnetic Fields, NCRP Report No. 86, National Council on Radiation Protection and Measurement, Bethesda, Maryland, 1986). Several key aspects of the NCRP recommendations were incorporated subsequently into the ANSI/IEEE C95.1-1992 guidelines. These guidelines and recommendations formed the basis for mandatory standards adopted by several federal agencies, including the FCC. They are also used extensively by standards-development organizations throughout the world. The FCCs guidelines establish separate maximum-permissibleexposure (MPE) limits for general population/uncontrolled exposure and for occupational/controlled exposure. The general population/uncontrolled limits set the maximum exposure to which most people may be subjected. People in this group include members of the general public not associated with the installation and maintenance of the transmitting equipment. Higher exposure limits are permitted under the occupational/controlled exposure category, but only for persons who are exposed as a consequence of their employment (e.g., wireless radio engineers or technicians). To qualify for the occupational/controlled exposure category, exposed persons must be made fully aware of the potential for exposure (e.g., through training), and they


237

must be able to exercise control over their exposure. In addition, people passing through a location, who are made aware of the potential for exposure, may be exposed under the occupational/controlled criteria. The MPE limits adopted by the FCC for occupational/controlled and general population/uncontrolled exposure incorporate a substantial margin of safety and have been established to be well below levels generally accepted as having the potential to cause adverse health effects. Determining whether a potential health hazard could exist with respect to a given transmitting antenna is not always a simple matter. Important questions must be considered in making that determination the following: • What is the frequency of the RF signal being transmitted? • What is the operating power of the transmitting station and what is

the actual power radiated from the antenna?

• How long will someone be exposed to the RF signal at a given dis-

tance from the antenna?

• What other antennas are located in the area, and what is the expo-

sure from those?

For all frequency ranges at which FCC licensees operate, the FCCs rules establish MPE limits. The MPE limits vary by frequency because of the different absorptive properties of the human body at different frequencies when exposed to whole-body RF fields. The FCC establishes MPE limits in terms of electric field strength, magnetic field strength, and far-field equivalent power density. For most frequencies used by wireless services, the most relevant measurement is power density. The MPE limits for power density 2 are given in terms of milliwatts per square centimeter, or mW/cm . In terms of power density, for a given frequency the FCC MPE limits can be interpreted as specifying the maximum rate that energy can be transferred (i.e., the power) to a square centimeter of a persons body over a period of time. In practice, however, since it is unrealistic to measure separately the exposure of each square centimeter of the body, actual compliance with the FCC limits on RF emissions should be determined by spatially averaging a persons exposure over the projected area of an adult human body. Electric field strength and magnetic field strength are used to measure near field exposure. At frequencies below 300 MHz, these are typically the more relevant measures of exposure, and power density values are given primarily for reference purposes. However, evaluation of far-field equivalent

238


power density exposure may still be appropriate for evaluating exposure in some cases. For frequencies above 300 MHz, only one field component need be evaluated, and exposure is usually more easily characterized in terms of power density. Transmitters and antennas that operate at 300 MHz or lower include radio broadcast stations, some television broadcast stations, and certain personal wireless service facilities (e.g., some paging stations). Most personal wireless services, including all cellular and PCS, as well as some television broadcast stations, operate at frequencies above 300 MHz. As noted above, the MPE limits are specified as time-averaged exposure limits. This means that exposure can be averaged over the identified time interval (30 minutes for general population or uncontrolled exposures, or 6 minutes for occupational or controlled exposures). However, for the case of exposure of the general public, time averaging is usually not applied because of uncertainties over exact exposure conditions and difficulty in controlling time of exposure. Therefore, the typical conservative approach is to assume that any RF exposure to the general public will be continuous. The FCCs limits for exposure at different frequencies are shown in Table 6.3. Finally, it is important to understand that the FCCs limits apply cumulatively to all sources of RF emissions affecting a given area. A common example is where two or more wireless operators have agreed to share the cost of building and maintaining a tower and to place their antennas on that joint structure. In such a case, the total exposure from the two facilities taken together must be within the FCC guidelines. It has been determined through calculations and technical analysis that due to their low power or height above ground level, many facilities by their very nature are highly unlikely to cause human exposures in excess of the guideline limits. Operators of those facilities are exempt from routinely having to determine compliance, and facilities with these characteristics are considered categorically excluded from the requirement for routine environmental processing for RF exposure. If a facility is categorically excluded, an applicant or licensee may ordinarily assume compliance with the guideline limits for exposure. However, an applicant or licensee must evaluate and determine compliance for a facility that is otherwise categorically excluded if specifically requested to do so by the FCC. No radio or television broadcast facilities are categorically excluded. Thus, broadcast applicants and licensees must affirmatively determine their facilitys compliance with the guidelines before construction, and upon every facility modification or license renewal application. With respect to personal wireless services, a cellular facility is categorically excluded if the total ERP of all channels operated by the licensee at a


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Table 6.3 Limits for the Maxium Permissible Exposure (MPE) (A) Limits for Occupational/Controlled Exposure

Frequency Range (MHz)

Electric Field Strength (E) (V/m)

Magnetic Field Strength (H) (A/m)

Power Density (S) (mW/cm2)

Averaging Time |E|2, |H|2 or S (minutes)

0.33.0

614

1.63

(100)*

6

3.030

1,842/f

4.89/4

(900/f2)*

6

30300

61.4

0.163

1.0

6

3001,500

f/300

6

1,500100,000

5

6

(B) Limits for General Population/Uncontrolled Exposure

Frequency Range (MHz)

Electric Field Strength (E) (V/m)

Magnetic Field Strength (H) (A/m)

Power Density (S) (mW/cm2)

Averaging Time |E|2, |H|2 or S (minutes)

0.31.34

614

1.63

(100)*

30

2

1.3430

824/f

2.19/f

(180/f )*

30

30300

27.5

0.073

0.2

30

3001,500

f/1500

30

1,500100,000

1.0

30

f = frequency in MHz. * = Plane-wave equivalent power density. Note 1: Occupational/control limits apply in situations in which persons are exposed as a consequence of their employment provided those persons are fully aware of the potential for exposure and can exercise control over their exposure. Limits for occupational/controlled exposure also apply in situations when an individual is transient through a location where occupational/controlled limits apply provided he or she is made aware of the potential for exposure. Note 2: General population/uncontrolled exposures apply in situations in which the general public may be exposed, or in which persons that are exposed as a consequence of their employment may not be fully aware of the potential for exposure or cannot exercise control over their exposure.

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site is 1,000W or less. If the facility uses sectorized antennas, only the total effective radiated power in each direction is considered. In addition, a cellular facility is categorically excluded, regardless of its power, if it is not mounted on a building and the lowest point of the antenna is at least 10m (about 33 ft) above ground level. A broadband PCS antenna array is categorically excluded if the total ERP of all channels operated by the licensee at a site (or all channels in any one direction, in the case of sectorized antennas) is 2,000W or less. Like cellular, another way for a broadband PCS facility to be categorically excluded is if it is not mounted on a building and the lowest point of the antenna is at least 10m (about 33 ft) above ground level. The power threshold for categorical exclusion is higher for broadband PCS than for cellular, because broadband PCS operates at a higher frequency where exposure limits are less restrictive. At present, the relevant safety guideline for human exposure in the 2 United States has been reduced to f /1,500 mW/cm , where f is expressed in MHz. For example, in case of 2,450 MHz the safety guideline translates to 1.67 mW/cm2. In point-to-point microwave radio systems, the only danger for the technicians is during the installation and testing of live systems, and only in cases where they spend extended time directly in front of the microwave parabolic antenna. Due to the fact that the microwave antennas are installed high above ground level and are highly directive, there is a very low (practically insignificant) level of the microwave radiation on the ground affecting any living creatures. In other words, point-topoint microwave radios use low-power transmitters and narrow, directed antenna beams (parabolic dish antennas) to reduce radiation levels of many orders of magnitude to well within safety limits. Therefore, MW systems are not dangerous to the public and do not present a safety or a health hazard.

6.6 Coordinate Systems, Datums, and GPS 6.6.1

About Datums

Geodetic datum (singular of the Latin word data, meaning given things) defines the reference systems that describe the size and shape of the Earth. Hundreds of different datums have been used to frame position descriptions since Aristotle made the first estimates of Earths size, and datums have evolved from those describing a spherical body to ellipsoidal models derived from years of satellite measurements. Geodetic datums and the coordinate reference systems based on them were developed to describe geographic


241

positions for surveying, mapping, and navigation. Through a long history, the figure of the Earth was refined from flat models to spherical models of sufficient accuracy to allow for global exploration, navigation, and mapping. True geodetic datums were employed only after the late 1700s, when measurements showed that the Earth was ellipsoidal in shape. Modern geodetic datums range from flat Earth models used for plane surveying to complex models used for international applications, which completely describe the size, shape, orientation, gravity field, and angular velocity of the earth. Cartography, surveying, navigation, and astronomy all make use of geodetic datums. Referencing geodetic coordinates to the wrong datum can result in position errors of hundreds of meters. Different nations and agencies use different datums as the basis for coordinate systems used to identify positions in geographic information systems, precise positioning systems, and navigation systems. The diversity of datums in use today and the technological advancements that have made possible global positioning measurements with submeter accuracy requires careful datum selection and careful conversion between coordinates in different datums. Datum types include horizontal, vertical, and complete datums. The Global Positioning System (GPS) is based on the World Geodetic System 1984 (WGS-84). Coordinate values resulting from interpreting latitude, longitude, and height values based on one datum as though they were based in another datum can cause position errors in three dimensions of more than 1 km. Datum conversions are accomplished by various methods. Complete datum conversion is based on seven parameter transformations that include three translation parameters, three rotation parameters, and a scale parameter. Simple three-parameter conversion between latitude, longitude, and height in different datums can be accomplished by conversion through Earth-centered, Earth-fixed, XYZ Cartesian coordinates in one reference datum and three origin offsets that approximate differences in rotation, translation, and scale. Parameters for simple XYZ conversion between many datums and WGS-84 are published by the Defense Mapping Agency.

6.6.2

Geometric Earth Models

Early ideas of the figure of the Earth resulted in descriptions of the Earth as an oyster (the Babylonians, before 3000 B.C.), a rectangular box, a circular disk, a cylindrical column, a spherical ball, and a very round pear (Columbus, in the last years of his life).

242


Flat Earth models are still used for plane surveying, over distances short enough so that the Earth, curvature is insignificant (less than 10 km). Spherical Earth models represent the shape of the Earth with a sphere of a specified radius. Spherical Earth models are often used for short-range navigation (VOR-DME) and for global distance approximations. Spherical models fail to model the actual shape of the Earth. The slight flattening of the Earth at the poles results in about a 20-km difference at the poles between an average spherical radius and the measured polar radius of the Earth. Ellipsoidal Earth models are required for accurate range and bearing calculations over long distances. Loran-C and GPS navigation receivers use ellipsoidal Earth models to compute positions. Ellipsoidal models define an ellipsoid with an equatorial radius and a polar radius. The best of these models can represent the shape of the Earth over the smoothed, averaged seasurface to within about 100m. 6.6.3

Reference Ellipsoids and Coordinate Systems

Reference ellipsoids are usually defined by semimajor (equatorial radius) and flattening (the relationship between equatorial and polar radii). Other reference ellipsoid parameters, such as semiminor axis (polar radius) and eccentricity, can be computed from these terms. The Earth has a highly irregular and constantly changing surface. Models of the surface of the Earth are used in navigation, surveying, and mapping. Topographic and sea-level models attempt to model the physical variations of the surface, while gravity models and geoids are used to represent local variations in gravity that change the local definition of a level surface. The topographical surface of the Earth is the actual surface of the land and sea at some moment in time. Aircraft navigators have a special interest in maintaining a positive height vector above this surface. Sea level is the average (methods and temporal spans vary) surface of the oceans. Tidal forces and gravity differences from location to location cause even this smoothed surface to vary over the globe by hundreds of meters. Gravity models attempt to describe in detail the variations in the gravity field. The importance of this effort is related to the idea of leveling. Plane and geodetic surveying uses the idea of a plane perpendicular to the gravity surface of the Earth, the direction perpendicular to a plumb bob pointing toward the center of mass of the Earth. Local variations in gravity, caused by variations in the Earths core and surface materials, cause this gravity surface to be irregular.


243

Geoid models attempt to represent the surface of the entire Earth over both land and ocean as though the surface resulted from gravity alone. Bomford described this surface as the surface that would exist if the sea was admitted under the land portion of the Earth by small frictionless channels. The WGS-84 geoid defines geoid heights for the entire Earth, and the U.S. National Imagery and Mapping Agency (formerly the Defense Mapping Agency) publishes a 10° × 10° grid of geoid heights for the WGS-84 geoid. By using a four-point linear interpolation algorithm at the four closest grid points, the geoid height for any location can be determined. The same grid can be used to produce a contour map of geoid heights for the globe. The National Imagery and Mapping Agency publishes a 0.25° model of the WGS-84 geoid. Coordinate systems to specify locations on the surface of the Earth have been used for centuries. In western geodesy, the equator, the tropics of Cancer and Capricorn, and then lines of latitude and longitude were used to locate positions. Eastern cartographers used other rectangular grid systems as early as A.D. 270, and various units of length and angular distance have been used throughout history. The meter is related to both linear and angular distance, having been defined in the late eighteenth century as one ten-millionth of the distance from the pole to the equator. The most commonly used coordinate system today is the latitude, longitude, and height system. The prime meridian and the equator are the reference planes used to define latitude and longitude. The geodetic latitude (there are many other defined latitudes) of a point is the angle from the equatorial plane to the vertical direction of a line normal to the reference ellipsoid. The geodetic longitude of a point is the angle between a reference plane and a plane passing through the point, both planes being perpendicular to the equatorial plane. The geodetic height at a point is the distance from the reference ellipsoid to the point in a direction normal to the ellipsoid. Earth-centered, Earth-fixed X, Y, and Z Cartesian coordinates are also used to define three-dimensional positions with respect to the center of mass of the reference ellipsoid. The Z-axis points toward the North Pole. The X-axis is defined by the intersection of the plane defined by the prime meridian and the equatorial plane. The Y-axis completes a right-handed orthogonal system by a plane 90° east of the X-axis and its intersection with the equator. Hundreds of geodetic datums are in use around the world.

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6.6.4


GPS

6.6.4.1 About GPS

GPS is based on the information users receive from GPS satellites. The purpose of GPS is to provide its users with the ability to compute their location in three-dimensional space [19, 20]; in order to do that, the receiver must be able to lock on to signals from at least four different satellites. Moreover, the receiver must maintain a lock on each satellites signal long enough to receive the information encoded in the transmission. Achieving and maintaining a lock on four or more satellite signals can be impeded, because the signal is transmitted at 1.575 GHz, a frequency too high to bend around or pass through solid objects in the signals path. This is the reason that GPS receivers cannot be used indoors. Outdoors, tall buildings, dense foliage, or terrain that stands between a GPS receiver and a GPS satellite will block that satellites signal. It has to be noted that the inherent accuracy of the GPS is better than 100m. Also, the signals from the GPS satellites have been degraded intentionally by the U.S. Department of Defense (DOD) for the purpose of national security. This performance degradation is known as selective availability (SA), and only DOD-approved users have access to satellite signals without SA. However, a policy statement recently issued by the White House indicates that SA will be turned off before 2006, and the accuracy of GPS position fixes will improve significantly. Authorized users with cryptographic equipment and keys and specially equipped receivers use the Precise Positioning System (PPS). U.S. and Allied military, certain U.S. government agencies, and selected civil users specifically approved by the U.S. government can use the PPS. PPS Predictable Accuracy • Horizontal accuracy of 22m; • Vertical accuracy of 27.7m; • Time accuracy of 100 ns.

Civil users worldwide use the Standard Positioning Service (SPS) without charge or restrictions. Most receivers are capable of receiving and using the SPS signal. Today, the DOD still intentionally degrades the SPS accuracy by the use of selective availability.


245

SPS Predictable Accuracy • Horizontal accuracy of 100m; • Vertical accuracy of 156m; • Time accuracy of 340 ns.

These GPS accuracy figures are from the 1994 Federal RadioNavigation Plan. The figures are 95% accurate; for horizontal accuracy figures, 95% is the equivalent of 2 rms (two-distance root-mean-squared), or twice the radial-error standard deviation. For vertical and time errors, 95% is the value of two standard deviations of vertical error or time error. SA was officialy turned off as of May 2001. Typical commerical GPS positioning accuracy will be 10 to 20m depending on the GPS receiver and the configuration of the satellite constellation. 6.6.4.2 Differential GPS Techniques

The idea behind differential positioning is to correct bias errors at one location with measured bias errors at a known position. A reference receiver, or base station, computes corrections for each satellite signal. Because individual pseudoranges must be corrected prior to the formation of a navigation solution, differential GPS (DGPS) implementations require software in the reference receiver that can track all SVs in view and form individual pseudorange corrections for each SV. These corrections are passed to the remote, or rover, receiver that must be capable of applying these individual pseudorange corrections to each SV used in the navigation solution. Applying a simple position correction from the reference receiver to the remote receiver has limited effect at useful ranges. This is due to the fact that both receivers would have to be using the same set of SVs in their navigation solutions and have identical GDOP terms (not possible at different locations) to be identically affected by bias errors. Differential corrections may be used in real time or later, with postprocessing techniques. Real-time corrections can be transmitted by radio link. The U.S. Coast Guard maintains a network of differential monitors and transmits Differential Code GPS (DGPS) corrections over radio beacons covering much of the U.S. coastline for the purpose of navigation. DGPS corrections are often transmitted in a standard format specified by the Radio Technical Commission Marine (RTCM). Corrections can be recorded for postprocessing. Many public and private agencies record DGPS corrections for distribution by electronic means. Private DGPS services use leased FM

246


subcarrier broadcasts, satellite links, or private radio beacons for real-time applications. To remove SA (and other bias errors), differential corrections should be computed at the reference station and applied at the remote receiver at an update rate that is less than the correlation time of SA. Suggested DGPS update rates are usually less than 20 seconds. DGPS removes common-mode errors, those errors common to both the reference and remote receivers (not multipath or receiver noise). Errors are more often common when receivers are close together (less than 100 km). Differential position accuracies of 1 to 10m are possible with DGPS. Differential carrier GPS is used for surveys. All carrier-phase tracking is differential, requiring both a reference and remote receiver tracking carrier phases at the same time. In order to estimate the number of carrier wavelengths at the reference and remote receivers correctly, they must be close enough to ensure that the ionospheric delay difference is less than a carrier wavelength. This usually means that carrier-phase GPS measurements must be taken with a remote and reference station within about 30 km of each other. Special software is required to process carrier-phase differential measurements. Newer techniques, such as real-time-kinematic (RTK) processing, allow for centimeter relative positioning with a moving remote receiver. 6.6.4.3 GPS Techniques and Project Costs

Receiver costs vary depending on capabilities. Small civil SPS receivers can be purchased for under $200; some can accept differential corrections. Receivers that can store files for postprocessing with base station files cost more ($2,000$5,000). Receivers that can act as DGPS reference receivers (computing and providing correction data) and carrier-phase-tracking receivers (and two are often required) can cost many thousands of dollars ($5,000$40,000). Military PPS receivers may cost more and can be difficult to obtain. Other costs include the multiple receivers (when needed), postprocessing software, and specially trained personnel. Project tasks can often be categorized by required accuracy, which will determine equipment cost: • Low-cost, single-receiver SPS projects (100m accuracy); • Medium-cost, differential SPS code positioning (110m accuracy); • High-cost, single-receiver PPS projects (20m accuracy); • High-cost, differential-carrier-phase surveys (1 mm to 1 cm accuracy).


247

Useful Facts to Remember • Coordinates can be taken from topographical maps, surveys, or GPS. • In North America, coordinates are based on NAD27 or NAD83

(North American datum).

• NAD27 is based on Clarke ellipsoid 1866 and uses a stone in Meades

Ranch in Kansas as a reference point.

• The NAD83 Geodetic reference system is based on satellites. • All older North American topographical maps are based on NAD27. • NGVD29 is a National Geodetic Vertical Datum. • NAVD88 is a North American Vertical Datum. • No universal formulas exist for the conversion between NAD27 and

NAD83.

• Ordinary GPS is usually not precise enough for the MW design, fre-

quency coordination, and licensing process. • Three satellites can be used for a two-dimensional position fix, but four are required for height above terrain (HAT).

6.7 Managing the MW Radio Network 6.7.1

Introduction

Microwave radios are sometimes used in large communications networks where network management must deal with particular network configurations: • Several hundred sites to be monitored; • Several hundred subnetworks not interconnected; • Fast notification of alarms when problems occur on any site of any

subnetwork; • The ability to manage radios from other NMS providers.

The domain manager application, hereafter referred to as the domain manager, is an on-site software tool for management and support of the digital radio relay link. It is also often called craft interface. Its software runs on a PC-based workstation connected to the indoor MW unit via a serial communications interface. The domain manager provides extensive element-

248


management functions on site and, via the microwave radio link, can be used to access any other link in a chain of stations. The domain manager can also be connected to the on-site Ethernet LAN, with transparent communications to and from any particular link whose indoor unit (IU) is connected to the LAN (as in the case of a splitconfiguration microwave system). The usual application software modules are as follows: • Element-management application; • Alarm-management application; • Configuration-management application; • Performance-monitoring application.

The element-management application allows the user to access, monitor, and control the radio stations in the network domain. In a multihop link system, any of the associated stations in the domain can be accessed. In a simple, single-hop link, only the near- and far-end stations can be accessed. All elements of information maintained within the management information base (MIB) of the selected radio station are accessible to the operator. The operator is alerted to alarms occurring within the domain by means of color changes affecting the relevant radio stations shown on the display list. The radio station signaling an alarm can then be accessed at will to establish the cause of the alarm condition and to determine appropriate steps to correct the condition. The alarm-management application monitors alarms from all radio stations within the network domain to which it has access. It also logs these alarms in a local or network database. Proper alarm-management procedures are enforced by this application in the sense that operators will have to formally acknowledge each alarm and report on the treatment thereof before being allowed to clear it. In this manner a full alarm audit trail can be established for all radio stations within the particular network domain. The alarm-management application contains a subsystem specifically designated to be adapted to control external alarm annunciation interfaces that may be specific to a particular service providers TMN or any other environment. The configuration-management application allows an operator to back up or restore the configuration of any particular radio station within the network domain to which it has access. The configuration information maintained by this application includes operational configuration parameters as


249

well as hardware and software build state parameters. The configurationmanagement application, besides allowing operators to restore the operational configuration state of radio stations in case it should become corrupted, is indispensable to service providers in whose organizations proper asset management is a prerequisite. The performance-monitoring application monitors measured, as well as statistical, performance parameters of all radio stations within the network domain to which it has access. It then logs this information in a local/ network database in a format that will facilitate the graphical display of this information using standard spreadsheet application programs. The variation in the values of parameters such as power supply voltages, outdoor MW unit temperature, received signal strength, and so on, over time can thus be subjected to graphical trend analysis. The ability to perform trend analysis is especially useful for preventative maintenance. The microwave radio craft-interface system can be integrated into another open network management platform that oversees the overall telecommunications transmission network and has full remote management capabilities. The optional network-management features include a simple network management protocol (SNMP) network agent in the microwave equipment. This allows a remote NMS to monitor and control the microwave system and to access its MIB. This includes G.826 performance monitoring information. In a similar manner, a proxy server can be configured to support other networking standards, such as Q3. Network links to the MW system can be any of the following: dial-up modem, leased line, remote via a radio link, via an on-site LAN, or by a WAN. 6.7.2

Managing a Microwave Network with SNMP

Typical microwave networks are composed of one or a combination of the network topologies shown in Figure 6.6, including meshed networks, where each node can be connected to any number of other nodes. 4 1

2

3

3 1

3 5

Figure 6.6 Linear, ring, and star topologies.

2 4

250


Although three nodes are theoretically sufficient to form the ring, due to the high-low violation, there must be even numbers of MW hops in the ring. In order to pass traffic and telemetry information, each radio has up to three connection points: the radio frequency (RF) link and the two wired connections (repeater in, repeater out). When three or more radios require interconnection at the same node, this may be implemented by daisy chaining the IDUs to form a ring. In typical wireless deployments, cells are established through an urban city. Each of these cells carries the call traffic of only its cell. In order to link the cells together and to the PSTN, microwave radios are used. The radio equipment is installed in the RBS at the center of the cell (hub site). Network management is used to monitor and control the radios of the network (Figure 6.7). In order to access and communicate with radios for control and monitoring functions, each radio within the network can be configured to have a unique identification number. This number can be in the form of the network element (NE) number or the IP address.

RBS 1

RBS 6 RBS 4 RBS RBS

3

RBS

2

5 6 1

Network management

Figure 6.7 Microwave network NMS.

1

2 2

3

3

4 8

12

9

5

4 7

6

5

10 11


251

SNMP Overview

The SNMP protocol was developed to provide a basic, easy-to-implement network-management tool that could be common to the network equipment and, yet, less restrictive of the management system used. SNMP allows the monitoring and controlling of radios by any network manager that supports the protocol (Figure 6.8). In addition, SNMP provides a quick and dynamic response to alarm conditions, which accelerates alarm notification and reduces the need for polling, but requires embedded radio software. The second method of implementing SNMP in a network of radios using a proprietary protocol is by inserting a proxy agent between the network manager and the radio network. This proxy agent is usually in the form of a computer and performs the following three functions: • Translates the proprietary protocol messages into SNMP (and vice

versa);

• Sequentially polls each radio of the network to obtain status

information;

• Reports alarm conditions to the network manager.

This type of arrangement is usually slower and still requires polling of every radio in the network, but the advantage is that it can convert legacy networks into an SNMP network. The SNMP network is made up of the following: • SNMP manager;

SNMP network

SNMP network manager

Embedded SNMP agent in all radios

Figure 6.8 SNMP network.

252


• SNMP agent; • MIB.

The SNMP manager uses SNMP to manage devices equipped with an SNMP agent such as microwave radios as well as other types of products. The management function is in the form of controlling and monitoring the network devices in a centralized fashion. It also serves as the man-machine interface to all network devices. The SNMP agent is the implementation of a network management protocol that exchanges network-management information with the network management station. The SNMP agent resides in each element (radio) of the network. The function of the agent is to respond to requests, such as the following, from the manager: • GET: Enables the management station to retrieve the value of

objects at the agent, such as status or configuration information;

• SET: Enables the management station to set the value of objects at

the agent, such as configuration parameters;

• TRAP: Enables an agent to notify the management station to signify

events; these messages are used to report unsolicited events, such as alarms.

SNMP traps, which are unsolicited alerts (alarms), are generated under the conditions listed below. It should be noted that traps are not generated for all of the radio alarms. A minor alarm does not generate a TRAP message. This is to avoid congestion of network messages that do not require immediate intervention. MIB is a structured collection of variables that can be monitored and managed by the network manager. This database is made up of all of the radios parameters, each parameter identified by a unique variable name that is used by the network manager. During the configuration phase, the MIB file is first executed on the network manager station. This action informs the network manager of the parameters that can be accessed. Direct Connection

When the network manager is colocated with a radio terminal (Figure 6.9) and only one radio subnet is to be monitored, then a direct connection (as illustrated) between the manager and the radio is possible. A different type of NMS direct connection, utilizing IP addressing, is shown in Figure 6.10.


253

MW radio network SNMP network manager

Direct connection or dedicated line RS-232

Figure 6.9 Direct NMS connection.

198.105.19.32

198.105.16.218 198.105.16.219 198.105.16.217 198.105.16.220 (router) 3 1

2 5

NMS

198.105.16.222

198.105.16.32 PPP 198.105.16.9

4

5

6

198.105.16.222

1

2

6

IP

IP

IP

IntNet

IntNet

IntNet

198.105.16.217

198.105.16.218

198.105.16.222 MW radio subnetwork

Figure 6.10 Direct connection and IP addressing.

Other NMS Connection Methods

When the network manager is not located near the radio terminal equipment, then a connection via modems (PSTN connection) is possible. A terminal server with built-in modems permits the access to several subnetworks.

254


The network manager can also be linked to radio terminal equipment via a LAN or WAN. In the case of the LAN connection, a terminal server is required at the radio end to convert the Ethernet to RS-232.

References [1] ITU-Recommendations of the CCIR, Fixed Service Using Radio-Relay Systems, Vol. 9, Part 1, 1990. [2] ITU, Handbook of Radiometeorology, Geneva, Switzerland, 1996. [3] ITU HandbookDigital Radio-Relay Systems, Geneva, Switzerland, 1996. [4] Vigants, A., Space Diversity Engineering, Bell System Technical Journal, Vol. 54, No. 1, January 1975. [5] ITU-R Rep. 338-6, Propagation Data and Prediction Methods Required for Terrestrial Line-of-Sight Radio-Relay Systems, Geneva, Switzerland, 1990. [6] Crane, R.K., Prediction of Attenuation by Rain, Proc. IEEE Trans. on Communications, Vol. Com-28, No. 9, September 1980. [7] ITU-R Rec. PN.837-1, Characteristics of Precipitation for Propagation Modeling, Geneva, Switzerland, 1994. [8] Mojoli, L.F., and U. Mengali, Propagation In Line of Sight Radio Links (Part II Multipath Fading, Telletra Review, 1988. [9] Mojoli, L.F., and U. Mengali, Propagation In Line of Sight Radio Links (Part I Visibility, Reflections, Blackout), Telletra Review, 1988. [10] ITU-R Rec. 836, Surface Water Vapor Density, Geneva, Switzerland, 1992. [11] ITU-R Rec. F.557-3, Availability Objective for Radio-Relay Systems, Geneva, Switzerland, 1994. [12] Rummler, W. D., Advances in Microwave Radio Route Engineering for Rain, ICC 87, Seattle, WA, June 1987. [13] Andrew Corporation, Catalog 38. [14] Rec. ITU-R SF.1006, Determination of the Interference Potential Between Earth Stations of the Fixed-Satellite Service and Stations in the Fixed Service. [15] GTE Lencurt, Inc., Engineering Considerations for Microwave Communications Systems, 1970. [16] Ivanek, F., Terrestrial Digital Microwave Communications, Norwood, MA: Artech House, 1989. [17] Henne, I., and P. Thorvaldsen, Planning of Line-of-Sight Radio Relay Systems, NERA Telecommunications, Second Edition, 1999.


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[18] Contract Telecommunications Engineering, PathLoss 4.0, User Manual. [19] Abbott, E., and D. Powell, Land-Vehicle Navigation Using GPS, Proceedings of the IEEE, Vol. 87, No. 1, January 1999. [20] Herring, T. A., Geodetic Applications of GPS, Proceedings of the IEEE, Vol. 87, No. 1, January 1999.

7 Transmission-Network Planning and Design 7.1 Overview The U.S. telecommunications industry was changed forever on February 1, 1996, with the passage of the Telecommunications Act. Telecommunications providers are now offered much more opportunity with fewer governmental restrictions, and yet this opportunity comes with the threat of increased competition and the need for greater focus on improved customer acquisition and support. Todays deregulated, competitive environment calls for solutions that provide a sustainable advantage in acquiring new customers and retaining existing customers. This is especially important to wireless carriers who bid large amounts of money for licenses, compete with entrenched and new competitors, and provide innovative services that are supported by increasingly sophisticated networks. Today, many wireless telecommunications providers are thinking of how to do the following: • Expand their footprint by finding and securing viable new markets

through licenses or strategic partnerships; • Plan, build, and maintain a cellular, PCS, enhanced specialized mobile radio (ESMR), also known as intergrated dispatch enhanced network (iDEN), or other broadband network; 257

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• Provide improved customer service while reducing churn and pre-

venting fraud;

• Market and sell their network and services to new and existing

customers.

In order to do these things, telecommunications providers, regardless of the market or country, must take the following steps: • Quickly and accurately estimate market potential to generate greater

return on investments made in costly licensing and infrastructure;

• Focus deployment on those areas within a market that generate

higher returns sooner;

• Increase quality and number of services offered by designing and

building networks faster;

• Reduce costs and increase speed of finding, acquiring, and building

cell sites;

• Visualize competitive threats or opportunities and identify potential

alliances in existing market areas;

• Decrease the amount of time between reported trouble and penalties

(from immediate credit to improvements in the network);

• Target marketing campaigns based on geographic profiles of cus-

tomers in a wireless network to attract new customers and increase sale of products;

• Share data among marketing, customer service, and engineering

departments for faster, more integrated decision making that benefits the entire enterprise.

Wireless networks are expensive in terms of the amount of labor needed to plan and build them, the amount of information that has to be generated, and the cost of equipment. Telecommunications companies will have to try to save time and money and ensure that the network is built cost-effectively and efficiently. Outsourcing engineering design and implementation services is becoming more popular every day. Wireless networks are increasingly complex (Figure 7.1), and different types of expertise are involved in their planning, design, and implementation (deployment).


PSTN SS7 ISUP links (STP, tandem)

PSTN (LEC, IXC)

SS7 gateway

LE, AT

MSC-to-PSTN voice trunks

Enhanced service platforms (VMS, PPS) ESP

MSC-to-BSC A-Interface

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VLR

MSC

SS7 network

STP Signaling links HLR

BSC BSC

ANSI-41 links to other MSC/HLR

WIN

Wireless Digital TCP/IP communication Intelligent network (data) Network (SCP, SN/IP)

Transmission network

Service regions

Microwave radio links Fiber rings Copper cables

RBS

Figure 7.1 Wireless network architecture.

7.2 General Wireless-Network Planning and Design Principles 7.2.1

Identifying the Opportunity and Strategic Planning

The two most important elements of strategic decision making are determining the opportunities available to the enterprise and implementing a plan to take advantage of those opportunities. This is most critical to wireless carriers that may face as many as eight competitors in their geographic market offering wireless voice and data. The first order of business is finding unfulfilled needs or services, and identifying products inadequately supplied in the marketplace. In the wireless industry, it may be markets showing rapid growth in population, where

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local wire line telephone companies are unable to provide basic telephone service (WLL). It may also be populations where the shift is away from static jobs and workplace environments to more mobile, professional types of employment (mobile wireless systems). The actual installation of a telecommunications networkthe wireless access links to users, the fiber-optic backbone, the microwave systems, the switchesis only one part of delivering a profitable and reliable service. Equally important are all the support systems that actually turn that network into a viable business. These include systems to bill users, to create and manage services across the network, to monitor failures of the network and expedite repairs, and to plan traffic capacities to ensure the optimum use of equipment. Previously, each support system operated independently and could only communicate with others with great difficulty, causing major inefficiencies or a reliance on expensive human labor. New systems using object-oriented software techniques mean that changes in one part of the system or network cause automatic updates across all related systems, creating a nearly seamless environment for service creation, delivery, billing, and customer relationship management. In most countries, telecommunications was traditionally seen as of such strategic importance to the nation that services were provided by state-owned utilities. As the political philosophies of much of the world shifted away from state control and ownership and toward private enterprise in the 1980s and 1990s, these existing telecommunications monopolies were challenged, and regulatory regimes were liberalized. In addition, many developing countries recognized the importance of telecommunications in underpinning economic and social growth and encouraged inward investment by new operators. Competition has usually been introduced to once-closed markets in a series of standard phases beginning with the opening up of the commercial data-communications services market. The next step would be the entry of mobile operators, the development of competition in long-distance and international services, and the freedom for cable TV (CATV) entertainment providers to offer telephony services over cable networks. In many parts of the world, the final stage has been reached, with operators of all types able to compete in providing domestic voice telephony services to the mass market. Complementing this, parts of the radio spectrum have been opened up from military and other closed-group use and are being auctioned off by governments for fixed and mobile voice, data, and multimedia communications services. The political shift toward open markets has also driven the increasing globalization of world trade and the development of multinational corp-


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orations. For these to operate effectively across different time zones and regions, telecommunications is crucial. This factor has driven technological, regulatory, and infrastructure development, with business location often dictated by the quality of local communications. In turn, increasing competition has led to a greater emphasis on speed of response, time to market, and the integration of the activities of distributed workgroups, workers, and outsourced suppliers. Whether it is the highly mobile international executive or the home-based teleworker, it is telecommunications that has made these new ways of doing business possible. Incumbent telecommunications operators, while often facing part privatization, have also confronted intense pressures as a result of the emergence of leaner and fitter competitors, usually using new technologies, while they themselves have the inheritance of aging legacy networks and systems. Major changes in their strategies and organizational structures have had to be engineered at short notice, and the traditional public service utility culture of most existing operators has undergone a shift toward a business-oriented, customer-focused environment. As open-market concepts have spread, a growing array of alliances, partnerships, and investments has sprung up to provide both national and international telecommunications services to business and domestic users. Many operators are now seeking to add value to their services to customers and differentiate themselves in increasingly crowded markets. Once opportunities are identified, it is important to identify and assess the competition. For example, PCS licenses were auctioned based on definitions of major trading areas (MTAs) and BTAs, while cellular licenses were awarded via lottery based on metropolitan statistical areas (MSAs) and rural statistical areas (RSAs). No single wireless company provides ubiquitous national coverage. A wireless operator must establish roaming agreements with other carriers to provide subscribers coverage while roaming. To complicate matters, there are multiple technology standards including GSM, WCDMA, TDMA, CDMA2000, and others. By layering ownership information with choice of technology, a wireless operator can learn what possible relationships are viable and the relevance of those relationships. This can lead to better decisions on the choice of the operators own technology and create opportunity rather than the alternative: isolation and failure. Once market goals for the future wireless network have been established, a thorough analysis of potential technologies is performed. Technology assessment compares each technologyvendor performance scenario against the established market goals. Once completed, the technology assessment is a strategic plan involving a complex

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trade-off between the advantages, limitations, and risk inherent in each scenario. 7.2.2

Customer Requirements Analysis

Crucial to successful market analysis is the identification of customers most likely to purchase service packages. Naturally there are many types of customers, but the most important is the one offering the greatest revenue potential in the shortest time frame. As an example, a given PCS operator may elect to target households for second line PCS service. The companys research may show that residents with $40,000 or more per year in income and having one or more children would be more likely to subscribe to that service. With this information, the operator can analyze the demographic variables on a geographic basis and visualize the results and quickly identify those license areas with people most likely to buy the service. A wide range of corporate and consumer applications are enabled not only by voice services, but by nonvoice wireless services, such as the following: • Internet chat groups have proven a very popular application of the

Internet, enabling groups of like-minded peopleso-called communities of interestto use nonvoice mobile services as a means to communicate.

• Textural and visual information with a wide range of content, from

share prices to sports scores, to weather and flight information, to news headlines, to traffic information and location-sensitive services, can be delivered to mobile phone users. This information need not necessarily be textualit can consist of maps, graphs, or other types of visual information.

• Still images such as photographs, pictures, postcards, greeting cards

and presentations, and static Web pages can be sent and received over the wireless network as they are across fixed telephone networks.

• Moving images are very important, since over time, the nature and

form of mobile communication is getting less textual and more visual. Videoconferencing applications, in which teams of distributed sales people can have a regular sales meeting without having to go to a particular physical location, is another application for moving images.

• Web browsing has never been an enduring application for mobile

users. Because of the slow speed of circuit-switched data, it takes a


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long time for data to arrive from the Internet server to the browser. Alternatively, users switch off the images and just access the text on the Web, and end up with difficult-to-read text layouts on small screens. • Document sharing and collaborative working let different people in

different places work on the same document at the same time. Multimedia applications combining voice, text, pictures, and images can even be envisaged. These kinds of applications could be useful in any problem-solving exercise, such as firefighting, combat (to plan the route of attack), medical treatment, advertising-copy setting, architecture, journalism, and so on. Anywhere somebody can benefit from being able to comment on a visual depiction of a situation or matter, such collaborative working can be useful.

• Corporate e-mail is another very important nonvoice application.

With up to half of employees typically away from their desks at any one time, it is important for them to keep in touch with the office by extending the use of corporate e-mail systems beyond an employees office PC.

• Internet e-mail services come in the form of a gateway service where

the messages are not stored, or mailbox services in which messages are stored. In the case of gateway services, the wireless e-mail platform simply translates the message from SMTP, the Internet e-mail protocol, into SMS and sends it to the SMS center. In the case of mailbox e-mail services, the e-mails are actually stored and the user gets a notification on his or her mobile phone and can then retrieve the full e-mail by dialing in to collect it, forward it, and so on. Upon receiving a new e-mail, most Internet e-mail users do not currently get notified on their mobile phone. When they are out of the office, they have to dial in speculatively and periodically to check their mailbox contents. By linking Internet e-mail with an alert mechanism, however, users can be notified when a new e-mail is received.

• Vehicle positioning applications integrate satellite positioning systems

that tell people where they are with nonvoice mobile services that let people tell others where they are. The GPS is a free-to-use global network of 24 satellites run by the U.S. Department of Defense. Anyone with a GPS receiver can receive his or her satellite position and thereby find out where he or she is. Vehicle positioning applications can be used to deliver several services including remote vehicle diagnostics, ad hoc stolen-vehicle tracking, and new rental car fleet

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tariffs. SMS is ideal for sending GPS position information such as longitude, latitude, bearing, and altitude. GPS coordinates are typically about 60 characters in length. • Wireless LAN access comes into play when mobile workers are away

from their desks and they clearly need to connect to the LAN in their office. Wireless LAN applications enable access to any applications that an employee would use when sitting at his or her desk, such as access to the intranet and corporate e-mail services. The mobile terminal, such as the handheld or laptop computer, has the same software programs on it as the desktop, cut-down client versions of the applications accessible through the corporate LAN. This application area is therefore likely to be a conglomeration of remote access to several different information typese-mail, intranet, and databases accessible through Web browsing tools or that require proprietary software applications on the mobile device.

• File transfer applications encompass any form of downloading size-

able data across the mobile network. This data could be a presentation document for a traveling salesperson, an appliance manual for a service engineer, or a software application such as Adobe Acrobat Reader to read documents. The source of this information could be one of the Internet communication methods, such as File Transfer Protocol (FTP), telnet, http, or Java, or a proprietary database or legacy platform. Irrespective of source and type of file being transferred, this kind of application tends to be bandwidth intensive. It therefore requires a high-speed mobile data service to run satisfactorily across a mobile network.

• Home automation applications combine remote security with remote

control. Basically, users can monitor their home from wherever they areon the road, on holiday, or at the office. If the burglar alarm goes off, the user is not only alerted, but can go live and see who the perpetrators are, and perhaps even lock them in. Not only can users see things at home, they can do things as well. It is possible to program the video recorder, switch the oven on so that the preheating is complete by the time the user arrives home (traffic jams permitting), and so on. As the IP will soon be everywherenot just in mobile phones but in all manner of household appliances and in every machineremote devices can be addressed and instructed. A key enabler for home automation applications will be Bluetooth, which allows disparate devices to interwork.

Transmission-Network Planning and Design 7.2.3

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Interconnection

In current usage, most wireless calls are terminated to wire-line networks, typically the local LEC, on the PSTN. However, with recent telecommunications legislation, wireless carriers can opt for competitive access providers (CAPs) that also provide local transmission in North America. In establishing connectivity to an LEC or CAP, a wireless operator must understand the ability of that wire-line carrier to facilitate interconnection. For example, most CAPs concentrate their networks in only the largest cities. By mapping incumbent LEC facilities and the facilities of available CAPs, planners can choose who handles calls terminating to wired telephone networks. Ultimately, this can lead to cost savings as LECs and CAPs begin to compete for business and interconnection rates drop. Here, it is important to identify markets with high concentrations of customers most likely to purchase a specific service package. Service options can be customized and marketing dollars precisely targeted in those areas. The Erlang B distribution is used for dimensioning trunk routes. It is also used in wireless networks, and it is based on the following assumptions: • There are an infinite number of sources. • Calls arrive at random. • Calls are served in order of arrival. • Blocked calls are lost. • Holding times are exponentially distributed.

To calculate the capacity (number of T1/E1 lines) required for the interconnect (PSTN lines), it is important to calculate the BSC-MSC traffic and add a few percent for additional services like prepaid services, voice mail, and so on. The BSC-MSC traffic depends entirely on the total number of Erlangs for the market and can be calculated in two ways. One of the methods is to use Erlang B tables, which give a slightly more precise result. In case we want to use Excel spreadsheet, the best way is to use the following formula:

(

N = E +

)

E /M

where N = Number of T1 or E1 lines; E = Total number of Erlangs (Erlangs/sub × number of subs);

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M = number of DS0s (24 for T1, 30 for E1). It is obvious that this formula does not take GoS into account. The results of the calculations for a very low number of Erlangs (a few hundred) follow closely 1% GoS, while higher numbers of Erlangs (a few thousand) follow closely 0.1% GoS. In any case, the maximum error that is made by using this formula instead of Erlang B tables and the given GoS is less than 10%, and in most cases close to 5%. For example, the number of required E1 circuits for the offered traffic of 500 Erlangs based on the tables and 1% GoS is 18 E1s, and based on 0.1% GoS it is 19 E1s. At the same time, the formula gives 18 E1s as a result. For the offered traffic of 5,000 Erlangs and 1% GoS, the required number of E1s is 165 and 172 E1s for the 0.1 GoS. The formula gives 170 E1s as a result. These results are somewhat overdimensioned, since not all the traffic will terminate on PSTN. In mobile networks, about 10% or less of traffic is between subscribers within the same network, and those calls do not get terminated on the PSTN. On the other hand, the call model is used for PSTN trunk design, BSC, and MSC capacity calculations. The call model is in compliance with the traffic formula: Erlang/Sub = ( AHT (s ) × BHCA /Sub)/3600 where AHT = Average call-holding time (sec). The call volume includes complete and incomplete calls. BHCA = Busy-hour call attempt. Again, this includes all call types. The customer usually defines AHT, but typically it is between 90 and 180 seconds. In the interconnection to the public network (PSTN), we design the network to connect to two or more tandem offices for local, national, and international traffic. In the design, the minimum trunk and signaling link requirement is two for reliability and redundancy reasons. 7.2.4

Spectrum Auctions

While the FCC (http://www.fcc.gov) has completed several sets of spectrum auctions, more spectrum auctions are scheduled for the months and years ahead. Congress has realized that auctioning radio spectrum is lucrative, and will continue pushing the FCC to make more spectrum available to


267

companies. Participation in any spectrum auction should be based on ration and the ability to generate substantial financial returns within a set period of time. Given forecasted penetration rates and estimated per-subscriber revenues, planners can identify the cost per POP and can use this information as a gauge in further bidding. In some countries (Mexico for example) there have been a number of auctions for the microwave spectrum as well (18, 23, 38 GHz). New wireless operators can lease spectrum directly from the government or in this case from the new spectrum owners and use it for the microwave systems connecting RBSs with the BSC (access transmission network or backhaul). 7.2.5

Clearing Spectrum and Microwave Relocation

One of the greatest obstacles faced by new PCS licensees is an incumbent microwave operator in the auctioned spectrum (2 GHz). These users are typically municipalities, pipeline and utility companies, or private firms. In order to carry wireless voice and data in the same spectrum, PCS licensees must provide compensation or comparable facilities to incumbent microwave operators. The FCC has undertaken a series of rulemakings that reallocated 220 MHz of fixed frequency bands near 2 GHz from fixed microwave to various PCS applications. The existing 2-GHz microwave links are permitted to move into several migration bands higher in frequency (4 GHz, 6 GHz, 10.5 GHz, and 11 GHz). The process will also provide narrower channelization in the destination bands to more efficiently handle the low-capacity links commonly in use at 2 GHz. The initial step in this process is identifying who the incumbent microwave users are and where they are located. This is accomplished by displaying the routes that microwave signals traverse in the PCS carriers license area and who the transmitter is. Overlaying this with the BTA or MTA boundary and cell site location can reveal impacts and help in decision making. If the microwave radio signals are in remote areas, the PCS carrier may be able to begin operation while negotiations continue with the incumbent microwave user. Otherwise, adjustments to network build-out will be needed, and negotiations with the incumbent will take a higher priority. Different countries have different approaches in resolving this issue. Of course, getting a license areas spectrum ready for use in PCS means more than just relocating microwave usersfor example, how much noise exists and will remain to cause problems for the future PCS network and how low this floor can be brought to ensure high quality of the wireless network.

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7.2.6


RF Design

RF design is a complex process that includes analysis of physical topography and morphology, propagation analysis, and different equipment scenarios. Before and during network build-out, it is important to model how RF propagation will occur to and from subscribers using their handsets. In doing this, many geographic characteristics have to be considered, including the following: • Building heights and their locations; • Topography; • Population densities; • Traffic volumes; • Zoning laws; • Green areas.

The end result is that cell sites are selected based on propagation models. These results are provided to site acquisition personnel who can proceed to negotiate for use of, and access to, each site. Once cell sites have been acquired and constructed, base station equipment can be tested to confirm results and to improve RF propagation (drive testing and optimization measurements). This information can be analyzed and compared to original plans for further network refining. An important aspect of this activity is that it can also be conducted on a competitors network. This can reveal coverage lapses and weaknesses in signal strength that could play to a competing wireless operators advantage in securing subscribers. RF design for a new wireless network starts with the definition and understanding of the customer requirements. The initial design has to satisfy the signal coverage of the specified square mileage; the design to satisfy capacity requirements will follow in the next phase of the network build-out. Erlangs represent the average activity during a period of time. An Erlang requirement on a system node represents, then, the ability of that node to allocate enough resources to serve as many customers as the requirement and still meet its performance goals. Performance goals are expressed in the form of blocking probability or delay. What is referred to by resource depends on the network type and call type. Required GoS is also one of the factors determining how many base stations, sectors, FDMA frequency channels, TDMA timeslots, or CDMA code channels will be deployed in a given situation. It is common for the wireless systems to require GoS, usually


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specified as a blocking probability using Erlang B formula, of 2% in North America and between 1% and 2% in Europe. A circuit-switched network establishes an end-to-end circuit involving various network facilities that are held for the duration of the call. From a network point of view, it is the holding of these resources that is important when specifying traffic capacity, not the flow of information within individual circuits. On the other hand, packet-switched networks are directly concerned with the actual flow of information, since in these systems, traffic on a transmission link is directly related to the activity of the sources. The trafficcarrying capacity of a packet-switched network is a function of the packetgeneration characteristics of the sources and transmission link capacity. In 3G networks things are more complicated since the traffic is a mixture of voice and different data services. Network resources used by a packetswitched call are not necessarily held for the duration of the call. Wireless technology is realizing a smooth evolution from circuit-switched technology to packet-based switching. This evolution can proceed over a period of time with ATM cell-based switching. ATM is an established and well-proven technology that is used by network operators to typically carry circuitswitched voice. The QoS mechanisms in ATM allow it to roll out multiservice (voice and data) networks with extremely tight delay requirements in combination with low-bandwidth radio links. In Universal Mobile Telecommunication System (UMTS), ATM is the prescribed transport mechanism in the radio access network. Here base stations are connected with many E1/T1s or higher-bit-rate connections. The small and delay-sensitive voice packets have to coexist with bursts of large IP data packets. Today, classic IP grabs all of the bandwidth, but ATM provides the required QoS for the delay-sensitive voice packets, even during handoffs from one network to another network. In the future, 3G standardization will most likely move toward the 3G real-time all-IP networks, where all services, transport and access alike, use the IP protocol stack. Initially IP will be used for simple data communication services, such as Web browsing, e-mail, and file transfer, later for streaming data, and finally for conversational services with real-time requirements. Examples of real-time services are voice, video, and multimedia IPall will be IP end-to-end. All-IP requires developments in the transport network and in managing the end-to-end QoS. Regardless of whether it is a 2G or 3G wireless network, the result of the RF plan is a number of cell sites, their location, and the type of the cellsite infrastructure required for the project. Calculations are based mainly on the given number of Erlangs per subscriber (or total number of Erlangs for the given market) and GoS.

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7.2.7


Transmission Media and Topology Planning

Transmission planning is a subset of the total wireless-network planning, both when creating new networks and extending existing ones. Modern networks allow a very great flexibility in routing and intelligent switching features. It is important, however, that transmission aspects are not forgotten in the planning process. For complicated networks one should keep in mind the abilities of the particular signaling that is implemented. Advanced signaling systems could, in addition to performing their normal functions, convey information about certain transmission parameters in connections. Examples of transmission parameters of interest are accumulated delay, existence of echo cancellers in the path, terminals not needing network echo control, accumulated impairments, choice of particular routes for calls with special requirements for high-quality connections, and so on. The goal is to build the network that will provide a reliable transmission network capable of delivering enough capacity for present needs as well as ensuring seamless expansion in the future. The cost of transmission for base station access and inter-MSC backbone networks is not insignificant, typically 20% to 25% of infrastructure costs. Its impact on operator profitability is also influenced by the way the network rollout speed and transmission-related network downtime affect the customers overall perception of service quality. Usually, transmission (transport) facilities are either leased or owned (copper, microwave, fiber optic), or most likely a combination of both. In many cases, project managers and those involved in the time-to-market assessment conclude that a faster way to build the network is to lease T1/E1 circuits from the local telephone companies rather than build their own microwave system. That may not be the case in every situation and may prove, for a number of reasons, to be a much more expensive and lengthy process than originally anticipated. Transmission-system requirements vary throughout the network life cycle and the most successful new and expanding operators offering mobile services take future transmission requirements into account in the initial service launch phase. A successful transmission platform includes necessary network functionality for the entire network life cycle, supporting network growth (scalability) and reconfiguration (flexibility) as well as fast multiservice deployment. The transmission network can be presented either with a logical or physical view. The logical view is useful for initial network dimensioning. It shows where a connection is needed in order to transport traffic from source to destination. It is, however, important also to look at the physical implementation in


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order to build an optimized network. Transmission architecture in 3G wireless networks will be slightly different than in 2G wireless networks. The main reason for this is required flexibility for expansion and increased capacity that will be mandated by new and bandwidth-consuming applications and also a number of new nodes that will be introduced in 3G networks. Even before the RF network design has been completed, it is possible to estimate roughly if the available microwave band will allow a simple microwave network or whether microwave repeaters or other microwave bands will be required. For example, in the 800-MHz cellular band, CDMA cell sites have a typical radius of 2 km for dense urban morphology, 3.5 km for urban, 7.5 km for suburban, 20 km for rural, and 25 km for highway cell sites. That will mean that in most parts of the world (except in areas with an extreme amount of rain) we could use 15- or 18-GHz bands to connect dense urban, urban, and suburban cell sites. For rural and highway sites, lower bands will be required, and most likely in the 6- or 7-GHz band or 5.8-GHz spreadspectrum (ISM band), microwave radios will be used. 7.2.8

Mobile Positioning

Location-related products are the next major class of value-added services that mobile network operators can offer their customers. Not only will operators be able to offer entirely new services to customers, but they will also be able to offer improvements on current services such as location-based prepaid or information services. The FCC adopted a ruling in June 1996 that requires all mobile network operators to provide location information on all calls to 911, the emergency services. The FCC mandated that by October 1, 2001, all wireless 911 calls must be pinpointed within 125m, 67% of the time. On December 24, 1998, the FCC amended its ruling to allow terminal-based solutions as well as network-based. There are a number of regulations that location-based services must comply with, not least of all to protect the privacy of the user. In 2000, Ericsson, Nokia, and Motorola joined forces to create a common standard for positioning for wireless networks and creating the Location Interoperability Forum (LIF). Members of this forum will be a mix of operators, suppliers, and application developers. By providing open interfaces as standardized by LIF, applications will be able to operate identically in systems from different suppliers. Different location-based applications require different levels of accuracy and clearly it is important for ships to know exactly how far they are from shore and the water depth, whereas for people location accuracy of a

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hundred meters or so would often be acceptable. Driving directions, roadside assistance and tracking of fleet, packages, and people will each become more prevalent with the availability of improved location accuracy. While some early adopters in vertical markets will be satisfied to know in what part of town their trucks or packages are located, most customers in the horizontal market will want to know exactly where their assets, or even their children, are. Similar dynamics surround roadside assistance. Although roadside assistance providers can gain efficiency from knowing in what part of town motorists are stranded, most subscribers believe they need to be located within 125m to feel safe. The provision of turn-by-turn driving directions, the most complicated location service, requires even higher accuracy and frequent updates. The key to incorporating accuracy into a network-based location service is the ability to combine improved location finding technology with less accurate methods, such as cell of origin data, so that network operators can achieve scale quickly. In addition, they can market the services broadly to existing customers instead of targeting only subscribers with enhanced phones. Customer care, billing systems, and switches can be integrated with the location applications without waiting for networkwide coverage or complete handset penetration. Finally, customers can immediately use their new location services in 100% of their coverage area and receive service 100% of the time, regardless of the inherent inconsistencies in location accuracy and availability. More information about different methods of mobile positioning is provided in other chapters of this book. 7.2.9

Toll QoS

QoS is defined as the collective effect of service performances that determine the satisfaction of a user of a service. These quality of service performance factors are applicable to all services, such as service support performance, service operability performance, service accessibility performance, service retainability performance, service integrity performance, and other factors specific to a given service. Network performance is defined as the ability of a network or network portion to provide the functions related to communications between users; it contributes to service accessibility, retainability, and integrity. Network performance parameter values are usually derived from quality of service parameter values [1]. Connections including the wireless system should, under error-free conditions, achieve subjective ratings comparable to those of connections in


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the ordinary PSTN. This is a necessary, but in no way sufficient, condition for being considered PSTN quality or toll quality. To realize the widespread acceptance, from a quality perspective, of the PSTN, many other requirements should be satisfied by wireless systems. For example, the received speech should sound natural, and users should be able to recognize the voices of people with whom they are familiar. In addition, wireless systems should be robust to transcodings (such as when used in tandem with a far-end wireless system) and robust to a reasonable level of bit and frame error. It also has to be robust to the wide variety of ambient noise conditions (e.g., offices, outdoors, highways) under which such systems will be used. Also, highly interactive conversations should be possible with minor effort, meaning that excessive delays cannot be introduced and no annoying effects should be imposed on call progress tones, network announcements, or music-on-hold. Severe channel impairments, such as signal dropouts shall not be frequent or regular, and speech processed through such systems shall be recognizable by network-based speech recognition systems (that already work well with PSTN-originated speech). Failing to support these capabilities, or providing performance levels that prove unsatisfactory to the user, may cause wireless systems not to have the widespread user acceptance that is the clear goal for these systems. 7.2.10 Network Performance

In todays telecommunications networks, wireline, wireless, and Internet communications converge. To be able to provision, test, and monitor the network (and of course bill the customer), it is necessary to have remote access to every piece of equipment in that network. The key to providing service quality lies in the successful integration of network technologies that provide both voice and data. Networks are becoming more intelligent, geographically dispersed, and larger every day. We continue to deploy thousands of intelligent software NEs throughout wireless networks, from the backbone to the periphery of the WAN and the last mobile unit in the network. These NEs will intelligently administer tasks across the network, such as accounting functions, user authentication, inventory control, and configuration and monitoring of remote devices, while also generating new revenues and reducing network operation and management costs. The integrated NMS should be vendor-independent and able to do the following: • Provide an impartial view of the service quality provided by each

vendors equipment;

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• Extract service quality information from all of the protocols and

technologies in use in the network; • Provide a view of end-to-end delivered-service quality.

In todays complex multiservice environment, the customer care and billing system must be flexible to handle the multitude of services that may be billed on different usage schemes. Whereas traditional telephony calls are billed on a duration basis, this charging model is, in many cases, not appropriate to the new band of services, which may require different models such as usage or volume-based. Additionally, services may need to be bundled to remain competitive and so the billing system will need to support the complex rulesbased rating models needed to bill for these. The time-to-market for new services is essential, and therefore the customer care and billing system must be flexible to support this. It is important to remember that superior customer service is the key factor in retaining subscribers. While this has become a cliché in many industries, including telecommunications, it is of critical importance now more than ever to wireless carriers. Fast resolution of customer complaints and the ability to quickly correlate multiple troubles is part of this customer service initiative. When a customer calls in, the customer service representative (CSR) must be able to record the trouble, the time when it occurred, and the location of the trouble. The CSR can also assess the problem based on specific categories: 1. Service quality Coverage Dropped calls Blocked calls 2. Voice quality 3. Special features SMS Voice mail Call waiting Call forwarding 4. Equipment problems No connection Handset problems 5. Fraud


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Subsequently, these problems can be mapped and correlated to problems other customers are encountering. This information provides engineering with the data it needs to spot recurring problems and to resolve them. Voice quality is usually based on subjective tests performed in the lab environment as well as the opinion of actual users of the system. The average of the scores obtained from the number of listeners is called the mean opinion score (MOS). The toll-quality voice usually has MOS > 4 on a scale of 1 to 5, with 5 being the best (speech is perfectly understandable) [2]. It is important to note that all of the problems listed above could be caused by over-the-air interface (RF), subscriber units, network elements (switching), or transmission facilities (E1/T1 circuits). When engineers start troubleshooting the network problems, all the components of the network architecture are the suspects. The network performance improvement and optimization process examines dropped calls, interference, handoff performance, coverage verification, customer churn, and other competitive factors. It focuses on data collected from the network in live tests. Drive tests using wireless measurement equipment, actual phone quality messages, switch data collection, and network performance-reporting software are important elements of the optimization process. Using this data, it is possible to troubleshoot problems and correct them accordingly. Network management involves assuring quality service for subscribers wireless voice and data communications. However, no network is completely immune to problems, natural or man-made. While cell sites can frequently impact call quality, there may also be problems in handling traffic from cell sites to the switching center during peak periods of the day. A group of cell sites may need to be upgraded from a T1 line to a T3 line to handle the volume of calls. To support network operation and maintenance and fast problem resolution, many wireless operators are creating site information databases. These databases detail information on each cell site including location, topography, RF propagation, equipment inventory, and physical structure. In turn, this information can be made accessible to field personnel who have responsibility for maintaining and inspecting each cell site. It is a tool that helps in identifying and deploying network components to begin operations and to sustain them with quality call completion. An increasing problem in providing wireless service is fraud. It accounts for close to $1 billion in cellular costs per year, and this is expected to grow. While providing quality customer service, a wireless operator must also pay attention to those end-users who are trying to exploit a wireless operators network illegally. The first step in combating fraud is identifying

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those parts of an operators coverage area where much of it happens. Measures can then be taken to reduce the problem; preventative efforts such as requiring personal identification numbers (PINs) or security codes can be used to intercept calls and to validate subscribers. Sales to these areas can be restricted and certain types of calls (e.g., international) can be blocked. On a more active basis, investigators can be assigned to identify and prosecute perpetrators. Also, users are most vulnerable when they leave their home areas. Users may roam using dual mode analog-digital handsets, and may travel to areas where the provider has not yet implemented authentication. Authentication is a method by which analog and digital networks exchange algorithms between the switch and a handset to insure secure transmission before sending information. For network operators, fraud represents a very real problem. An effective fraud strategy can save millions of dollars in lost revenue and protect your relationships with your key customers. Faced with increasingly sophisticated fraudsters and hackers, operators must take additional steps to ensure the security, reliability and profitability of their services across all their networks: fixed, mobile, and intelligent. 7.2.11 Sales and Marketing

The key ingredient to near-term success for any wireless operator is the level of sales and marketing expertise that it concentrates on attracting subscribers. As a carrier builds out its network, it must match its network and capabilities to end-user needs and the likelihood of end users to purchase. Customer profiles must be built and analyzed, competitive offerings must be understood, and sales materials must be generated. As an example, those subscribers having high long-distance usage may tend to be located in certain license areas. These end users could be offered the operators branded long-distance for wire-line and wireless calls that, when bundled, provides cheaper overall monthly bills. Identifying this would assist in filing the appropriate tariffs and forms and arranging for connectivity with a wholesale long-distance carrier. Lastly, information on calling patterns can be used to create tailored rate plans. For instance, if a firm currently resells cellular service prior to operating its own PCS network, it can draw upon billing records to identify cell sites where its cellular customers are making calls. The correlation of where cellular calls are placed and where those end users live and work can result in PCS rate plans that are zoned and more attractive than cellular plans. Also, by obtaining information on competitors license areas and the status of their network,


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a wireless operator can match those competitors or decide to focus on underdeveloped or neglected areas. In some cases, multiple players may fight aggressively in the same big city for large revenues while ignoring smaller cities with higher profit margins. An operator can determine this through competitive analysis and refocus its energies on the smaller cities. Over the next few years, many of the existing operators will migrate from the 2G toward 3G wireless networks. The following issues concerning that process stand out: • How to increase average revenue per user; • How to minimize costs of technology deployment and to adopt as

simple a deployment process as possible;

• How to deploy viable commercial services in a timely manner; • How to maintain a satisfactory end-user experience throughout this

process.

The latter two items on this list are especially important. If operators cannot deploy viable commercial services in a timely manner, they run the risk of losing competitive position. If the network or handsets deliver a negative end-user experience, they will not only fail to generate revenue, but will encourage churn as well. Not to be forgotten is the importance of voice. Using compression techniques and more sophisticated bandwidth management, 3G will eventually enable nonvoice applications such as full-motion video and multimedia, all in real time. This is the beginning of the mobile Internet era with possibilities to increase revenues from existing subscriber bases as well as from new market segments, segments that have thus far not been economically or technically possible to address. An indication of the business potential is presented by predictions that there will be approximately 600 million mobile Internet users by the end of 2004, with 3G leading us into a phase of mobile Internet hypergrowth. It is predicted that within a few years, more end users will access the Internet via handheld mobile terminals than wireline connections. Subscribers today are more sophisticated when choosing their wireless service provider. Increased subscriber dependency on the Internet and information services is creating new demands on wireless systems. In the near future, IP telephony has the potential to become the dominant technology for the provisioning of real-time wireless telephony services. The migration of all elements in the present circuit-switched domain (basically voice and video

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applications) toward the IP-based packet-switched domain is expected in the medium to long term. The circuit-switched domain will, however, continue to grow for some years mainly due to the fact that this technology is currently the only one that will ensure the service performance and reliability required by the operators. Although wireless IP telephony is more flexible in its ability to provide a variety of innovative services, the strong consensus today indicates it may be several years beyond 2002 before it can compete with CS services in terms of radio spectrum efficiency, error robustness, and voice quality. Smooth migration toward an all-IP network means preparing for this evolution by basing the launch system on a future proof backbone and at the same time maintaining the support for circuit-switched domain traffic. The focus of a subscribers evaluation criteria when choosing a wireless service provider is based on value-added services and features. A sophisticated blend of information, voice, data, education, entertainment, multimedia, anywhere, anytime, will provide wireless operators with opportunities in offering differentiated services for targeted customer segments.

7.2.12 Regulatory Issues

The FCC is an independent federal regulatory agency that is responsible directly to Congress. Established by the Communications Act of 1934, it is charged with regulating interstate and international communications by radio, television, wire, satellite, and cable, and its jurisdiction covers the 50 states, the District of Columbia, and U.S. possessions. The general objectives of federal telecommunications regulations are to provide efficient use of the electromagnetic spectrum, which is considered a public resource and to develop a domestic telecommunications infrastructure able to provide service on the national level and compete on a global level. Especially in recent years, its task is also to provide a highly competitive economic market that spurs technological advances in the telecommunications industry. The FCCs Wireless Telecommunications Bureau (WTB) handles all FCC domestic wireless telecommunications programs and policies, except those involving satellite communications. Wireless communications services include cellular telephone, paging, personal communications services, public safety, and other commercial and private radio services. The WTB regulates wireless telecommunications providers and licenses. The bureau also serves as the commissions principal policy and administrative resource with regard to federal auctions for the private use of the public airwaves.


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The WTB regulates the three primary wireless communications services: cellular, ESMR (also known as iDEN), and PCS. Portions of the frequency spectrum are allocated to specific uses (such as TV broadcast band, cellular), and specific frequencies within that part of the spectrum are assigned to licensed operators. These procedures are intended to prevent interference or conflicts among various operators or services at a given location attempting to use the same portion of the frequency spectrum. One of the functions of the FCC is to issue licenses to wireless communications carriers. The FCC issues licenses for certain frequency bands of the electromagnetic spectrum and effectively limits the number of wireless communications providers in a specific geographic service area. In addition to regulating licenses, the FCC establishes performance standards for cellular and PCS providers. The FCC requires cellular and PCS licensees to provide, within a specified period of time, a coverage ratio of a minimum quality for either a composite geographic service area or a percentage of an areas population. The FCC currently requires cellular, ESMR, and PCS providers to comply with the ANSI/IEEE standards for radio frequency electromagnetic fields as a condition of licensure. Under authority granted by the Federal Aviation Act, the FAA has jurisdiction over the following communication facilities: (1) towers that exceed 200 ft in height; (2) towers that are located within 20,000 ft of a major commercial or military airport; and (3) towers that are located within 10,000 ft of a general aviation airport. The FAA reviews the location and height of such towers and may require them to be painted or illuminated to prevent possible interference with nearby airport operations. The FAA also reviews possible interference issues with aircraft-to-ground communications that may be caused by transmission facilities located in or near airport flight paths. Under the requirements of the FAA, wireless communications providers are responsible for filing a notice with the FAA if their facilities are subject to FAA review. Most local governmental agencies regulate wireless communications facilities via land-use regulations contained in respective zoning ordinances and general plans and are responsible for reviewing and processing applications for discretionary and ministerial permits for these facilities. Local governments also have the broad authority to ensure the public health, safety, and welfare of their citizens. Local jurisdictions regulate wireless communications facilities through the permitting process. Most agencies require a discretionary permit, such as a conditional use permit, in order to construct a facility. Whether a permit is processed administratively or requires a public hearing varies among local

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agencies. In general, administrative processing entails lower permit fees and shorter processing times, while the public hearing process involves higher permit costs and a longer permit turnaround time. Frequency spectrum is a valuable resource and is generally subject to appropriate planning and management to prevent misuse and interference between the many and varied applications. National administrations will allocate some or all of these bands for fixed microwave radio use in line with local requirements. Before microwave network planning commences, the wireless operator must determine available frequency bands and channel plans specific to that country. Often, and preferably, an operator may be able to obtain a number of frequency allocations as a block, enabling him to perform his own network planning in advance without risk of interference from other users. Most regulatory authorities also operate a local link length policy, where the length of a particular path will determine what frequency bands are available for the operator to choose from. Typically, the shorter the path the higher the frequency required. Frequency planning is the coordination of link frequencies to minimize any interference between links within the network and those operated by other users. In some instances, the local regulatory authority undertakes frequency planning. If, however, a block allocation has been obtained, then planning will be the responsibility of the wireless operator. The local requirement for equipment type approval will also vary from country to country, ranging from a simple paperwork exercise to a full product test program to local standards. Type approval is generally the responsibility of the radio supplier, but an operator should ensure that all requirements are satisfied before microwave links are deployed. Other limitations imposed by authorities can also have an impact on microwave radio deploymentfor example, tower height restrictions or limitations on antenna size. These factors can restrict effective radio lengths at the planning stage and should be ascertained in advance of the detailed link design stage. In every wireless and wire-line network there must be a provision for the lawful intercept (LI). The LI is used to intercept unobtrusively telephone calls to or from designated subscribers and send the information, composed of the activity on the connected line as well as call-related information (CRI), to Monitoring Center (MC). The LI is equipped to handle calls that require full monitoring, where the entire conversation is recorded and can be monitored by the appropriate law enforcement agency, and calls that only require statistical monitoring, where only CRI is kept in a database. CRI consists of the following information:


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• Called and calling parties phone numbers; • Priority of the call (high or low priority); • Level of monitoring (full or statistical); • Call start time; • Call end time; • Call duration.

The LI receives a call to be intercepted from the switch on one trunk, makes an outgoing call to the switch on another trunk, and connects both legs of the call to carry the conversation. Additionally, in the case of fully monitored calls, the LI establishes another call to the MC where the call can be recorded. In the case of statistically monitored calls, the LI sends to the MC information such as time call started, time call ended, called and calling phone numbers involved, and other pertinent information to be stored in the data files. 7.2.13 Life Cycle of Wireless Networks

The life cycle of successful wireless networks involves development through steps: initial service launch, high growth phase, network changes and optimization, and new revenue opportunities involving the launch of new services. This stepwise development implies changes in focus for network optimization. Essential to competitive development through transmission technologies, these changes in focus drive changes in network planning and configuration throughout the network life cycle. Increasingly, new operators strategically create their own unique portfolio of value-added services designed to attract a targeted subscriber potential, specializing either in cost minimization or revenue maximization. The operators choice of strategy naturally varies from one country to another, depending on local or regional regulatory terms and charges. Successful mobile network operators launch their services by providing adequate geographical coverage in key areas of the network, including major cities main roads and selected special tourist or business locations. This coverage tends to be implemented through a network consisting of a few macrocellular base stations and an MSC. Fast service rollout for a new network is of vital importance to enable cash inflow and investment payback. The transmission network in such a launch phase is not usually optimized for link capacities or cost, but the transmission system and its management system

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are advisable to be selected for their ability to meet potential requirements of future network growth. During later phases, many new base stations will share the main transmission links of the initial launch network. As the subscriber base increases, successful operators add new base stations, BSCs, and switches to serve all cities and improve roadside and indoor coverage. The number of macrocellular base stations can grow swiftly to hundreds, and ultimately to several thousands, and the number of switches could grow to between 15 and 30, resulting in a rapid increase in the number of transmission links. Managed network growth is essential in this phase, as installation and commissioning successfully require the minimum number and duration of site visits for a smooth and cost-effective transition. Cost optimization plays an increasing role during this phase, as grooming to carry traffic from several base stations over fewer T1/E1 links optimizes transmission links. Transmission hubs located in selected base station sites perform the grooming, and new operators who lease the transmission capacity also benefit from savings enabled by grooming. As the number of base stations grows larger, successfully expanding operators rearrange transmission links more frequently by adding capacity or rerouting the traffic for one or more reasons: • Macrocellular base stations are added; • Transmitter/receiver units (TRX) are added to existing base stations

to upgrade capacity;

• Microcellular cell systems are added; • New BSCs are installed, and base stations reconfigured to work

under new BSC serving areas;

• New switches are installed, and base stations reconfigured to work

under new switch serving areas;

• Automatic rerouting and repair capabilities are introduced; • New services, mobile and fixed, are introduced with minimum

rollout, with the same transmission infrastructure used for multiple services.

As the competition for a regions cellular market grows, service differentiation and introduction of new value-added services play an increasing role. Operators who use their own transmission networks tend to begin offering fixed services such LAN-interconnect, voice, IP and other services in any


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phase. As a cost-effective strategy, the most successful operators may use the same transmission network for both mobile and fixed services.

7.3 Transmission System Design 7.3.1

Transmission System Design Process and Requirements

Process is, by definition, a linked set of activities, which defines an object and is directed toward achieving a specified and measurable result. Processes should be defined and described to improve performance within the organization and its projects. A process description includes descriptions of activities in the process as well as entry and exit criteria for these activities. Entry criteria are predefined criteria that should be fulfilled before a certain task can be started. Exit criteria are also predefined criteria that should be fulfilled for a certain task to be completed. The purpose of this chapter is to provide an overview of the transmission-network design methods, processes, and procedures used in the wireless networks and in transmission networks in general (Figure 7.2). It will also point out the need for the RF, switch, and transmission engineers to work closely together in defining, planning, and deploying the most reliable and cost-effective network possible. During the first stage of the network design process, a few initial questions need to be asked with regard to such areas as economics, the area topology, the existing network, and the services that the customer wishes to offer. The following are a few examples: • Is this an upgrade of the existing wireless network or a completely • • • • • • • •

new network (green field project)? What economic resources does the operator have? Are we planning 10 years ahead or just dealing with todays demand? Is the operator going to cover small areas (islands) or the entire country? Are we primarily aiming for the business or private subscriber? Is the main purpose of the new network for data traffic or speech traffic? Is CS traffic to be maintained on the existing network? Is there spare capacity on this existing network? What kinds of services is the operator going to offer?

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Transmission Systems Design Handbook for Wireless Networks Site visit results

Customer/marketing input Other inputs

Draft RF network plan output

RF engineering

System engineering Transmission Switching

Customer input Draft Tx network plan output

Tx engineering

Switching/Network Architecture input

All Yes Final RF/Tx requirements network plan satisfied? No Deployment

Figure 7.2 Transmission-network design process.

Before the transmission-network planning can start, some basic activities have to take place in order to define the wireless networks requirements and expectations: 1. Review the proposed coverage area (or RF network plan, if completed) and identify the need for the transmission (backhaul)leased lines, fiber, or microwave facilities. 2. Determine the responsibilities for the backhaul (leased lines, fiber, MW) design and deployment. 3. Meet with customer(s), contractor(s), vendor(s), and partner(s). 4. Sign off the nondisclosure agreements with all parties (customer, vendors, partners) involved in the project. 5. Identify potential microwave sites and MW capacity requirements. 6. Identify need for the MW repeaters or SDH high-capacity MW backbone. 7. Identify MW frequency bands or channels to be used. 8. Identify availability of the unlicensed microwave spectrum. 9. Identify existing MW systems in the area and the source of information (microwave frequency coordination).


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10. Provide information (drawings, maps) of the existing transmission facilities in the area (MW, fiber-optic, copper), as well as PSTN offices and POPs of the local telecommunication companies. 11. Determine availability numbers, reliability, and mean time to repair (MTTR) of the existing transmission facilities. 12. Identify backhaul requirements for the switch location. 13. Identify and visit potential switch site location(s). 14. Complete the site survey on some or all of the potential cell-site candidates and determine feasibility of the sites to be used as MW sites or hubs (determine LOS with the adjacent sites). 15. Determine existing tower and other antenna mounting structure capabilities, sufficient space for the MW radio equipment and antenna installation (provide site layouts and tower profiles) and access to those sites. 16. Determine power and battery back-up requirements. 17. Find out all the customer-specific requirements. 18. Organize and schedule detailed MW path or site survey if required. 19. Complete a scope and task delineation list (showing who is doing what). 20. Provide preliminary high-level transmission-network design (TND) and identify equipment required. 21. Identify equipment and service resources (for international projects, try to find local companies). 22. Develop preliminary transmission-network build-out schedule. The role of the transmission-network planner is no longer restricted to simply network and technology requirements. It is important for the network planner to be able to optimize existing and new network resources to create a high-quality and cost-efficient network. It is vitally important that the transmission-network planner understands the business objectives of the company and is able to respond to business requirements effectively. To plan a network effectively, a transmission-network planner needs to have a complete understanding of the whole network. At all stages, the emphasis should be on designing simple network architecture. This will be beneficial in terms of deployment and network management and will provide flexibility to allow for easy network expansion.

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7.3.2


SDH and PDH Transmission Systems

In PDH, older T-carrier and E-carrier networks are plesiochronous; backhaul microwave systems for small to medium wireless networks are always PDH. One of the problems with the PDH is that it doesnt have standardized optical interfaces. SDH is a set of international standards for broadband communications over single-mode fiber-optic transmission systems. The standards were originally developed in the United States, where they are known as SONET. Basic signal levels of E1 (2 Mbps) and E3 (34 Mbps) can be multiplexed into an STM-1 or STM-4 signal. The SDH signal, being synchronous, allows direct access to all of the component signals without the need to demultiplex the complete signal; thus, only equipment sufficient to add or remove the required signal is needed at any location. SDH networks are usually configured to be highly redundant, with dual fibers providing backup. They can be laid out in a linear fashion; however the optimum network topology is a ring. In a ring configuration, the first fiber transmits in one direction, with the second fiber transmitting in the opposite direction. Through this layout, it is highly unlikely that any device on the network can be isolated through a catastrophic failure; this feature is known as self-healing. Digital microwave high-capacity networks today are designed with these same principles in mind. 7.3.3

SDH Transmission-Network Protection

A description of the generic protection types in an SDH network is provided in ITU-T Recommendation G.805. This ITU-T recommendation indicates how these generic types are applied in the case of the SDH. A detailed description of the implementation of some of these schemes is provided in ITU-T Recommendations G.783 and G.841. SDH multiplex section protection is a type of trail protection as described in ITU-T Recommendation G.805. Failure events are detected by the multiplex-section-termination (MST) function, and the reconfiguration uses the protection switching functions that are in the multiplex-sectionprotection sublayer. The resultant reconfiguration may involve protection switching in multiple SDH network elements. The coordination of such switching in multiple SDH network elements is by means of an automatic protection switching (APS) protocol [3]. In a 1+1 SDH multiplex-section-protection system, two multiplex sections are provided: One carries the traffic, and the other acts as a standby. A description of multiplex section 1+1 protection is given in ITU-T Recommendation G.783.


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A 1:N SDH multiplex-section-protection system consists of N trafficcarrying multiplex sections that are to be protected, together with an additional multiplex section to provide protection. When not required for protection, this additional multiplex section capacity can be used to support lower-priority extra traffic. This extra traffic is not itself protected. A description of multiplex section 1:N protection together with the APS protocol is given in ITU-T Recommendation G.783. Multiplex-section shared-protection rings are characterized by dividing the total payload per multiplex section equally into working and protection capacity. That means that for a two-fiber STM-N ring, there are N/2 administrative unit groups (AUGs) available for working and N/2 AUGs for protection, while in a four-fiber STM-N ring, there are N AUGs available for working and N AUGs available for protection. Any multiplex section of a multinode ring under a section or node failure condition can access the ring protection capacity. Thus, the protection capacity is shared between multiple multiplex sections. This sharing of protection capacity may allow a multiplex-section shared-protection ring to carry more traffic under normal conditions than other ring types. Under nonfailure conditions, the protection capacity can be used to support lower-priority extra traffic. This extra traffic is not itself protected. A description of multiplex-section sharedprotection rings including the definition of the APS protocol, is provided in ITU-T Recommendation G.841. A multiplex-section dedicated-protection ring is a 1:N protection scheme where N = 1. A system consists of two counterrotating rings (each transmitting in opposite directions relative to the other). Under failure conditions, the entire working channel is looped to the protection channel. The APS protocol required for this scheme is not provided in ITU-T Recommendation G.841, since the maximum capacity of this type of ring is the sum of the capacity on each span, and therefore the applications for this type of protection scheme are limited. Subnetwork connection protection is described in ITU-T Recommendation G.805. It may be applied to either a SDH higher-order path or lower-order path. To support subnetwork protection, two dedicated subnetwork connections are provided: One carries the traffic, and the other acts as a standby. This protection mechanism can be used on any physical transport structure (e.g., meshed, rings, or mixed). It can be used to protect a complete end-to-end network connection or a portion of a network connection. Further details of the application of this scheme in the SDH are provided in ITU-T Recommendation G.841.

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7.3.4


Ring Protection in the Wireless Network

For a larger transmission network, it is recommended that ring configuration be used as a high-capacity backbone (SONET or SDH) carrying traffic from the cell sites to the switch location. In this configuration BSC is always one node on this high-capacity ring. Ring configuration provides for a reliable communication facility, since it offers automatic protection from the following: • Site hardware (batteries, towers, antenna system) failures; • Radio and MUX equipment failures; • Propagation failures in the microwave network; • Cable cuts in the fiber-optic network.

It also provides basic user features such as simple operation, fault location, and maintenance, enables alternate routing of E1/T1 traffic automatically, and protects against loss of E1/T1 traffic due to single failure. The main rule is that each E1/T1 circuit must be dedicated completely around the ring. Reuse of the same E1/T1 in the opposite direction is not possible, and for ultimate reliability, both directions can be 1:1 hardware protected. All the sites that belong to the ring are considered hub sites and have to be completed first in order to provide protection. Figure 7.3 shows the principle of operation of the self-healing bidirectional ring configuration. In microwave systems, additional protection (e.g., space or frequency diversity) at lower frequencies may be required against short-term multipath outages. Added service reliability can be achieved even without the ring configuration, using automatic rerouting. Many successful mobile operators protect transmission by using automatic traffic rerouting on their most important links, assuring additional reliability in normal situations, such as when access microwave radio links suffer cutoff due to poor weather conditions or possible fiber-optic cable cuts or any other human error. With a flexible rerouting transmission system such as T1/E1 trunk rerouting, backup capacity can pass via physically separate routes, as the problem is not likely to interrupt both routes simultaneously. The base station trunk is the entire physical transmission link between two base stations or sites or between a base station and its base station controller, typically a T1/E1 or nxT1/E1 link. In case of traffic failure, trunk rerouting switches all traffic in the main trunk simultaneously to the backup trunk (Figure 7.4). Large base stations comprising a number of circuits are


E1/T1

E1/T1

RBS3

E1/T1 RBS2

Repeater

3

T1/E1 Protection-only direction

4

DS0 21 RBS4

1

RBS1

Repeater

Red alarm

RBS4

1

E1/T1

Rain outage

3 T1/E1 Wrap

RBS2 2

4

Red alarm

RBS3

E1/T1

2 T1/E1 Normal direction

289

RBS1 BSC

E1/T1

BSC

E1/T1

E1/T1

a.

b.

Figure 7.3 Ring protection in MW network: (a) normal (CW rotation) operation, and (b) failure between sites 3 and 4.

switched simultaneously for minimum service downtime. Rerouting can be arranged for all sites or only critical sites, such as base stations that are labeled as higher priority (e.g., hub sites). 7.3.5

Description of TND Deliverables

A deliverable is any output from the process of planning, design, testing, and so on. It could be a drawing, written document (report, white paper, spreadsheet), or just about anything two sides agree upon. This section describes briefly each deliverable during the transmission-network planning, design, and deployment stage and its contents. The preliminary network report is a description of the adopted transmission strategy. The topologies chosen for the different parts of the network are presented along with background for the choice. The report will also present the proposed media for the various parts of the network and the reasoning

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RBS3

Traffic interruption RBS2

RBS1 BSC

RBS4

RBS5

RBS3

RBS2

RBS1 BSC

RBS4 Traffic rerouting RBS5

Figure 7.4 T1/E1 link (trunk) protection.

supporting the proposal. Furthermore, the estimated capacities in the network are presented. The report is written on an overview level. It does not account for each individual link. The preliminary network layout is a network drawing showing an overview of the access and backbone transmission networks. Links routing as well as estimated capacities and proposed media are indicated. If the nominal network planning is based on a channel-loading plan only, the layout will not show each individual link as the site positions are not known. In this case the layout will detail the main link routing such as the backbone transmission network, and the higher capacity links in the access network connecting a group of base stations and hub sites.


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The preliminary equipment list is presented on a network basis, so the list is not broken down to show equipment per subnetwork or lower levels. The equipment list will detail all transmission equipment included in the preliminary network plan. The level of detail is sufficient to enable forecasting and ordering of equipment. Prices are not part of the equipment list. The list of approved sites presents which of the proposed sites have been approved from a transmission point of view. The list will also present rejected sites together with a short description of why they were rejected. Every site has to have either access to leased T1/E1 facilities or LOS for the microwave system. The MW LOS survey report contains all relevant site data for surveyed sites from a transmission point of view, such as site name, geographical position, street address, site height, and proposed tower or pole height. It also includes line-of-sight information on all surveyed paths detailing path name, bearing (azimuth), distance, a short path description, and LOS confirmation. Photographs for each surveyed path are included in the report. The transmission-network plan describes the final transmission-network design based on actual site positions. It includes trunk routing diagram, base station to switch logical connection diagram, timeslot routing diagrams and the network management system plan. The report is written on a detailed level and it accounts for each individual link. The diagrams are normally supplied as attachments to the report. The timeslot routing diagrams, are only produced for links connected to DACS equipment, if any. The layout is a network drawing showing the access and backbone (core) transport networks. Link routing as well as capacities and media are indicated and the layout will show each individual link. The final equipment list is presented according to customer preferences and on a network basis, subnetwork basis, per transmission site, or any combination thereof. The equipment list will detail all transmission equipment included in the final network design. The level of detail should be as close to the final requirements and final transmission-network plan as possible. The MW path and frequency-planning report (for microwave systems) includes a specification of recommended channels with exact frequency, output power, polarization, and antenna size and height. The report will also present the estimated threshold degradation for each link. The degradation value is a theoretically calculated value. Furthermore, the report includes performance prediction for each link. The calculated quality and availability figures are presented along with corresponding objectives. The calculations are based on North American and International (ITU) theoretical models.

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7.4 Leased Lines in Wireless Networks Leased lines (known also as leased circuits or leased facilities) in the transmission network (usually T1/T3, E1/E3) are the optimal solution in the short term for the fast transmission-network deployment. Minimal resources are required from an engineering, implementation, project management, and operations perspective when compared to a microwave or fiber-optic build, but represent high recurring costs due to monthly fees. For example, orders are placed with a carrier when a site location is confirmed. Careful scheduling with carrier is needed to ensure minimal disruption or noise when installing facilities in a building and to meet the overall objective of delivering DS1s on time for turnup. Once in operation, a maintenance problem is simply addressed by placing a phone call to the carrier to report the problem. Unfortunately, there are a number of caveats when using leased facilities. First and foremost is the high recurring cost. It will last ad infinitum unless the intention is to replace it with a privately owned system. In addition to the recurring cost, there is a typical service chargeusually several thousand dollars per facilityand a construction chargefrom several thousands to hundreds of thousands of dollarsif the carrier is required to build facilities to a site. Another problem with leased facilities is the limited capacity and also the fact that leased transmission facility cost is linear with respect to bandwidth. For example, if twice the capacity is desired then twice the facility must be leased, thereby doubling the cost. Leased facilities are also inflexible in network design and it is a lengthy and costly process to reconfigure leased lines to address ongoing changes in the network. Finally, for any maintenance problems on the leased line, the time to repair is at the mercy of the carrier. Therefore, it is important to establish a good working relationship with the facility provider (telecommunication provider) in order to understand the interworkings of the company and hopefully to influence key departments when help is needed. Owned facilities in the wireless network usually require planning, building, and maintaining the microwave network. This process starts almost immediately after the launch of the wireless network, and by year three, most of the leased facilities should be replaced by the completely owned and operated transmission network. Details of the microwave network planning design and build-out are discussed in Chapters 6 and 9 of this book. Today, customers are also able to lease different services based on ATM from the local telecommunications company. Recommendation ITU-T G.176 provides transmission performance planning guidance to network and service planners who are responsible for the integration of ATM


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technology (e.g., digital processing equipment, multiplexers, and switches) into the PSTN [4]. This recommendation recognizes and addresses the interconnection of other networks (e.g., private networks and digital cellular networks) with the PSTN and the continued need to support existing voiceband services. 7.4.1

Leased Access and Core Transmission Networks

Base station access networks in most parts of the world (except in North America) are most often both owned and operated by the mobile network operator as a strategic asset. The main reasons for this are profitability, together with the control it gives over rollout and services in terms of quality and the timely availability of new connections. Microwave access dominates in base station access network implementations, as it is often the fastest means for network rollout and capacity expansion. Using microwave transmission, an operator saves on operational expenses compared to laying his own cables or leasing connections. At least two-thirds of all base station connections are based on microwave. Access is copper-based transmission, when copper lines are widely available at an attractive price. This is particularly attractive if the operator owns copper lines. The access technology for the dark copper case has been high-bit-rate digital subscriber line (HDSL). However, copper-based connections may not always provide the same flexibility and controllability in rollout as wireless (microwave) alternatives. Optical fiber is constantly gaining a greater foothold and has a clear role in future (3G) transmission-network implementations, providing transmission capacity to regional hub sites, from where the capacity is further distributed by using wireless or copper media up to individual base stations. Optical fiber network may be used in the future to provide access even up to the base station sites. Hub sites are needed in base station access transmission network for grooming traffic and managing protection, especially when distances between the BSC and base station increase. The connections between switches (core network transmission), extending also to BSC connections, are of higher capacity than the base station access connections, and the distances to cover are longer. Not every mobile network operator has so far had the resources to build its own transmission for these long circuits, so these may be leased from a country-wide transmissionservices provider as protected E1 (2 Mbps) or T1 (1.5 Mbps) connections. BSC-to-MSC (sometimes called fronthaul) and interswitch BSC-BSC traffic (sometimes called sidehaul) is already carried mostly over optical fiber, even if this may be invisible to a customer subscribing to a protected N × 2

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Mbps (1.5 Mbps) leased-line service. The existence of fiber-optic networks, and SONET/SDH technology in particular, compared to N × 2 Mbps (or 1.5 Mbps) leased-line-service prices, has led many operators to reconsider their core transmission sourcing policies. In Europe, many of the existing N × 2 Mbps leased lines have been converted into SDH leased transmission, providing 63 × 2Mbps capacity. The main reasons for this change are the savings achieved in operational expenses and the fast upgrades in capacity. The only changes involved for mobile network operators have been the purchase of an SDH multiplexer, located at their central transmission sites, to provide 2-Mbps electrical interfaces with the leased SDH stream. If the leasing alternative is chosen for backbone services, for greatest cost-efficiency, leasing should take place at the STM-1 (VC-4) level, not the N × 2Mbps level. With current leasing prices, the SDH alternative may bring the annual operating cost of leased connections down by 70% to 80% for full SDH lines, easily justifying the modest investment needed in SDH terminal multiplexers. With partially filled lines, the savings are naturally less than this. The second step, already directly taken by a number of mobile network operators, has been to acquire their own SDH terminal equipment and lease dark fiber to carry the signal. In areas where leased dark fiber is rare, the solution has been a rollout of owned or partially owned fiber-optic cable infrastructure between key locations in the networkincluding the central switching sites and a number of other strategically important hub sites. 7.4.2

Dedicated Leased Service

Dedicated service is a circuit between fixed end points. The circuit is leased (or owned by an end user) from a common carrier. A dedicated line is obtained from the carrier on a monthly basis for unlimited usage between the individual points to be connected [5]. The service is available 24 hours a day, 7 days a week, 52 weeks a year, for exclusive use by that user. This service may also be referred to as a leased line, private line, or nailed-up circuit, and its primary features include the following: • Point-to-point service; • Exclusive use 24-7; • Always available; • Payment for mileage.


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T1 digital communications were introduced in the early 1960s to reduce the amount of copper cable needed to carry the same number of telephone conversations as analog communications. The term T1 circuit is commonly used to identify a multiplexed 24-channel, 1.544-Mbps digital data circuit providing communications between two facilities or from a local service provider. T1 refers to the transmission of a DS1-formatted signal onto a copper, fiber, or wireless medium for deploying voice, data, or videoconferencing services. The T1 is part of an extensive digital hierarchy that starts with 24 DS0s at 64 Kbps. These individual DS0s are used to provide voice or digital data to support point-topoint or network applications. By combining multiple DS0s, a high-speed interface can be provided to support a synchronous interface to a LAN router or voice PBX. For distances longer than one mile, a repeater is placed every mile to regenerate the signal. E1 is the European equivalent of the American T1. Although both E1 and T1 use 64-Kbps channels, they differ in many aspects; E1 is a point-topoint dedicated 2.048-Mbps communications circuit that carries 32 channels while T1 has 24 channels. Out of these 32 channels, 30 channels transmit voice and data. Unlike T1, E1 always provides clear 64-Kbps channels. Timeslot 16 (TS16) is used for signaling and carrying line supervision (such as whether the telephones are on-hook or off-hook) while timeslot 0 (TS0) is used for synchronization, channel control, and framing control. There are two options for the physical media: • 120-ohm twisted-pair cabling, typically foil shielded. This is called a

balanced interface and uses a DB-15 or 8-pin modular connector. T1 always uses twisted-pair. • 75-ohm coaxial cable. This is called an unbalanced interface because the two conductors do not have equal impedance to ground, and uses a BNC connector. E1 can be either coax or twisted-pair. T3 is a physical transmission medium, which varies from DS3, which is the actual data, video, and voice that is transmitted over the medium. DS3 can be transmitted over microwave radio, fiber optics, or 75-ohm coaxial cable (in-house wiring with distance limitations of 450 ft). The line coding for a T3 line is bipolar three-zero substitution (B3ZS). The framing formats available are M12/M23, M13, or C-bit parity. DS3 can be delivered as channelized or nonchannelized. Channelized DS3 is delivered as 28 individual DS1s and 672 individual DS0s. Each DS1

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may come from a remote location. The telephone companys central office will do the subdivision of the DS3 to each site. The nonchannelized DS3 does not involve DS2 or DS1 multiplexing. This service is delivered as a T3 pipe with the bandwidth being 44.2 Mbps. It is generally used in point-to-point applications (one customer sending data to one remote site). Any subdivision of bandwidth is performed at each customer site rather than the central office. It is expected that in 3G wireless networks not only hub sites, but also some other heavily loaded cell-sites in the wireless network, may require close to this amount of bandwidth (capacity).

7.4.3

xDSL

xDSL is the generic name used to represent a wide variety of digitalsubscriber-line technologies including HDSL, ADSL, and IDSL. HDSL is the most widely available and used xDSL service in North America today. HDSL technology has been developed to allow the transmission of a standard DS1 signal over the outside plant wiring. With HDSL electronics at both the central office and the customers premises, it is possible to extend a full-duplex 1.544 Mbps (voice, data, and video) applications over two pairs of copper wire across private or leased copper facilities to distances of 9,000 ft (26 AWG) or 12,000 ft (24 AWG). All this can be achieved without redesigning the copper loop and without expensive repeaters. The specific rates achievable with DSL depend on factors such as the DSL technology used, the distance between end points, and the wire size. ISDN digital subscriber line (IDSL) is also commonly used for applications that require ISDN BRI signaling in a dedicated mode. IDSL can be extended up to 18,000 ft and can transmit digital data at rates up to 144 Kbps. HDSL technology also provides T1 capability for private networks by utilizing existing copper loop plant originally designed to carry much slower signals (usually POTS). Not only did the copper cable itself have limitations, but also things were done to this cable to make it even more unsuitable for high-speed data transmission. These actions primarily took two forms, like loading and bridge taps. Loading is a procedure where load coils were frequently added to loops longer than 18,000 ft. These load coils were essentially low-pass filters. That is, they passed without attenuation all voice frequencies but effectively blocked frequencies above the voiceband. This is disastrous for data communications, which depend on high frequencies to


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achieve the desired speed of transmission [6]. The xDSL system cannot be supported on a loop with load coils. Bridge tap is any unterminated portion of a loop not in the direct talking path. A bridge tap may be a used cable pair connected at an intermediate point or an extension beyond the customer. For example, a drop wire that provided a second line to a home is left in place even after the second set of CPE is removed. Records of this were not always kept and assigning a particular copper pair to a high-speed data circuit is far from a sure thing. Bridge taps can bring many problems to data transmission and xDSL installations. 7.4.4

Switched Leased Service

Switched service is a circuit for which the end points may vary with each usage. The circuit is provided by a common carrier, which is routed through a switched network, providing circuit switching between public end users. There are two types of switched technology: circuit switched and packet switched. In a switched circuit, a call is established only for as long as needed and then the session is disconnected. Other characteristics of the switched leased services are as follows: • Physical path may vary; • Network is shared; • Customer connects on demand; • Customer pays for usage; • Service provider can oversubscribe capacity. 7.4.4.1 Circuit-Switched Network

A circuit-switched network is one where the user has full access to the connection when needed. The connection is often established by placing a call to another party. The user has exclusive use of the path. Switched 56 is a 56-Kbps digital data service that is purchased from some local exchange carriers or IXCs. It may be deployed on either two or four wires depending on the local carriers capabilities. In both four-wire and two-wire switched-56 applications, 56 Kbps is the standard network operating rate accessed, or access is required on an as demanded basis and 24-

298


hour-a-day connectivity is not required. The maximum distance is 18,000 ft on a 26-gauge wire without a repeater. ISDN was first established by the CCITT (now ITU-T) in 1980 to integrate an all-digital public switched telephone network. This is accomplished through ISDN that offers a full range of enhanced services supporting voice, data, and video applications through standard interfaces over a single twisted pair of copper. ISDN provides a means of integrating these services and modernizing communications networks to provide information movement and management efficiency. The two types of ISDN service are basic-rate ISDN (BRI) and primary-rate ISDN (PRI). BRI can transmit data up to 128 Kbps. PRI (which is transmitted over a T1 line) can transmit data up to 1.536 Mbps. A local directory number (LDN) is a customers seven-digit ISDN phone number. A service profile identifier (SPID) is a unique identifier that is used to represent the service and feature identifiers of a particular ISDN line or service provider. This number generally is 10 or more digits long and includes the LDN. 7.4.4.2 Packet-Switched Network

In a packet-switched network, data is carried in the form of packets. This data would be given an ID on a per-packet basis and sent across the network in the most efficient way. Frame relay is a digital-packet network that provides all the features and benefits of a dedicated DDS or T1 network, without the expense of multiple dedicated circuits. Frame relay is deployed over the same services used to deploy DDS and T1. In a frame relay network, circuits are connected to a packet switch within the network which ensures that packets are routed to the correct location. Frame relay is optimized for use over higher-speed and very-low-errorrate data circuits. To reduce latency in FR networks, the switches do not perform error correction (other than discarding corrupted frames) or flow control (other than setting forward explicit congestion notification and backward explicit congestion notification bits in the frame header). If the user equipment (UE) does not react to those notifications, then the network discards bits when it gets congested. All other functions of error congestion and flow control are left to the equipment on the customers premises. One of the main advantages contributing to growth in the frame-relay industry is the cost effectiveness of this type of service. Frame-relay service is a cost-effective solution for networks with bursty traffic requiring connections to multiple locations and where a certain degree of delay is acceptable. It also allows a voice circuit to share the same virtual connection as a data circuit, again saving money.

Transmission-Network Planning and Design 7.4.5

299

Higher-Speed Switched and Nonswitched Services

Until recently, the primary method by which businesses and institutions obtained nonswitched private line connections between their locations was to use dedicated 1.5-Mbps T1 lines and dedicated 56-Kbps digital private lines leased from telecommunications carriers, including the local exchange carriers. Some larger businesses and institutions have used higher-speed 45Mbps private lines for point-to-point connections. Recently, new types of digital services, including frame relay, SMDS (a public network service designed primarily for LAN-to-LAN interconnection), and ATM, have been introduced by telecommunications carriers, including local exchange carriers. SMDS is a packet-switched connectionless data service that allows the destination to be specified independently for each packet. Frame relay and ATM are currently nonswitched services that utilize predetermined destinations for traffic; switched versions of these services are under development. All these services offer the advantages of improved sharing of facilities (fibers, terminations on electronic equipment) through statistical multiplexing. These new services, particularly ATM, can also support advanced multimedia applications that require high data rates and low delay variability between communicating end points. 7.4.6

Leased Lines Network Build Out

The major steps in the process of any transmission network include determining the total transmission requirements for each site in the core and access network. After creating a transmission demand matrix (i.e., a matrix of all traffic nodes), identifying the transmission capacity requirements (typically in terms of the number of T1/E1 links required) between nodes would be the next logical step. From the topology and demand matrix, it is important to create a transmission link matrix, a matrix of all transmission nodes showing transmission-capacity requirements between these nodes. This matrix is different from the demand matrix in that it reflects the chosen topology; that is, it identifies the actual transmission links (physical connection), whereas the demand matrix identifies only the demand between two nodes (logical connection). The most important thing in designing the transmission network is to understand the existing network and design the best transmission network with rings, linear systems, and digital-access cross connects (DACS) to satisfy the demand matrix, diversity, flexibility, and topology for the wireless network that is being built. If the network is owned, this information will be well understood; however, if the transmission network is leased, information

300


on the providers network may be harder to get. Information on the tariffs, distance, and point-of-interconnect information from the provider of leased capacity, as well as local rules and regulations, will be required. It is important to determine the requirements for owned and leased transmission links (e.g., type, size, quantity) from the transmission link matrix. From these it is possible to specify the bill of requirements (BOR), which should specify the equipment quantities and configurations and leasedline quantities and configurations. These activities take place during the planning and design stage of the leased-lines network build-out. Other activities are listed below: System Planning

1. Identify private and public carriers and facilities. 2. Identify regulatory issues. 3. Establish standards and objectives. 4. Establish nondisclosure agreement (NDA) with carriers. 5. Conduct intercity interconnect analysis. 6. Establish procedures for circuit ordering. 7. Establish trouble reporting and escalation procedure. System Design

1. Analyze search ring information (from RF plan) and create a preliminary network design. 2. Perform primary (candidate) site-engineering analyses. 3. Select carrier(s) in each market. 4. Finalize carrier agreements. System Deployment and Testing

1. 2. 3. 4.

Prepare site package. Order leased facilities. Track carriers progress. Prepare ATP.


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System Optimization

1. 2. 3. 4.

Interface with construction contractor for site engineering; Coordinate leased-circuit construction; Perform construction walk-through; Prepare site completion package and as-built documentation.

It is very important to notice that even when the transmission network is completely leased, it still requires a great deal of engineering effort from the wireless operators side. Transmission engineering cannot be left to the carrier(s) to make decisions on the leased-circuits routing, installation, project management and testing without any supervision from the wireless operator. One of the leading causes of delay in getting T1/E1 lines installed is an unprepared equipment room [2]. It is important to coordinate with carriers and cell-site construction teams the requirements for the equipment room or shelter. This could include ducts or cable ways for the lines, ac and dc power, adequate space, heating and air-conditioning, solid grounding, and a very common requirement by carriers for a plywood board on the wall for terminal blocks, NIUs, punch-down blocks, and so on. 7.4.7

Owned Versus Leased Transmission Networks

Every wireless-network project is different and requires detailed business case typeanalysis of owned verses leased transmission facilities. Microwave networks are usually the way to go if the wireless operator opts to build its own transmission network. Some of the advantages of microwave radios are listed below: • MW system meets superior reliability, higher security, and more

demanding performance and quality standards;

• User has total control over site access and restoration time; • From the performance perspective, carriers provide best effort and

average (not specific) availability for all the customers;

• Easy expansion, or future relocation, or both; • Overhead channel and protection (path, hardware, or both); • MW radio has an operational life long after the leased-line payback

time (24 years).

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Some of the disadvantages of microwave networks include the following: • High capital investment on day one (unless the MW equipment

supplier is financing); • Microwave system design required;

• Frequency coordination and spectrum licensing required; • Microwave network operations and maintenance required.

Let us assume that the 4T1 microwave-system cost, fully engineered, installed, and tested, is $56,000 per hop (see Table 7.1). Not only is the total cost after four years still the same, but it provides for system expansion for an additional three T1s, and thus the cost per T1 is actually only $14,000. On the other hand, a leased T1 has recurring costs of $500 per month (typically, in the United States), and after four years the cost for the leased T1 would grow to $26,000, or almost double that of the microwave T1 circuits. In many countries where leased T1/E1 lines are not readily available or cost a lot more, it would be even easier to prove the advantage of building a microwave network.

7.5 SynchronizationStratum, BITS, and GPS 7.5.1

Introduction and Historical Overview

Digital network connectivity is dependent upon the availability of a reliable synchronization source to provide a timing reference to the network elements. This chapter discusses the hierarchical clocking scheme, and its applications within public and private networks. Networks that use asynchronous Table 7.1 Leased Lines Versus Microwave Comparison MW 4T1 ($)

Leased 1T1 ($)

Up-front charges

56,000 (turnkey)

2,000

Monthly charges

0

After 12 months

56,000

8,000

After 12 months per T1

14,000

8,000

After 48 months

56,000

26,000

After 48 months per T1

14,000

26,000

500


303

digital multiplexers, synchronous multiplexers (SONET), or both, are briefly discussed. Synchronization networks provide timing signals to all synchronization network elements at each node in a digital network. These timing signals are traceable to a highly accurate primary reference source (PRS) clock of Stratum 1 quality [7]. The aim is to ensure that all outgoing transmissions from a digital network node have the same average frequency. Buffer elements are used at important transmission interfaces to absorb differences between the average local frequency and the actual short-term frequency of incoming signals, which may be affected by phase wander and jitter accumulated along the transmission paths. A sync network has two major partsinteroffice and intraoffice. The interoffice network consists of a primary and a secondary DS1 link, carrying timing between offices in a hierarchical relationship. Intraoffice timing distribution is based on the concept of a Building Intergrated Timing Supply (BITS) master clock, providing timing to all other digital equipment in the office. Telephone companies and long-distance carriers started using digital technology in the 1960s to improve service and lower the costs associated with the transmission of analog services. The first digital system installed was known as the T1 transmission system, and it was used primarily for pair gain, or the reduction in the number of pairs used to carry voice traffic. Private companies usually have constructed telephone systems to obtain more efficient communications between locations. Originally, these networks were small copies of the analog transmission and switching systems used by telephone companies. The typical data network was constructed using analog modems in a separate point-to-point network. This scheme often led to two separate networks, one for switched voice and one for data transmission. Unfortunately, the voice quality of these analog private networks proved to be less than desirable due to the low bandwidth and losses of the analog channels. Data transmission over the analog network suffered from error bursts, which resulted in lower speeds. The new digital system provided better voice quality, fewer wires between offices, higher reliability, and improved data transmission. Initially, T1 was an asynchronous system. Each pair of end terminals ran at its own clock rate, and each terminal used its receive timing to demultiplex the incoming signal. The transmit and receive sides were independent of one another. Later, when digital channel units were used, one end terminal was designated as the master and had its own timing reference. The other end terminal was a slave and derived timing for its transmit side from the

304


data being received (looped timing). This arrangement worked as long as the end terminals were no more complex than a channel bank. The growth of T1 (DS1) networks to include higher multiplexing rates and long-distance DS1 connectivity introduced various synchronization problems. Digital switches with DS1 port interfaces exemplified the shortcomings of an asynchronous system. If the two switch clocks were not at the same frequency, the data would slip at a rate dependent on the difference in clock frequencies. A slip is defined as a one-frame (193 bits) shift in time difference between the two signals in question. This time difference is equal to 125 microseconds. Slips were not considered a major impairment to trunks carrying voice circuits. The lost frames and temporary loss of frame synchronization resulted in occasional pops and clicks being heard during the call in progress. With advances in DS1 connectivity, however, these impairments tended to spread throughout the network. To minimize this problem, a hierarchical clock scheme was developed, whose function was to produce a primary reference for distribution to switching centers in order to synchronize the toll switches. Local switching in that era was primarily analog, so that synchronization was not required at the end offices. Later, digital switches and direct digital services or networks (DDS or DDN) became common at the end offices, providing digital services to customers. This meant that timing had to be distributed to local levels.

7.5.2

Strata

As shown in Figure 7.5, the resulting stratum hierarchy evolved into four levels. Stratum 1 is the primary reference. Stratum 2 is used at toll switches, while Stratum 3 is used at local switches. Channel banks and end terminals that use simple crystal oscillators are known as Stratum 4 devices. Recently, SONET networks have created the need for a clock stratum level better than Stratum 3, which is called Stratum 3E. When the Bell System broke up into the local service providers and long-distance carriers, the timing hierarchy became less well defined. Now each local company could no longer take its timing from the long-distance carrier, but had to engineer a system, either a hierarchy or otherwise, to distribute timing to their offices. This made everything more difficult because failures in the transmission systems may cause islands, or areas without a reference to Stratum 1. In such a case, the island is in holdover at whatever stratum level has been provided. Even if a particular network is still traceable to Stratum 1, the traffic of concern may be coming from such an island and

Transmission-Network Planning and Design Stratum 1 Telco (primary reference source)

Toll switching center

Toll switch

Stratum 2 Telco

305

Toll switch

Toll switch to other local switches

Stratum 3 Telco

Stratum 4 Telco or end user

Local switch

Local switch

to other digital terminating locations Channel banks

Local switch

T1 multiplexers

DACS

Digital switches

Figure 7.5 Digital network hierarchy.

will therefore slip at some rate. Having a Stratum 1 source is, in itself, no guarantee of a slip-free network. Voice equipment tends to reacquire frame synchronization quickly, resulting in a pop or click, which is not usually a problem. Data circuits lose some number of bits depending on the data rate being transmitted, and on whether or not FEC is being used. Some multiplex equipment that provides add and drop services interrupt all output trunks while a new source of synchronization is acquired. Such interruptions, if due to circuit noise, may render a network temporarily useless, as the slip causes further slips downstream (error or slip multiplication). A clock system provides a stable frequency source during circuit impairments. The connected equipment will not be affected until the clock holdover drift results in a slip. A stable clock will change a network that experiences problems two or three times a day to one that maintains timing through a major trunk outage. The network will continue to operate without impairment until the outage is repaired, as long as the repair time is comparable to the time of the first frame slip. Since occasional slips will always occur, the best one can do is to minimize their rate of occurrence. Through careful network engineering of the clock systems, near

306


perfect timing may be achieved at a reasonable cost with excellent reliability and maintainability. The ANSI standard titled Synchronization Interface Standards for Digital Networks (ANSI/T1.101-1987) was released in 1987. It was revised in Committee T1X1 for release as ANSI/T1.101-1998. This document defines the stratum levels and minimum performance requirements. The requirements for the stratum levels are shown in Table 7.2, which provides a comparison and summary of the drift and slip rates for the strata clock systems [8]. It is important to note the following in Table 7.2: 11 1. Stratum 2 is typically less than 2.5 × 10 /day; 2. Stratum 2 drift results in more than two weeks to a frame slip; 3. Stratum 3E enhanced; The typical performance of which is less than one slip in 36 hours or 9 × 10 10/day.

To calculate slip rate from drift, one assumes a frequency offset equal to the above drift in 24 hours, which accumulates bit slips until 193 bits have Table 7.2 Clock Strata Requirements

Stratum Accuracy Adjustment Range

Pull-In Range

Stability Time to First Frame Slip

N/A

72 days

1

1 × 10−11

2

1.6 × 10−8 Must be capable of synchronizing 1 × 10−10/day (1) 17 days (2) with clock with accuracy of +/−1.6 × 10−8

3E(3)

34 × 10−7 −6

N/A

4.6 × 10−6

5 × 10−9/day

Must be capable of synchronizing 1 × 10−8/day with clock with accuracy of +/−4.6 × 10−6

7 hours (4)

3E

1.0 × 10

3

4.6 × 10−6 Must be capable of synchronizing 3.7 × 10−7/day with clock with accuracy of +/−4.6 × 10−6

6 minutes (255 in 24 hours)

4E

32 × 10−6

Must be capable of synchronizing Same as with clock with accuracy of accuracy +/−32 × 10−6

Not yet specified

4

32 × 10−6

Must be capable of synchronizing Same as with clock with accuracy of accuracy +/−32 × 10−6

N/A

3.5 hours


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been accumulated. Drift rates for various atomic and crystal oscillators are well known and are not usually linear or not necessarily continually increasing. Stratum 1 is defined as a completely autonomous source of timing that has no other input, other than perhaps a yearly calibration. The usual source of Stratum 1 timing is an atomic standard or reference oscillator. The minimum adjustable range and maximum drift are defined as a fractional fre−11 quency offset of 1 × 10 or less. At this minimum accuracy, a properly calibrated source will provide bit-stream timing that will not slip relative to an absolute or perfect standard more than once every 4 to 5 months. Atomic standards, such as cesium clocks, have far better performance. A Stratum 1 clock is an example of a PRS as defined in ANSI/T1.101. Alternatively, a PRS source can be a clock system employing direct control from coordinated universal time (UTC) frequency and time services, such as GPS navigational systems. The GPS may be used to provide high-accuracy, low-cost timing of Stratum 1 quality. A Stratum 2 clock system tracks an input under normal operating conditions and holds to the last best estimate of the input reference frequency during impaired operating conditions. A Stratum 2 clock system requires a −8 minimum adjustment (tracking) range of 1.6 × 10 . The drift of a Stratum 2 −8 with no input reference is less than 1.6 × 10 in 1 year. The short-term drift −10 of the system is less than 1 × 10 in 24 hours. If one interprets this specifica−10 tion as a drift of 1 × 10 each 24 hours, this amounts to a frame slip rate of approximately one slip in 7 days when the Stratum 2 clock system is in the hold mode. Stratum 3 is defined as a clock system that tracks an input as in Stratum 2, but over a wider range. A Stratum 3 clock system requires a minimum −6 adjustment (tracking) range of 4.6 × 10 . The short-term drift of the system −7 is less than 3.7 × 10 in 24 hours. This amounts to approximately 255 frame slips in 24 hours while the system is holding. Some Stratum 3 clock equipment is not adequate to time SONET network elements. Stratum 3E, which was defined in Bellcore documents, is a new standard created as a result of SONET equipment requirements. Stratum 3E tracks input signals within 7.1 Hz of 1.544 MHz from a Stratum 3 or better source. The drift with no input reference is less than 1 × 10−8 in 24 hours. Stratum 4 is defined as a clock system that tracks an input as in Stratum −5 2 or 3, except that the adjustment and drift range is 3.2 × 10 . Also, a Stratum 4 clock has no holdover capability and, in the absence of a reference, free runs within the adjustment range limits. The time between frame slips can be as little as 4 seconds. Stratum 4E is a proposed new customer premises clock

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standard which allows a holdover characteristic that is not free running. This new level, intended for use by customer-provided equipment in extending their networks, is not yet standardized. 7.5.3

General Timing Planning Rules in Transmission Networks

Inadequate timing may produce problems in any digital network so the objectives have to be set very early in the planning process: • The number of frame slips per day/month/year to be allowed under • • • •

impaired network conditions; The timing sources available from the carrier or local telephone company to help solve the problem; The degree of redundancy required; The locations of the timing systems and the stratum levels desired; The management and maintenance of the system.

One major problem encountered after designing a timing network is evaluating its performance. Standard tests for large switching systems and clock distribution systems require the use of a cesium or rubidium standard. A much simpler method of verifying performance is to use test equipment that displays bit slip, but reference is required in order to use such equipment effectively. A Stratum 1 clock may control Stratum 2, 3E, 3, 4E, or 4 clocks while a Stratum 2 clock may drive Stratum 2, 3E, 3, 4E, or 4 clocks. A Stratum 3E clock may drive Stratum 3E, 3, 4E, or 4 clocks. A Stratum 3 clock may drive Stratum 3, 4E, or 4 clocks. A Stratum 4E or 4 clock is not recommended as a source of timing for any other clock system. Because of the narrower capture and adjustment range of the higherstrata clock systems (2 is higher than 3, and so on), driving a Stratum 2 clock from a Stratum 3E or 3 clock is not recommended. In fact, it will not work under some transmission-impaired conditions. Also, extreme care must be taken in network applications where more than one Stratum 1 source is used to ensure that these sources are accurate and traceable to some other standard. Another standard commonly used to check a Stratum 1 clock sources accuracy is the GPS system. A GPS receiver can also be used directly as a source of Stratum 1 quality (commonly used for the cell-site equipment timing in cdmaOne and CDMA2000 networks).


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Stratum 1 clock administration, operation, and maintenance can be a costly effort. Atomic sources may not have long maintenance-free operating intervals and may experience failures without giving an indication that the source is off frequency. In addition, if a Stratum 1 source of timing is shown to be inaccurate, the network must be able to accept another networks timing until the problem is corrected. Thus, GPS is attractive in order to assure accuracy and minimize cost. Initially, private digital networks were arranged for point-to-point, single trunk. A channel bank at one end provided the source of timing for the distant end terminal bank, known as the slave. The slave terminal derived or extracted timing from the transmitted data. With no intervening transmission equipment to provide a source of timing, this arrangement rarely experienced synchronization problems. When more than one trunk was connected point-to-point, all equipment at one end used the same clock, which was established through a clock-distribution system. The other end was slaved as before, and timing problems were not encountered. The public-switched network gets more complex every day, comprising numerous long-distance carriers, local operating companies, as well as the competitive access providers. In addition, networks are no longer point-topoint, or PMP. They now include add and drop, as well as ring configurations. Designating the location of master and slave terminals becomes quite difficult when point-to-point circuits are used for add and drop services, switched services, or services passing through certain customer multiplexing equipment. However, reliable network timing can be achieved if one location is designated as the master, and Stratum 2 or 3E clock systems are installed at all other sites. While todays point-to-point services and tariffs are typically untimed, telephone companies and long-distance carriers can install equipment to administer, operate, and provide service observation (such as DACS). Without network timing, the timing is that of the customers equipment (leased lines), and may create jitter and wander problems with SONET transmission equipment in the network. The advantage of having the local or long-distance carriers pass the DS1 circuit through a DACS is that the trunk is timed by the carrier. This then provides a source of timing directly traceable to a Stratum 1 reference. In a network with carrier imposed timing on the trunk or trunks, all private network sites should be configured as slaves, and equipped with Stratum 3E clocks to provide timing to the equipment.

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A Stratum 2 clock system should not be used in networks using carrier-derived timing. This arrangement may not work when the transmission system timing is in its holdover condition at the Stratum 3 level at the serving central office or long-distance terminal. In all cases, careful examination of the network and equipment is required. A Stratum 1 clock system may always be used, since it should be identical to the network providers clocks. Only in the case of satellite networks will a Stratum 1 remote clock lead to problems. Clock-system arrangements are based on intended use. A long-distance carrier may elect to provide and administer Stratum 1 clock sources at all major toll switch locations. Another less costly method is to provide one or two Stratum 1 sources, followed by a distribution of the timing information on dedicated DS1 paths to Stratum 2 systems at each switching center. It is possible to use GPS to control rubidium sources. These sources would then distribute timing to Stratum 2 clock systems. In some cases, providing each location with a Stratum 1 GPS system is the most efficient solution. Local telephone companies may choose one DS1 path from one incoming group as a primary reference, and another DS1 path from another incoming group as a secondary reference. These two references feed a Stratum 2 system at the toll level and a Stratum 3 system at the local-switching system level. Such a system provides reliable timing at the local level. As an alternative, many local telephone companies, because of problems they have experienced while using someone elses timing, are beginning to consider the use of Stratum 1 systems of their own. Another method of distribution timing is via older analog microwave links. Many of these furnish a source of synchronization, which is transmitted as subcarriers for synchronizing the single-side band (SSB) and FDM equipment. These pilots may be converted to 1.544-Mbps DS1 framed 1s signals for use in distribution and timing of newer digital equipment, which may be using supergroups on the existing microwave to provide digital services. A private network may require a Stratum 3E clock at each location to provide a local source of timing. The Stratum 3 clocks are referenced to the telephone company by bridging one of the incoming DS1 signals to extract timing information. 7.5.4

Interoffice Distribution

Timing information is distributed to offices through a hierarchical series of levels, starting with a PRS. Clocks are grouped into stratum levels, based on


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their ability to maintain accurate timing if their reference fails (holdover mode). Stratum 1 is the most accurate; it is by definition a stand-alone PRS, which does not have an external reference. Stratum 4 clocks are the least accurate; they have no holdover requirements and are typically found in DS1 terminal equipment, such as D4 channel banks. A given clock must be able to track a reference from a free-running clock of the same or higher stratum level. BITS clocks in central offices are Stratum 2, Stratum 3, or Stratum 3E, with Stratum 2 being used in the larger offices where holdover drift can affect hundreds or thousands of outgoing trunks. A sync network is designed so that a clock always receives timing from a clock of equal or higher stratum level. This ensures that if an upstream clock enters a hold mode, the downstream clocks will be able to track it. Stratum 2, 3E (3E Enhanced), and 3 clocks are provided with primary and secondary timing reference inputs, with automatic switching between the two if either one fails. Timing distribution is typically done with traffic-carrying DS1 signals over paths selected for best availability. However, using traffic-carrying DS1s for this purpose is discouraged when using SONET transmission equipment. The following are the major rules used in designing a synchronization network: • An office BITS clock can receive its reference only from another

office or offices of the same or higher stratum level. A higher stratum is preferred, provided reliable diverse DS1 paths from that office exist, but is not necessary.

• For a given BITS clock, a DS1 facility with the highest availability

should be selected for the primary reference input from an upstream clock. A facility with the next highest availability should be selected for the secondary reference, preferably on a diverse route from a different upstream clock. Its historical failure record, installation or rearrangement activity, facility length and type, protection switching, and the number of repeaters or multiplexers on the path determine the availability of a path. There may be specific equipment types that must be avoided.

• No timing loops are allowed in the sync network for any combina-

tion of primary and secondary facilities. The potential for loops exists when either primary or secondary reference signals are passed between clocks of the same stratum. Loops are avoided within an

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office by distributing timing from the BITS clock in a star network. SONET rings, at present, cannot operate with redundant timing, as they are by definition a loop. • There are no fixed rules for the number of BITS clocks that can

be used in cascade. Any clock can track any other clock of equal or higher stratum level, and it will filter out most jitter and shortterm wander introduced along the timing reference path. However, long-term wander (over periods of hours) and possible phase transients will accumulate along the cascaded paths from the PRS to a given clock. Furthermore, the failure of a reference path will affect all downstream clocks. For these reasons, the number of cascaded clocks should be minimized as far as possible and be consistent with the use of whatever highly reliable reference paths are available in the network. As an example, it would be preferable to time a Stratum 3 clock from another Stratum 3 clock upstream, which is in turn timed from a Stratum 2 clock through reliable facilities, rather than use a direct path of questionable reliability to that Stratum 2 clock. The objective should be to maximize the overall availability.

• According to [9] it is recommended that no more than two Stratum

3 or 3E offices be timed in tandem, and where possible, the timing distribution should be limited to only one Stratum 3 office. This reference applies to large telecommunications synchronization networks.

7.5.5

Intraoffice Distribution

The BITS clock system is the preferred method of distributing timing within an office (Figure 7.6). Redundant hardware and automatic switching between primary and secondary reference inputs provide a high degree of availability. The BITS clock supplies timing directly to all digital equipment in the office requiring synchronization, usually by means of DS1 framed 1s, or a 64-Kbps composite clock. Many network elements have primary and secondary timing ports; the signals to these should be taken from different output cards of the BITS system. The BITS clock may also provide primary and secondary timing signals, through interoffice sync network paths specially selected for high availability to other BITS clocks in downstream offices.


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GPS

DS1 BITS

64 Kbps composite clock

DS1

DS1

Digital switches

DCS

DS1 SONET

Channel banks

DDS

Traffic Timing Figure 7.6 BITS clock system.

In the wireless network, the BITS clock is placed at the BSC/MSC location (switch office) and receives the reference clock from the PSTN (T1/E1 line) or GPS receiver installed at the premise. 7.5.6

SONET Network Timing

SONET networks are designed to operate in a maintenance mode when not timed. In this mode, the jitter may be larger than that of a properly timed network. It has been noted by many that for voice traffic, and even for digital traffic between switches and digital cross connects, no impairments are noticed, even when the SONET system is untimed. However, it is recommended that all SONET networks be timed to minimize phase noise (jitter and wander). With SONET, the multiplex equipment itself must be timed. If the timing of the dropped DS1s is different from the multiplex timing, there will be unsatisfactory jitter and wander performance, which may or may not affect the connected equipment. By timing all network sources of DS1 signals from the same reference and making sure that the reference chosen is the same as the SONET

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reference, problems can be minimized. Some of the more important considerations for synchronizing SONET network elements are as follows: • External timing for SONET NEs from a BITS clock is the preferred

mode, where BITS is available. • When BITS timing is not available, other SONET timing modes (line, through, or loop timing) should be chosen in such a way as to avoid timing loops and to minimize the lengths of timing paths. Rings may only be fed in one direction with a single reference. • If a string of more than one SONET ADMs exists between BITS timed offices, no secondary references should be used. This also applies to ring configurations; timing should be passed in one direction only, between line-timed ADMs. • Caution is advised when using DS1s carried on SONET for synchronization distribution; these DS1s are subject to phase transients (pointer adjustments) that may not meet short-term stability requirements. It may be necessary to pass timing this way, because customers may have no other choice of a timing source. A typical private network timing is shown in Figure 7.7. It is necessary to know if the carrier can supply a source of Stratum 1 traceable timing. For purposes of this example, the carrier identifies two T1 trunks at locations A and B, which will be timed by the network and which can be traced to the carriers Stratum 1 source. At location A, a synchronization timing system connects to one of the incoming trunks timed by the network. The outputs of the synchronization system drive the clock inputs of the multiplex equipment. Another synchronization timing system at location B is connected in the same manner as at location A. 7.5.7

Synchronization: Issues in PCS Networks

New digital network technologiesCDMA, TDMA, and GSMhave stringent requirements for network synchronization and timing accuracy. In PCS and cellular systems, these requirements appear both in the wireless and in the wireline parts of the network, as well as at the points of interconnection with other public networks required to complete a call. The over-the-air portion of every call requires accurate timing and the most obvious example is found in the CDMA world. To achieve the timing


MUX

315

Local Telco CO

Location A Timing Private network

MUX Location B

Longdistance carrier

Local Telco CO

Timing

Figure 7.7 Private network timing.

accuracy required for CDMAs soft handoffs, the IS-95 standard calls for synchronization using the GPS. The precise timing allows more calls to be put through the network, because a very narrow window can be used for timing the handoff between cells, thus freeing network facilities to handle other calls. Problems at the wireless-to-wireline interconnect could appear when the digital information being transmitted leaves the airlink to pass through a switching office. Unless the call is to another subscriber from the same network, it normally enters the PSTN and experience has shown that problems can originate here in several ways. If the wireless network uses its own clock, and if the timing has variations, a slip can occur at the interface between the wireless and the wired networks, and the call can be lost. Wireless networks often take their timing from the PSTN, however, and this can be the source for a second kind of problem. If the timing of the PSTN is inaccurate, it will affect the performance of the wireless network. Moreover, the problem may not show up at the interface to the PSTN from which the timing was taken,

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but at a later point, perhaps at the interface to another operators network. A problem such as this can be subtle and hard to identify. Another risk of taking timing from a PSTN involves holdover. If the PSTN has a fiber network, signals must pass through numerous repeaters and regenerators. This constant regeneration of the signals can cause difficulties in some clocks, causing them to enter a state of holdover. For those wireless networks that use the PSTN timing to generate its radio frequencies, the effect can be a drifting of their frequencies into adjacent channels, causing channel interference and problems with regulatory agencies. The ideal solution for the wireless-network operator is to implement a highly accurate timing system based on synchronization with the GPS. This solution eliminates the uncertainties and holdover problems associated with reliance on timing from the PSTN, and it offers independence and control of the quality of the network synchronization. It has to be noted that this solution, GPS timing, is preferred in North America, but could be difficult or impossible to implement in some other parts of the world because of strategic, political, or other reasons. 7.5.8

Cell-Site Timing in Wireless Networks

As wireless communications systems move from analog to digital operation, their timing subsystems become both more complex and more critical. Operational timing in each cell of a wireless network is based on the GPS location and time standards. Timing references are downlinked from a satellite by the GPS receiver at an RBS. The clocks are then used by the RBS for timing mobile station calls and T1/E1 channel support. CDMA (cdmaOne and CDMA2000) is dependent on precision timing for its performance advantages over current analog systems. Accurate timing enables advanced features like soft handoffs, which allow a CDMA cellular user to move freely throughout a service area and be transferred from one base station to another without interruption in service or deterioration of signal quality. In the final stages of the encoding of the radio link from the base station to the mobile, CDMA adds a special pseudorandom code to the signal that repeats itself after a finite amount of time. Base stations in the system distinguish themselves from each other by transmitting different portions of the code at a given time. In other words, the base stations transmit time offset versions of the same pseudorandom code. In order to assure that the time offsets used remain unique from each other, CDMA stations must remain synchronized to a common time reference. All sector air interface transmissions are referenced to a common systemwide timing reference that uses the GPS time,


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which is traceable to and synchronous with UTC (in the United States, the offical UTC is kept by the U.S. Naval Observatory). GPS and UTC differ by an integer number of seconds, specifically the number of leap-second corrections added to UTC since January 6, 1980. The start of CDMA system time is January 6, 1980, 00:00:00 UTC, which coincides with the start of GPS time. CDMA system time keeps track of leap-second corrections to UTC, but does not use these corrections for physical adjustments to the CDMA system time clocks. Customarily, cellular base stations receive their timing signals from the worldwide GPS, an array of satellites that beam timing signals accurate to within 300 ns to receiving stations located around the globe. Should the link fail, a backup system must be in place to maintain timing accuracy; if there is no backup, the cell goes down and communication is lost, which is an unacceptable condition. The GPS time standard is downlinked from the satellite as long as it is in range of the GPS receiver at the RBS. There are, however, intervals of time when a satellite is not in range (overhead), and no GPS time standard is available. This period without any GPS time update is called the GPS holdover time.

7.6 Transmission-Network Optimization 7.6.1

Daisy Chaining and Traffic Grooming

By definition, the term daisy chain in telecommunications refers to a wiring method where each telephone jack in a building is wired in series from the previous jack. Daisy chain is not the preferred wiring method, since a break in the wiring would disable all jacks downstream from the break. The same analogy goes for the wireless network, which has a large number of cell sites; some of the sites will be connected directly to the switch location (to BSC) and some will be connected through other cell sites (daisy chained). A problem on a certain T1/E1 circuit will cause problems not only on that immediately affected cell site, but on all of the cell sites downstream from it. It is important to note that the transmission-network planners have to be aware of the risk when the decision is made to use daisy chaining in order to save on transmission facilities. Grooming, or segregation as it is sometimes called, is done when E1/T1 lines contain a mixture of different types of service circuits. The (X) circuits are segregated to one E1/T1 line, while the other circuits (O) are

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cross-connected to another E1/T1 line (Figure 7.8). DACS or mini-DACS equipment is usually used to perform this kind of functionality. 7.6.2

Voice Compression

7.6.2.1 Voice Coding Overview

One of the principal technologies enabling explosive growth in the field of digital communication is voice coding, in which an analog speech signal from a microphone is digitally sampled via an A-to-D converter and then efficiently compressed into a digital bit stream for transmission or storage. A corresponding voice decoder receives this bit stream and decompresses it back into a series of digital speech samples suitable for playback through a D-to-A converter and a loudspeaker. Voice coders can take a number of different forms each of which involves tradeoffs in terms of bit rate (i.e., degree of compression), complexity, and voice quality, as well as robustness. Voice coders are normally divided into two broad classes referred to as waveform coders and model-based speech coders. In a waveform coder the objective is to reproduce at the decoder the original speech samples on a sample-by-sample basis. A simple PCM waveform coder accomplishes this by quantizing each speech sample to one of a fixed number of levels. Assuming 8 bits (256 levels) are used per sample and the signal is sampled at 8 kHz, the overall data rate is 64 Kbps. More involved adaptive differential pulse code modulation (ADPCM) waveform coders apply prediction with differential quantization to reduce the data rate to 24 to 32 Kbps. In any case, the process of quantizing the speech samples adds quantization noise, which is usually audible as distortion in the decoded signal. The primary advantage of waveform coders is that if the data rate is kept sufficiently high the amount of DACS or mini-DACS XOXOXOXOX XXXXXXXXX XOXOXOXOX

XOXOXOXOX

Figure 7.8 Digital access cross connect.

OOOOOOOOO


319

distortion can be kept reasonably low. Hence waveform coders have historically been prevalent at rates over 16 Kbps. Another advantage of traditional waveform coders is that their complexity is typically modest, allowing them to be more readily implemented on early DSP devices. In contrast to waveform coders, model-based speech coders or vocoders use a parametric model to approximate short (1040 ms) segments of speech. For each segment, a set of model parameters are estimated and converted into a bit stream. The decoder converts this bit stream back into model parameters and then uses these parameters to synthesize a speech signal that is perceptually close to the original. In this approach no attempt is made to recreate the original speech samples, instead only the perceptual content as approximated by the model parameters is maintained. The use of a parametric model allows vocoders to operate at lower data rates (under 8 Kbps) than waveform coders; however, they require an accurate speech model to obtain good performance. Early vocoders such as the channel vocoder, homomorphic vocoder, and Linear Predictive Coding (LPC) vocoder all demonstrated the ability to produce intelligible speech at low to medium data rates. A good example is the 2,400-bps LPC-10 vocoder used as a U.S. government standard for secure (i.e., encrypted) telephony, but resulting in poor voice quality. Over the last decade, continued work has improved the performance of voice coders. The challenge with waveform coders has been to try to maintain adequate voice quality while lowering the bit rate. Generally this effort has focused on techniques commonly referred to as code-excited linear prediction (CELP), in which vector quantization is combined with adaptive linear prediction. This approach borrows several concepts from model-based coders in that an all-pole model is used to approximate the speech spectrum and a long-term predictor is used to represent the pitch (i.e., local periodicity) of the speech signal. However, in a CELP coder, an error signal or residual is computed to compensate for the shortcomings of the linear predictive model. This residual is quantized using vector quantization which typically requires a search for the best vector from a large codebook of candidates. While this approach has made headway, yielding good algorithms at 8 Kbps, the performance generally degrades rapidly at lower rates. In addition, the vector search employed in CELP coders (and its many variants) has significantly increased the complexity of these algorithms. In contrast to waveform coders, the challenge in model-based coders has been to improve the speech model to allow higher voice quality at low bit rates. One approach that has made a significant contribution is the

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multiband excitation (MBE) speech model. In this model, which is fundamentally different from the linear-predictive methods found in traditional vocoders as well as CELP, speech is modeled with a fundamental frequency, a set of spectral coefficients, and a set of frequency-dependent voicing decisions. The inclusion of multiband voicing information plus new algorithms to analyze and synthesize speech has resulted in new MBE-based vocoders that can provide very high-quality speech at rates between 2 and 5 Kbps. Achievement of high-quality speech at such low bit rates is facilitated by the lack of any residual signal in the MBE-based approach. Instead, increased emphasis is placed on high-fidelity estimation and quantization of the model parameters so that voice quality can be maintained without the need for such an error signal. 7.6.2.2 Voice Compression in Wireless Networks

Development of advanced model-based coders has had a significant impact in a number of fields, including wireless communications, voice storage, and digital telephony, which require high quality combined with efficient bandwidth utilization. In a wireless system, this added coding efficiency is typically used to increase the number of users which can be supported across a fixed bandwidth. The rapid growth in personal communications has led to a critical need for increased capacity in voice communication networks. This need typically takes the form of a broad requirement to handle more simultaneous voice connections or messages in some constrained bandwidth or bit rate, without degrading power, weight, and range or voice quality. The typical solution to this requirement is convert to a digital communication network and to then employ voice compression to reduce the amount of digital data (i.e., bits) which must be handled. One of the applications where increased capacity has been required is in wireless networks. In this case the large increase in subscribers has overstretched the capabilities of the original analog cellular systems deployed in the United States and around the world. The combination of voice compression and digital communications is key component in most modern mobile voice communication systems and in many nonmobile systems as well and the basic reasons are cost and performance. In a digital system using voice compression, the number of users, which can be supported in some available bandwidth, is greatly increased relative to analog or uncompressed digital systems. The result is that the cost of developing and operating the system can be spread over many more users which typically leads to lower charges per user.


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In addition, lowering the data rate in a wireless application generally leads to smaller, less expensive mobile equipment which uses less power and, with additional FEC, is more robust to bit errors found in a typical mobile environment. Transmission engineers have been using voice compression as a tool to increase capacity of the transmission (backhaul) networks for years. Voice circuits between cell sites and base station controllers, quite often carried over the leased T1/E1 facilities, can benefit from the addition of voice-compression hardware, which is usually built into the RBS and BSC utilizing the manufacturers proprietary compression scheme. In 3G wireless networks, IS-127 enhanced variable rate codec (EVRC) vocoder allows increased network capacity or cell site coverage without sacrificing voice quality. EVRC offers 13-Kbps voice quality at an 8-Kbps rate per IS-127. With EVRC, the vocoder rates are decreased during low speech activity to preserve bandwidth, and increased during high speech activity. The MOS for EVRC is 4.01, and as reference, the 64-Kbps PCM has a MOS of 4.27. EVRC has an average rate of 5.16-Kbps. Dynamic allocation of vocoders is supported in the hardware-software architecture of the BSC. The new selective mode vocoder (SMV) will also be supported in the future as the standards continue to mature. SMV promises EVRC-like voice quality with a capacity increase of ∼27%. Higher-capacity gains are also possible with SMV (up to ∼49%), but at the expense of slightly lower voice quality when compared with EVRC. It must be noted, though, that additional external voice compression can be used on the transmission network successfully only in case of channelized (individual) DS0s within the T1/E1 circuit that does not have compression already applied to it.

7.6.3

Signal Propagation Delay

Signal propagation delay, or latency, describes the delay of a transmission from the time it enters the network until the time it leaves the network. Low latency means short delays while high latency means long delays. Latency may occur in the handset or in the network. Latency that occurs in the handset or between the handset and the base station is called access latency. Latency that occurs from the base station through the network is called network latency. Low latency is essential for real-time transmissions. These include live voice conversations (but not voice mail messages, which are time insensitive) and live twoway video (but not entertainment video clips, which also are time insensitive).

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Latency is not a phenomenon only of mobile networks. It is an outcome of all the networks, terminals, and devices through which transmissions may pass and the bottlenecks (and, therefore, delays) they may encounter. At home, users of broadband Internet connections experience latency as delays in downloading Web sites during peak traffic hours (often in early evenings and during inclement weather). Such delays are due to overloading bandwidth at the network periphery. More important are delays due to overloading bandwidth at the network core. A user in New York may download a Web site that is hosted in San Diego. Depending on traffic loading and transmissions costs at the moment, the download may travel from San Diego to Los Angeles to Denver to Houston to Chicago and finally to New York. It may use fiber networks owned by a number of different carriers. At each switching point, and in particular at the juncture of each network, it will encounter delay. Each of these delays increases the latency. This means that even if a mobile network is configured to provide low latency, the operator cannot guarantee low latency for end users who use their mobile devices to access other networks or who use their terminals in a noisy, and thereby latency inducing, RF environment. Also, differential delay between E1/T1 timeslots that may be routed over different paths (for fractional services) can cause protocol time-outs, retransmissions, and disruptions in video circuits as well as inhibit voice transmissions. Many types of transmission equipment such as multiplexers, packet assemblers, microwave radios, and DACSs add small amounts of delay due to their internal buffering while satellite links add significant delay to a signal. That is the reason satellite links are never used for backhaul in typical wireless networks. Testing signal delay is done during an out-of-service E1/T1 facilities bit error test. By establishing a loopback of the test signal at the desired point in the network, the test set can be used to measure the round-trip (RT) delay. It is also useful for fractional (N × 64 Kbps) E1/T1 services to measure the differential delay between the individual 64-Kbps timeslots, particularly when one application uses several contiguous or noncontiguous timeslots. For example, CDMA networks are very sensitive to delays and some vendors recommend that backhaul delay between the cell site and the BSC be as follows: Cell-site1 to BSC < 12 ms Cell-site 2 to BSC < 12 ms ∆ Delay = (Cell site 1 to BSC) − (Cell site 2 to BSC) < 10 ms


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While designing wireless networks, transmission engineers have to be careful since CDMA networks, as well as other packetized data-transmission systems, do not tolerate noncontiguous timeslots. 7.6.4

Example of Optimized Network Design

A plan is in place to deploy multiple cell sites to support six rural service areas (RSAs) in city A. Each cell site is initially 8 to 12 channels over a T1 facility, served by a single MSC. All calls to and from landline use the switched network. To cover the RSAs without incurring significant long-distance phone charges, a multiple T1 private network is planned to be extended from the MSC to LEC COs in each of the six RSAspromising to result in a very costly overlay network. One solution is to use mini-DACS at the cell sites to groom and fill the cellular service with the switched-network backhaul access, making an overlay network unnecessary (Figure 7.9). The equipment consists of T1 multiplexers at each cell site with dual 4-wire channel cards and fractional T1 channel cards, as needed, to connect locally to the T1 and to the LEC CO. The MUX installed at the cell site has the flexibility of multiplexing channels onto a network T1 to the MSC location. These channels can be

T1 microwave

LEC central office Radio cabinet

Fractional T1

T1 Telco

MSC

T1

Te lc

Radio cabinet

LEC central office

oo

rF

ibe

r

Radio cabinet

LEC central office

Fractional T1

Long haul T1 network Local T1 access

Figure 7.9 Groom and fill network design example.

Fractional T1

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allocated to voice and control data to support the cellular service, or can be allocated as a partially filled local-access T1 to the nearest CO to support back-haul phone calls locally. This partially filled T1 is often referred to as a fractional T1. By allocating the network channels to either the cellular voice and data or the fractional T1 channel allocation, the digital bandwidth can be shared between the various services. This configuration is called groom and fill. At the MSC location, the cellular and switched voice channels are connected using the mini-DACS. The mini-DACS usually can terminate up to 16 T1s in a small chassis, reducing space, power, and overall cost of the solution. The MSC has voice tandem trunks that connect on a T1. These trunks are connected to the mini-DACS together with the cellular T1s. The T1s to each of the cell sites also connect to the mini-DACS. The mini-DACS is then programmed to allow any DS0 channel originating from the MSC to connect to any DS0 of the long-haul T1 that is connected to the cell site. This grooms the network at the MSC, while the mini-DACS redistributes the DS0 channels at the cell site. An RSA cell site can, for example, be set up to support 12 DS0s for cellular and 12 DS0s for backhaul switched-voice access. At the MSC, the mini-DACS programmed on a DS0 basis fills the connecting long-haul T1 with channels connected to various T1 trunks of the MSC. The cellular base station trunks use 11 DS0s for cellular voice and one DS0 for cellular data control to the base station. The remaining 12 DS0 channels are crossconnected to an MSC voice trunk T1. At the cell site, the mini-DACS is programmed to connect the 11 cellular voice channels to six dual-channel fourwire VF channel cards, each connected to cellular transceivers. One channel is available for cellular data at 56/64 Kbps, which connects to a BSC system. The remaining 12 channels are programmed to provide T1 access to the local CO over a fractional T1 facility.

7.7 Transmission Network: Design Examples 7.7.1

Small PDH Microwave Transmission Network

Figure 7.10 shows an example of a small wireless network that has a total of four cell sites, out of which one is colocated with the switch (BSC). All the sites contained RBS, and they were connected to the switch by means of microwave radios. Connection to the local telephone company (PSTN) was achieved over a new short run of fiber-optic cable laid just for this purpose.

Transmission-Network Planning and Design 4-ft dish 6-ft dish 2 ODUs 2 coax. 1 waveguide 2 racks

Site 4

325

Site 3 7 GHz MW system 16xE1 1+1

15 GHz MW system 8xE1 1+1

4-ft dish 2 ODUs 2 coax. 1 racks

Switch site Site 2 4-ft dish 2 ODUs 2 coax. 1 racks

15 GHz MW system 8xE1 1+1

Site 1 Fiber-optic cable cca 800m

PSTN

Figure 7.10 PDH microwave network example.

If we assume a requirement of 1E1 per cell site (RBS), and calculate required capacity over every individual link (1E1 between Sites 3 and 4, 2E1 between Sites 3 and 1, and 1E1 between Sites 2 and 1), it is obvious that the system was designed for increased capacity and expansion in the future. The cost difference between installing 4E1 and 8E1 microwave systems is negligible in comparison with adding additional capacity or deploying a new microwave system later on. It makes sense to install an 8E1 microwave system from day one if there is even a remote chance that the capacity might be increased sometime in the future. Also, Site 3 is planned to be used as a hub site for any future expansion and since it is located on the top of the hill overlooking the city, it has the LOS with almost every potential new site location in the city. Thus, a 16E1 microwave system connecting Site 3 and the BSC location is not overkill. 7.7.2

Complex Transmission Network

Shown in Figure 7.11 is an example of the large network (over 300 cell sites) and as such, only a small part of the network is shown. This transmission

326


End link New microwave SDH ring

Spur hub

End link

Existing fiber-optic ring

SDH ring node

New microwave SDH ring

BSC MSC Fiber-optic POP Cell site MW SDH site (ideally colocated with the cell site) Microwave repeater

Figure 7.11 Transmission system design example for the complex network.

network is designed to meet service demands, but with the most economical routing in mind. It was designed and implemented in the one of the biggest cities in the world using multiple circuit providers, different technologies, and a combination of leased copper and fiber facilities and the new microwave systems. Survivability and reliability of the network are achieved by means of transmission loops (ring configuration) or a combination of star and ring configurations that provide the following:


327

• Increased system reliability due to less equipment; • Protection against catastrophic site failure; • No loss of E1/T1 traffic due to a single failureprovides alternate

routing of E1/T1 traffic automatically;

• Alternate routing of orderwire and alarm system automatically; • Easy and rapid maintenance, since individual E1/T1 failures are

switched and can be tested without taking the E1/T1 out of service;

• Isolated paths for maintenance and fault location purposes.

Predesigned protection, which refers to the fact that recovery from network failures is based on preplanned schemes and usually relies on resources dedicated to protection purposes (fibers, wavelengths, switches, etc.). A better and more efficient system is dynamic restoration, in which the discovery of spare capacity is done dynamically, as needed [10]. In SONET networks, intelligent DACS and controllers are used as the main components to realize dynamic restoration. Both protection schemes have their advantages and disadvantages and have to be chosen based on network requirements and topology. It is important to notice that even leased-lines (facilities) networks require careful planning and routing, the philosophy of the entire network (present requirements and future expansion) design cannot be left to the discretion of the telephone company providing the T1/E1 leased lines. This is especially important when multiple service providers and multiple equipment vendors are used for the project. Study of the existing transmission facilities and laying the RF network over it determine the number and type of the required and newly constructed facilities. As usual, in the network of this size, site acquisition is the most important bottleneck to take into consideration when trying to schedule the build out.

7.8 Overview of RNC Dimensioning in the 3G Wireless Network The purpose of this section is to describe briefly how to dimension the RNCs in a UMTS network for WCDMA (FDD mode). The intention here is to describe the dimensioning process without actually going into the details of the RNC hardware, which is vendor specific. This is a very generic, highlevel design that only covers some very basic assumptions.

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7.8.1


Traffic Classes in the UTRAN Network

According to the UMTS Terrestrial Radio Access Network (UTRAN) system description, the ALL5 will be used for carrying signaling traffic and O&M traffic, and the AAL2 will be used to carry user traffic over Iub (packet and voice data traffic). The signaling traffic can be further divided into control signaling traffic and O&M signaling traffic. Since the voice data and the packet data may travel through different routes in the network (mainly at Iu and Iur interfaces), for dimensioning purposes we should further divide the AAL2 traffic into voice-data traffic and packet-data traffic. From the dimensioning point of view, therefore, the traffic in a UTRAN network can be classified as follows: • Packet-data traffic; • Voice-data traffic; • Control and signaling traffic; • O&M (network management) traffic.

We can further divide the packet data into two categories, one being real-time packet data and the other non-real-time packet data. Real-time traffic is usually generated by some kind of data application, such as a multimedia conference application, real audio player, and so on. Non-real-time packet-data traffic is usually generated by such applications as Web browsers, e-mail, and FTP. Burstyness is the main characteristic of this kind of traffic. Voice data will probably have peaks of voice-data traffic and packetdata traffic at different times. The UTRAN network nodes ATM traffic multiplexing ability can fully utilize this peak time difference (multiplexing gain). The amount of O&M traffic in a UTRAN network is dependent on how the O&M is implemented in the system, and what functionality it will cover. Usually, an O&M system (network management system) for a packet network just performs the following functions: • Set: initializes a specific management function in a specific point; • Get: gets some wanted statistical data to further analyze the perform-

ance of the specific network element. This element may be a piece of software, an interface or node, link, and so on.

• Trap: automatically reports (alarm) when critical events happen.


329

When dimensioning a network, traffic generated by the above O&M functions usually does not take too much bandwidth and can be performed by the system administrator at nonpeak hours. The network operator or administrator usually chooses nonpeak hours to do the corresponding operations to analyze the network performance. Counting might be a very important function for the O&M subsystem. The use of ATM in UTRAN networks provides a way to allow the O&M system to perform byte-based counting of user traffic. So, another function might be to count how many bytes, kilobytes, or megabytes the user has transferred over the network, so they can charge the user. All the control and signaling traffic will be transferred over a logical network. Permanent configured signaling VCs between all the RNCs and RBSs will form this logic network. From a transmission-network dimensioning point of view, the control and signaling of traffic volume are important. How the control and signaling traffic will be treated when they travel through a network is irrelevant for this discussion. Since the control and signaling traffic usually has higher priority than the user traffic, network nodes usually implement some kind of QoS mechanism to differentiate them. Since a system usually does similar housekeeping for every user, the volume of the signaling traffic from an RBS to an RNC should be roughly proportional to the number of subscribers in the corresponding cell. 7.8.2

Description of RNC Interfaces

The RNC has four interfaces toward different parts of the UMTS network. They are shown in Figure 7.12 and explained as follows: • The Iub interface is the interface between the RBS and RNC. The traf-

fic on this interface is user data, signaling, and network management.

• The Iur interface is the interface between different RNCs. The main

part of the traffic on Iur is soft handoff traffic. In addition, signaling and management are carried over the Iur interface.

• The Iu interface is the interconnection point between the RNS and

the CN. The main traffic on the Iu interface is user data, but signaling is carried over the Iu interface as well. The CN often is divided into a CS domain and a packet-switched (PS) domain. For this reason, Iu often is divided into a circuit-switched part (Iuc, Iu-CS) and a packet-switched part (Iup, Iu-PS).

330

Transmission Systems Design Handbook for Wireless Networks Core network

lu

lur RNC

Mur

lub Mub

RBS1

RBS2

Mobile terminals

NMS called Ranos

RBS3

Physical connection Logical connection

Figure 7.12 UTRAN network interfaces.

• Mur interface is the interface between RNCs and Radio Access Net-

work Operations System (RANOS) and handles the O&M traffic.

• Mub interface is the interface between base stations and RANOS and

handles the O&M traffic.

• Node B is a logical node responsible for radio transmission and

reception in one or more cells to and from the UE. It terminates the Iub interface toward the RNC.

• Soft handoff is the state where a UE is connected to two or more

cells in different base stations.


331

• The radio network subsystem (RNS) typically consists of an RNC

and the RBSs connected to this RNC.

• Serving RNS is a role an RNS can take with respect to a specific con-

nection between a UE and UTRAN. There is one serving RNS for each UE that has a connection to UTRAN. The RNS is in charge of the radio connection between a UE and the UTRAN. The serving RNS terminates the Iu for this UE.

• Drift RNS is a role an RNS can take with respect to a specific con-

nection between a UE and UTRAN. An RNS that supports the serving RNS with radio resources when the connection between the UTRAN, and the UE needs to use cell(s) controlled by this RNS is referred to as a drift RNS.

• Controlling RNC is a role an RNC can take with respect to a spe-

cific set of RBSs. There is only one controlling RNC for any RBS. The controlling RNC has the overall control of the logical resources of its RBSs.

• Inter-RNC soft handoff is soft handoff with more than one RNS

involved. An inter-RNC soft handoff can occur if a user travels from an area controlled by one RNC to an area controlled by another RNC (from one RNS to another).

The RNC locations are dependent on many factors, such as transmission cost, existing infrastructure, and location of CN nodes. The RNCs can be located or distributed near the base stations or centralized in a few major switch locations. Whether centralized or distributed, RNC topology must be determined from case to case. The reason for distributed RNC topology is to reduce the need for transmission resources. It is, however, not true that distributed RNC topology automatically leads to reduced need of transmission resources and the following should be considered: • The sum of Iur and Iu traffic is often higher than Iub traffic. Especially

in urban areas, the traffic is high and the number of base stations supported by each RNC low. A low number of base stations per RNC leads to a large proportion of the UE involved in RNC handoff.

• Statistical multiplexing can be achieved in the RNC, but there are

alternatives that might be more cost-effective. For example, the RBS

332


is ATM-based and it is possible to get efficient statistical multiplexing using the base stations as hubs. • The RNC capacity is used more efficiently with centralized RNC

resources than if the resources are distributed around many locations.

• The location of MGWs is of importance for the location of RNCs. • In the future, most of the traffic most likely will be IP-based. If some

of the functionality contained in the MGWs is moved into the RNCs, it will be possible to connect the RNC directly to ISPs. This would reduce the traffic between the CN and the UTRAN considerably.

• Many RNC locations might give extra costs for buildings and

personnel.

7.8.3

User Traffic Modeling

So far we have discussed cell capacity, service classes, and their possible impacts on transmission-network dimensioning. In real life, many more factors must be taken into consideration while designing the network. For example, in a CDMA system, users may conduct a voice call, and at the same time run several data applications (e.g., while checking e-mail, browsing the Web, talking with a friend). A common scenario is that when some users are using speech service, some other users are using data service. This indicates that the traffic volume generated by a user and all the users in a cell is a variable, which depends on the environment, the portion of users using different services, and so on. This means that a parameter describing the average user traffic behavior is required so that reasonable average traffic can be deduced from it to indicate the bandwidth requirement of a corresponding transmission link(s). This parameter(s) is called the user traffic model. A user traffic model is a collection of information that describes the average and common user-traffic behavior in a system. These parameters should be decided according to the operators suggestion, the service supported by the system, experimental statistics, former experience (which could be difficult to define in new 3G systems), and so on. They may include the following: • Vm: Mb per busy hour (or Erlang) voice-data traffic generated by

user;

• Pm: Mb per hour packet-data traffic generated by a user;


333

• Tvm: Connection holding time in minutes for voice-data connection; • Tpm: Connection holding time in minutes for packet-data connection; • Td: Data traffic peak hour(s); • Tv: Voice traffic peak hour(s); • S : Percentage of users using speech service; • C d: Percentage of users using circuit-data service; • P d: Percentage of users using packet-data service.

It is important to realize that these parameters might be given in another form. For a specific traffic class, it can be expressed with megabit (or Erlangs) per user per busy hour but also in sessions per day per user and average session length to describe the above Vm or Pm. Usually, the operator will provide a kind of user traffic model. Sometimes, the network planner will provide help to the wireless operator to define user traffic model. Different operator might provide their user traffic model in different forms. The user traffic model provided by the operator is likely to be a mixture of many other parameters, which may look very confusing. It is imporatant to understand that the user traffic model is the first step to calculating the per-site traffic throughput correctly and we should focus on the useful information, and skip the rest. With these parameters, we can roughly decide the average user traffic behavior in a system or a specific coverage area. For example, it is possible to estimate the user call frequency and mean traffic generated per user during busy hour, and so on. With these parameters, the number of cell sites, number of subscribers, a signaling and management traffic model, it is possible to roughly calculate the possible traffic load on a link between an RBS and an RNC. Usually, this is the starting point of dimensioning the transmission-network part of a UTRAN radio access network. The following formula can be used to convert voice traffic Erlang into bandwidth requirement on a specific ATM link: Bandwidth (Mbps) = (Erlang(A, blockingpob))(53/48/0.9) Here, 53/48 is an ATM overhead, and 0.9 is a 90% link load. The Erlang(A, blockingpob) represents the number of 64-Kbps links needed.

334

7.8.4


Traffic Calculation Guidelines

The calculation of the traffic on each RNC interface can be done based on the traffic model, RF design, and some basic assumptions. The main input for the dimensioning of RNCs is the number of base stations, user data, and RNC type. The number of base stations and user data should be given for the area served by each RNC. User data on the Iu Interface is one of the parameters used for RNC dimensioning. It is of importance to notice that only the traffic in one direction (uplink or downlink) is used for the dimensioning of the RNC. The user data on the uplink (UL) and downlink (DL) should be calculated and the highest figure will be used for the RNC dimensioning. U = U v + U cs + U ps where U = User data; Uv = User voice; Ucs = User circuit-switched data; Ups = User packet-switched data. The user data is a combination of different services. The services can be either circuit data or packet data and have a number of different bit rates. The amount of voice normally is expressed in Erlangs. In order to calculate the amount of voice expressed in bytes per second, knowledge of the voice codec and the voice activity is needed. Voice in WCDMA uses a codec called adaptive multirate (AMR). AMR supports eight different source codec bit rates ranging from 4.75 to 12.20 Kbps. In addition, a codec mode for periods of silence is available. During periods of silence, one five-octet-long silence descriptor (SID) is sent every 160 ms. A commonly used model for voice activity is that each subscriber talks 50% of the time. During periods of talk, 5% of the time is considered as silence. This gives an actual time of talk of 45% and of silence 55%. An investigation shows that the voice activity in reality is around 60%. The reason for a higher figure is the influence of background noise from traffic, other people, and so on. For this calculation we can assume 50% voice activity. Table 7.3 shows the average bit rate for different AMR modes. For the calculations, 50% voice activity is used. The SID descriptor is considered as user data and, thus, included in the calculations.


335

Table 7.3 Average Voice Bit Rate AMR mode (Kbps)

Average User Bit Rate (Kbps)

04.75

2.50

05.15

2.70

05.90

3.07

06.70

3.47

07.40

3.82

07.95

4.10

10.20

5.22

12.20

6.22

Calculation of user circuit data is as follows: U cs, c = ∑U cs,c where Ucs,c = User data for circuit-switched data with bit rate c; c = All used circuit-data bit rates. Calculation of user packet data is as follows: U ps = (1 + Ω r )∑U ps, p where Ωr = Overhead caused by transmission (10%); Ups,p = User data for packet-switched data with bit rate p; p = All used packet-data bit rates. The retransmission rate of 10% is commonly used, although due to customer demand or other factors, different values might be used. 7.8.4.1 Calculation of Iu Traffic Load

The traffic load on the Iu interface is made up of user data, signaling, and frame overhead. All traffic is transported using ATM, and the user circuit

336


data and voice use AAL2, while user packet data uses ATM AAL5. The overhead is different for different services and, therefore, can make dimensioning less precise. The traffic load on the Iu interface is as follows:

(

T u = 1 + Ω sign

)(T

v

+ T cs + T ps

)(1 / l)

where Tu = Total Iu traffic load; Tv = Iu voice-traffic load; Tcs = Iu circuit-switched-data traffic load; Tps = Iu packet-switched-data traffic load; Ωsign = 10% (signaling). In order to cope with traffic variations, a load factor ( l) has been introduced. The load factor is dependent on the traffic load and low traffic load leads to larger traffic variations. The traffic variations on the Iu interface are smaller than the traffic variations on the Iub interface. Usually a load factor of 80% is used. Iu Voice Traffic Load

Table 7.4 shows the average bit rate for different AMR modes. For the calculations, 50% voice activity is used. The SID descriptor is considered as user data and, thus, included in the calculations. The multiplexing efficiency is considered to be 100%. Iu Circuit-Switched Data Traffic Load

T cs = (1 + Ω fr )U cs where Ucs = User data for circuit-data service; Ωfr = 21% for ATM, AAL2, UP, GTP frame overhead (fixed rate 64Kbps circuit-switched data). The frame overhead for fixed-rate 64 Kbps and for variable rate 57.6 Kbps is 21% and 28%, respectively. Higher bit rates are not yet standardized, but it can be assumed that the bit rate for fixed-rate 64 Kbps can be also used for higher bit rates. Again, 100% multiplexing efficiency is assumed.


337

Table 7.4 Average Iu Voice Bit Rate Including Frame Overhead

AMR Mode (Kbps)

Average User Bit Rate VA = 50% (Kbps)

14.75

4.62

15.15

5.07

15.90

5.30

16.70

5.75

17.40

6.20

17.95

6.65

10.20

8.01

12.20

9.13

Iu Packet-Switched Data Traffic Load

The frame overhead on the Iu interface is heavily dependent on the size of the user data packets. T ps = (1 + Ω fr )U ps where Ups = User data for packet-data service; Ωfr = 72% for ATM, AAL5, UP, GTP frame overhead. The frame overhead is only valid if the average length of packets is 125 octets. Larger packet lengths give a smaller percentage overhead and vice versa. The user data protocols are assumed to be TCP/IPv4 or protocols of similar length. Again, 100% multiplexing efficiency is assumed. 7.8.4.2 Calculation of Iub Traffic Load

The traffic load on the Iub interface is made up of user data, signaling, O&M, and frame overhead. All traffic is transported using ATM with user data using AAL2. Soft handoff causes a large part of the overhead. Making a correct calculation of the user data traffic from one or a few base stations is currently a problem. The average traffic per RBS can be used for calculation of the traffic on the RNC level, as many RBSs are connected to one RNC. The

338


assumption is that the connected RBSs will have peak traffic at different times and the combined busy hour (BH) traffic probably will be near the average BH traffic due to statistical multiplexing. Iub Total Traffic Load

(

)(

T = 1 + Ω sign + Ω O& M T v + T cs + T ps

)(1 / l)

where T = Total Iub traffic load; Tv = Iub voice traffic load; Tcs = Iub circuit-switched-data traffic load; Tps = Iub packet-switched-data traffic load; Ωsign = 10% for signaling; ΩO&M = 2% for O&M. In order to cope with traffic variations, a load factor has been introduced. The load factor ( l) is dependent on the traffic load. Low traffic load leads to larger traffic variations, and normal traffic load is at least 1 Mbps per RBS. For most applications, a load factor of 80% is used. Iub Voice Traffic Load

T v = (1 + Ω SH )U v where Uv = User voice including frame overhead (see Table 7.5); ΩSH = 30% for soft handoff. Iub Circuit-Switched-Data Traffic Load

T cs = (1 + Ω SH )(1 + Ω fr )U cs where ΩSH = 30% for soft handoff factor; Ωfr = 23% for ATM, AAL2, UP frame overhead;


339

Table 7.5 Average Iub and Iur Voice User Bit Rate

AMR Mode (Kbps)

Average User Bit Rate VA = 50 %, Uplink (Kbps)

Average User Bit Rate VA = 50%, Downlink (Kbps)

14.75

15.61

5.10

15.15

16.06

5.55

15.90

16.29

5.78

16.70

16.74

6.23

17.40

17.19

6.68

17.95

17.58

7.13

10.20

19.22

8.71

12.20

10.35

9.84

Ucs = Circuit-switched data on Iub The frame overhead is calculated for fixed-rate 64 Kbps and variablerate 57.6 Kbps circuit-switched data. It is assumed that all CS RABs have a similar-size frame overhead. Again, 100% multiplexing efficiency is assumed. Iub Packet-Switched-Data Traffic Load

T ps = (1 + Ω SH )(1 + Ω fr )U ps where ΩSH = 30% for soft handoff; Ωfr = 24% for ATM, AAL2, PDCP, UP frame overhead; Ups = Packet-switched user data on Iub. The frame overhead varies with the packet size. The frame overhead is approximately 24% if header compression is used and the average length of the packets is 125 octets. If the packet length is larger, the overhead will be higher due to less impact from header compression. The user data protocols are assumed to be TCP/IPv4 or protocols of similar length. Multiplexing efficiency of 100% is assumed.

340


7.8.4.3 Calculation of Iur Traffic Load

On the Iur interface, user data is transported for users that are in soft handoff with more than one RNC involved. The user data is of the same type as on the Iub interface with the same type of overhead. The traffic load on Iur interface is made up of user data, signaling, O&M, and frame overhead. All traffic is transported using ATM and all user data use the AAL2 protocol. The amount of the Iur traffic load depends on how large the probability is that an UE is in soft handoff with two or more RNSs involved. If an RNS does not have any borders towards another RNS, there is no traffic on the Iur interface. For the case in which the RNS has a homogeneous cell pattern and is surrounded by other RNSs on all borders, an RNC handoff factor has been introduced. Calculation of Traffic Load on the Iur Interface

(

)(

T = 1 + Ω sign + Ω O& M T v + T cs + T ps

)(1 / l)

where T = Total traffic load on the Iur interface; Ωsign = 10% for signaling; ΩO&M = 2% for O&M; Tv = Iur voice traffic load; Tcs = Iur circuit-switched-data traffic load; Tps = Iur packet-switched-data traffic load. In order to cope with traffic variations, a load factor (l) has been introduced. The load factor is dependent on the traffic load, and for normal traffic, a load factor of 80% is used. Low traffic load leads to larger traffic variations. The Iur interface normally has smaller traffic variations than Iub interface. Iur Voice Traffic Load

T v = aΩ SHU v where a = Inter-RNC soft handoff factor;

ΩSH = 30% for soft handoff;


341

Uv = User voice including frame overhead (see Table 7.5). Iur Circuit-Switched-Data Traffic Load

T cs = aΩ SH (1 + Ω fr )U cs where a = Inter-RNC soft handoff factor;

ΩSH = 30% soft handoff factor; Ωfr = 23% for ATM, AAL2, UP frame overhead; Ucs = Circuit-switched user data on Iub. The frame overhead is calculated for fixed-rate 64-Kbps and variablerate 57.6-Kbps circuit-switched data. Again, 100% multiplexing efficiency is considered. Iur Packet-Switched-Data Traffic Load

T ps = aΩ SH (1 + Ω fr )U cs where a = Inter-RNC soft handoff factor;

ΩSH = 30% for soft handoff; Ωfr = 24% for ATM, AAL2, PDCP, UP frame overhead; Ups = Packet-switched user data on Iub. The frame overhead varies with the packet size. The frame overhead is approximately 24% if header compression is used and the average length of packets is 125 octets. If the packet length is larger, the overhead will be higher due to less impact from header compression. The user data protocols are assumed to be TCP/IPv4 or protocols of similar length. Multiplexing efficiency of 100% is assumed. The simplified inter-RNC soft handoff factor is calculated as follows: a = 2.5 / n RBS where nRBS = Number of RBSs connected to current RNC.

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7.8.5


Example of Traffic Calculations

The user traffic model shown in Table 7.6 is very straightforward. It gives the Erlangs and BH data volume, so it is not very difficult to calculate the traffic volume generated per subscriber during busy hour. Once the network planner fully understands the user traffic model, it is easy to calculate the average traffic volume generated on a per-subscriber basis during busy hour: Voice traffic per subscriber: 8 × 0.5 × 15 = 60 bps Circuit-data traffic per subscriber: 64 × 1 × 4 = 256 bps Packet-data traffic per subscriber: 500 bps So, the average pure user data rate generated per subscriber during BH is 60 + 256 + 500 = 816 bps Note that the data rate is the pure user data rate; this means that it does not include any protocol overhead. Based on the raw data rate, in the next step, we can calculate data throughput per site. To calculate the traffic throughput per site, we need to know the number of subscribers per site. Table 7.6 Example of User Traffic Model Circuit Services Call type

Voice

Data

Average data rate (Kbps)

8

64

Activity factor

0.5

1

BH traffic per user

15 mErlang

4 mErlang

Packet Services Uplink

Downlink

Average data volume

20 kB/day/sub

200 kB/day/sub

BH data volume

50 kB/day/sub

500 kB/day/sub

Average packet size

128 bytes/packet

128 bytes/packet

Average number of connections

2 times/day

2 times/day

Number of handoffs

2 times/connection

2 times/connection


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This usually can be calculated from the total number of subscribers the system is going to support and the number of the RBSs (provided by the radio planner). If the Ttp is traffic throughput from a site, then we can calculate T tp = T + Ω + Ω O&M where T = Pure user traffic in a site multiplied by the number of users in a site; Ω = Protocol overhead (assume an added 20%); ΩO&M = Operation and maintenance of traffic for the site (assume 128 kbps). We can add 20% for the ATM protocol overhead. The O&M traffic also remains as an open issue but a good approximation is to add 128-Kbps O&M traffic to each site. With a given user traffic model, one usually can calculate the mean (average) traffic generated by the number of users in a given area (cell or site). Usually, this is the first step in dimensioning the radio access network. However, this mean traffic volume sometimes does not reflect the whole picture of the real traffic. In TDMA networks, the capacity of a radio timeslot is fixed. This means that the traffic generated by one user is limited by this capacity. However, in a UTRAN (CDMA-based) network, since different services correspond to different data rates, one user may at the same time run several applications. The traffic burstyness characteristics require peak traffic to be taken into consideration when dimensioning the transmission network. One interesting scenario that happens frequently is that the mean data rate (site traffic throughput) calculated according to some user traffic model is lower than the data rate required by a single service. For example, the network might be designed to support 384-Kbps data service, but according to the user traffic model and the number of users covered by one site, when one calculates the mean traffic volume generated by the site, the result is lower than 384 Kbps. If we directly use the calculated mean traffic volume to dimension the transport link, in this case, the resulting transport link may not even be able to support a single user running a single 384-Kbps application with required quality. To solve this problem, peak traffic might be considered while dimensioning transmission links.

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7.9 Alternative Solutions in Transmission Networks 7.9.1

Considering Dark Fiber and Dark Copper

Sometimes wireless operators, instead of leasing E1/T1 circuits or fiber-optic facilities, can lease copper or fiber lines without terminal equipment (dark fiber or copper), then design the system and provide their own terminal equipment. This solution for providing transmission facilities is obviously more cost efficient, but requires a lot more time and resources on the wireless operators side to do the engineering, equipment procurement, and installation, as well as operation and maintenance. Electrical utility companies, cable companies, and railways are good candidates from which to lease dark copper or fiber. Sometimes offering services in return, wireless operators could make a business case and build a partnership that would be of interest to all parties involved. Electrical utilities design their networks with high reliability and survivability as well as very low network delay. Their telecommunications networks are required to carry not only data traffic for LAN connectivity and administrative data needs but also teleprotection traffic. Teleprotection traffic is a reason that network delays must be very low for correct operation of protective relaying equipment. 7.9.2

Partnership with Utilities

Most data today travels over a network that was built for voice traffica process that is unreliable and inefficient because the loss of a single bit stream of data will adversely affect the entire communication. System providers are offering several different devices to transmit and receive data, but regardless of how data is transmitted, the wireless infrastructure will have to be improved to ensure consistently reliable communications. For ISPs and other data-centric businesses, it is critical that wireless providers work quickly and effectively to broaden their coverage areas. They also must provide an infrastructure that is more suited to wireless data transmission in order to get to market more quickly and deliver their own services and applications more reliably. Required improvements include upgrading existing cell sites with new electronics and placement of new towers and antennas to ensure coverage across the nation. However, these activities often require wireless providers to solve zoning regulation issues, civic resistance, and technical obstacles. By creating strategic partnerships with local utilities, which have a core competency of infrastructure management, communications providers can overcome these


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problems. While the wireless communications provider maintains the role of network manager and supervisor, the utility leverages its existing relationships, equipment, and expertise to construct the infrastructure for delivery of wireless communications. Whenever possible, wireless providers are using existing buildings and communications infrastructure (colocation) to build out new networks. Adding antennas and upgraded electronics to towers already in place is both cost effective and more easily accepted by the public. However, existing infrastructure is not always available to cover new areas. There are a number of ways wireless providers can work with local electric utilities to place antennas and associated electronics on electric transmission and distribution facilities and, thereby, expand the network, improve coverage, and increase reliability. 7.9.2.1 Electric Transmission Towers

Large electric transmission towers provide a corridor between generation stations and substations. They tend to be in remote, out-of-the-way locations and run through less expensive territories. Typical voltages on these transmission towers range from 138,000 to 500,000V. Their location and 150-footplus height allow for placement of larger antenna arrays and excellent antenna elevation. These towers are ideal as antenna locations for propagating wireless signals over large areas, but precise engineering and extreme care must be used when placing RF or microwave antennas near transmission conductors to avoid the dangers of electric arcing. From a strictly microwave perspective, these towers as well as power lines are not an obstacle to install a microwave system even if they obstruct the LOS of the microwave system. Signal loss due to the partially obstructed Fresnels zone would be only 1 to 2 dB, depending of the type of the electrictransmission tower and its construction. 7.9.2.2 Distribution Poles

Colocating antennas on wooden distribution poles enables the wireless provider to enter neighborhoods that otherwise would not be served. The lower pole heightstypically 50 to 90 ftmean that signal propagation will be reduced, requiring installation of more antennas, new poles, and other structures. When pole locations and physical constraints prevent the use of existing towers and distribution poles, there may be no alternative but to erect a new tower. Utilities own property in their coverage areassubstations, offices, and maintenance and storage sites. In addition, they often hold right-of-way easements that provide access to areas otherwise unattainable. Developing a relationship with the local electric utility could offer the service

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provider hundreds of potential sites. Wireless service providers attempting to gain similar access must negotiate individual leases with individual ownersa process that is time consuming, frustrating, and expensive. Another option is negotiating a master lease with a company that owns multiple sites in the neighborhood. School districts typically resist antennas on their properties, but local mass-transit (rail and bus) companies, city governments, libraries, and churches are often open to the idea of leasing space for a tower or other antenna mounting structures. 7.9.2.3 Technical Issues

Some utilities have resisted the idea of adding wireless infrastructure development to their workload because of the potential effects on the electric transmission and distribution (T&D) system. Installing antennas on electric transmission and distribution equipment affects wind resistance and torsion stresses on the towers. The design of these structures must account for conditions such as conductor weight, wind loading and ice loading, bracket design, attachment methods, and dead-weight loads. In some cases, utilities simply may not be willing to take these risks with their infrastructures. Research has shown, however, that proper design and clearances between wires and antennas produce no ill effects on either the communications network or the T&D system. It is critical that electrical and physical constraints on individual installations be carefully considered. Electric arc effects can allow flashover between the grounded antenna and T&D wires, so antennas must be kept 20 ft from 13,000V distribution lines and a minimum of 28 ft from 500,000V transmission lines to provide appropriate clearances and safe working distances. Distances must also be maintained between antennas that are comounted on a tower. Analyses must carefully review initial weight, wind loading, and torsion effects. Weld methods, bracket types, and even the types and sizes of bolts must be appropriate for the installation. Perhaps the most critical element is the towers foundation. Depending on the utilitys philosophy, safety factors of 10% to 50% are factored in to prevent tower failure under all but the most unusual circumstances. If the existing safety margin is inadequate, the carrier and the utility often negotiate the cost of replacing the foundation and associated structure. Installation and maintenance of wireless antenna systems are more painstaking than T&D line work because of the precision required to position and orient the antenna. Additional training will be required, but utility line workers are generally receptive because they know that broadening their skill set increases their employee value. Economies of scale have been realized in


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wireless maintenance activities because they are integrated into maintenance practices for the electrical infrastructure. It is important to understand that electric transmission towers are dangerous locations prone to a phenomenon called ground potential rise (GPR). The communications cables (E1/T1 circuits) are susceptible to damage from lightning surges, since they can develop high shield-to-pair voltages even with low lightning currents on the shield; these voltages could cause lengthy and potentially catastrophic outages. GPR usually occurs during a power fault when the fault current returns to the power neutral source through the earth. Communications cables are considered exposed to GPR when the possibility exists that the local ground (at the cell site) differs from remote ground by 300V or more. Optical isolators (optocouplers) are generally used to treat each circuit (including T1/E1 circuits) going into the power station to protect the circuits and facility, as well as personnel, from electric hazard associated with GPR. Design of any communications circuits into the power location has to be in accordance with ANSI/IEEE 367, Recommended Practice for Determining the Electric Power Station Ground Potential Rise and Induced Voltage from a Power Fault, and ANSI/IEEE 487, Guide for Protection of Wireline Communications Facilities Serving Electric Power Stations. The only other alternative would be a letter, on letterhead and appropriately signed, from the power utility owner, stating that at no time will the GPR at the site or sites ever exceed 1,000V peak-asymmetrical, as calculated per ANSI/IEEE 367. The power utility owner must be aware that issuance of the letter constitutes an assumption of liability for injury or damage brought about by electrical fault conditions. Installation of the microwave systems using electrical utility poles and towers has to be carefully examined, since they may not fulfill requirements for the twist and sway of the microwave antenna mounting structure, especially on the higher microwave frequencies. One major benefit of partnership with the local utility is the utilitys experience and established credibility with the community and town zoning boards. Enhancing communications services that will benefit local residents and businesses, while diminishing the need for new roads to reach new towers and minimizing installation of new structures, is a win-win situation. The communications network improvements may be invisible to the neighborhood but the benefits are not. Wireless providers may prefer to erect new towers and install antennas on targeted buildings as they grow their networks. When entering a new territory, it is much faster and easier to conduct a single negotiation by partnering with one local utility than hundreds of negotiations to get the same area coverage. When wireless providers seek to

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expand their network infrastructure and broaden coverage, partnering with utilities speeds deployment, reduces costs, and increases efficiency. Utility expertise in zoning, construction, and maintenance, as well as access to existing antenna-ready structures, can be put to good use without expanding the carriers payroll. Using existing structures to locate antennas allows faster deployment of equipment and improves maintenance cost-efficiency while the wireless carrier enjoys fast, cost-effective deployment of a new network. The ISP or wireless operator receives a higher QoS, more reliable coverage and transmission, and faster time to market. 7.9.2.4 Power Line Carrier and Optical Power Ground Wire

Voice and limited data for supervisory control and data acquisition (SCADA) over the power lines [power line carrier (PLC)] has been used by electrical utilities for years. SCADA systems are used extensively by power, water, gas, and other utility companies to monitor and manage distribution facilities. Very narrowband in its nature, it was not suitable for any serious application outside transmission of the monitoring and control signals. Over the last few years and with the introduction of digital PLCs, its bandwidth is approaching 1E/T1. Aerial fiber-optic cables can be suspended on the telephone poles just like any other copper cable, wrapped around existing power lines, or be a part of the ground wire on top of the high-voltage transmission tower [optical power ground wire (OPGW)]. OPGW is described in more detail in the chapter dealing with fiber-optic equipment. Utilities may be willing to lease out some of the capacity of their fiber-optic system, and usually every tower has a junction box with access to the fiber-optic cable. 7.9.3

Optical Laser Communications

The primary application for high-capacity laser communication system, sometimes called laser transmission systems (LTS), is extending fiber-optic backbones to the customers premises. The products optical fiber interfaces must be fully compatible with the existing carriers SONET/SDH backbones, and the system can meet both the voice and data bandwidth needs of the business customer. Typically, in North America only 10% to 15% of the carriers customers and only 3% to 5% of commercial buildings are on fiber with highbandwidth capabilities. These numbers are even lower in Europe and other parts of the world. In general, for a building, a minimum of 10 to 15 T1s


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(DS3 break-even point) of existing revenue was required for a carrier to even consider the long and expensive process of deploying fiber. Laser (wireless) optical networking is an innovative technology that improves upon the concept of free-space optics. A free-space optical link consists of two optical transceivers accurately aligned to each other with a clear line of sight. Typically, the optical transceivers are mounted on building rooftops and use the light near the infrared region of the spectrum. The optical transceiver consists of a laser transmitter and a detector to provide full duplex capability. Free-space optics enables very fast deployments of broadband access services to buildings; the time-consuming and expensive process of getting permits and trenching city roads is completely avoided. This approach is, however, problematic in that atmospheric conditions have a significant impact on the optical link performance. Availability of a free-space optics link is generally determined by the link length and fog patterns in a specific location. The laser optical network uses mesh configuration of short, redundant links between optical transceivers. The mesh network functionality is provided by compact nodes mounted on the rooftops of various buildings and connected by LOS. Beam attenuation, depending on bandwidth configuration, can cover distances up to 2.5 mi (4 km) with transmission rates up to 622 Mbps (OC-12 equivalent). Infrared-based laser optical transmission systems are not subject to any licensing requirements in the United States and probably the rest of the world. Because this transmission system is immune to electro-magnetic interference (EMI) and does not cause RF interference, it is also virtually interceptionproof, exceeding methods employed in microwave and copper cable products. Safety issues are regulated by international standards IEC 825-1 and EN-60825-1 regulating health issues and especially eye safety of optical sources used in laser communications systems. This classification is based on concerns for the potential risk to human health (such as eye damage) caused by high-power optical transmission systems. LTS could become much more interesting in 3G wireless systems, where huge capacity links to every cell will be required. Heavily loaded sites will be in dense urban and urban areas where the distance between cell sites is short and can range from a few hundred meters to a few kilometers; they are therefore good candidates for alternative backhaul media solutions.

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References [1] ITU-T G.174,Transmission Performance Objectives for Terrestrial Digital Wireless Systems Using Portable Terminals to Access the PSTN, June 1994. [2] Lee, W. C. Y., Mobile Cellular Telecommunications: Analog and Digital Systems, New York: McGraw-Hill, 1995. [3] ITU-T G.803, Architecture of Transport Networks Based on the Synchronous Digital Hierarchy (SDH), March 2000. [4] ITU-T G.176, Planning Guidelines for the Integration of ATM Technology into Networks Supporting Voiceband Services, April 1997. [5] Sherman, K., Data Communications: A Users Guide, Third Edition, New York: Prentice Hall, 1990. [6] Flanagan, W. A., The Guide to T1 Networking, Telecom Library, Inc., New York, 1990. [7] GR-2830-CORE, Primary Reference Sources: Generic Criteria, Bellcore, Issue 2, December 1995. [8] Larus Corporation, Digital Network Timing and Synchronization, San Jose, CA, 1997. [9] Dixon, R. C., Spread Spectrum Systems with Commercial Applications, New York: Wiley, 1994. [10] Zhou, D., Survivability in Optical Networks, IEEE Network, November/December 2000, pp. 1623.

8 Transmission Equipment 8.1 Digital Microwave Radio 8.1.1

PDH and SDH Microwave Radios

The PDH microwave radio provides a transmission medium for digital traffic of standard capacities typically ranging from 1.544 Mbps (1T1) to 45 Mbps (1DS3) in North America and from 2.048 Mbps (1E1) to 34 Mbps (16E1) based on ITU standards. Sometimes high-capacity MW backbones in North America are built for 3DS3 and require at least 28 MHz of bandwidth. These radio links may be established between any two points with the line of sight and depending on the frequency, geographical region, and rain statistics the typical link distance can be up to 25 mi (40 km). For the longer microwave link hops, additional measures have to be taken to ensure required reliability of the system (e.g., space or frequency diversity). In todays wireless networks, these PDH microwave systems are used for the low- to medium-capacity links (e.g., RBS-BSC backhaul connectivity) in wireless networks. When it comes to high-capacity transport of information, network planners have two key technologies at their disposalfiber-optic systems and digital microwave radio. The new generation of digital microwave systems, based on SONET/SDH, is able to meet the requirements for the highcapacity backbone transmission systems. SDH/SONET radios provide economical solution when existing infrastructure (towers and shelters) can be reused, and when rights of way or adverse terrain make fiber deployment very costly or time consuming. In existing 2G wireless networks, these high351

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capacity microwave systems are only used for the connection of the hub sites to the fiber-optic ring or directly to the BSC. Also, they can be used for the PSTN connection in third-world countries where existing telecommunications infrastructure is limited. In the future 3G wireless systems it is expected that more cell sites will require this kind of connectivity due to the increased backhaul capacity requirements. To maximize the benefits of the SONET/SDH microwave radio, the radio must be capable of complementing a synchronous fiber-optic network. This means that in order for the microwave radio to integrate with fiberoptic network elements, its design must address a number of parameters, including capacity and growth, network management, maintaining pace with SDH/SONET standards evolution, interface, and performance. Providing microwave radio with the optical interface will allow the microwave network to integrate with the fiber-optic network without the use of multiplexing equipment (unless drop or insert of the traffic is required). Todays SONET/SDH radio technology is capable of delivering bandwidth-efficient 8 bits/s/Hz of bandwidth. For example, 512-state QAM technique can pack two STM-1 streams into a single 40-MHz channel using a single carrier [1]. By adding channels in a 1:N configuration, system capacities of up to 14 protected STM-1s can be achieved within one frequency band (e.g., in the upper 6-GHz band, eight bidirectional channels are available). By deploying a dual-band configuration (such as lower 4- and 5-GHz bands), system capacities of STM-16 and greater are achievable. Although theoretically possible, this can be done only if the spectrum governing bodies allow more than one channel to be used by the same user. SDH/SONET microwave radios have to meet very stringent error performance objectives defined in ITU and ANSI standards. Sophisticated and powerful countermeasures, including FER and ATPC, are used to combat propagation anomalies.

8.1.2

Standard Microwave Radio Configuration

Standard microwave radio configuration consists of the entire microwave and digital modem part being placed indoors, microwave antenna mounted outside on the tower, and a waveguide connecting the transceiver of the radio with the antenna. This solution is shown in Figure 8.1 and it is acceptable for the lower frequencies but quickly becomes unacceptable as frequency increases. This is due to the high losses in the transmission lines (coaxial or waveguide) which become unacceptably high at higher frequencies. Most


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PCS antennas

Microwave antenna

Equipment shelter

A standard microwave system includes an antenna, MW radio waveguide, installation and mounting accessories, and pressurization

Figure 8.1 Standard-configuration microwave system.

commonly used waveguides today for terrestrial microwave point-to-point systems are Helliax elliptical waveguides. This configuration is still used today for lower-frequency bands below 10 GHz and for high-capacity (backbone) microwave systems. Whenever possible, split configuration is replacing the standard configuration and this is particularly true for the cell-site microwave systems in wireless networks. 8.1.3

Split Microwave Radio Configuration

8.1.3.1 General Description

To reduce losses between the transceiver and antenna, the outdoor unit (ODU) containing all the RF modules can be mounted near the antenna

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(Figure 8.2). The ODU is connected to the indoor unit (IDU), containing baseband circuitry, modulator, and demodulator, by means of one single coaxial intermediate frequency (IF) cable. The distance between indoor and outdoor equipment can usually be up to 300m (1,000 ft). SDH microwave radio will be described as an example of the highcapacity, split-configuration microwave system. Baseband interfaces for the equipment are STM-1 electrical or optical, 2-Mbps wayside channel and 64Kbps insert channels. The basic block diagram for a digital microwave system (split configuration), including the main blocks, is shown in Figure 8.3. The block diagram includes marked interface points that serve as reference points for several technical parameters used in the text. Some main technical characteristics of the European version of the split-configuration STM-1 microwave radio in 18- and 23-GHz bands are presented in the following sections. 8.1.3.2 Equipment Main Characteristics

Table 8.1 shows two very common frequency bands, traditionally used in wireless networks18 and 23 GHz. Channeling plans are according to ITU-R recommendations. MW Antenna Outdoor module

Indoor module

Single coaxial cable

Figure 8.2 Split-configuration MW radio.

Transmission Equipment IDU OUT

IN

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ODU

Modulator and baseband

Transmitter

RF Rx filter

Branching network

Feeder

Modulator and baseband

Receiver

RF Rx filter

Branching network

Feeder

Figure 8.3 Principal block diagram of the MW radio.

Nonprotected systems (1+0) consist of one IDU and one ODU interconnected with a single coaxial cable. In the case of failure of any of the electrical or mechanical components, the entire microwave hop would fail. In hot standby (protected) configuration (1+1), the IDU, the transceiver unit, and the coaxial cable between IDU and ODU are duplicated. The two transceivers share the same branching unit. A switch at radio frequency level, included in the branching unit, allows for switching between the two transmitters. A splitter and switching unit placed between the two IDUs is added. The unit splits and switches between the STM-1 interfaces. For each terminal one antenna is used. Also, there is a version of the protected system with two ODUs and only one antenna. These radios were also designed with ease of installation in mind. The equipment is designed to enable very easy and quick installation within a few hours. It is designed for split-mount installation but can also be all-indoor mounted. One coaxial cable between IDU and ODU is used for 1+0 systems and two cables for hot standby systems (1+1). Installation of the splitTable 8.1 Frequency Bands, 18 and 23 GHz Frequency Frequency Band (GHz) Range (GHz)

ITU-R Adjacent Channel Tx-Rx Duplex Recommendation Spacing (MHz) Spacing (MHz)

18

17.719.7

ITU-R F.595-3

55

1,010

23

21.223.6

ITU-R F.637-2

56

1,232

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terminal configuration, assuming that all of the infrastructure is available (tower, shelter, racks, pipe-mounts), should not take more than one day per hop and two people to do the work. Indoor installation consists of one small module for nonprotected and three modules for protected systems (two modems plus the switching module). Figure 8.4 illustrates the usual telecommunications rack, layout with the fuse or breaker panel on the top, space for the small rectifier and battery backup at the bottom of the rack, and microwave radio and the DSX panel mounted in the middle of the rack.

Rack units 44 (RUs; 1RU=1.75 in) 40

Dual fuse panel 1+10 IDU (nonprotected)

35

Channel A Channel B 1+1 IDU

30

Modem A Prot. unit Modem B

25

20

1+1 protection

DSX panel

15

10

6

Figure 8.4 Microwave equipment rack.

Rectifier and battery backup


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Mechanical Characteristics

The ODU consists of the antenna (Ø = 0.45m, 0.6m, or 1.2m), a branching unit, and one or two transceivers, depending on the selected configuration. The unit may be attached to a vertical pipe-mount (Ø = 115 mm or Ø = 75 mm). A pipe-mount arrangement for tower, wall, and roof mounting will depend on the location where the ODU and antenna are mounted. The IDU can be installed as a stand-alone unit or it can be mounted within an ETSI standard cabinet (ref. ETS 300 119) or 19-inch standard cabinet (ref. IEC 297-2 and IEC 297-3). Sometimes IDUs can be wall-mounted as well. Dimensions and weight are as follows: • Dimensions (1 + 0 Terminal) •

IDU: 483 mm (W) × 250 mm (D) × 50 mm (H)

•

ODU: 135 mm (W) × 310 mm (D) × 430 mm (H)

• Weight for a complete 1 + 0 terminal •

IDU: approximately 4.0 kg

•

ODU (without antenna): approximately 9.0 kg

Power Supply

The equipment operates from a battery supply between −40.5V and −57V, nominally −48 VDC according to ETS 300 132-2. The primary dc power is supplied to the IDU through a main fuse and a filtering function, which includes input filter to attenuate the common mode noise. The power to the ODU is supplied from the indoor unit via the intermediate frequency (IF) coaxial cable. Power consumption is as follows: IDU:

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