Information Technology Network and Internet: Innovative Single Window Book from Base to Research

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DEDICA TED to DEDICATED My late parents parents,, Satish Chandra Bhunia and Santa Prabha Bhunia Shyamsundar pur, Sabong Shyamsundar pur, East Midnapore, West Bengal, India

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Preface

I am by character quite unsystematic. That has cost me many things. Yet I do not like to change. Because I find our mother nature is unsystematic. It is only we who have gone out of nature’s way to become systematic to meet our own objectives, the natural consequences of which are environmental crisis, lack of imagination and increased disharmony, conflict and wars. Water flows from hill to plain following haphazard paths. We make dams to make some systemic control, which is the cause of floods to my mind. We destroy un systemic natural forest to build systematic towns and cities, the consequence of which is environmental problem. There need to have science and technology for development and growth as we want but in nature’s way. This will give a better society. I sincerely believe students should be given guidance in natural way, and he has himself to become mastery of systematic integration of knowledge so earned and acquired subsequently. Lots of scopes need to be left out for students to imagine and think. I do not like to teach a kid, as is done in all cases first +, then –, but I prefer to leave—to the kid after teaching him/her +, so that the himself/herself may discover—for which some tips may be given. This is exactly what I try in high-level teaching in institutes/universities. I was wondering to prepare a manuscript following this philosophy of teaching that creates imagination in the students. Accordingly the present book is not that way a so called systematic book, but a book for my way of teaching. The chapters in the book have been prepared in a way to make them independent to each other as far as possible. This will make readers to read chapter as chosen, being independent on other chapters. Besides, I wonder the present practice of writing books that on a same subject there will be separately one for B Tech level, one for M Tech level and another one for research level. This practice is not appropriate for the students of developing countries. For our students, a concise book for all levels is appropriate, and is surely possible to prepare. My attempt does not belie this motivation in this book too. However I do believe my different strokes in writing this book will be best judged by readers. The criticism is welcome. In order to complete this book, I could not give adequate time to my family members as they expected. Their sacrifice is greatly acknowledged. As I write this preface, the remembrance of Late Paresh Maity, science teacher of my higher secondary school suddenly floats in my mind alongwith those of Pabitra Sir and Satish Sir. This may be called telepathy. But I must sincerely acknowledge Paresh Sir’s way of teaching physics to me that is suddenly one of the pillars that has made me what I am today. Indian School of Mines C T Bhunia Deemed University


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Contents

Chapter 1: • • • • • • • • •

1

Introduction ....................................................................................................................... 1 Recent Progress of Computer Technologies .................................................................... 2 Current and Future Communication Technologies ...................................................... 22 Local Loop Transport Technology .................................................................................. 67 Multimedia Communication and Conferencing Standards .......................................... 76 UTN Personal Communication ...................................................................................... 81 From 2G to 3G ................................................................................................................. 82 e-business etc. .................................................................................................................. 84 Knowledge Age and Management .................................................................................. 86

Chapter 2: • • • • • • • • • • • • • • • • • • • •

Information Technology in 21st Century

Network and Internet Technology

99

History Behind Computer Network ............................................................................... 99 Objectives of Networking .............................................................................................. 115 Functions and Scopes of Networks .............................................................................. 115 Different Networks ....................................................................................................... 120 Introduction of Internet ................................................................................................ 137 Protocol .......................................................................................................................... 139 Switching Techniques ................................................................................................... 140 Layered Protocol ............................................................................................................ 142 OSI Protocol ................................................................................................................... 143 HDLC/SDLC Frame of Data Link Layer ..................................................................... 148 Error Control ................................................................................................................. 149 Control Field .................................................................................................................. 191 Other Protocols .............................................................................................................. 193 TCP/IP Protocols ........................................................................................................... 206 IP Header Descriptions ................................................................................................. 207 Unregistered or Private Address Space ....................................................................... 217 Subnet Mask .................................................................................................................. 217 Classless Addressing and Routing ............................................................................... 221 Option Fields of IP ........................................................................................................ 224 Description of TCP Headers ......................................................................................... 228

(ix)


5

(x) • • • • • • • • • • • • • • • • • • • •

UDP Headers ................................................................................................................. 233 Address Resolution Protocol ......................................................................................... 241 Proxy ARP ...................................................................................................................... 245 Reverse Address Resolution Protocol ........................................................................... 246 IPv4 to IPv6 ................................................................................................................... 248 How Did IPv6 Come Up ................................................................................................ 252 IPv6 in Details ............................................................................................................... 253 Details of Use of Extension Headers ............................................................................ 256 ICMP .............................................................................................................................. 258 BOOTP ........................................................................................................................... 261 DHCP ............................................................................................................................. 262 IPv4 and IPv6 Address Compatibility ......................................................................... 262 Dual Stack/Tunneling ................................................................................................... 271 Objective Type Questions ............................................................................................. 280 Domain Name Service .................................................................................................. 287 Voice Over Internet or Internet Telephony ................................................................. 289 Technical Problems of Voice Packet Transmission Over Internet ............................. 290 IPv6 for Real Time Services ......................................................................................... 292 Physical Layer Interface ............................................................................................... 293 Transmission Media Option ......................................................................................... 299

Chapter 3: Advanced Error Control Techniques in Network • • • • • • • • • • • • • • • • • • •

306

Introduction ................................................................................................................... 306 Basic BEC Techniques .................................................................................................. 309 Different Modified Techniques ..................................................................................... 312 Sastry’s Scheme and Morris Modification ................................................................... 313 Other Modifications ...................................................................................................... 313 Two Level Coding .......................................................................................................... 314 Parity Selection in Two Level Coding .......................................................................... 316 Packet Combining Scheme ........................................................................................... 317 Modified Packet Combining Scheme ............................................................................ 318 ARQs for Variable Error Rate Channels ..................................................................... 321 YAO Technique .............................................................................................................. 321 Chakraborty’s Technique .............................................................................................. 322 New Schemes ................................................................................................................. 323 ARQ Schemes Under Practical Situations .................................................................. 326 GBN and SRQ Under Different Schemes .................................................................... 327 Issues of Sending Different Signal Waveforms for Repeated Retransmitted Copies .... 328 Application of Multilevel Coding Scheme in Variable Error Rate Channel .............. 329 Majority Technique ....................................................................................................... 330 Analysis of the Majority Scheme for SW ARQ ............................................................ 330


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(xi) Chapter 4: • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • •

Data/Network Security Techniques and Approaches

355

Data or Information Security Introduction ................................................................. 355 Cryptography ................................................................................................................. 358 Conventional Encryption .............................................................................................. 359 Classical Cipher ............................................................................................................. 359 Substitution Codes ........................................................................................................ 359 Transposition Codes ...................................................................................................... 360 Cryptanalysis of classical ciphers ................................................................................ 360 General Attacks ............................................................................................................. 362 Secret and Private Key Cryptography ......................................................................... 362 Stream Cipher ............................................................................................................... 363 Block Cipher .................................................................................................................. 363 DES ................................................................................................................................ 364 Modes of Operation of DES .......................................................................................... 370 Automatic Variable Key ................................................................................................ 373 Proof of DES .................................................................................................................. 375 Merits and Demerits of DES ........................................................................................ 375 Quantification of Performance ..................................................................................... 377 Triple DES ..................................................................................................................... 379 International Data Encryption Algorithm .................................................................. 380 Advanced Encryption Standard ................................................................................... 382 Comparisons of Secret Key Cryptosystems ................................................................. 385 Modes of Operation of AES ........................................................................................... 386 Limitations of AES ........................................................................................................ 387 Limitations of Secret or Private Key Cryptography ................................................... 388 Key Transport Protocol ................................................................................................. 389 Needham—Schroeder Protocol ..................................................................................... 389 Key Agreement Protocol ............................................................................................... 390 Diffie-Hellman Protocol ................................................................................................ 390 Station to Station Protocol ............................................................................................ 390 Merkles’s Puzzle Technique of Key Agreement .......................................................... 391 Quantum Security ......................................................................................................... 392 Public Key Cryptography .............................................................................................. 395 RSA Algorithm .............................................................................................................. 395 How Secured is RSA ...................................................................................................... 398 Limitations of RSA Algorithm ...................................................................................... 399 Trapdoor Knapsack Problem ........................................................................................ 401 McEliece’s Public Key ................................................................................................... 402 Comparison of RSA and TRAP DOOR Public Key Cryptosystems ............................ 402 Public Key Cryptographic Mechanisms ....................................................................... 402 Digital Signature ........................................................................................................... 404 Digital Signature under RSA algorithm ...................................................................... 404 Check Functions for Authenticity, Integrity and Norepudiation of the Message Content ........................................................................................................... 404


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(xii) • • • • • •

Illustrate the Non-repudiation by Digital Signature of RSA .................................. ...411 Strength of Mechanism ................................................................................................. 412 PGP (Pretty Good Privacy) ........................................................................................... 412 Modern Crypto Systems................................................................................................ 416 Integrated Solution for Error and Security ................................................................. 416 Internet Security ........................................................................................................... 417

Chapter 5: • • • • • • • • • • • • •

Reviewing Information, IT and Looking into Future IT

427

Information and Knowledge ......................................................................................... 427 Proof of Tom Stonier’s Theorem ................................................................................... 431 Tom Stonier’s Theorem With Shannon’s Theorem ..................................................... 432 Proposing Laws of Information .................................................................................... 433 Mass Energy Equivalencey ........................................................................................... 434 Present Imbalance in IT Era, Digital Divide ............................................................... 434 DD Between the Developed and the Developing ........................................................ 436 DD Trend in Future ...................................................................................................... 441 DD Betwen India and China ........................................................................................ 444 DD Within a Country .................................................................................................... 445 DD in Language Zone ................................................................................................... 447 Looking Differently ....................................................................................................... 447 Looking into Future IT ................................................................................................. 452


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1 1.

Information Technology in 21st Century

INTRODUCTION

The basic motivations behind all scientific and technological inventions and discoveries are two: (1) man’s inherent desire to live with the principle of least action and (2) man’s inherent desire to be a master like nature for which they quest for to know what are there in nature’s actions and designs. All the discoveries from the fires to computers conform to the going with the principle of least actions. Man’s aim of becoming the creator or master of all has lead to design or redesign himself or herself which has been manifested in the recent development of cones in laboratory, in continuing research on high speed computing, autonomic computing, quantum computing and in possible designing of intelligent or brainy computer in near future. In the field of communication engineering, its trends of development duly conform to these two basic motivations of discoveries and inventions. To achieve all sorts of communication with least action, the developmental phase of communication has proceeded as: connecting geographically separated but location-fixed machines (conventional wired telephones/fax) to connecting geographically separated but movable machines (chord less /mobile phones) to connecting people rather than machines (communication that supports both man and machine mobility which is personal communication). This is how the total wireless communication is the lust of tomorrow’s communication. In order to achieve the nature like communication, the communication we do in our day-to-day life, PTN/UTN (Personal Telecommunication Number)/ (Universal Telecommunication Number) has evolved out. In the existing communication the connection number changes from location to location and from service to service. We are having separate telephone numbers while at Calcutta than that from while at Delhi. This is not the case in the natural communication. A person is called by his name where he is in Calcutta or in Delhi. A person is called or addressed by his unique name whether it is voice communication or letter communication. Basic motivations behind scientific and technological development have moved the communication research and development on the footings of TOTAL WIRELESS COMMUNICATION and PTN/UTN—in the combined form of Personal Communication Network/Service(PCN/PCS). There are several other parameters including techno-economic and socio-economic aspects that have caused the total wireless communication becoming pillar of tomorrow’s communication; and to name a few are the lower maintenance cost of wireless, easier up gradation and reconfiguration of wireless networks, easier installation of wireless network over difficult regions like over hills and seas, and avoiding threats of theft of costly copper wire used in wired communication. Only existing disadvantage of wireless

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INFORMATION TECHNOLOGY, NETWORK AND INTERNET

communication is the higher initial deployment cost of wireless networks over wired networks and high error rate probability of the wireless links. But over the time and once the maturity of the wireless technology and its systems is attained, these disadvantage will undoubtedly be the past issues. High-speed communication and integrated services are other two important directions of communication technology. High bit rate carriers like SONET and integrated transport technology ATM are future power of communication technology. In the same conformity of principle of least action and man’s earnest desire to be a master of nature, the knowledge age is believed to follow the current information age. The technical capability and the technology are readily available to transform data into knowledge and that is how there emerge challenges of expanding vision to turning from data to knowledge. Actually knowledge age is the next natural consequence of networked age. In the knowledge age, knowledge workers, knowledge factories, knowledge organizations and knowledge economy will be the rule of law. The main wealth of the knowledge age will be knowledge rather than any physical wealth. The subject knowledge management (KM) is therefore will be key issue in the 21st century. This chapter reviews the growth of computer and communication technologies along with knowledge management that are all trying to merge with human axis (Fig. 1)[1], critically analyze the problems thereon, attempts for possible solution and predicts what is there after knowledge age.

2. RECENT PROGRESS OF COMPUTER TECHNOLOGIES Since the inception of electronic classical computer in the year 1948 by the brand name ENIAC, computer has undergone four generations. Present age is of the fifth generation. Hectic research is going on to make “brainy computers”[2,3]. Worldwide research on optical, chemical and quantum technology is being reported [4-6]. Classical computer was the brainchild of Von Neumann. Classical computer is also known as serial computer. Problems of classical computer were two folds: • How to use its power for general purpose small computing jobs thereby having a cost effective solution and raising the system productivity. • How to raise its power, performance and capacity to tackle extensive, complex numerical jobs (for example design of supersonic aircraft, modeling of global weather etc.) where if a serial computer is used, it may take even a year to many years to solve the problem. The solution to the first problem came in the year 1960, with the introduction of timesharing multi-user concept. This was based on the philosophy of utilization of slowness of human as compared to computer, so that if one user is thinking, the computer can be used by other users (resources sharing by time slice). This provided a means of distribution of the cost of computation over many users. Other early solution to the first problem is “batch system” which remained dominant where large amount of data was processed with minimum human interaction (one operator). But as it was not of interactive type, it lost itself to time-sharing system. One of the answers to the second problem gave the birth of parallel computers, which is the ultimate aim of the fifth generation computing system. A few parallel computers are in operation in world. Parallel computing is to speed up operation. With this in mind the concept

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of optical computer was developed. In optical computers it is the light that will carry the signals; and in universe it is the light that has the ultimate speed. Accordingly, non-linear optics emerged as the new frontier of science and technology. The other important deviation from classical computer, that emerged due to technological growth and demand, was the design of “brainy computers”. The chemical computer is a bold step in formulating the “brainy computer”. Optical and chemical computers are now merged under a new field of electronics known as molecular electronics. There are several empirical laws that correlate, govern and predict the technological progress and growth in the last few decades [7-9]. These are: 1. Joy’s law, which states that the computing power, expressed in MIPS (Millions of Instructions Per Second), doubles every 2 years, 2. Ruge’s law estimates that the communication capacity necessary for each MIPS is 0.3-1 Mbps (Million of Bits Per Second), 3. Metcalfe’s law which states that if there are ‘n’ computers in a network, the power of the computers in a network like Internet is multiplied by ‘n’ square times. The law has been applied on Table (1) that lists the growth of Internet users over several years; assuming year 1988 as the reference year, and assuming that in that year the power of a computer was one unit (used for normalization). In that case the power of a computer over different years would be as shown in the table. Assume that each user on average uses only a computer for world access through Internet. Applying the Metcalfe’s law to a lowest extent that the power of individual computer in the Internet is multiplied by square of the number of users in the Internet, the power of computer would be as shown in the last column of the Table (1). From a figure of 0.25 × 1012 in 1988 to 2433600 × 1012 in 2000, a 9734400 (≅107) times increase over a gap of only 12 years! What a future is ahead of ! Super information power or infinite information power! Due to this power, the flexible transport technology, ATM and very high rate carriers like SONET/SDH (Table 2), the requirement of any services at any time at anywhere with a single device and with a single communication number may be possible even through modest Internet, which was basically designed to carry data only. Table 1: Trend in Internet/Computer power Year

Internet Users in 106

1988

0.5

1

0.25

1989

1.3

1.5

2.535

1990

2.4

2

Computer power normalized to year 1988 on standalone condition

Computer power on networking in 1012

11.52

1991

4.4

3

58.08

1992

8.7

4

302.76

1993

14.8

6

1314.24

1994

26.1

8

5449.68

1995

49.2

12

29047.68

2000

195

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2433600

4


Computer Technology Progress

Brainy Computer Human Axis Year

Personal Communication

Communication Technology Progress

Fig. 1: Trends of Computer and Communication Technology.

Table 2: Bit Rates of Digital Hierarchy North American Type

European Type

Bit rates

Type

E1

2.048 Mbps or 2 Mbps

DS0

T1 or DS1 1.544 Mbps

E2

8.448 Mbps or 8 Mbps (≅ 4 × 2 Mbps)

J1

1.544 Mbps

T2 or DS2 6.312 Mbps (or 4 × 1.5 Mbps)

E3

34.368 Mbps or 34Mbps (≅ 4 × 8 Mbps)

J2

6.312 Mbps (≅ 4 × 1.5 Mbps)

T3 or DS3 44.736 Mbps (or 7 × 6 Mbps), sometimes referred to as 45

E4

139.264 Mbps or 140 Mbps (≅ 4 × 34 Mbps)

J3

32.064 Mbps (≅ 5 × 6 Mbps)

T4 or DS4 (1) 139.264 Mbps (or 3 × 45 Mbps) (2) 278.176 Mbps (or 6 × 45 Mbps)

E5

564.992 Mbps or 565 J4 Mbps (≅ 4 × 140 Mbps)

DS0

Bit rates

Other (used predominantly by Japan)

64 Kbps

Bit rates 64 Kbps

97.728 Mbps (≅ 3 × 32 Mbps)

4. Moore’s laws state that (a) the number of components on an IC would double every year (this is the original Moore’s law predicted in 1965 for the then next ten years), (b) the doubling of circuit complexity on an IC every 18 months (this is known as revised Moore’s law), (c) the processing power of computer will double every year and a half (Moore’s second law which closely resembles to Joy’s law). 5. Law of “Price and Power” that states that over the years the computing, processing, storage and speed up power of computers will continue to increase whereas the price of computers will continue to fall. 6. For a new law of communication, readers may refer to Appendix-A. In table (3), a list of computer generations with power in terms of information processing, storage and speed up factor is given. It is seen that first three laws fit well into the list. In

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pace with increased processing power in terms of volume and speed, and the wide and flexible use of computers, the communication transport technology and transmission media have been developed. Table 3: Computer power over years Generation of Intel processors Processor

Number of Transistors in the chip

Word length in bits

Internal bus size in bits

External bus size in bits

8080

8

8

8

8088

16

16

8

8086

16

16

16

80286

134,000

16

16

16

i386

275,000

32

32

32

i486

1,600,000

32

32

32

32

64

32

P24T Pentium

3,300,000

32

64

64

Celeron

4,000,000

64

64

64

Pentium Pro

5,500,000

64

64

64

Pentium with MMX (multimedia) Technology

4,500,000

64

32

64

Pentium II

7,500,000

64

64

64

In the chip level integration till date, Moore’s laws say the last word. From SSI to ULSI, the trend set (Table 4) by Moore’s law is followed. But beyond ULSI, what is there? The extrapolation of the trend predicts that the future will be the age of molecular dimension inherited by the already established subject of molecular electronics that is based on organic materials rather than inorganic semiconductor. Beyond ULSI, the further integration on a chip will face serious problem from physical constrain like the quantum effect. This may lead to the death of Moore’s law. But another interesting dimension may be added to the cause of the death of Moore’s law. This is based on the law of “Price and Power”. It is said that: “ The price per transistor will bottom out sometime between 2003 and 2005. From that point on, there will be no economic point in making transistors smaller. So Moore’s law ends in” a few years. “In fact, economies may constrain Moore’s law before physics does.” Table 4: Generation of IC integration Generation

Number of components

Small Scale Integration (SSI)

2–64

Medium Scale Integration (MSI)

64–2000

Large Scale Integration (LSI)

2000–64,000

Very Large Scale Integration (VLSI)

64,000–2,000,000

Ultra Large Scale Integration (ULSI)

2,000,000–100,000,000

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2.1 Newer Technologies To go beyond the conventional laws mentioned above, the computer technology has taken a few new direction: (1) recently reported Intel’s Terahertz Transistor, (2) molecular electronics, (3) autonomic computers and (4) quantum computers. 2.1.1 Terahertz Transistor The Intel’s terahertz transistor is reported to be a new method of making transistors with a new class of material to overcome the problem of heat dissipation and quantum effect. This transistor will save power and provide miniature chips. The transistors will stay cooler and be smaller in size but with faster operating speed. This is made possible, as the new method will be based on innovative design that will eliminate leakage. It is believed that by 2007, this transistor will be available in the market. 2.1.2 Molecular Electronics The subject of molecular electronics has emerged as an important area of research and application during 1980’s [10]. The definition of molecular electronics is not unique and simple. Even within a country scientists differ. A leading scientist of the field [11] “molecular electronics can be divided into two main themes: these are molecular materials for electronics (MME) and molecular scale electronics (MSE). The topic of molecular materials for electronics deals with the use of macroscopic properties of organic materials in devices, and includes current and near-term application. In the near-term it seems likely that conductive polymers will offer the prospect of novel electronic devices and that organic materials with pronounced non-linear optical properties will find application in upto electronics. A simplistic extrapolations of the reduction of time leads eventually to the molecular scale i.e. molecular scale electronics.” Prof. Bloor further observed [12] “many regard the quest for molecular scale devices as true molecular electronics. However it can be argued that the distinction between MME and MSE is somewhat arbitrary and that both need to be considered as constituent parts of molecular electronics if the topic is to grow and prosper.” Ashwell, Sage and Trundle [13] defined that “its definition has broadened from electronics at molecular level to include molecular materials with potential electronics and photonic applications.” Peterson defined that “in the most general sense, molecular electronics covers the use of molecular (and hence essentially organic) materials to perform signal processing or transformation function.” However, the famous Link programme of Britain defines molecular electronics as [14]” systematic exploitation of molecular, including macro molecular, materials in electronics and related area such a photo-electronics.” The molecular electronics is therefore to explore the potential application of organic materials and non-linear optics in the field of electronics. It is a highly interdisciplinary field and prospects lies on the successful interaction and co-operation of scientists of different fields like biology, chemistry, computing, physics and electronics. 2.1.2.1 History of molecular electronics In history the concept of molecular electronics dates back to the last century. The familiar example is the use of organic materials in displays. The use of liquid crystals display found in watches, calculators and TV sets in historically patented over fifty years ago [3]. As Prof. Bloor pointed out [15] that “molecular exhibit great variety in their structure and properties from simple diatomic species through to very large synthetic and bio-macro-molecules. It is not surprising therefore that molecules can be found that process unique combination of properties which find application in fields of electronics and opto-electronics.” This idea stimulated work on MME since 1950s. The reduction of size of active electronic device compound problems in

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regard to quantum effects.. At this juncture, molecular electronics, the application of molecular materials in electronics, started exploiting some of the new advanced technologies that may be beyond the scope of the silicon chop. Prof. Bloor explained [16-17] that “the continuing development of silicon micro-electronic devices of smaller size and grater complexity has brought more compact and powerful instrumentation and computing facilities into the laboratory and office. Though silicon technology holds a dominant position the continuing reduction in dimensions of an individual device creates problems both at the fundamental and systems level. On one hand quantum effects must ultimately come into play dissipation and the design of testable architectures are already with us. These pressures lead inevitably to a search for alternatives to current technology that can offer prospects for the realization of devices with even higher densities of active components. MSE is one avenue which is being explored with these targets in mind.” The research and the interest in molecular electronics were mainly initiated by the late Forest Carter who conducted a series of international conferences on molecular electronics [18-20] in 1980s. Prof. Bloor wrote that [21] “organic solids have attracted the interest of materials scientists and solid-state physicists since the 1950s both as alternative semiconductor and because of their optical properties. Strong research groups grew up in the USA, Russia, Germany and France at this time.” Although the progress of molecular electronics has not always been smooth, yet the prospects for the future are good. In this article, we shall review the present position and future aspects of molecular electronics. 2.1.2..2 Molecular Materials for Electronics (MME/M2E) The study of MME is to see the use of molecular materials in key and active roles in electronic and opto-electronic devices and systems. It is based on understanding and use of macroscopic properties of the bulk molecular materials i.e. of the organic materials. The main categories of MME are[22] • Organic semiconductors and metals • Liquid crystalline materials • Piezo/pyro-electric materials • Photo/Electro-chromic materials • Non-linear optical materials/photonics. Organic Semiconductors and Applications Organic semiconductors and metals have been much less studied than their inorganic counterpart. Under MME, a good study is gradually emerging. The major applications of organic semiconductor are in (1) electronic active devices and (2) xerography. Therefore before going to organic semiconductors, the process in amorphous materials is required to be studied. What are amorphous materials? In crystal, atoms or molecules are arranged in a regular structure with periodicity. But in amorphous materials there is no ordered structure. The developments of electronic devices in last few decades were tremendous because the electrical conductivity of crystalline semiconductors such as silicon can be controlled over much order of magnitudes by doping. But [23] “there are a number of areas where the expenses of preparing. These crystals and where the limited size to which they can be grown (at present about 25 cm in diameter) have prevented any very large-area applications. For example,

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crystalline silicon solar cells are widely used in space vehicles for converting sunlight into electrical power, but the economics of their production is such that their use here on earth is relatively limited. Silicon can be prepared very cheaply in large areas by vacuum evaporation or by sputting, but the materials is then amorphous rather than crystalline ……… sine (the) work on doping amphorous silicon (a-Si) was published, there has been a considerable research into and development of this materials, leading to a member of commercial products.” Table 1 [46] shows a progress list. MME makes a study with electronic processes as distinct from ionic processes, in organic crystals. What are organic crystals? By organic we usually mean a compound containing carbon. Almost 90% of 2 millions compounds known to us are organic. But for MME, there is choice and limitation that need a careful study. Till today organic materials have not presented to be a real competitor to the silicon/ inorganic material in terms of active electronics devices. However, during last five years the progress in the synthesis of high purity semiconductor polymers and oligomers is note worthy. Experiments showed that conductive polymers could be employed as either metallic or the semi conducting component of metal-semiconductor junction devices [14]. “semi conducting polymers can be used to produced Schottky diodes [6]. Where the polymer has temperature dependent properties have been observed, with rectifying behavior at room temperature changing to ohmic behavior above 100°C [15]. Burroughes et al. first reported an active polymer transistor in 1988 [16,17]. The important characteristic of this device were: (1) no chemical doping or side reactions and (2) the characteristic of the polymers device was insensitive to disorder. But the major disadvantage of the device was that its maximum operating frequency was limited. This is because the carrier mobility in the amorphous polyacetylene layer is very low. The mobility’s of electrons in semi conducting polymers, amorphous silicon and crystalline silicon are of the order of 10–4, 1 and 103 cm2 /Vs respectively. One can see the large gap between properties of polymers and silicon. However a dramatic lead was done by Frincis Garnier and co-workers [18-19]. They reported a totally organic transistor. This transistor is known as thin film transistor (TFT) or organic FET. This transistor is a metal insulator semiconductor structure comprising an oxidized silicon substrate and a semi conducting polymer layer. It has grater flexibility and can even function when it is bent (disorder is acceptable). The operating speed is still poor. The problem of low carrier mobility of insulating polymer is under active research. The diodes made of semiconductor with rectification’s ratios in excess of 103 have been reported in [23], and light emitting diodes, made in organic semiconductor with external quantum efficiencies in excess of 1% photons per electrons are reported in [16-22]; and organic photovoltaic cells are reported in [19-22]. However, within a short period, a rapid progress has been observed on use of semi conductive polymers and oligomers in electronic devices. If this progress is maintained, in near future it could be competitive to silicon. The field of optical computation starts with the search of a bi-stable optical switch based on non-linear optical properties of materials. Non-linearity can be used for device basically by two techniques: frequency conversion and reflective index modulation. The frequency conversion technique, which is due to second order non-linearity, may be used to second harmonic generation frequency mixing and parametric amplification etc. Refractive index modulation

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particularly Kerr effect which is due to third order non-linearity may be used for optical bistable switches and parallel processing. Till date a few optical gates and all optical bi-stable switches have been reported, but the field is still confined in the laboratories. Yet optical computation is a promising field. Optical computing and processing of information are the important application of photonic. The gain of photonics switching speed (of order of femto second 10-12 ) is many order of magnitudes over that of electronic switching. Optical processing is free from interference from electrical or magnetic sources. “Based on the prospect of three dimensional interconnectivity between sources and receptors of light concepts of optical neural networks that mimic the fuzzy algorithms by which learning takes place in the brain have been proposed and experimentation has begun. Integrated optical circuits, which are counterparts of electrical circuits photons, can provide for various logic, memory, and multiplexing operations. Utilizing non-linear optical effects, analogs of transistors or optical bistable devices with which light controls light have also been demonstrated” [23]. So far nlo materials are concerned, all materials in forms of gases, liquids or solids, exhibit nlo phenomena. However broadly we can defined two classes of nlo materials : (1) molecular materials or organic materials which “consist of chemically bonded molecular units that interacts in the bulk through weak van der waals interactions” and (2) bulk materials and traditional inorganic materials. Today rapid progress and research in organic nlo materials proved to be attractive. The nlo devices utilize two different techniques: frequency conversion and refractive index modulation. Based on letter effects, the developments of frequency converter and light modulator have been reported in [23]. However organic materials are seen to be quite attractive for electro-optic light modulation as “their low -frequency dielectric constant is quit low leading to a small RC time constant, thus permitting a higher bandwidth for light modulation compared to that achievable using inorganic materials.” The application of second order non-linearity needs that the crystal must not be centrosymmetric structure. In centrosymmetric structure the non-linearities, which are vectorial, cancel each other to give zero microscopic effect. This is a stumbling block in the progress of application of second order non-linearity. To solve the problem two approaches are being examined: 1. Use of LB films “with either alternating layers of a polar molecule or molecules which inherently from polar multi-layers, 2. Inclusion of “non-linear optically active molecules in polymer films which are poled with an applied electrical field.” In a single way, a materials with a bulk where, its molecules are non-centrosymmtric nature may be defined as anisotropically oriented over volumes measure in cm3 . “These conditions are best achieved by growing a crystal. The Langmuir-Blodgett (LB) technique is a comparable high tech organic fabrication method, appropriate when the implementation of the function requires a high degree of molecular anisotropy in an extremely thin layer of uniform thickness. For OICs, particularly for single processing, L-B technique offers the possibility to orient molecules with in a thin layer of highly precise thickness. It has thus become an attraction. However films are not the final answers. There are many drawbacks with films namely mechanical softness, limited high temperature range and extremely slow rate of deposition etc. But rapid research is going on L-B film technology and its application in molecular electronics materials both for ME and MSE .

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2.1.2.3 Molecular Scale Electronic (MSE) The quest for an ever decreasing size but more complex electronic components with high speed ability gave the birth of MSE. The concept that molecules may be designed to operate as a selfcontained device was put forwarded by Carter, and he only proposed some molecular analogous of conventional electronic switches, gates and connections [9]. Accordingly Aviram and Ratner first advanced a molecular P-N junction idea. MSE is a simple interpolation IC scaling. Scaling is an attractive technology. Scaling of FET and MOS transistors is more rigorous and well defined than that of bipolar transistor. But there are problems in scaling of silicon technology. In scaling on the one hand propagation delay should be minimum and packing density should be high; on the other hand these should not be at the expenses of the power dissipated. With these scaling rules in minds, scaling technology of silicon is to reach a limit. Another thing is that scaling can due to the quantum nature of physics. At this junction molecular scale scaling technology. Dr. Barker reported in [9] that “change, spin, conformation, color, reactivity and lockand-key recognition are just a few examples of molecular properties, which might be useful for representing and transforming logical information. To be useful, molecular scale logic will have to function close to the information theoretical limit of one bit on one carrier. Experimental practicalities suggest that it will be too easiest to construct regular molecular arrays, preferable by chemical and physical self-organization. This suggests that the natural logic architectures should be cellular automata: regular arrays of locally connected finite state machines where the state of each molecule might be represented by color or by conformation. Schemes such as spectral hole burning already exist for storing and retrieving information in molecular arrays using light. The general problem of interfacing to a molecular system remains problematic. Molecular structures may be the first to take practical advantages of novel logic concepts such as emergent computation and ‘floating architecture’ in which computation is viewed as a selforganizing process in a fluid-like medium.” MSE spans several disciplines and requires a co-ordination of scientists of different group if the subject is to grow and prosper based on cross fertilization of ideas of different subjects. But problem is how can the properties of individual molecules and/or small aggregates be studied? Fortunately day-by-day we are evolving new techniques and methods to tackle this problem. At present we are having technologies like STM (scanning tunneling microscope) AFM(atomic force microscope) and NFOM (near field optical microscope) etc. In addition, submicron lithography, L-B films and adsorption/reaction in 2D/3D are also there. L-B technique is particularly important because it provides one of the few ways of marketing separate electrical connection to two ends of a molecule. A very good illustration of molecular electronics logic and architecture can be seen in [10]. 2.1.2.4 Bio/Chemical Computer A new radical information processing system is being thought of where organic cells or bacteria are to act as the basic element. Living organisms are made of organic compounds. As such thinking function can be easily realized in such system. As scaling will be at biological level, very high-density circuit can be at biological level, very high density circuit can be achieved. Our average brain comprises 1011 neurons ranging in size from 0.2mm linear dimension to

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about 100 mm, each with an average connectivity of 104 giving a crude bit-count of 1011 to 1015. An equivalent artificial brain may therefore be of such dense circuit. Enzymes and proteins are being studied. We should not forget that an example of a natural molecular device. Is the bacterial photo-reaction center. Recent research to produce analogous have been successful through the synthesis of single and complex molecules, which release charge on photo-excitation. This subject of molecular electronics has moved from conjuncture to experimental study and scientific development. With the rapid growth of research and development of few liquid crystals, polymers, L-B films and NLO materials; molecular electronics is now with us. With advances in Physics, Chemistry, Materials Science, Biology and Engineering as our understanding of molecular materials both at microscopic and microscopic level with grow; the field of molecular electronics will prosper. The better understanding of natural system and processes and living organisms, will enhances the capability and potentiality of molecular electronics particularly in terms of its application in radical new computational machines and engineering. Much more work remains to be done. It needs scientific, intellectual and technological challenges on one hand; and Government and Industrial supports on the other hand. The progress of all these will determine actually whether molecular electronics if so, when. But research in molecular electronics and device technology it, will emerge as exciting and frontier fields of science and technology in the current century. The molecular electronics is a revolutionary idea. To attain maximum miniaturization, it is proposed that instead of using transistor’s states, namely ON and OFF to implement 1s and 0s, the characteristics of electrons may be used for the same. For example, the positive and the negative spin be respectively used to implement 1s and 0s. The idea is new. It will take lots of time to mature and to develop the technology. This will be the last resort of miniaturization. The molecular electronics is believed to be based on new organic material technology that may lead to bio or chemical computer. A new radical information processing system is being thought of where organic cells or bacteria are to act as the basic element. Living organisms are made of organic compounds. As such thinking function can be easily realized in such system. As scaling will be at biological level, very high density circuit can be achieved. Our average brain comprises 1011 neurons ranging in size from 0.2mm linear dimension to about 100 mm, each with an average connectivity of 104 giving a crude bit-count of 1011 to 1015. An equivalent artificial brain may therefore be of such dense circuit. Enzymes and proteins are being studied. We should not forget that an example of a natural molecular device is the bacterial photo-reaction center. Recent research to produce analogous have been successful through the synthesis of single and complex molecules, which release charge on photo-excitation. However while the above new technologies aim to attain miniaturization going in line and/or beyond Moore’s law, the autonomous computing technology aims at the economic aspect of technology. 2.1.2.5 Autonomic Computing Consider the computing paradigms of the Internet. Fig. 2 and Fig. 3 show the exponential growth of Internet users and Information Technology. It is therefore understood the need of huge technologists to keep on running Internet without much disruption of services. A statistics says: “At current rates of expansion, there will not be enough skilled IT people to keep the world’s computing systems running. Even in uncertain economic times, demand for skilled IT workers is expected to increase by over 100 percent in the next six years.”

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Growth Rate of Internet Users

Internet users in thousand

250000 200000 India

150000

USA 100000

UK

50000 0 1997

2002

2004

Year

Fig. 2: Growth of Internet Users. IT as % share of GDP in India : Source NASSCOM 3.5 3.15

3

2.87

2.66 2.5 2 1.87 1.5 1

1.45 1.22

0.5 0 1997-98

1998-99

1999-00

2000-01

2001-02

2002-03

Fig. 3: IT growth related to economy.

Under such a scenario, it is not unbelievable to believe that there might be an exponential relationship between the growing complexity and power of the computing systems and the technical manpower required to manage and administer them. A new paradigm to relieve humans of the burden of managing, administering, and maintaining the computer systems, and thereby passing these back to computers is to design “Computers that help themselves”, now known as Autonomic Computers. Consider how we, the humans do act when we face problems. When we are physically attacked, we protect ourselves. This solution uses a biological metaphor. Just as the autonomic nervous system of our bodies monitors, regulates, controls, repairs and responds to hazardous conditions without any conscious effort on our part, so the autonomic computer systems. The autonomous computers are to self control, self monitor, self regulate, self-repair and respond to problematic conditions, again without any conscious effort of humans.

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The autonomous computing technology therefore is a major deviation from the conventional rules like Moore’s law. The aim is not to attain more complex, more integrated, more powerful computers but self healing computers that will be economic in terms of maintenance and operation. The key characteristics of an autonomic computer systems system are: • They should be able to fix failures, and able to configure and reconfigure themselves under varying, undefined and unpredictable conditions so that they prevent system freezes and crashes • The systems should known themselves fully and comprise components with proper identity • The systems should work always in optimize conditions and adopt itself accordingly to varying conditions • The systems should be self healing, self correcting and capable of recovering from common, routine and extraordinary, known and unknown events that might cause some of its parts to malfunction or crash • The systems should be self protective against unwanted intrusion • The systems should be expert to know its environment and the surrounding activity, and act accordingly in order to easy recovery from crashes and interoperations • The systems should adhere to open standards to ensure interoperability among myriad devices • The system should better prevent themselves from failures at first place • The systems should optimize resource in anticipation while keeping its operation hidden to users. The self-managed computers will have four major components (Fig. 4): • Self optimized—components and devices of the system will automatically and continually check their performance and seek to improve the same • Self configurable—components and systems will automatically configure and reconfigure to required adjustments seamlessly • Self healing—system will automatically detects and repairs localized problems • Self protected—System automatically protects itself from intentional attacks

Self Healing Self Optimized Self Configurable Self Protected

SELF MANAGED /AUTONOMOUS COMPUTER

Fig. 4: Autonomous Computer.

2.1.2.6 Quantum Computing The conventional computing is based on the concepts of bits. The bits in the classical computation may have two possible states 0 and 1. The fundamental concept of the quantum computing is

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the quantum bits, referred to as qubit. Two possible states of qubit are |0> and |1>. Like binary bits of the classical computing, all possible superposition of qubits are possible. Therefore, a two qubit system has four computational states, namely |00>, |01>, |10> and |11>. With Moore’s Law being saturated, it is expected that quantum computers will be one of the future solutions for high speed and high power computing. A few theoretical work has been reported but practical implementation is yet to reach. However an important milestone in application of quantum computers has been achieved due to pioneer work of Bennett et al in quantum cryptography in the area of data security. BOX 1 Quantum Computing: a bit review QUANTUM GATES The information processing in the quantum computing has a component of qubit manipulation. The qubit manipulation is performed by unitary operations. A quantum logic gate is a device that performs a particular unitary operation on the selected qubits at a given time. There are infinite numbers of single-qubit quantum gates unlike only two (identity and the logical NOT) in classical information. The quantum NOT gate performs |0> to |1> and vice versa analogous to classical NOT Two-qubits quantum gates performs many possible unitary operation, an interesting subset of which is |0> or |1>. This gives to define controlled NOT, CNOT gate as: |00> |00> |01> |01> |10> |11> |11> |10> this shows that: (a) second qubit undergoes NOT if and only if the first qubit is in state |1>; (Fig. 1) (b) the effect of CNOT on states |x> |y> may be written as : x ® x, y → x⊕y for the reason of which this gate is also called. XOR gate Fig. (1). X

Xo = X

Y

Yo

Y

Notes: (a) x-wire means NOT but controlled by o-wire. (b) Each horizontal line represents a single qubit evolving in time from left to right A symbol on a line represents a single qubit gate. (c) A vertical line connects two or more qubits. Symbols on two qubits connected by a vertical line represent a two-qubits gate on those two qubits. CNOT GATE: The output, Yo at x-wire is controlled by the input, X of the o-wire. When input to o-wire is |1> , the output Yo of the x-wire is NOT of its input state, Y. XOR GATE: Whatever the first qubit, the output second at the x-wire is always XOR of the two input qubits. Fig. 1: CNOT/ XOR gate.

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Other logical operations do require additional qubits. The most popular three qubits gate is Controlled- Controlled NOT gate/CCN or C2NOT gate (Fig. 2). This gate is also known as Toffoli gate that demonstrated that the classical version is universal for classical reversible computation. A gate is reversible when for a given output; one can reconstruct the input(s). The output of the gate on o-wire can be described as: (a) if third qubit is in state |0>, then output is AND of two other qubits. The effect on the input states |x> |y> |0> is x → x, y → y and output → x.y. (b) the effect on the input state |x> |y> |1> is that output is XOR of x and z, (c) the effect on |1> |1> |z> is that output is not of z. X

Xo = X

Y

Yo = Y

Z

Zo = Z

Note:

Sum generation

(X.Y)

to mean toggle control. Others as in Fig. 1. Fig. 2: Controlled Controlled NOT/ CCN gate.

It has been argued that any logic circuit can be made of only CN and CCN gates only. For example, Fig. 3 illustrates a half adder circuit.

X

Xo = X

Y

Yo = X

Z

Zo = (X.Y) Carry generation

Fig. 3: Half adder using CN and CCN gates.

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Table 2 Superposition

In general this means that two things can overlap with each other with interfering with each other. In quantum mechanics two electrons can overlap with each other making a combined waveform that is a set of amplitude probabilities.

Principal ideas of quantum physics

Energy is in discrete units. Photons are each a discrete bundle of energy. A photon of characteristics frequency,n carries a quanta of energy equals to h.n where h is the Planck’s constant. The particles in quantum physics behave both a particles and waves. The state vector of particles obeys Schrodinger wave equation.

Uncertainty principle

It is impossible to measure both the position and the momentum of the particles at the same time. More accurately one is measures, less precisely the other is known.

Entanglement

With entanglement the systems are correlated in a way that does not involve force and the restriction of the speed of light is not applicable .

QUANTUM TELEPORTATION Teleportation is by which an object or person while physically remains present in one place, is made to appear as a perfect replica somewhere else. The classical or conventional approach of teleportation is illustrated in Fig. 4. Fax machine is an example of teleportation machine. Till recently the quantum teleportation was assumed impossible as it would violate the uncertainty principle of quantum mechanics. The uncertainty principle prohibits any scanning or measuring process to extract all the information in an atom or such object. As the more accurately an object is scanned, the more accurately the object is disturbed that may ultimately lead to complete change of the original state of the object even before the whole of information is extracted to make a perfect replica of the original one. But quantum mechanics has an aspect known as entanglement. If outside force is applied on two atoms, the aspect of entanglement occurs whereby the second atom can take the properties of the first atom. Thus if left alone, an atom will spin in all directions; but the instant it is disturbed it chooses one spin, or one value; and at the same time, the second entangled atom will choose an opposite spin or value. This allows learning the value of qubits without actually looking at them, which could collapse them back into 1’s or 0’s. Sending Station Original object, A physically present at location, P Receiving Station A replica of the original Object A is Generated / Received at a location, Q away from P

A is Scanned or Processed

Send Data

Original, A remains intact at the sending location

Fig. 4: Classical Teleportation/FAX.

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Apply treatment

Raw Material


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The property of the EPR (Einstein Podolsky Rosen) or “entanglement’ has made the quantum teleportation possible hurdling the principle of uncertainty. Fig. (5) illustrates the quantum teleportation. In the process, part of the information of the original object is scanned out. The un scanned part of the information is passed viz EPR effect into anther object C. the object C was never in contact with the original object A. the intermediary object or the delivery vehicle, B conveyed the un scanned part of information from A to C. It is now possible to apply treatment on C to make it as A before A, was disrupted by the scanning process. So a real transportation is achieved in C rather than replica. Sending Station Original object, A physically present at location, P Receiving Station A replica of the original Object A is Generated / Received at a location, Q away from P

A is Scanned or Processed

Apply treatment

Send Data B

Original, A becomes completely disrupted

C

Entangled pair, B and C

The intermediary object, B

Fig. 5: Quantum teleportation.

QUANTUM CRYPTOGRAPHY The disadvantage of key distribution in secret key cryptography can be removed with the aid of quantum technology. If key distribution problem is solved, the use of Vernam technique will be best technique of security. In order to solve distribution problem, use of quantum channel for sending information about key is being explored. In quantum mechanics one cannot measure something without causing noise to other related parameter. For example Hysenberg’s uncertainty principle state that ∆x.∆m.= constant. Thus if ∆x. is changed, ∆m is bound to change. An ideal quantum channel supports transportation of the single photon. Thus a single photon can represent a bit 0 (zero) or 1 (one). The phase or state of polarization of photon may be used for identifying the 0 or 1. For example. Photons with 0° and 90° of polarization may therefore be treated as bit 0; and photons with 45° and 135° (also known as – 45°) of polarization may be assumed as bit 1. Data security through quantum channel is under active research in the UK and USA. Some positive breakthroughs have been made by Charles Bennet of IBM Research at Yorktown Heights, New York, and by Gilles Brassard at the University of Montreal. If, in the example discussed earlier, Alice wants to send Bob the secret key as required in the Vernam cipher, she can send the key, say of N bits, through quantum channels. Bob will be instructed by Alice to detect the photons (bits) from the quantum channel starting from a given time. There may be some transmission loss, and Bob may be able to detect some fraction of photons or bits. Bob will have to inform Alice over a telephone as to which photon he has seen. For this, they may share both a common and variable key. For instance, if Alice sends

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11110000 as the key, and Bob replies that he has seen the first, seventh and eighth photons (starting from the leftmost bit), then their common key shall be 100. Alice can send data haphazardly using different polarized photons. Alice can do so (Fig. 6) either on rectilinear basis: When a horizontal polarized photon represents a 0 and a vertical polarization represents a 1 Or on diagonal basis: When a – 450 polarized photon represents a 0 and a + 450 polarized photon represents a 1.

=1

=0

Fig. 6: Use of polarization for representing 1s and 0s typically.

Alice haphazardly uses both to send qubits (Fig. 7). Bob will haphazardly try to filter out the qubits. For the purpose of qubits detection Bob will use a polarization beam splitter. The polarization beam splitter is a device that allows the photons of orthogonal polarization to pass through but shunts the photon of other polarization. The quantum nature dictates that: (a) the same basis beam splitter will pass the received same basis polarized photons, but (b) the rectilinear beam splitter will pass the received diagonally polarized photons either as vertical or horizontal polarization with equal probability and the diagonal beam splitter will pass the received rectilinear polarized photons either as vertical or horizontal polarized photons with equal probability. This will provide the different combinations of Alice’s sent photons and Bob’s detected photons. Therefore when both Alice and Bob use the splitter on same basis they with correctly communicate qubits, but when they use on different basis, the chance of matching between sent and received qubits is 50%. Bob now tells Alice (over conventional method, say telephone, as there is no need to keep secret these) how he used the beam splitter to detect received qubits. Assume Bob’s choice was as rectilinear, rectilinear, diagonal, rectilinear, diagonal (Fig. 7). Bob does not announce the results of detection. Alice replies publicly (means over conventional method as there is no need to keep this secret) Bob, which times her choices of base match with Bob’s choices. Then they use the qubits of those instant when they use same base (in those instant they correctly communicate the bits), and ignores the bits of other instants. The matching bits (Fig. 7) generate the secret key for the session.

(Rectilinear) (Diagonal) (Rectilinear) (Rectilinear) (Diagonal)

(a) Alice sends qubits to Bob randomly (we have taken only 5 qubits for illustration)

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(Rectilinear) (Rectilinear) (Diagonal) (Rectilinear) (Diagonal)

(b) Bob measures the received photons using random polarization basis Same Base

Different Base

Different Base

Same Base Same Base

(Uncertain) (Uncertain) (Correctly (Correctly (Correctly detected by detected by detected by Bob) Bob) Bob)

© Alice and Bob Communicate and identify locations whether they correctly used the polarization base. COMPARE (a) with (b). BUT THEY KEEP SECRET THE POLARIZATION OF SENT OR RCEIVED PHOTONS. 1

Ignored

Ignored

1

1

(d) Correct bits are taken for key. Bits of other positions are ignored. So the key in this example is 111. Fig. 7: Key exchange between Alice and Bob.

Should any eavesdropper attempt to intercepted photon transmission; there shall be garbage with the key accepted by Alice and Bob. This is because the quantum theory ensures that, without changing the phase of the photon, an intercepted photon cannot be retransmitted. Therefore, a change in the polarity of the photon will let Alice and Bob immediately known of an interception. The scheme of sending information at the one-photon-per bit level as proposed by IBM research and research of university of Montreal reported that “to send the key, the transmitter (Alice) tells the receiver (Bob) that the plans to send n bits (photons) starting at a given time. Alice than sends the bits by randomly switching the phase in the transmitter between 00 to 1800; this switches the output in the receiver between “0” and “1”. Although transmission and detection losses mean that Bob will only see a small classical communication channel (the telephone, for example) to tell Alice which photons he has seen—but not which detector he has seen than in. This allows Alice and Bob to share the same random number. For example, Alice uses ten photons to send the random number 1001011101; Bob replies that he only received the second, fifth and last photon; therefore they have shared the random number 001. However, it is conceivable that an eavesdropper could intercept the signal, copy Alice’s message, and send it on to Bob without either Alice or Bob realizing. One way to overcome this, and ensure absolute security, is for both the transmitter and receiver to use non-orthogonal measurement bases. In other words, Alice sends parts of the message by switching the transmitter phase between 900 and 2700, say, and other part by switching between 00 and 1800. When the Bob and Alice are using the same base, the system works as before. However, if Alice is using 00/1800 and Bob is using 900/2700 (or vice versa), the message is meaningless

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- a photon that Alice sends as a “0” has a 50% chance of being received as a “1” and vice versa. Therefore when Bob tells Alice which photons he has received, he now also says which base he was using and Alice must tell him if that is a valid photon (i.e. one which was sent and received when they were both using the same base). Paul Townsend of British Telecom, working with the Malvern group, recently demonstrated self-interference of short light pulses, containing on average 0.1 photons, down 10 km of standard communications fiber using the technique.” There is anther technique to minimize the hacking by Eve. The technique is known as privacy amplification protocol. In the protocol, Alice randomly chooses pair of bits from the key they have got over quantum channel. Then she performs XOR on the pairs. She then tells publicly to Bob on which bits the XOR operation was made but not the results. Bob then performs the XOR operation on the bits that Alice informed him. Alice and Bob then replace the pair with XOR results to design the new key. The is illustrated as below: (a) Alice and Bob have secret key 111 as in Fig. 7. (b) Alice chooses first and second bit as pair and she informs these to Bob publicly. She gets XOR result 1 ⊕ 1 = 0 and keeps it secret. (c) Bob performs XOR on the informed bits and get the result 1 ⊕ 1 = 0. (d) Alice and Bob both replace the pair by XOR result. So their new key = 01. (e) Note that even if Eve definitely knows one bit of the chosen pair, until & unless she gets the result of XOR (which Alice and Bob never communicates) she can not replace the pair for hacking the key. Quantum computer is very promising. It has numerous advantages over classical computers, namely in terms of speed (parallelism inherent in quantum computer), power consumption (nearly at the half of classical computer due to superposition), and tackling of computational problems here to impossible with conventional computers. The quantum computer will be based on quantum logic gates based on quantum circuit, and the technology for these is even prior to the infancy stage. On the other two problems of the quantum computers have been identified. It is estimated that the quantum error correction will generate more power than the chips can dissipate; the technology of quantum computer may not be so easy to develop. The problem of decoherence intervals that measure how long a qubit can maintain synchronized waveform to represent either 1 and 0 simultaneously. The decoherence time is estimated on average to be less than 1 microsecond. The challenge remains how to increase this interval time. Yet there is no stop, and shall not be a stop in development of quantum computer. We will be wrong to think that the quantum computers will replace classical computers. The quantum physics has not replaced the classical physics. They co exist each within their own parameter.

2.2 Quantum Security The disadvantage of key distribution can be removed with the aid of quantum technology. If key distribution problem is solved, the use of Vernum technique will be best technique of security. In order to solve distribution problem, use of quantum channel for sending information about key is being explored. In quantum mechanics one cannot measure something without causing noise to other related parameter. For example Hysenberg’s uncertainty principle state that Dx.Dm.= constant. Thus if Dx. is changed, Dm is bound to change. An ideal quantum channel supports transportation of the single photon. Thus a single photon can represent a bit 0 (zero) or 1 (one). The phase or state of polarization of photon may be used for identifying

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the 0 or 1. For example. Photons with 0° and 90° of polarization may therefore be treated as bit 0; and photons with 450 and 1350 of polarization may be assumed as bit 1. Data security through quantum channel is under active research in the UK and USA. Some positive breakthroughs have been made by Charles Bennet of IBM Research at Yorktown Heights, New York, and by Gilles Brassard at the University of Montreal. If, in the example discussed earlier, Alice wants to send Bob the secret key as required in the Vernam cipher, she can send the key, say of N bits, through quantum channels. Bob will be instructed by Alice to detect the photons (bits) from the quantum channel starting from a given time. There may be some transmission loss, and Bob may be able to detect some fraction of photons or bits. Bob will have to inform Alice over a telephone as to which photon he has seen. For this, they may share both a common and variable key. For instance, if Alice sends 11110000 as the key, and Bob replies that he has seen the first, seventh and eighth photons (starting from the leftmost bit), then their common key shall be 100. Eavesdropping can be tackled by sending photons with different phases. For example, the bit 0 may be represented by a photon having a phase of 0° or 180°, and the bit 1 can be denoted by a photon with a 90° or 270° phase. When Bob uses, he will be able to detect the bits correctly. Alice can send data haphazardly using different polarized photons. Bob will haphazardly try to filter out the bits. After the operation, Bob will inform Alice over the telephone of the timings and the state of filter used by him. Alice can then inform him at what instances they have used the same state of filters. Based on this exchange of information. Bob and Alice will get to know their keys. Should any eavesdropper attempt to intercepted photon transmission; there shall be garbage with the key accepted by Alice and Bob. This is because the quantum theory ensures that, without changing the phase of the photon, an intercepted photon cannot be retransmitted. Therefore, a change in the polarity of the photon will let Alice and Bob immediately known of an interception. The scheme of sending information at the one-photonper bit level as proposed by IBM research and research of university of Montreal reported that “to send the key, the transmitter (Alice) tells the receiver (Bob) that the plans to send n bits (photons) starting at a given time. Alice than sends the bits by randomly switching the phase in the transmitter between 00 to 1800; this switches the output in the receiver between “0” and “1”. Although transmission and detection losses mean that Bob will only see a small classical communication channel (the telephone, for example) to tell Alice which photons he has seen— but not which detector he has seen than in. This allows Alice and Bob to share the same random number. For example, Alice uses ten photons to send the random number 1001011101; Bob replies that he only received the second, fifth and last photon; therefore they have shared the random number 001. However, it is conceivable that an conceivable that an eavesdropper could intercept the signal, copy Alice’s message, and send it on to Bob without either Alice or Bob realizing. One way to overcome this, and ensure absolute security, is for both the transmitter and receiver to use non-orthogonal measurement bases. In other words, Alice sends parts of the message by switching the transmitter phase between 90° and 270°, say, and other part by switching between 0° and 180°. When the Bob and Alice are using the same base, the system works as before. However, if Alice is using 00/1800 and Bob is using 90°/2700 (or vice versa), the message is meaningless—a photon that Alice sends as a “0” has a 50% chance of being received as a “1” and vice versa. Therefore when Bob tells Alice which photons he has received, he now also says which base he was using and Alice must tell him if that is a valid photon (i.e. one which was sent and received when they were both using the same base). Paul Townsend of British Telecom, working

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with the Malvern group, recently demonstrated self-interference of short light pulses, containing on average 0.1 photons, down 10 km of standard communications fiber using the technique.” But remember Moore’s laws here to stay for at least anther decade! BOX 2 ILLION-TRANSISTOR IC—Hope or Hype Since the inception of digital electronic in the brand name of ENIAC in 1948, the computer has gone through a number of generations, and it is now in the fifth generation. The so vast and rapid changes of five generations of the computer technology just over a period of 50 years results in one hand the reduction of size & cost of computers and on the other hand the tremendous increase in the processing power & capacity of computers. The credit for these is due to IC (Integrated Circuit) technology. Out of many others the famous empirical laws known as Moore’s Laws, basically govern the pattern of growth of computers and that of IC technology. Mr Gordon Moore, Head of Research & Development of Fairchild coined these laws around 1965. Moore’s laws state that (a) the number of components on an IC would double every year (this is the original Moore’s law), (b) the doubling of circuit complexity on an IC every 18 months (this is known as revised Moore’s law), (c) the processing power of computer will double every year and a half (Moore’s second law). Presently ICs are made of around 250 million transistors. If Moore’s law continues to hold good, it is predicted that by 2010 ICs will be made of billion transistors. The threats to the survival of Moore’s laws are heat dissipation and quantum effect that is a physical limit to IC integration. Several predictions were therefore earlier made for imminent death of Moore’s laws. Contrary to these predictions, Moore’s laws are surviving and hold true for IC integration. Recent two research reports have further showed confidence of survival of Moore’s laws for al least another few years. A survey conducted jointly by IEEE (Institute of Electrical and Electronics Engineers) and the Response Center Inc of USA (a market research firm) over the fellows of IEEE showed that 17%, 52% and 31% respondents respectively predicts the Moore’s laws continuation for more than 10 years, 5-10 years and less than 5 years. The average predicted life term for the laws is then about 6 years. Moore’s laws existence if then guaranteed up to 2009, by the time of which following the laws the billion transistors IC will be a reality. The expectation of realizing billion transistors IC by 2010 has been further brightened by the current research of Intel expanding Moore’s laws. Mr Pat Gelsinger ‘s vision of expanding Moore’s laws includes Intel’s 90-nanometer fabrication process. Although a several alternative technologies, namely quantum computing, bio computing, molecular electronics and chemical computing are under investigation as possible replacement digital computing, the year 2010 may achieve the landmark of billion transistor IC, an another leap forward in IC technology— really a high hope and not a hype.

3. CURRENT AND FUTURE COMMUNICATION TECHNOLOGIES 3.1 Personal Communication Personal Communication is poised to bring a revolution in communication. Personal Communication shall be wireless, service independent and like natural communication. It shall support all sorts of mobility. Active research is going on in this field all over the world. As

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of today, personal communication is seen as a sum total of existing wireless communications like cellular communication, paging, mobile satellite services, VSAT (Very Small Aperture Terminal), wireless LAN, Wireless Internet etc., although ultimately personal communication shall be a UTN (Universal telecommunication Number) service. Personal communication is believed to be a total wireless communication. It is aimed to provide global coverage and to serve any sort of information like voice, data, messaging etc., to anywhere, at any time[24]. At any location or anywhere could imply home, office or in-transit or any other place. Personal communication has two different attractions. First it is total wireless and thereby it supports both man mobility and machine mobility[25]. Personal (or man) mobility and terminal (machine) mobility have distinct and separate characteristics[26]. For personal mobility a person need not to carry a terminal and needs to have a personal communication number. Personal communication number is typically a UTN. (UTN is discussed later). For terminal mobility, a person needs to carry a terminal and needs to be within its radio coverage. With personal mobility, all sorts of communication can be made through the personal number. A caller getting connection of a callee through callee’s personal number may opt for a particular terminal like telephone or fax, for the session. In terminal mobility different types of communication need different numbers and different call sessions. For example for mobile fax, we need to have separate numbers, and for mobile telephone we need to have another separate number. For personal communication, any device like conventional home phone, cellular phone, key phone, fax and pager can be used. Service wise therefore, personal communication is much more flexible, portable, accessible, and reachable compared to wired communication. The philosophy behind personal communication is unique. Personal communication is for connecting people rather than machines. Personal communication is believed to provide a single Universal Telephone Number (UTN) or Universal Personal Telecommunication (UPT) number to a subscriber for all sorts of communication at any time, anywhere. Today, we cannot reach many people most of the time, at most places even though they have a number of telecommunication devices like telephone, fax, telex, e-mail, etc. With single UTN, a subscriber can communicate all over the world. In to-day’s communication scenario, a person has many different numbers for communication at different locations (one’s telephone number in Kolkata is different from that in Delhi) and for different uses (fax number is different from home telephone number). With the two of the above stated characteristics, personal communication will seem like almost natural communication. In our day-to-day natural communications (acoustic communication), we use wireless mode and single name for addressing caller (or callee). A person’s name does not change whether he/she is in Kolkata or in New York. Personal communication is postulated as universal communication. Technology development, standardization, system development, performance analysis and spectrum allocation etc., for personal communication is actively underway. Experts and scientists view personal communication from different angles. One group views personal communication as a distinct and separate total mobile communication solution. Others see personal communication as a migration of the existing conventional wireless communication with enhanced features. The latter view is quite balanced one. Therefore, as of today personal communication can be viewed as a combination of various existing wireless services and new proposed services like UTN[27-28]. Personal communication includes the platforms of each of the existing services like cellular, wireless PBX, Centrex, cordless both home and public, CT2 and wireless LAN etc. Partial application of existing services by personal communication includes paging, SLMR (Special Land Mobile Radio), PSTN (Public Switched Telephone Network), VSAT, and Common Channel Signaling System-7 and ISDN (Integrated Services Digital Network)[29-30]. Personal communication shall fully include future services

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like next generation cellular or TGMS (Third Generation Cellular System) and UTN services. As it is expected that personal communication shall operate globally using the concept of UTN, the required switching and processing systems for personal communication shall be huge and complex. Intelligent capabilities of switching and nodes are a must. On the basis of this, we can define personal communication as an intelligence-based and natural-like communication. Wireless transmission can take place using different frequency bands. An overview of different frequency bands is given in table 5. The frequency allocation to some of the wireless communication is given at table 6. Table 5: Different Frequency Bands and their applications Frequency Band

Wavelength

Name of the Band

Usual Transmission Line Covering the band

Application

10 km

Very Low Frequency (VLF)

Twisted Pair

30-300 KHz

10-1km

Low Frequency (LF)

Twisted Pair/Coaxial Long Radio waves/ Used in Cable submarines because these waves can penetrate waters and follow earth’s surface

300 KHz-3 MHz

1 km-100 m

Medium Frequency (MF)

Twisted Pair/Coaxial Radio Waves/AM between Cable Long 520 KHz to 1605.5 KHz

3-30MHz

100-10 m

High Frequency (HF)

Twisted Pair/Coaxial Short Radio Waves/ AM Cable/Radio waves with 5.9 MHz to 26.1 MHz

30-300MHz

10-1 m

Very High Frequency (VHF)

Twisted Pair/Coaxial FM between 87.5-108 cable/Radio waves MHz/TV between 174-230 MHz

300MHz -3 GHz

1 m-100 cm

Ultra High Coaxial cable/Radio Frequency (UHF) waves/Micro waves

TV between 470-790 MHz

3-30 GHz

100 cm-1 mm

Super High Frequency (SHF)

Micro waves

Analog mobile phone (450465 MHz)/ Digital GSM (890-960 MHz)/DECT at 1880-1900 MHz/Fixed Satellites Service in C-band (4/6 GHz), Ku band (11/14 GHz) and Ka band (19/29 GHz)/Digital TV is planned at 470-862 MHz.

>30GHz

Less than 10 micrometer

Extra High Frequency (SHF)

Optical fiber/Infrared links

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Table 6: Frequency Bands in some of the important wireless applications US Mobile Phones

AMPS, TDMA, CDMA 824-849 MHz 869-894 MHz GSM, TDMA, CDMA 1850-1910 MHz 1930-1990 MHz

Europe

Japan

GSM 890-915 MHz 935-960 MHz 1710-1785 MHz 1805-1880 MHz

PDC 810-826 MHz 940-956 MHz 1429-1465 MHz 1477-1513 MHz

Cordless Telephones

PACS 1850-1910 MHz 1930-1990 MHz 1910-1930 MHz

CT + 885-887 MHz 930-932 MHz CT2 864-868 MHz DECT 1880-1900 MHz

PHS 1895-1918 MHz JCT 254-380 MHz

Wireless LAN

IEEE 802.11 2400-2483 MHz

IEEE 802.11 2400-2483 MHz HIPERLAN 1 5176-5270 MHz

IEEE 802.11 2471-2497 MHz

3.2 Cellular Communication The world of wireless communication actually began in the USA around 1930s when the American police started using radiotelephones for communicating with the field offices. Public radio applications like PLMR (Public Land Mobile Radio) and SLMR (Special Land Mobile Radio) gradually developed. In early 1980s four more wireless services were introduced. These are AMPS (Advanced Mobile Phone Systems) developed in Bell laboratory in 1980, airphone services, cordless services and telepoint. AMPS is the earlier example of cellular communication. AMPS and for that purpose, cellular communication, migrated from wide area radio communication system. Wide area radio transmission technology is the earliest form of mobile communication. The objective of wide area radio transmission technology of the early days is to cover as large area as possible, with a single base station. The single base-station of the wide area cell is equipped with high tower antennae. The transmitter of the base station is very high powered. The configuration of the system once designed is fixed. On the other hand, in cellular concept, smaller cells, each with a low powered base station are used. In cellular concept, the objective is to increase the customers (specifically subscriber density per MHz of allocated spectrum) rather than the coverage area. This objective is met by the concept of cell-splitting and frequency re-use. These have been illustrated with examples in reference. Cell-splitting and frequency re-usable plan are changeable; and hence configuration is flexible and changeable. In cellular system many cells are used. Each cells cover relatively small coverage radii of the order of 0.5 km to 10 km, compared to 50 km to 100 km of the early day mobile communication system. Small cells of cellular communication are formed in splitting large cells of previous mobile systems. For frequency re-use, the cells are clustered into a group with say, k number of cells per cluster. Allocated band of cellular may be divided into k, and each of the divided bands may be allocated to a cell for communication of a mobile base pair. Frequency re-use may be defined as use of some carrier frequency to cover different cells separated by a distance

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so that co-channel interference does not cause problems. Carrier re-use follows well-defined rules described in standard literature.

3.3 First Generation Cellular The first generation cellular is analog. AMPS is the standard analog cellular used in USA, Canada and Australia etc. Other first generation cellular standards are TACS (Total Access Communication System) use in UK, Austria, Spain and Italy; C-450 of Germany and RTMS (Radio Telephone Mobile System) to Italy etc. All the first generation cellular systems use frequency modulation for speech and frequency shift keying technique for signaling. Band sharing among users is done by frequency division multiple access (FDMA) technique. In USA a total of 50 MHz is allocated in bands of 824-849 MHz and 869-894 MHz for analog cellular communication. In AMPS system, each channel is 30 kHz wide. Hence 832 channels are provided in AMPS. Frequency modulation with 8 kHz deviation is used in speech, and frequency shift keying with 10 kbps is used for signaling. In AMPS, cluster size is either 12 with omni directional antennas or 7 with directional antennas per cell. In Japan, a total of 56 MHz is allocated for analog cellular communications in band of 860-885/915-940 MHz and 843-846/898-901 MHz. NTT (Nippon Telephone and Telegraph) employed a system in 1979 using bands of 925-940 MHz and 870-885 MHz respectively for uplink and downlink. 25 kHz channel spacing was used and 600 duplex channels were provided. The signaling rate was 300 bps. This system was upgraded in 1988 with reduced channel spacing of 12.5 kHz and increased signaling rate of 2400 bps. Frequency interleaving technique was used. Number of channels increased to 2400. For farther and more information, reference can be seen.

3.4 Second Generation Cellular Second generation cellular systems were evolved with digitization, digital technology and digital signal processing. With digital techniques in hand and application, it was seen that TDMA (Time Division Multiple Access) and CDMA (Code Division Multiple) could be other viable and potential alternatives to FDMA. Digital techniques offer a number of advantages over analog techniques, namely flexibility (digital systems can support mixed and/or integrated communication and wide range of services), reliability (digital systems are less noise/error prone; can support security easily), cost effective (one transceiver can be used in base station to serve a number of users in digital systems whereas in FDMA, this number increases with number of users) and reduced complexity etc. Digital cellular is known as second generation cellular. Digital cellular technology and techniques are well standardized. GSM (Global System for Mobile Communication), ADC (American Digital Cellular), IS-54 (developed by Electronic Industries Association–TIA of America and JDC (Japanese Digital Cellular) are examples of second generation cellular standards. They are respectively used in Europe (and some parts of Asia including India), USA and Japan. GSM was actually standardized in 1982 as ‘Group Special Mobile’ by CEPT (Conference European Post & Telecommunication). In GSM, 50 MHz band is allocated for cellular communication in the bands of 890-915 (mobile transit) and 935960 MHz (base transmit). Each radio channel is allocated 200 kHz. Thus there can be maximum 25 MHz/200 kHz = 125 carriers. As a convention, only 124 carriers are used. First 200 kHz in uplink and last 200 kHz in downlink are not used. Minimum and maximum number of carriers per cell can be respectively 1 and 15. TDMA is used with 8 slots per radio channel. Each mobile transmits periodically in its slot and receives in the corresponding slot. Each slot is of 0.577 msec duration. Each frame duration is 0.577 × 8 = 4.615 msec. GSM supports full rate operation

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at 22.8 kbps with 8 slots per frame as well as half rate operation at 11.4 kbps with 16slots per frame. For voice communication speech coders compatible with both the rates are available. For data communication various asynchronous and synchronous services at different rates of 9600, 4800 and 2400 bps are specified for both full and half rate service operation. These data services interface to audio modems (like V.22 bis or V.32) and ISDN (Integrated Services Digital Network). GSM can also support connectionless packet switched network X.25, Internet and group 3 FAX (Fly Away Xerox). GSM has recently extended to include ‘group calls’ and ‘push to talk’ services. Extension bands of GSM which are yet to be explored are 880-890 MHz for uplink communication and 925-935 MHz for downlink communication.

3.5 DCS 1800 DCS 1800 is an extension of GSM. In DCS 1800 standard uplink and downlink bands are respectively 1710-1795 MHz and 1805-1880 MHz. It is working at around 1800 MHz which is higher than that of GSM. Higher frequencies always have more penetration power. Therefore, compared to GSM, DCS 1800 system is better in terms of interference and fading. DCS 1800, besides third generation cellular is preferable to in personal communication.

3.6 CDMA Cellular Code Division Multiple Access (CDMA) cellular is another example of second generation cellular. It is a good competitor of TDMA cellular. In TDMA cellular, different users’ signals use the same frequency band, but are distinguished by different codes. The codes are spread codes. This is done basically by two techniques: frequency hopping and direct sequence. In frequency hopping, the transmitter jumps from one narrow band frequency to another according a sequence mutually known to transmitter and receiver. Thus several data bits may be sent at different frequencies. In direct sequence each bit (‘yes’ or ‘no’) of data is represented by a sequence of bits to be transmitted in the same time. The length of sequence is known as chip ratio. For example, one user may code ‘yes’ and ‘no’ states respectively as ‘0000’ and ‘1111’; whereas another user may do so by codes ‘0101’ and ‘1010’ respectively (chip ratio is four). With such spreaded code, signal is disturbed over a wide band. As signal is spread over wide band, effectively signal looks like a noise and becomes indistinguishable from noise. The CDMA system is a secure communication; and this makes it advantageous over TDMA cellular. Another plus point of CDMA is capacity. The capacity of CDMA is more than that of TDMA cellular. Other aspects of CDMA cellular are parallel with TDMA cellular and for that purpose GSM. In [14] it was shown that narrow band propagation, path loss (path loss of say, TDMA) should be applied to wide band path loss (path loss of CDMA). IS-95 is EIA/TIA standard of CDMA cellular system used in AMERICA. The basic user channel rate is 9.6 kbps. It is spread by a factor of 128 and channel chip rate becomes 1.2288 Mchips/sec.

3.7 Wide Area Connectivity In cellular, the wide area coverage or the world wide coverage is done through a basic connectivity scheme. In this scheme a group of basic stations are connected to a MSC (Master Switching Center). MSC is connected to other public and national or inter national networks. Through base stations the mobiles access the network over radio links. Base stations provide overall management and controls switching between radio channels and TDMA time slots in order to connect the mobile to MSC. Through MSC a mobile can connect to other mobiles of other cells as well as connect to subscribers of all the public national or international networks

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connected to MSC. A mobile of one MSC can connect to any other mobile of other MSC s via MSC-MSC switching.

3.8 Continuous Operation Originally a mobile belongs to a base station and is assigned a number for communication. This original number is its number for communication kept stored in HLR (Home Location Register) of MSC. When the mobile is within the coverage area of his original base location permanent number is made use of for communication. A mobile can cross its base region and enter into other base regions (foreign) while talking or communicating. In such situations to maintain continuous operation it is required that foreign base stations should take control of visiting mobile. That is, for continuous operation the control of mobile shall pass from original base station to visiting foreign base station. The pass over technique is called ‘hand off’ operation. Hand off is decided upon comparing the signal strengths received by mobile from the original base station and the foreign base station. As the mobile proceeds to cross the area of original base, the received signal strength from original base gradually diminishes; while the received signal strength from the foreign base station gradually increases. The cross over instant of the signal may be taken as the time of hand off operation. However to avoid falls due to noise some hysteresis is often used for cross over decision. On hand off operation a visiting mobile is assigned a temporary number for communication; and the said information is kept stored in VLR (visiting Location Register) of MSC for farther and future management and control. The hand off operation and technique are equally applicable when a mobile roams from one foreign location to other foreign location i.e. when a mobile crosses over boundaries to boundaries.

3.9 Cordless Telephone First generation cordless is analog. In USA, analog communication cordless is allocated 46.647.0 MHz (base transmit) and 49.6-50 MHz (handset transmit). Ten frequency pairs are used in these bands. Frequency modulation is used for voice. In Europe first standard that was used for cordless telephone is known as CT0. Eight channel pairs are used in this standard near 1.7 MHz (base transmit) and 47.5 MHz (handset transmit). The CEPT developed a standard for analog cordless, which is known as CT1. The bands used in CT1 standards are 914-915 MHz (base transmit) and 959-960 MHz (handset transmit). Forty 25 kHz duplex channel pairs are used in these bands. CT1+ standard was later developed with bands 885-887 and 930-932 MHz with provision of 80 channel pairs. It may be noted that CT1+ bands are chosen to avoid overlapping with GSM bands. In Japan, for analog cordless telephones using FM, 89 duplex channels are provided near 254 MHz (handset transmit) and 380 MHz (base transmit). Digital cordless is known as second generation cordless. Digital cordless is like digital cellular to some extent. Cordless is usually for walking indoors and outdoors. Naturally, cell size, antenna height, mobile speed, handset design complexity and handset transmitter power, in cordless are less compared to those of cellular. CT2 is the first standard of digital cordless in Europe. CT2 is allotted bands 864-868 MHz; and it can support 40 FDMA channels with 100 kHz spacing. In CT2 voice is digitized with 32 kbps ADPCM (Adaptive Differential Pulse Code Modulation) encoder. CT2 can also support data up to 2.4 kbps through speech codec, upto 4.8 kbps with increased error rates and higher data rates using 32 kbps voice channel. Telepoint concept is a migration of CT2 technique. Telepoint is wireless pay phone service. Another standard of digital cordless is DECT(Digital European Cordless Telecommunication). It uses TDMA with 12 slots per carrier for each upward and downward

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communication. With TDMA we have earlier seen in case of cellular communication that multiple users can simultaneously communicate with a single transceiver. The same is true for DECT also. It uses 32 kbps ADPCM technique for voice digitization. In addition DECT can support telepoint, wireless PBX and RLL (Radio Local Loop). In Japan, HS (Personal Handy phone System) is the main standard for digital cordless. PHS uses TDMA. Each channel has a width of 300 kHz. 77 channels are permitted in the band of 1895-1981.1 MHz. 37 carriers within band1895-1960.1 are allocated for home and office cordless; and 40 carriers within band of 1906.1-1918.1 MHz are allocated to public cordless. Digital cordless in USA was developed by Bellcore (Bell Communication Research) with a title WACS (Wireless Access Communication System). Actually PACS (Personal Access Communication Service) is now in use. It is a combination of WACS and PHS. In North America, ISM (Industrial, Scientific and Medical) bands like 902-928 MHz, 2400-2483.5 MHz and 57255850 MHz are in use for digital cordless.

3.10 Wireless Data Trend is towards wireless. Wireless communication offers a number of advantages including high performance-cost ratio. Cellular communication and cordless communication are basically for voice communication although they can be used for data communication for voice communication messaging. Wireless data networks are basically designed for packet mode communication. Cordless is for synchronous services (uses circuit switched techniques), whereas wireless data communication is asynchronous in nature (uses packet switched techniques). But we shall see in IEEE 802.11 standard, that wireless LAN has been proposed to provide both the asynchronous and synchronous services. BOX 3 Know thy Elegant Ethernet

1. INTRODUCTION One of the hottest topics of IT is the Local Area Network (LAN). LAN bears an indispensable role of service to information community. LAN provides basically a shared data access of an organisation, which has several systems, and nodes distributed geographically, logically and physically. The three main physical attributes–limited geographic scope (in the range of 0.1–10 KM [1], low delay or very high data rate (over 1 MBPS [02], and user’s ownership, make LANs substantially different from conventional computer networks. Moreover, while Wide Area Network (WAN) and Metropolitan Area Network (MAN) allow user in network to access the shared databases, LANs go a step ahead and allow users to have shared access to many common hardware and software resources [3] such as storage I/O peripherals and communication devices. For example, a costly high resolution laser printer is usually shared by users in a LAN, and all users in a LAN use an inexpensive single transmission medium in a multidrop environment as well as they use whenever required a single bridge or gateway to communicate with other homogeneous or heterogeneous network respectively. LAN is hence a resource sharing data communication network that is usually used to connect computers, printers’ terminal controllers (servers), terminals (keyboard NDU), plotters, mass storage units (hard disk) and any other piece of equipment (exam. Word-processing machine) that has some form of computer connectivity. LAN is to solve for “MY” problem [4] of “80/20 rule” [5] of communication in a cost-effective scope in an office, factory, university and such relevant environment.

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However PABX (Private Automatic Branch Exchange) differs from LAN in that unlike LAN, PABX user a separate pair of wires (transmission medium) to connect each device (or extension), low bandwidth (limited to that of telephone line) and rugged hardware switching for interconnection. The communication in LANs is peer to peer and not via intermediaries as with WANs and MANs. MAN’s coverage is from a few miles to 100 miles and WAN’s coverage is from 100’s miles to 1000’s miles [6]. These entire three networks follow layered architectural standard protocol like 7 like 7-layer ISO-OSI protocol or SNA protocol etc. for interconnection strategies [6]. LANs continue to be driving force to implement future’s white hope of digital wall socket [7], which will act like today’s electricity socket and telephone socket. The digital wall socket is to be used in handing explicitly low or high data rate devices like copying machines, word processing machines, facsimile displays. VDU, keyboard, microcomputers/PC, large computers etc. This may ultimately lead to 100 percent paper less “Office-of-the-future” and 100 percent automated “factory-of-the-future” with “diskless” managers, administrators and engineers etc. One of the most successful LANs is Ethernet. Ethernet was the most popular LAN in 1987. As per Forrester Research Inc, [5], in U.S.A Ethernet covers 33 percent of LAN market with IBM token ring lagging behind at 22 percent. Dataquest estimated that Ethernet had covered 52 percent of installed LANs U.S.A is Ethernet hottest now? Whatever may be the answer to this question, it is a fact that Ethernet is still today very popular and will continue to be so at least for some time to come. This paper will make a thorough review of Ethernet.

2. Ethernet Historically, Ethernet was developed by the Xerox Corporation on an experimental basis [8] around 1972. Based on this experimental experience, the second-generation system was soon developed by the Xerox Corporation in late 1970’s [9]. Around 1080-81, under a joint effort of DEC (Digital Equipment Corporation), Intel and Xerox, an update version of Ethernet specifications (table I) [8] was designed. This historically leads to development of IEEE (Institute of Electrical and Electronics Engineering Inc) 802 standards (table II) [4,6] of LAN in reference to 7-layer OSI-ISO (Open System Interconnection of International Standards Organisation) the LIC (Logical Link Control) is covered by IEEE 802.3 standard at MAC actually specify the accessing mechanism, physical level covers the electromechanical connectivity at network medium, LIC and MAC of LAN jointly form the data link of OSI-ISO protocol standard. Nowa-day Ethernet is available from many vendors [10]. Such Ethernet is as per IEEE 802.3 standard. These are actually “Ethernet-like” [11] networks. However, all LANs covering IEEE 802.3 standard are not Ethernet. But all Ethernets cover IEEE 802.3 standard. Table 1: Specification of Ethernet Parameters

Experiment Ethernet

Industrial Commercial Ethernet

1. Data rate

2.94 MBPS

10 MBPS

2. Maximum end-to-end length coverage using repeaters/bridge

1 KM

2.5 KM

3. Maximum segment length

1 KM

500 M

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4. Data encoding technique

Manchester

Manchester

5. Co-axial cable impedance

75

50

6. Co-axial cable signal level

0 to +3 volt

0 to – 2 volts

7. Transceiver cable connectors’ size

25 and 15 pin D series

Only 15 pin D Series

8. Preamble

1 byte of a pattern of 10101010

1 byte of a pattern of 10101010

9. Size of CRC (Cycle Redundancy Check)

2 byte

4 bytes

1 byte

6 bytes

10. Size of address field

31

Table 2: IEEE 802 Standard Standard for MAC & Physical layer

Access Technique and Topology

Transmission medium with allowed data

Basic application area

802.3

CSMA/CD with BUS topology

Broad band: Co-axial cable with 1 MBPS/ 5 MBPS/10 MBPS/ 20 MBPS Base Band : Co-axial Cable with 1 MBPS

Office Automation (OA)

802.4

Token passing with BUS topology

Broad : Co-axial cable with 1.5444 MBPS/ 5 MBPS/ 10 MBPS 20 MBPS. Base band: Co-axial cable With 1 MBPS/ 5 MBPS 10 MBPS.

Manufacturing Automation (MA)

802.5

Token passing with RING topology

Base band: Shielded twisted wire pair with 1.4 MBPS. Co-axial cable with 4 MBPS/ 20 MBPS 40 MBPS.

Process real time application

802.6 802.7 802.8 802.9

Yet to be finalized -do-do-do-

Yet to be Finalized -do-do-do-

MAN Broadband LAN LAN with fiber optical LAN in ON (Integral service digital network)

802.2 standard is for LIC of LAN. 802.10 is for network security.

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2.1 Features of Ethernet Why Ethernet is so popular? This is due to some of its important features. The most appealing features of the Ethernet are its protocol simplicity, and the relative low-cost and elegant implementation of LAN system which meets the following desirable characteristics [6,7] of a local networking facility. • High flexibility i.e easily adaptability when devices are system to be added or removed. This is due to the bus topology and the cable tapping facility of Ethernet • The transmission medium and access control is easily extensible with minimum service disruption. • High reliability, which assures the continuation of the operation of the network in failure of one or more active element (node) like PC, terminal or workstation etc. This is due to the passive feature of Ethernet cable. Moreover, there is no centralized control but distributed control in Ethernet. • The traffic will be bursty in nature. In office and engineering environment, nature of data is infrequently bursty [10] and ironically Ethernet, was specially made for office automation, although not in general.

2.2 Components and Operation of Ethernet The Ethernet is itself a hardware system. Ethernet can connect typically a maximum number of nodes of 100 per segment [5] and 1024 per total Ethernet [10]. An Ethernet LAN must have Ethernet cable, transceiver, and interface unit, control unit, the user system (Fig 1) and terminals. Two types [12] of ‘co-axial cable popularly known as “thick Ethernet” and “thin Ethernet” is used, mainly as backbone Ethernet. On this back bond cable, the communicating systems and peripherals are attached (tapped). Taps may be intrusive where the cable is cut for tapping or may be non-intrusive where the cable is cut drilled and a tap added without hampering the operation of the network. The most common Ethernet, the baseband Ethernet is tapped nonintrusively, whereas the broadband Ethernets used intrusive tapping using T-junctions scheme. Baseband Ethernet is an implementation where the entire bandwidth of backbone cable is used only for Ethernet communications. Singles in cable are not modulated signal. Thick Ethernet cable resembles a marking every 02.4 meters usually by black ring around cable to show where the taps go. However, thick wire co-axial cable has maximum length limitation of 500 meters and thin wire co-axial has limitation in the range from 189 meters to 1 km depending upon the vendors of transceivers and controllers. Ethernet may also run on twisted pair under certain restriction and on fiber. The length of twisted pair may range from 20 meters to 100 meters. Ethernets on fiber optic medium have length restriction in the range of 30 meters to 5 km. In some cases, thin wire Ethernet may be required to be connected to a thick wire Ethernet. Thin wire cable may be connected to thick wire though a barrel connector. In such case, the restriction on segment length will follow the formula [5]. (3.28* thin wire length) + thick wire length p). Any other station which is desirous to send urgent message, if sees that transmission is going on, may check the priority is less than its priority, it will distort the priority. The on going transmitting station not getting back the proper priority byte will stop, immediately transmission to allow the higher priority station to access. However, if checked priority is greater than its priority, it has to wait for free carrier. A modified and deterministic Ethernet is already there in France-defence department [5]. This is of course a proprieratory item.

3.2 Ethernet for Data and Voice ISDN (Integrated Service Digital Network) is becoming more and more attractive to communication engineers. In spirit with goals of ISDN, a concept of ISLN (Integrated Service Local Network) [17] was introduced. But why? Statistics show that about 15 percent of their office time is spent by senior managers on telephone, and not more than 3 percent of the same is used in handling data oriented jobs. Besides, “real managers don’t use terminals” [19]. But today, of course, they do. Therefore in a complete and cost effective OA system, the integration of voice and data is an essential requirement. In any organization, why shall be there one PAPX (for telephone) and one LAN (for data). However, early problem of LAN design was to communicate data, but the real problem is to provide users’ requirements of both data and voice communication. The pioneer vendors of Ethernet can examine whether Ethernet can be extended to cover ISLN requirement in either of two technique [19] : (i) conventional voice + data upto facsimile or (ii) upto full moving video.

4. EXTENDED ETHERNET A number of Ethernet segments may be connected together (Fig. 1) via repeater or bridge [5,12,20]. A repeater consists of some sort of microprocessor (like Intel 8088, Motorola MC 68000) and memory etc. They are standalone units. They repeat everything what is received from any segment to other segment and vice versa. They connect two Ethernet segment via transceiver. Bridge, on the other hand, store and forward the intended data only from a source segment to a destination segment. Bridge is made of some sort of processor, storage, buffers and a set of software.

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User device

Segment-2 TR

Terminal Controller

TR

CI

Repeater

Controller Interface

maximum 2.5 meter TR

CI

Transceiver cable (max50 m/15 ft)

Printer Top

Terminal

TR Original segment

Terminal TR

User device

TR

CI Controller Interface

TR

terminal either (may be twister wire pair, Co-axial cable, Fibre or radio link)

Terminal Service Like DELN

TR = Transceiver PC

User

TTY

Single Ethernet Segment, max. 500 m/1500 ft

Bridge

TR

TR Segment-3 TR

TR

Gateway

CI

WAN or MAN

Plotter

Fig. 1

5. CONCLUSION A number of important considerations of Ethernet have been highlighted. Ethernet is seen to be very effective for OA. If the next generation of Ethernets is to be developed, they must be done in a direction to extend the application to MA utilizing the proposed suggestions in paper.

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References 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11.

C. David Tsqo, “A local area network architecture review”, IEEE Communication Magazine, Vol. 22, No. 8, pp 7º, Aug.’1984. D.D. Clark, K.T. Pogran and D.P. Reed, “ An introduction to local area network”, Proc. IEEE, Vol. 66, No. 11, pp. 1497-1517. Nov. 1978. John E. McNamara, “Local area Network”, Prentice Hall of India, Ch. 1, 1991. Stephen P.M. Bridge, “Low cost local area Networks”, Galgotia Pub. Pvt. Ltd. Ch 1, 1990. Bill Hancock, “Designing and implementing Ethernet Networks”, QED information Science, Inc, 1989. Paul J. Fortier, “Handbook of LAN Technology”, McGraw Hill Inc., NY, 1989. James Martin, “Computer networks and distributed processing”, Prentice Hall, Inc, Ch. 26, 9181. John F. Shoch, Young K. Dalal, David D. Redell and Ronald C. Crane, “Ethernet”, Advances in Local Area Networks, IEEE Press, NY. pp 29-48, 1987. Timothy A Gonsalves, “Measured Performance of the Ethernet” Advances in Local area Network, IEEE Press, pp. 383-387, 1987. William L Schweber, “Data Communication”, McGraw Hill, Intl. Ch. 11, 1988. Neil Willis, “Computer Architecture and Communications”, Paradigm Pub. Ltd., U.K., Ch. 14, 1988.

3.11 Wireless LAN Wireless LAN offers wireless data communication for a limited geographical area. Wireless LAN is like wireless PBX for data. It solves ‘My Problem’] of any organization utilizing the 80/ 20 rule of communication. Wireless LANs are meant for private or organizational uses where wired communication is impossible or impractical or not desirable or expensive ( examples are historic building, trading floors, manufacturing floors, conventions etc.) and/or where some sort of mobility is required (examples university environment, conference room, hospital environment). Wireless LANs are aimed at data rates of 1 Mbps or more. Basically there are two forms of wireless LAN : radio LAN and Infrared LAN. IRLAN is less popular than radio LAN due to : IRLAN can cover wide area but it is almost twice in cost of that an equivalent radio LAN. IRLAN requires license for use of spectrum. Radio LAN uses ISM spectrum for which license may not be required. Unlicensed radio LAN s are available in USA in the bands of 902928 MHz, 2.4-2.4835 GHz and 5.725-5.85 GHz. IEEE committee for standardization of radio LAN has proposed to use 2.4 GHz band (discussed later). IRLAN is only for point to point communication. Radio LAN is much more flexible and can be used in multiple users communication. IRLAN is short ranged (if it is made wide ranged, it becomes costly), radio LAN is long ranged. Architecture wise, both radio LAN and IRLAN assume one of the two basic technologies : infrastructure or ad hoc. Infrastructure topology is the most common in radio LAN. Under infrastructure network stations (computers fitted with adapters/transceivers) communicate with each other under a coverage area of an access point, as well they communicate with any other station in the network through backbone wired network. Backbone network is accessed via access points. In IEEE 802.11 standard, access points are known as base points; and backbone network is known as distribution system. Access point is a combination of transceiver and data bridge. Each access point provides a certain coverage area. Number of access points required for an infrastructure network, thus depends on the required coverage area. Infrastructure topology is useful in covering a building or campus or an institute under radio communication.

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In ad hoc network stations independently communicate with each other and there is nothing like access point for communication through backbone network. Ad hoc network can be either temporary or semi-permanent. Semi-permanent networks are used for a few months and useful for companies, which move frequently. Field construction companies, military camps on war days may use semipermanent ad hoc networks. Temporary networks are used for a day or for a few hours of business. They may be used in sharing files, databases in a company meeting or convention. There are two important standards wireless LAN. IEEE is developing IEEE 802.11 standard, which is proposed to be used in USA. HIPERLAN (High Performance Radio LAN) is the standard developed by European Telecommunications Standard Institute and is for use in Europe. HIPERLAN standard has already been ratified by CEPT. IEEE 802.11 draft standard defines three different physical layers : a) 2.4 GHz ISM band with frequency hopping spread spectrum radio, b) 2.4 GHz ISM band with direct sequence spread spectrum radio and c) infrared light 2.4 GHz ISM band has been allowed both in USA and Europe for IEEE 802.11 version LAN; whereas Japan has allocated the band 2.471-2.497 GHz for IEEE 802.11 LAN. Japan has allowed such a narrow band in 2.4 ISM in order to provide radio LAN in medium data rates of 256 kbps to 2 Mbps where spread spectrum technique is used. Japan has allocated another band near 18 GHz for high rate of 10 Mbps or more radio LAN where QAM (Quadrature Amplitude Modulation), QPSK (Quadrature Phase Shift Keying) are used. In frequency hopping system, 79 and 23 different frequencies are used respectively in USA/Europe and Japan for data transmission under IEEE 802.11 scheme. In direct sequence the processing gain is proposed to be 10.4 dB in IEEE 802.11 draft standard. Frequency hopping system can support large number of channels compared to direct sequence scheme. Frequency hopping is also having superior performance when interference is high. However, direct sequence is simpler in design and implementation. Service wise, IEEE 802.11 has proposed to serve asynchronous and time sensitive (synchronous/isochronous) services. In radio LAN when an access point is shared by all stations, all stations use same hopping/sequence pattern. As such there is always a fair chance of interference and collision. Hidden node problem of radio network on the other hand has a tendency to increase collision. When two transmitters send data to a single receiver, single receiver can hear to transmissions; but transmitters cannot hear each other. This is known as hidden node problem of radio system that depends on physical sensing of carrier. Thus a good medium access control (MAC) strategy is essential In radio system. In IEEE 802.11 standard, MAC is CSMA/CA (Carrier Sense Multiple Access/Collision Avoidance) rather than CSMA or CD (Collision Detection) used in Ethernet. Radio Technique does not allow the collision detection mechanism. In CSMA/CD technique, when a station senses a free carrier it backs off transmission for a random amount of time. Thus if more than one station detects free carrier at the same time , due to random back off periods, collision may be avoided. For tackling hidden node problem, in IEEE 802.11 scheme, two controls frames, RTS (Request to Send) and CTS (Clear to Send) are used. Theses are like RS-232-C transfer protocol. HIPERLAN differs from IEEE 802.11 on number of accounts. IEEE 802.11 does not support multi-hop communication. No access point or station can act as a data router or relay point. HIPERLAN does not support multi-hop communication by the way of cellular architecture. It is targeted for higher data rates than IEEE 802.11 and may support 23.5294 Mbps. That is why a large and dedicated band of order of 150 MHz (5.150-5.300 GHz) near 5 GHz another band of 17.1-17.2 GHz near 17 GHz are allocated to HIPERLAN. HIPERLAN is also aimed to be indistinguishable from wired LAN of Ethernet and to support some sort of isochronous services. For modulation, Gaussian minimum shift keying is used. A (31,26) BCH mode is used for error control. It aims to achieve BER (Bit Error Rate) of 10-3 or less for fair service. MAC in

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HIPERLAN is different from both CSMA/CD of IEEE 802.11. In HIPERLAN accessing scheme, if a station senses free medium for 1700 bit times, it can transmit immediately. If not, channel accessing is done through three phases of prioritization, elimination and yield. HIPERLAN MAC can reduce chances of collision to a less than 3 per cent. IRLAN works on IEEE 802.3 and IEEE 802.5 protocols. IRLAN is based on line of sight technology, and hence it can support high data rates up to 10 Mbps for Ethernet configuration and up to 16 Mbps for token ring configuration. IRLAN is costly and hence some vendors are hopeful to go through the technology. An association of vendors has made their own standards for IRLAN. The IEEE standards for different LANs are 802.3 for CSMA/CD Bus LAN, 802.4 for Token Passing Bus LAN, 802.5 for Token Passing Ring LAN and finally 802.11 for WLAN (Wireless LAN). Of course in general IEEE standards, 802.3, 802.4 and 802.5 is commonly known as 802.x; and these standards are for wired LANs. As of today two basic transmission technologies those are in use to set up WLAN (Wireless Local Area Network) are: Infrared light at THz wavelength and Radio wave at GHz (2.4 GHz in the license-free ISM-Industrial, Scientific and Medical band). Infrared technology uses either diffuse light reflected at obstacle like furniture, walls etc or directed light if line of sight path exists between the sender and the receiver. Simple transmitter may be light emitting diodes or laser diodes; and the receiver can be a photodiode. But most of the wireless systems use radio waves. IEEE 802.11 LAN can use both Infrared and Radio wave, but HIPER LAN1 uses only Radio wave. A comparison of Infrared and Radio wave transmission technology is given in the table (7). Table 7: Comparison of Infrared and Radio waves Transmitter Receiver r

Data rate

Shielding

Infrared Technology

Very Simple

Very low as due to low bandwidth of Infrared. 115Kbps to 4 Mbps is the data rate

Infrared can be easily shielded. It can not penetrate obstacles like walls etc.

Radio Wave

Not simple as in Infrared

Higher than Infrared

Shielding is not so simple.

Like other 802.x standards, the standard 802.11 covers only physical layer and MAC sub layer. IEEE 802.11 supports three different physical layers: one layer on using infrared, and another two layers on using basically 2.4 GHz ISM band available free on world wide. ISM bands are: 902 to 928 MHz, 2,4000 to 2.4835 GHz and 5.7250 to 5.825 GHz. Radio LANs operate in the high UHF and low microwave range. Infrared LANs do transmission just below visible light. At physical level three different wireless specifications are: Infrared LANs, Frequency Hopping Spread Spectrum (FHSS) LANs and Direct Sequence Spread Spectrum (DSSS) LANs. FHSS and DSSS LANs belong to radio LANs. FHSS LANs are specified to support data rate of 1 Mbps with a faster specification of 2 Mbps. DSSS LANs are specified for 1 Mbps and 2 Mbps also. FHSS is a spread spectrum technique that allows for the coexistence of a multiple number of networks in the same area by allowing different networks the different hopping sequence. Under IEEE 802.11 standard, 79 hopping channels for North America and Europe; and 23 hopping channels for Japan are specified each with a bandwidth of 1 MHz in 2.4 ISM band. A particular channel is identified by a

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pseudo random hopping pattern. The maximum transmitter power is 1 watt EIRP (Equivalent Isotropic Radiated Power) in US and 100 mW EIRP in Europe. In DSSS, the separation is done by codes rather than frequency. Except this, all other like bit rate and transmission power remain same as in FHSS. The frame formats of physical layer of 802.11 are shown in Fig. (5). The figures in the bracket in the fields refer to the size of the fields in bits. In FHSS frame, the synchronization field is a bit pattern of 010101. The star Frame delimiter( SFD) is 0000110010111101. PLW refers to PDU Length Word i.e. length of payload including 32 bit error control CRC bits at the end payload. It ranges from 0 to 4,095. PSF is for signaling. Out of its 4 bits only one bit is specified to indicate either 1 or 2 Mbps. HEC is a 16 bit header error check field for which ITU-T CRC-16 standard is used. In DSSS frame, 128 synchronization field is made of only scrambled 1 bits. 16 bits start frame delimiter is 1111001110100000. Signal refers to bit rate. Service field is reserved for use. Length is used to indicate payload size with CRC field. HEC is used to check error on header with IUT-T CRC-16 standard. The MAC data frame of IEEE 802.11 is as shown Fig. (6). The figures in the bracket in each field refer to the size of the field in bytes. Frame control is used for several reasons like protocol version and the type of the frame etc. Duration ID indicates the virtual reservation mechanism. Address 1 to 4 which has 46 bits in each is used as they are done for 802.x LANs. Sequence control is used for acknowledgement and error and flow control. CRC is used as it is done 802.x LANs. Synchronization

SFD

PLW

PSF

HEC

Payload

-on (80)

(16)

(12)

(4)

(4)

(Variable)

(a): For FHSS Synchroni zation (128)

SFD (16)

Signal (8)

Service (8)

Length (16)

HEC (16)

Payload (variable)

(b) For DSSS Fig. 5: Physical frame format of IEEE 802.11 radio WLAN Frame

Control (2)

Duration ID (2)

Address 1 (6)

Address 2 (6)

Sequence 3 (6)

Address Control (2)

Data 4 (6)

CRC (0-2312) (4)

Fig. 6: Data format of IEEE802.11

The world is rapidly shifting towards wireless and faster network. In such a rapidly changing scenario, let us see how one of the oldest local area networks, namely Ethernet is keeping pace with the changes. Ethernet dominates as a LAN (Local Area Network), as it is time tested highly reliable, scalable, elegant and low cost network. IEEE 802.3 Ethernet is the established corporate LAN technology, and most of its implementations are with IEEE 802.3u or 100 Base T that defines a 100 Mbps data rate using four pairs of twisted wire pair wiring or Ethernet cable. Tree of the Ethernet is shown in Fig. (7). The Ethernet was originally wired network. It follows the IEEE standard 802.3 for logical link control by which the several nodes can share the single physical medium. The physical layer implementation is made with wires.

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Ethernet Wireless

Wired

802.3

802.11

Conventional Ethernet 10 Base5 Thick Co axial 10 Base2 Thin Co axial 10 Base T UTP

802.11b (11Mbps)

Fast Ethernet

802.11a (125 Kbps–54 Mbps

802.11g (54 Mbps)

100 Base T4 (CAT 3 UTP) 100 base Tx(CAT 5 UTP)

Gigabit Ethernet 1000 Base LX 1000 Base SX 1000 Base CX 1000 Base T (CAT 5+) 10 Gigabit Ethernet under IEEE 802.3ae

Fig. 7: Ethernet as grows.

The IEEE standards for different LANs are 802.3 for CSMA/CD Bus LAN, 802.4 for Token Passing Bus LAN, 802.5 for Token Passing Ring LAN and finally 802.11 for WLAN (Wireless LAN). Of course in general IEEE standards, 802.3, 802.4 and 802.5 is commonly known as 802.x; and these standards are for wired LANs. IEEE 802.11 mainly provides connectivity to corporate LAN. It is very costly for home LAN.

3.12 IEEE 802.11 Architectures In IEEE 802.11 LAN standard, there are two different configurations of a network: ad-hoc and infrastructure. In the ad-hoc network, there is no fixed structure to the network and no fixed access point. There are no fixed points and the computers are brought together to form a network “on the fly” as shown in Fig. (8) and usually every node is able to communicate with every other node. A good example of this configuration is the unscheduled meeting where officials bring laptop computers together to communicate and share information to arrive at a decision. In this type of configuration it is difficult to fix the type of the nodes, but algorithms such as the spokesman election algorithm (SEA) may be used to “elect” one machine as the master of the network with the others as slaves. To know who’s who in the ad hoc networks, a broadcast and flooding method may be used. The infrastructure network (Fig. 9) uses fixed network access points with which mobile nodes can communicate. These network access points may also be connected to landlines to widen the LAN’s capability by bridging wireless nodes to other wired nodes. As and when service areas overlap, handoffs can occur. The structure is very similar to the current cellular networks around the world.

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Computer

Computer

Computer

Computer

Fig. 8: Ad hoc Network.

Computer

Computer

Computer

Computer

Computer

Fig. 9: Infrastructure Network.

Wireless computing, Wireless communication and Wireless networks shall be the rule if future. In such a scenario, WLAN will play a major role. In the last few decades two important wireless technologies those emerge as viable and promising are LEO (Lower Earth Orbital satellites) and 3G (Third Generation) cell phones. But both the technologies fail to meet the expected aspirations. Here at, the WLAN has come out as an alternative. Presently under IEEE 802.11, two major WLAN standards are operating: 802.11a and 802.11b (Table 8). The first 802.11 standard is 802.11b that was approved by the IEEE in 1999. The 802.11b is the first standard that broke the wired brethren of 802.3 wired Ethernets. The 802.11b standard transports data at 11 Mbps using CCK (Complementary Code Keying) using 2.4 GHz band. The 802.11b has is a very successful track record as it is learnt that “ the sale of IEEE 802.11b wireless LANs has increased dramatically from 5000 to 70,000 units per month since early 2000.” It is also reported that: “ The growing popularity and ubiquity of WLANs will likely cause wireless carriers to lose nearly a third of 3G revenue as more corporate users begin using WLANs to connect to the Internet and office networks” Many analysts feel that “ the ease of installing and using WLANs is making it alternative to mobile 3G. In contrast to the reported $650 billion spent worldwide by carriers to get ready for 3G, setting up a WLAN hotspot requires only an inexpensive base station, a broadband connection and one of many interface cards using the 802.11b.” But the speed of 802.11b is one-tenth of wired Ethernets. Therefore the IEEE to have high speed wireless access approved the 802.11a standard concurrently. The IEEE 802.11a standard provides scalable data rates from 125 Kbps to 54 Mbps in increments of 125 Kbps with OFDM (Orthogonal Frequency Division Multiplexing) using 5 GHz band. Actually 54 Mbps is known as “turbo” rates. IEEE 802.11b standard defines only two lower levels of OSI (Open Systems Interconnection) reference model, the physical layer and the Data

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Link Layer Medium Access Control (MAC) sublayer. IEEE 802.11b uses two pieces of equipment, a wireless station, which is usually a PC or a Laptop with a wireless network interface card (NIC), and an Access Point (AP),which acts as a bridge between the wireless stations and Distribution System (DS) or wired networks. There are two operation modes in IEEE 802.11b, Infrastructure Mode and Ac Hoc Mode as discussed earlier in the IEEE 802.11 standard. The physical layer covers the physical interface between devices and is concerned with transmitting physical raw bits over the communication channel. IEEE 802.11b supports different data rates (Table 9). The problems of 802.11a are many: It does not support different devices with different speed, design and complexities. The standards 802.11a and 802.11b are not interoperable. 802.11a is presently used only in North America, and 802.11b is used in the whole of Europe and Asia. The IEEE 802.11e is tasked with a new protocol to non-guaranteed quality service in ad hoc connectivity. The IEEE task group “G” for 802.11 has now deliberating on the next generation standard for 802.11 that would transmit data at the speed of wired Ethernet. The new standard will be 802.11g. The mission of 802.11g standard is to have wireless access at the turbo speed of 54 Mbps while maintaining the interoperability. The IEEE task group “G” for 802.11 has now deliberating on the next generation standard for 802.11 that would transmit data at the speed of wired Ethernet. The new standard will be 802.11g. The mission of 802.11g standard is to have wireless access at the turbo speed of 54 Mbps while maintaining the interoperability.

3.13 GIGABIT ETHERNET Over the decades the speed of Ethernet has grown every time by a factor of 10 starting from 10 Mbps to 100 Mbps to 1000 Mbps (1 Gbps). The Ethernet that carries data at the rate of 1 Gbps or more is known as Gigabit Ethernet. The physical media initially recommended for Gigabit Ethernet is the fiber (table 10) But another IEEE committee is considering the use of UTP cable for Gigabit Ethernet called 1000 Base-T. Keeping the growth of speed, the next Ethernet will be 10 Gbps. The IEEE 802.3ae standardization is going on for 10 Gbps Ethernet. The goal is to achieve very high speed transport keeping maximum compatibility with already installed base of Ethernet 802.3. The 10 Gbps Ethernet will provide almost zero latency service to users. Thus even when coverage area is increased, the remote application and services will appear as local. Table 8: IEEE Standards for LAN IEEE Standards

Definition

802.0 802.1 802.2 802.3

Sponsor Executive Committee Higher Layer LAN Protocol Logical Link Control Medium Access Control (MAC) of CSMA/CD Bus LAN (Example: Ethernet) 10 Gbps Ethernet Data Terminal Equipment -electrical via balanced cabling for 802.3 interface MAC of Token Bus LAN

802.3ae 802.3af 802.4

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802.5 802.5t 802.5v 802.5z 802.6 802.7 802.8 802.9 802.10 802.11 802.11a 802.11b 802.11g 802.12 802.13 802.14 802.15 802.16

45

MAC of Token Ring LAN 100 Mbps Token Ring LAN Gigabit Token Ring LAN Link Aggregation MAN Working Group Broadband Technical Advisory Group Fiber Optic Technical Advisory Group Isochronous or Integrated Services LAN (ISLAN) Inter operable LAN Security Working Group Wireless LAN (WLAN) Working Group 56 Mbps WLAN 11 Mbps WLAN Next Generation WLAN Demand Priority Working Group Inactive Cable Modem or Cable TV Wireless Personal Area Network (WPAN) Broad Band Wireless Access (BBWA)

Table 9. Different data rates of IEEE 802.11b IEEE 802.11b Data Rate Specifications Data Rate in Mbps

Code Length

Modulation

Symbol Rate in MSps

Bits/Symbol

1

11 (Barker Sequence)

BPSK

1

1

2

11 (Barker Sequence)

QPSK

1

2

5.5

8 (CCK)

QPSK

1.375

4

11

8 (CCK)

QPSK

1.375

8

Table 10. Gigabit Ethernet Gigabit Ethernet

Mode Supported

1000 Base LX (Long wave laser over single mode and

Fiber diameter in micron

Maximum Distance in segment

Single

9

10 Km

Multi

50

550 m

Single

50

3 Km

Multi

62.5

440 m

1000 Base SX (Short Wave laser over multimode fiber)

Multi

50

550 m

Multi

62.5

260 m

1000 Base CX (Balanced shielded 150 ohm copper cable)

BALANCED SHIELDED CABLE

25 m

1000 Base T (UTP Cable)

UTP Cable

100 m

multimode fiber)

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3.14 Trade Off In the speed jargon, the wired Ethernet still outplays the wireless Ethernets. But Wireless LANs are also picking up speed. Trade off lies in wired LANs’ speed versus wireless LANs’ flexibility, reliability and low maintenance cost or looking at today IEEE 802.3ae versus 802.11g. Alan Mc Adams, Chair of IEEE-USA committee on communication and information policy (CCIP) said “ Gigabit Ethernet over fiber will allow the transfer of Ethernet technology, concepts and benefits from Local Area Networks to Metropolitan Area Networks and Regional Area Networks.” It is reported in the IEEE The Institute of Oct’2002 that “ The June approval of the IEEE 802.3ae standard for 10 gigabit per second Ethernet has the potential to allow Gigabit Ethernet over fiber (GEF) technology to supplant current telecommunications infrastructures with its cost, speed and distance advantages.”

3.15 Integrating Wireless Protocols In nature, only one thing is permanent and that is nature. This law of nature appears to be equally applicable to Ethernet, which is ever changing and growing.. The present scenario is that of cellular data and of wireless LAN data, respectively under protocols of TIA/EIA (Telecommunications Industry Association/Electronics Industry Alliances) IS-856 and IEEE 802.11.Whereas IEEE 802.11 meets short-range high speed data network, IS-856 is meant for wireless voice data. They are complement to each other. They may take advantage of each other and integrate to provide typical bridge (Fig. 6) to satisfy demand for access to the wireless Internet. On the other hand, the IEEE 802.11 will immensely take part in the seamless integration of total wireless access and networking in the next-G era (Figs. 10 and 11).

Computer

IEEE 802.11 access point

Wireless Station

Computer

Independent device

802.11

Fig. 10: Typical integration for a corporate connection.

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IS-856


Seamless Next G

47

Mobile wireless Integrat

3G/Data WLAN 801.11a

2G/Digital WLAN 802.11b

1G/Cellular WLAN/ Conventional

Fig. 11: Seamless integration of wireless access and networking—IEEE802.11.

3.16 IEEE 802.15.4 StandardLow Data Rate, Low Cost Wireless Home Networking Solution Due to applications of networking in almost everywhere, several attempts are being made to offer solutions that aim to be flexible, cost effective, reliable and consume less power, the features particularly so important for home or residential networking. In the wired communication, the DSL (digital subscriber loop) technology (discussed later) is one important driver. The cost effectiveness is achieved by utilizing the existing copper line in the local loop. But the wireless communication and networking has an edge over wired technologies for which a wireless local loop solution is needed. The wireless networking and communication technologies that have appeal in voice and data applications in residential or home services are among others cellular, cordless and IEEE 802.11b. The consideration of cost effectiveness and low power consumption has motivated for development of a new standard IEEE 802.15.4 for home networking with low data rate wireless solution. The initiative to develop a standard for low powered and low cost home wireless networking was taken by IEEE working group 15 in 2000. Besides home automation, the standard is poised to be applied in different services of industry like industrial control, automotive sensing (monitoring tire pressure, sensing of soil moisture, pesticide, pH levels), and disaster management (sensing and determining location of disaster) etc. In home applications, the services of the standard will be PC peripherals (Keyboard, PDA, Mouse), consumer electronics (TV, Radios, VCR, CD etc), automation (heating, air conditioning, ventilation, windows and doors lock), remote control, health monitoring, security and PC enabled

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services. These applications need data range ranging from a few kilobits per second (kbps) to 115.2 kbps. The acceptable delay or latency for these services ranges from 15 ms to 100 ms. The major features of the IEEE 802.15.4 proposed standard are: • like all other IEEE standards, the IEEE 802.15.4 refers to lower layer specification. In reference to OSI ISO 7 layer protocol, the IEEE 802.15.4 refers to DLL (Data Link Layer). DLL is split into two sub layers: LLC (Logical Link Control ) and the MAC (Medium Access Control) sub layer. The LLC is as per other specification of 802.3 etc. The IEEE 802.15.4 defines a separate MAC sub layer. • The IEEE 802.15.4 as recommended two versions of physical layers: (1) 868/915 MHz and (2) 2400 MHz • the IEEE 802.15.4 supports both star and peer to peer networks including ad hoc networks • the generic frame format of IEEE 802.15.4 is made of : frame control and sequence number that are respectively 2 and 1 bytes. The Address field is variable from 0 to 20 bytes. Payload is variable, but the full MAC frame is limited to 127 bytes. Frame check sequence is 2 bytes and it uses 16-bit CRC control. In the physical layer frame, the total header length s 6 bytes with preamble of 4 bytes, start of packet delimiter of 1 byte and physical header of 1 byte. The payload is limited to 127 bytes, being the MAC frame. • The header fields of the physical layer frame format are: 4 bytes preamble that is used for synchronization, 1 byte start of packet delimiter that is used to indicate the end of preamble, and 1 byte physical header used to specify the length of physical service data unit • The physical layers in IEEE802.15.4 uses DSSS (direct sequence spread spectrum) methods with different channel frequencies and modulation parameters. • The DSSS method is chosen in order to use low cost IC for implementation by which the cost of the system is made low • IEEE 802.15.4 aims to provide excellent battery life, low transmit power • IEEE 802.15.4 devices aim as much as 99.9 percent of sleeping time • The simplicity is the another attraction of IEEE 802.15.4

3.17 IEEE 1394 for Home Network A recognized definition of home network is “A home network interconnects electronic products and systems, enabling remote access to and control of those products and systems, and any available content such as music, video or data.” Several standards are in perspectives for application in home networks. For example, IEEE 802.11 is the most talked of standards for wireless interface. The standard has got several modification in order to meet with high speed requirements as well as other different requirements of home networking. Unfortunately the standard is still costly for home networks. Nonetheless the standard is yet to overcome many obstacles for wide spread deployment. Several modifications are proposed in the IEEE 802.11 standard for different requirements. For example, the task group “G” is proposing 802.11g that would transmit data at the speed of wired Ethernet. The IEEE 802.3ae standardization is going on for 10 Gbps Ethernet. The goal is to achieve very high speed transport keeping maximum compatibility with already installed base of Ethernet802.3. The 10 Gbps Ethernet will provide almost zero latency service to users. Thus even when coverage area is increased, the remote application and services will appear as local. But the cost factor is not considered in such modifications.

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The IEEE 1394 working group defined a standard known as IEEE 1394. Truly speaking, the standard was originated by Apple Computer Company for desktop LANs. IEEE1394 is a low cost digital interface that can work over existing copper, fiber and co axial cables too. The Broadband Home Company has used co axial cable to extend IEEE1394 interface beyond the local audio & video cluster. The solution so provided looks like a virtual IEEE1394 wire connection to other IEEE1394 networks. It supports hot plugging, thereby allowing users to add and/or remove devices when the interface bus is active. It provides both hardware and software specification for peer to peer connection at different operating speed of 100, 200 or 400 Mbps. The enhancement in speed may go to support 800, 1600 and 3200 Mbps. It supports a scalable architecture to meet with different speeds of different requirements, thereby providing a cost effective solution. That standard integrates communication, entertainment, and computing to provide a single digital interface for consumer multimedia. It supports both asynchronous and synchronous types of data transfer as required in home networks. Asynchronous transfer is related to conventional data/computer file transfer. But for multimedia application of voice and video where delay is the most sensitive issue, transport at the guaranteed delay is done by synchronous or isochronous technique that is duly supported by IEEE 1394. It supports high speed communication at low cost interface. IEEE 1394 has been recognized as digital interface by many organization for different purposes that includes entertainment, consumer applications, digital TV, Home multimedia, conventional file transfer, digital video conference etc. A typical integration of several networks including IEEE1394 in a single cable is as shown in Fig. (12).

Phones

0

2.5 MHz

TV Channels

Cable

Infrared

5 MHz

55 MHz

1000 MHz

IEEE 13 94

Ethernet

1500 MHz

Fig. 12: Typical several networks in a cable.

3.18 Paging Paging is a one way message system unlike two way interactive mode system of communication of cellular. Pagers transfers message on wireless network; and thereby support mobility. Person having a pager can be contacted anywhere at any time. Pagers are quite useful for doctors, journalists etc. Paging is basically a back up system to telephones. It enhances the productivity of telephones. They work on a simple technique. The caller may dial the paging center through usual telephone and leave the message with operator along with callee’s pager number. The operator shall send the message to the callee’s pager. The message will then be flashed out on the callee’s pager with an activating signal. There are two basic types of paging transmission standards : POCSAG (Post Office Code Standard Advisory Group) and RDS (Radio Data System). In India frequency allocation for POCSAG is 134-168 MHz whereas idle band of AIR’s (All India Radio) existing FM network is used to RDS. AIR is operating RDS paging. POCSAG is under DoT (Department of Telecommunication). Different types of pagers are available in the market like numeric, alpha-numeric and recently introduced English type. In advanced countries, two way paging is being developed.

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3.19 VSAT Satellite communication started with the pioneer work of Dr. Artheer C. Clarke. He showed that using just three satellites placed each at 1200 apart from each other and at a height of about 36000 Kms from earth surface, world wide communication is possible. The satellites placed in orbits of about 36000 kms away from earth surface are known as geostationary satellites as they rotate in their orbits once in 24 hrs’. therefore, from any point on the earth, these satellites appear stationary to any person on earth. Due to two big advantages of satellite communication over other means of communication, satellite communication has a big appeal to users. It is said that “Going for Satellite means Going for Wireless Communication.” Wireless communication is more reliable, flexible and adaptable than wireless communication. We do communication by acoustics in wireless communication. Auther saying is “Going for Satellite means going for wide area coverage.” Wide area coverage has a natural attraction. Over the years, hence, the satellite communication has diversified its area of application and technologies. One of the major technologies is VSAT. VSAT (Very Small Aperture Terminal) is a cost-effective technology meant for networking computers and terminates mainly for the purpose of data communication. The VSAT network may be a wide network and extended in any remote location easily. The basic components of VSAT networks are: 1. a geological satellite. 2. a master earth station or hub. 3. micro earth stations(VSAT stations or nodes) VSAT node is made of VSAT ports, VSAT controller and VSAT antenna. The size of the antenna is 1.2 m × 1.8 m. Basically two types of topology are used in VSAT communication. Star and Mesh. Star topology uses TDM-TDMA (Time Division Multiplexing–Time Division Multiplexing Access) CDMA(Code Division Multiplexing Access) and mesh topology uses DAMA–SCPC (Demand Assigned Multiple Access—Single Channel Per Carrier). VSAT communication is broadcast communication. VSAT nodes cannot communicate directly to each other. They communicate with each other via master earth station or hub. Naturally VSAT communication is known as two-hop-communication. This means a VSAT signal from node has to travel at least 36000 × 2 x 2 = 144000 kms to reach another node. The delay for which shall be around 480 msec at least. The quality voice and video communication do not allow more than 80 msec delay between transmitter and receiver. Delay is not a issue for data transportation. VSAT communication is, hence, most suitable for data communication. The characteristics of VSAT are: 1. cost effectiveness is a big advantage of VSAT communication. An STD call between Delhi and Mumbai can be around Rs. 40 whereas a VSAT call may be about Rs 10. 2. Reliability and flexibility are always present in VSAT communication as it is wireless communication. A leased telephone line can have at most 90% up time. A VSAT line shall have around 99.5% up time. Due to wireless, ease of expansion of VSAT is there. This is how flexibile is VSAT network. VSAT(Very Small Aperture Terminal)communication is useful for huge organization like DVC,ONGC, IOC, BHEL, etc., as a means of cost effective date communication system for within the organization VSAT is a small dish antenna 60 cm or 120 cm which communicates

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with central hubs and terminals via satellites VSAT is cheaper then conventional earth station communication using satellites. Power budget calculation [26] shows that in order to meet with required bit energy to noise ratio a large antenna is essential for covering a wide area. Cost increases with antenna size. For intra organizational communication a small antenna is justified. DoT has allocated extended C band for VSAT communication in India. Nowadays VAST can also provide cost effective telephony and fax services. VAST is a low speed 1200 BPS data communication system and employs TDMA for accessing.

3.20 Mobile Satellite Service MSS (Mobile satellite Service) is a form of cellular like wireless communication. With terrestrial wireless networks, wireless communication is either not economically viable (examples are :remote areas, semi-hill areas etc.) or may not be physically possible (overseas, over large mountains). In such cases, wireless communication via satellites is an alternative proposition. Satellites can be of three types: GEOs (Geo-stationary Satellites), MEOs (Medium Earth Orbital Satellites) and LEOs (Lower Earth Orbital Satellites). Orbital altitude of GEOs, is 35786 km; whereas the same for MEOs and LEOs are respectively of the order of 10,000 km and 1000 km. INMARST, INTELSAT5, INSATAs are examples of GEOS. One example of LEOs is Odyssey with 12 satellites at an altitude of 10600 km. Odyssey is proposed to use CDMA technology and ground based switching for voice communication. Project 21 is another example of MEOs. In project 21, 10 satellites are used and they are placed at an altitude of 10500km. Project 21 is proposed to use TDMA technology to support both voice and data. Examples of LEOs are Iridium with 66 satellites oat an altitude of 765 km., Globstar with 48 satellites at an altitude of 1389 km. and Ellipso with 24 satellites at an altitude of 429-2903 km. They are respectively proposed to use FDMA/TDMA, CDMA and FDMA/CDMA technology. Iridium and Globstar are to support voice and paging. A good account of GEOS, MEOS and LEOs can be seen in[29,34]. GEOs have two major advantages : they are costly systems (due to high transmitter power and large antenna size) and round trip propagation delay is about 270 ms. Large round trip propagation delay is unwarranted in voice and in real time interactive communication. LEOs and MEOs can overcome the problem. In these systems many satellites are required to be placed in the orbits. As the satellites are not geo-stationary, for continuous communication, hand off operation among satellites are required. MSS can therefore support mobility. In MSS, MEOs, LEOs are base stations and they are on motion. Here lies the difference between cellular communication and MSS. In MSS, actually base stations are assumed as mobiles. Another disadvantage of LEOs and MEOs is their short life span. HEOs (Highly Elliptical Orbital Satellite) may also be used in MSS for wireless communication. A good account of MSS in personal communication is found out in[30,31]. Satellites Communication (GEOs/MEOs/LEOs) In general “going for satellites” means going for two important philosophies - going wireless and going for large area, even upto whole world coverage. These greatly influenced the use of satellites in communication. Till date, the major satellites involved in communication are GEOs (Geo Stationary Satellite). GEOs are placed at orbit of high 35800 Kms away from earth surface, and therefore they move around the earth once in 24 hrs. Thus GEOs look stationary at anywhere relative to earth, and communication using fixed antenna is possible. A world wide coverage with just three GEOs is possible, if they are placed at equidistant apart of 1200. But there are number of disadvantages with GEO based communication: (1) GEO has poor elevation at higher latitudes and no coverage of the polar regions, (2) mobile communication using GEO under INMARST (International Maritime Satellite Organization) is possible to

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provide voice, data, telex and facsimile services to ships; but for this, there remains the requirements of very high power for both terminal and the space craft, (3) the large distance between earth and GEO causes a high propagation loss of about 200 db and a time delay of about 350 msec in one way. Such long delay is not acceptable to 80% of users. MEOs are placed at orbit height of about 10,000 Km or above; whereas LEOs are placed at orbit height of 2000Km or less. By this LEO overcome the disadvantages of GEO based communication pointed out above. Besides, there some major differences between LEO based and GEO based communications. # The communication in LEO is done through a constantly moving and tracking switching network and antenna rather than fixed system of GEO. Mobile communication in LEO is based on the relative mobility. LEO systems move and moving users appear stationary. For example in Iridium system, the LEO speed relative to earth is 26,676 Km/ hr, whereas average mobile speed is around 90 Km/hr. # The GEO based communication is single hop (earth–satellite–earth) communication; while LEO based communication is multi hop communication. # Under LEO, the communication across the world is low cost. For example while a typical GEO can provide about 10,000 channels for global services, the LEO can provide 7000 channels for regional services and 35,000 to 70000 channels for global services. This typical means the cost per channel for global services in LEO is about one half of that of the for global services in GEO. LEO is effective for global services rather than regional services. # LEOs are smaller than GEOs. The mass of LEO satellite range from 50 to 700 Kg (whereas that of GEOs range from 1800—2000 Kg). Therefore economic multiple launching of LEOs is possible. The variety of services offered by the satellites was divided into three groups by ITU (International Telecommunication Union). These are: (1) Fixed Satellite Service (FSS), which offers radio communication services between fixed location in earth through one or more satellites. (2) Broadcast Satellite Services (BSS) which provide direct reception of satellite broadcast by public and/or community and (3) Mobile satellite Service (MSS) which provide a communication between mobiles through one or more satellites. As in past twenty five to thirty years, FSS and BSS shall continue to be served with GEO mainly. LEOs shall dominate MSS. It is said, “The LEO and MEO systems offers an innovative approach to providing service to a country, a region, or to the whole world. Instead of transmission to and from a fixed point in sky (as for geostationary satellite systems) the user transmits to and receives from a network of lower altitude satellites, that move overhead with some satellites disappearing from view as others come over the horizon. The system can provide service to all parts of the world as the low altitudes satellites pass over different parts of the earth.” LEO Systems LEOs are classified into two groups: “Little-LEOs” and “Big-LEOs”. The little-LEOs group consists of satellites, which are small in size and low in weight. Little LEOs are expected to provide services of only low bit rates of the order of 1 Kbps (kilo bits per second) and they are placed near orbit height of around 1000 km. Naturally they are used for non-voice services. The frequency band allocated for mobile satellite services (MSS) under little—LEO group are: 148-150.50 MHz (uplink) and 137-138 MHz (down link). Big-LEO group of satellites are expected to provide near-toll-quality voice service and other related services like paging, data communication, facsimile, and position location. BigLEO group contains MEO (International Circular Orbit) satellites. The important three BigLEO systems are: Globalstar, Odyssey and Iridium.

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Iridium Iridium system was proposed by Motorola to provide global services of voice, data, fax, paging, RDSS; and was scheduled to operate in 1998. The cost of the system is about US$ 3.4 billion. The system is composed of 66 (77) satellites with 11 satellites in each of 6(7) polar orbits placed at the orbit height of 780 Km above earth surface. Satellite shall provide 3168 cells out of which only 2150 cells shall remain simultaneously active to provide global coverage of mobile/ cellular telephone service. In the system the same frequency band 1616–1626.5 MHz shall be used for both uplink and downlink communication on time-shared basis. Message on one telephone to another is transmitted from mobile to satellite using 23 GHz (22.55–23.55) intersatellite link until the satellite viewing the destination mobile is reached. The system uses FDMA (Frequency Division Multiplexing Access) and TDMA (Time Division Multiplexing Access) on uplink and downlink respectively. The connection to the terrestrial network is done via earth station gateway. Voice circuits per satellite are 1100. Voice service rate is 2.4 Kbps. Data service rate is 7.2 Kbps. Modulation technique used in the system is OPSK ( Quadrature Phase Shift Keying). Footprint diameter of each satellite is 4700 Km. and therefore satellite visibility is 11.1 minutes. Satellite life span is rather less and it is 5 years. Satellite antenna type is fixed and six feet in size. Beams per satellite is 48 and therefore total beams in the system are 3168. Feeder uplink and downlink frequency 27.5-30 GHz and 18.8-20.2 GHz. Minimum and maximum one way propagation delay are respectively 2.6 msec and 8.22 msec. Airtime charge per minute is US$ 3.0. Iridium System is working but not to the level of satisfaction expected before launch. Globalstar Qualcomm proposed Globalstar LEO system to provide services of voice, data, facsimile and RDSS. The Globalstar system shall use 48 satellite in 8 polar orbits. The orbit height is 1400 Km above earth. It provides global coverage and can work with existing PSTN (Public Switched Telephone Network). Calls are granted through satellites only when access is available to the terrestrial all network. The PSTN can be used via gateways for long distance communication. The system does not support intersatellite link. The gateways to the PSTN shall use 6.5 GHz and 5.2 GHz respectively for uplink and downlink communication. Access technology for MSS is CDMD (Code Division Multiplexing Access) via L-band (1610.0–1626.6 MHz) and S-band (2483.5–2500.0 MHz) for uplink and downlink communication. The modulation technique used in the system is QPSK. The system can support 2000 - 3000 voice circuits / satellite. The voice and the data service rate in the system are 2.4 to 9.6 KBPS respectively. Minimum and maximum one way propagation delays are respectively 4.63 msec and 11.5 msec. Mobile terminal cost is about US$ 750. Air time charge per minute is 30 cents. The satellite foot print diameter is 5850 Km. The satellite visibility is 16.4 minutes and lifespan is 7.5 years. The satellite on a orbit mass is 450 Kg. The system cost is US$ 1.8 billion. The satellite antenna is fixed and size is 03 feet. Feeder uplink and downlink frequencies are respectively 5.091 - 5.250 GHz and 6.875 - 7.875 GHz. The satellite output power is 1000 watt. Beams per satellite is 16 and total beams are 768. Comparing Iridium with Globalstar, a report says globalstar has capital costs (at $1 billion) one-half Iridium, circuit costs one-third Iridium’s and terminal cost (at $750 each) onefourth Iridium’s. With no intelligence in space, Globalstar relies entirely on the advance of intelligent phones and portable computer devices on the ground; it is the Ethernet of satellite architectures. Costing one-half as much as Iridium, it will handle nearly 20 times more calls.

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The advantages of Globalstar stem only partly from its avoidance of complex intersatellite connection and use of infrastructure already in place on the ground. More IMPORTANT is its avoidance of exclusive spectrum assignments. Originating several years before spread-spectrum technology was thoroughly tested for cellular phones, Iridium employs time division multiple access, an obsolescent system that requires exclusive command of spectrum but offers far less capacity than code division multiple access. “It is said that” Iridium’s voice service cannot complete with GlobalStar’s cheaper and more robust CSMA system. “It is also reported that “Iridium satellite together use 80% more power than Globalstar’s, yet employ antennas nearly twice as larger and offer 18.2 times less capacity per unit area”. Odyssey TRW proposed a system known as odyssey to provide voice, data, facsimile and RDSS services on global basis. In the system, 12 satellites and 3 polar orbits are used. The orbit height is 10370 Km above earth surface, and therefore this system is better known as MEO system. The orbital period of satellites is 359.5 minutes and visibility is 94.5 minutes. The satellite mass is 2207 Kg. Footprint diameter of each satellite is 10540Km. The access technology of the system is CDMA, and modulation technique is QPSK. The system operates at L and S brand. The mobile uplink and downlink frequencies are respectively 1601.0–1626.5 MHz (L brand) and 2483.5–2500.0 MHz ( S brand). The system supports to 3000 to 9500 voice circuits per satellite. Voice and data services are made with respectively 4.8 KBPS and 2.4 KBPS. Delay are respectively 34.6 msec and 44.3 msec. Airtime charge in the system is US$ 0.65 per minute. Satellite antenna type is steer able. Uplink and downlink feeder frequencies are 29.1– 29.4 GHz (ka - band) and 19.3–19.6 GHz (ka band) respectively. The system supports 61 beams per satellite, thereby supporting total 732 beams. Satellite output power 6177 watt. Ellipso Ellipsat proposed a LEO satellites system known as “ELLIPSP” to provide voice, data, facsimile, and RDSS, using 15 (9) satellites placed in 3(1) polar orbits. The orbit height is 7800 Km over earth surface and provides coverage over entire northern hemisphere and to southern hemisphere upto 50 south latitude. It uses L and C-band for communication. The mass of satellite on orbit is 300Kg. The system supports voice and data at 4.2 KBPS and 0.3 to 9.6 KBPS respectively. The satellite life span is 5 years. Air call charge per minute is US$ 0.50. Access technology is CDMA. ICO Hughes proposed ICO system to provide services of voice, data, fax, paging, massaging and position location employing 10 satellites in 2 orbits placed at 10355 Km. The system is MEO rather than LEO. The system supports voice and data at the rates of 4.8 KBPS and 2.4KBPS respectively. The satellite life time is 10 yrs. Air charge is US$ 1 to US$ 2. The system covers service all over world. Orbit period is 358.9 minutes and satellites visibility is 115.6 minutes. The down-link and the up-link frequencies for MSS are 1980.0–2010.0 MHz and 2170.0– 2200.0 MHz respectively. Satellite antenna type is fixed. Feeder uplink and downlink frequencies are respectively 5.091–5.250 GHz (C-band). The system supports voice and data service at the rates of 4.8KBPS and 2.4 to 9.6 KBPS respectively. Minimum and maximum one way propagation delays are respectively 34.6 msec and 48 msec. Air time charge per minute is 2.00.

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Teledesic Teledesic system of LEOs is a different class. Difference stems from the application point of view. The system is aimed at providing wireless broad band access and computer networking. Little LEOs are equivalent of paging. Big-LEOs like iridium, globalstar and ICO, are equivalent of fiber. The system comprises 840 small satellites in proposed 21 orbital planes and 20000 super cells on the earth in order to provide broadband-on-demand service by 2002 for 99% of the earth. The orbit height is 700Kms. Teledesic system is expected to use ha—band of frequencies, between 17 GHz and 30 GHz. And antennas of size 66 cms. The Teledesic system is Giga Band system. A comparative study says” in the long run Iridium could be trumped by Teledesic. Although Teledesic has no such plans, the incremental cost of cost of incorporating an “L” band transceiver in Teledesic, to perform the Iridium functions for voice would be just 10% of Teledesic’s total outlays, or less than $ 1 billion (compared with the $ 3.4 billion initial capital costs of Iridium). But 840 linked satellites could offer far more cost-effective service than Iridium’s 66. Iridium’s dilemma is that the complexities and costs of its ingenious mesh of intersatellite links and switches can be justified only by offering broadband computer services. Yet Iridium is doggedly narrow band system focused on voice. The evolutionary process of development of personal communication shall go on using existing cellular, cordless, satellites, wireless data networks, WLL ( Wireless Local Loop) VAST (very small Aperture Terminal), wireless centrex / PBX, and other GMS (Third General Mobile System/Cellular) and MSS (Mobile Satellite Service) etc. But the use of MSS in personal communication may be revolutionary of evolutionary, which remains to be seen. The MSS under different LEO projects-both big LEOs and small LEOs, is believed to be a high hope of implementing personal communication.

3.21 Wireless ATM ATM (Asynchronous Transfer Mode) technology is believed to be the only suitable presently available technology for integrated services-present and future, time sensitive and time insensitive, voice, video and data. Thus ATM can support multimedia and can be service or application independent technology for transport and switching. Therefore, a future direction for development of wireless ATM is initiated. WAND (Wireless ATM Network Demonstrator) is such a project of European unions. HIPERLAN standards of 5 GHZ shall aim towards wireless ATM. In USA, a high speed multimedia network using ATM is under development. It is targeted to operate at 25 Mbps using 25 MHz channel in 5 GHz band (5.15-35 and 5.7255.875 GHz). On the other hand, work has started to provide air interface to internet.

3.22 Changing Scenario of Internet Internet is going to have several changes and version like IPv4 to IPv6, VoIP, Internet2, Wireless Internet and under sea /Cable Internet. But most immediate is from IPv4 to IPv6 and VoIP. 3.22.1 IPv6 Presently Ipv6, Internet works on IPv4 (Internet Protocol Version 4) as defined in RFC791. By the middle of 1990s, by the time of which the IPv4 became about 15 years old, it was recognized that there are several limitations in the IPv4. Table (XI) lists the major studies on the run up of IPv4. Two important limitations are the inadequate address space available with 32-bit address space of IPv4 and inability of the IPv4 to support real time services or time-sensitive services. The 32-bit address space is not sufficient to cope up with the growing Internet users.

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Since it is estimated that the Internet has been growing by a factor of two every year, the underlying principles and assumptions based on which IPv4 was designed are going to be invalid. What was duly sufficient for a few million users or a few thousands of networks will no longer can support a world with tens of billions of nodes and hundreds of millions of networks. Inability of IPv4 to support real time services was the stumbling block to realize Internet telephone. IPng (Internet Protocol Next Generation) initiative (RFC 1752) was then, started by the Internet Engineering Task Force (IETF). By 1996, the IETF proposes IPv6 (Internet Protocol Version 6) under IPng initiatives, which is supposed to solve the problems of IPv4 including the two major limitations mentioned above. IPv6 is therefore the future replacement of IPv4. From the experience over IPv4, it was felt that new version should take care of: More addresses, Reduced overhead, Better routing, Support for address renumbering, Improved header processing, Reasonable security and Support for mobility. Under the IPng initiatives the main techniques investigated were: • TUBA that refers to TCP (Transmission Control Protocol) and UDP (Users’ Datagram Protocol) with bigger addresses • CATNIP that means common architecture for the Internet. The main idea is to define a common packet format that will be compatible to IP, CLNP (Connectionless Network Protocol) and IPX (Internet work Packet Exchange). CLNP has been proposed by OSI (Open System Interconnection) as a new protocol to replace IP, but never been adopted because of its inefficiency • SIPP (Simple Internet Protocol Plus) that proposes to increase of the number of address bits from 32 to 64, and to get rid of unused fields of IPv4 header As none of the above three was seen to be suitable. As such, a mixture of all these three along with other modifications was suggested in RFC 1883. The RFC 1883 suggested the modifications as below: • Expanded Addressing in suggesting 128 bits for address that may allow more levels of address hierarchy, increased address space and simpler auto configurable addressing • Improved IP header format by dropping the least used options • Improved support for Extensions that will bring flexibility in operations • Flow Label that will make the real time services possible over Internet Based on the experience gained in operation of IPv4 over about 20 years, the design of IPv6 has considered four major simplifications: • assigning fixed format to each header. This ensures the removal of header length field that is essential in IPv4 • removing header checksum. The main advantage in removing header checksum is to diminish the cost and the time delay in header processing. This may cause the data to get misrouted. But experience has shown that the risk is minimal as most of data pack is encapsulated by the packet checksum at other layers like MAC (Media Access Control) procedure in IEEE 802.X and in adoption layer of ATM (Asynchronous Transfer Mode) etc. • removing the hop by hop segmentation procedure • removing TOS (Type Of Service) field that IPv4 provides, since experience has shown that this field has ever been set by applications. On the other hand, IPv6 has considered two new fields, flow label and priority. These are included to facilitate the handling of real time services like voice, video and high quality multimedia etc.

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Thus the IPv6 was finally come up with packet format as in the Fig. (13). The final specifications of IPv6 were produced in RFC 1883. The new features of IPv6 are: Version (4 bits)

Priority (4 bits)

Payload Length (16 bits)

Flow label (24 bits) Next Header (8 bits)

Hop Limit (8 bits)

Source Address (128 bits) Destination Address (128 bits) Variable length TCP pack (which is TCP header + Payload)…….. Fig. 13: IPv6 Packet Format

• A fixed and streamlined 40-byte header: IPv6 is having fixed header bytes like that in ATM (Asynchronous Transfer Mode) cell. This makes the node processing delay to minimize, and thereby becomes more suitable for real time services like voice, video and multimedia. • Expanded addressing capabilities: A 128-bit address space in IPv6 instead of 32 bit as in IPv4, is believed to ensure that the world won’t run out of IP addresses. The 128 bit address size gives rise to a total of 256 × 1036 different addresses. It is expected the Internet under IPv6 to support 1015 (quadrillion) hosts and 1012 (trillion) networks. The Internet under IPv4 can support maximum 232 hosts. Therefore the IPv 6 address space is about 64 × 109 times more that that of IPv4. This is why it is expected that future and exponential growing demand for Internet connection be met with IPv6. • New Address Class: Besides unicast and multicast, IPv6 has the provision of anycast addressing. Anycast address allows a packet addressed to an anycast address to be delivered to any one of a group of hosts. • A single address associated with multiple interfaces • Address auto configuration and CIDR (Classless Inter-domain Routing) addressing • Provision of extension header by which special needs like checksum, security options may be introduced. • Flow labeling and priority: Flow level and priority headers are used to comfortably support the real time services. By assigning higher priority to the real time packets, the necessity of time sensitiveness is restored. Data packets and for that purpose time insensitive packets are assigned low priority and serviced by the best effort approach. As per RFC 1752 and RFC 2460, this new feature allows “ labeling of packets belonging to particular flows for which the sender requests special handling, such as a non-default quality of service or real-time service.” Hence video and audio may be treated as flows whereas traditional data, file transfer and e-mail may not be treated as flows. • Support for real time services • Security support that could be eventually seen as the biggest advantage of IPv6. Today, billion dollars business is done over Internet. To keep the business secure, public crypto system has emerged out as one of the important tools. IPv6 with its ancillary security protocol has provided a better communication tool for transacting business over Internet • Enhanced routing capability including support for mobile hosts. IPv6 as such is not simple extension of IPv4, but a definite improvement over IPv4 in order to meet growing demand of Internet connectivity and the services of real time communication via Internet.

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The functions of IPv6 headers that is of base headers of fixed 40 bytes are: • Version field (4 bits). It contains the version number. Versions are 4 and 6. For version 6, this field is 6 (i.e. 0110). The various assigned values for IP version label are shown in table (12). But it must be remembered that just putting a number “6” or “4” does not make the corresponding IP packet. For the corresponding IP packet the proper format is required to be made. • Priority (4 bits). The bits in the field indicate the priority of the datagram. The priority levels are 16 from 0 to 15. The first 8 priority levels (from 0 to 7) are for the services that provide congestion control. If the congestion occurs, the traffic is backed off. These are suitable for non-real time services like data. The different priority levels under the first 8 levels are: 0 that defines no priority, 1 that defines background traffic like Netnews, 2 that defines unattended transfer like e-mail, 3 remains reserved, 4 that defines attended bulk transfer like FTP (File Transfer Protocol), NFS, 5 remains reserved, 6 that defines interactive traffic such as Telnet, X-windows, and 7 that defines control traffic such as SNMP (Simple Network Management Protocol) and routing protocols. The higher 8 priority levels (from 8 to 15) are used for services that will not back off in response to congestion. Real time traffics are examples of such services. The lowest priority level of this group 8 refers to traffic that most willing to be discarded on congestion and the highest priority level15 is for traffic that is least willing to be discarded. · Flow level (24 bits). It is proposed to be used to identify different data flow characteristics, which will be assigned by the source and can be used to label packets. The packet labels may be required to provide special handling of packet by IPv6 routers, such as defined quality of service (QoS) or real time services. The combination of the sender IP address and the flow label creates a unique path identifier that can be used to route the datagrams more efficiently. The field is still being experimented. Flow is actually a sequence of packets coming from a particular source and destined for a particular destination. A flow may require a special handling by routers. Each flow is uniquely defined by the combination of the source address and a non-zero flow label. The flow label can be from (000001)H to (FFFFFF)H in hex. The packets having no flow label are given a zero label. All packets in the same flow must have same flow label, same source and destination addresses and same priority level. The initial flow label is obtained by the source by pseudo random generator, and the subsequent flow numbers are obtained sequentially. • Payload length (16 bits): The field indicates the total size of the payload of the IP data gram that excludes header fields. It can define up to 65,536 bytes of payload. • Next header (8 bits): The field indicates which header follows the IP header. The next header can be either one of the optional extension headers used by IP or the header for an upper layer protocols such as UDP or TCP. The field defines the type of extension header. For example 0 defines IP information, 1 defines ICMP (Internet Control Message Protocol) information, 6 define TCP information, 44 defines fragmentation header, 51 defines authentication header and 80 defines ISO (International Standard Organization) /IP information. Each extension header again contains an Extension Header Field and a Header Length Field (Fig. 14). When there is no other extension header, the next header will be TCP and hence the next header field will contain 6.The length of the base header is fixed 40 bytes. The extension header gives the

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functional flexibility to the IPv6 datagram. Maximum six extension headers can be used. The extension headers may be source routing, fragmentation, authentication and security etc. IPv6 has currently defines six extension headers: (1) Hop by hop option header, (2) Routing header, (3) Fragment header, (4) Authentication header, (5) Encrypted security payload header and (6) Destination options header. If one or more extension headers are used, they must the order in which they are presented above. For example, if Authentication header and routing extension header are to be used, the extension header fields must follow as: (1) main IPv6 header, (2) routing extension header (3) Authentication header and (4) TCP header with data. Each extension header must have one 8-bit next header field. For all extension headers except the fragment header(as in case of fragment header the flags and offset is 16 bits fixed), the next header field is immediately followed by a 8-bit extension header length that indicates the length of current extension header in multiple of 8 bytes. In the last extension header the next header field contains the value 59. The example that we considered earlier, the next header in main IPv6 packet will contain the routing extension header, the next header field in the routing header will show the authentication extension header, and the next header field of the authentication header will contain the value 59. • Hop Limit (8 bits): This field indicates the maximum number of hops that the datagram is allowed to traverse in the network before it reaches its destination. If after traversing this maximum number of hops the data gram does not reach the destination, the datagram is discarded from the network. The field is used to avoid the congestion that may be caused by the datagram. Each router decreases the hop limit by 1 while releasing the datagram to the network. When the hop limit reaches 0, it is deleted. The hop limit of IPv6 is exactly what is called Time To Live in IPv4. The new name of Hop Limit has been given as the name suits better to its function. • Source Address and Destination Address (Each 128 bits): Both the addresses can be called IP address and are described in RFC 2373. IP address that defines the original source of datagram is called source address. The IP address that defines the final destination of the datagram is called the destination address. The three main groups of IP addresses are: unicast, multicast and anycast. Unicast address defines a particular host. A unicast packet is identified by its unique single address for a single interface NIC (Network Interface Card), and is transmitted point-to-point. A multicast address defines all the hosts of a particular group to receive the datagram. The anycast address will be addressed to a number of interfaces on a single multicast address. The anycast packet therefore goes to the closer interface and does not attempt to reach the other interfaces with the same address. A multicast packet, like anycast packet has a destination address that is associated with number of interfaces, but unlike the anycast packet, it is destined to each interfaces with that address. Unlike IPv4, IPv6 addresses do not have classes. But the address space of IPv6 is subdivided in various ways for the purpose of use. The sub division is done based on leading bits of addresses. The present division of IPv6 address space is as shown in table(XIII). The IPv6 address space is huge enough. So a portion of the IPv6 is reserved for computer system using Novell’s Internet Packet Exchange (IPX) network layer protocol, as well as the Connection Less Network Protocol (CLNP).

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It is found that several fields present in IPv4 are no longer present in the IPv6; and notably among them are: • Checksum field. The main issue of designing IPv6 was the fast processing of packets. This results in designing with fixed header fields and removing the redundant fields. The error check is done at upper layers namely TCP/UDP. As such the check sum field further at IP layer was assumed as redundant and accordingly it was removed from IPv6. Again with check sum at IPv4 packet, the error checking at every node was essential. It was a very time consuming and costly thing and duly unwanted at IPv6. • Options field. Dropping of options field has made the IPV6 a fixed header packet. Of course if required the IPv6 packet may use next header field for the purpose of header extension. • Fragmentation. The IPv6 version has dropped the fragmentation and reassembly feature at intermediate routers. The data is fragmented for packetization at the source only. The reassembly is done at destination only. If a IP packet received by any intermediate router is found as too large to be forwarded on the outgoing link, the router simply drops the packet; and in turn send a ICMP error message of “Packet Too Big” to the sender. Sender on receiving the ICMP error message of “Packet Too Big”, retransmit the data with smaller packet size. Actually the fragmentation and the reassembling the datagram at routers is a time consuming matter; and removing these from routers’ functions to end users’ functions, makes the network to speed up. ICMP (Internet Control Message Protocol) ICMP for version IPv4 is used by hosts, nodes, routers and gateways to communicate network layer information to each other. ICMP is specified in RFC 792. ICMP information is carried as IP payload like TCP or UDP information. ICMP messages are basically used for error reporting among others (Table XIV). An ICMP message is made of a type field and a code field and also the first eight bytes of the IP datagram for which the ICMP message is to be generated in the first place so that the sender can know the packet that caused the error. A new version of ICMP defined for IPv6 in RFC 2463. The new ICMP has the reorganized existing types and codes as well as added new types and codes. The added new ICNP type includes “ Packet Too Big”, and “unrecognized IPV6 options” among others. Auto configuration and multiple IP addresses IPv4 address structure is a stateful address structure, which means that if a node moves from one subnet to another the user has either to reconfigure the IP address or to request for a new IP address from DHCP (Dynamic Host Configuration Protocol). With DCHP, an IP address is leased to a particular host or computer for a defined period of time. But IPv6 supports a stateless auto configuration whereby on moving from a subnet to another subnet a host can construct its own IP address. This is done by host on adding its MAC (Media Access Control) address to the subnet prefix. IPv6 also supports multiple addresses for each host. The addresses can be either valid, deprecated or invalid. With valid address new and existing communication may be done. With deprecated address, the existing communication may be done. With invalid address no communication is done. Address Notation Like IPv4, the IPv6 has special notation for representing the IP addresses. The IPv6 address is represented by hexadecimal colon notation. The 128 bits are divided into eight sections

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each of two bytes in length. Each of the eight sections is represented in four hex digits (or a pair of hexagonal numbers separated by a colon. A pair of hex means a byte) and is separated by a colon. One example is: AB12:0978:CF56:00FE: 1234:127E:CB65:7890 The notion allows to drop leading zeros. This means and for example 0045 can be just represented as 45, and 0A456 can similarly be represented as A456, and 0000 as simply 0. The notion also allows removing a zero leaving a colon, and therefore for example 2456:AC67:0:0:67:D4E5:A456:A678 can be written as 2456:AC67::67:D4E5:A456:A678. The stated double colon notation can be used at the beginning or at the end of an address but only once. The double colon at the start indicates leading zeros and that at the end indicates contiguous zeros at the end. If more that one location double colons are used, it will not possible to know how many zeros are there at a particular double colon location. This is why double colon notation is used only once. By counting the other bytes, the number of zeros at the single double colon location can be found out. IPv6 and IPv4 address compatibility For a long interim period, the IPv6 and the IPv4 have to coexist. During this period, an IPv4 address can be converted to an IPv6 address by pre pending 12 bytes of zero. For example, an IPv4 address 126.34.67.10 will be converted to an IPv6 address as 0:0:0:0:0:0:0:0:0:0:0:0:126.34.67.10 or::126.34.67.10. Similarly a host having an IPv4 address as 128.67.56.9 may be mapped (read as IPv4 mapped IPv6) could have an IPv6 address as ::AC45:128.67.56.9. The different special notations of version 4 and version 6 will make them separable. Version (4 bits)

Priority (4 bits)

Flow label (24 bits)


Next Header (8 bits)

Hop Limit (8 bits)

Source Address (128 bits) Destination Address (128 bits) Next header

Header length

Variable header fields …

………………………………………… Next Header

Header length

Variable header fields………….

………………………………………….. Variable length TCP pack (which is TCP header + Payload)……. Fig. 14: Illustration of use of Next Header Fields

Table 11: Reports of different studies on IPv4 address space run up Study group

Recommendation

Two leaders of IETF Address Lifetime Expectations (ALE)’s recommendation

IPv4 address space would be exhausted in 2008 and 20018 respectively

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Final recommendation of ALE in 1994

IPv4 address space will be exhausted at some time between 2005 to 2011.

American Registry for Internet Numbers (ARIN)’s report in 1996

All class A Address has been assigned; 62% of class B address and 37% of Class C address have been assigned

Table 12: Different IP version labels Value

Key

0

Description Reserved

4

IP

Internet Protocol (RFC 791)

5

ST

ST datagram Mode (RFC 1190)

6

SIP

Simple Internet Protocol (IPv6)

7

TP/IX

TP/IX: The Next Internet

8

PIP

The P Internet Protocol

9

TUBA

TUBA

10-14

Unassigned

15

Reserved

Table 13: IPv6 address space subdivision based on prefix assignments of bits Prefixed bits

Use of address space

0000 0000

Reserved

0000 0001

Unassigned

0000 001

Reserved for NSAP application

0000 010

Reserved for IPX application

0000 011 0000 1 0001

Unassigned Unassigned Unassigned

001

Aggregatable Global Unicast address

010 011 100 101 110 1110 1111 0

Unassigned Unassigned Unassigned Unassigned Unassigned Unassigned Unassigned

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1111 10 1111 110 1111 1110 0

Unassigned Unassigned Unassigned

1111 1110 10

Addresses for Link Local use

1111 1110 11

Addresses for Site Local use

1111 1111

Multicast addresses

63

Table 14: Selected ICMP messages ICMP type

CODE

Remarks

0

0

Echo reply (to ping)

1

0

Destination Network unreachable

2

1

Destination Host unreachable

3

2

Destination Protocol unreachable

4

3

Destination Port unreachable

5

6

Destination Network unknown

6

7

Destination Host Unknown

7

0

Source quench (Congestion Control)

8

0

Echo requested

9

0

Router advertisement

10

0

Router Discovery

11

0

TTL expired

12

0

IP header bad

3.22.2 Voice over Internet or Internet Telephony Internet has established itself as the most important and the single most tool of global information age. It was developed for transporting packet data, a non real-time service. But today, Internet telephony has emerged as an important technology. Internet telephony is supposed to carry real time and jitter-free voice over Internet. Active and hectic researches are being carried over the subject of VoIP (Voice over Internet Protocol). Generally speaking the use of Internet for all real time services, like voice, video, and multimedia is being explored. Table (15)[12] shows a growth estimation of VoIP traffic. There are several motivations[11] for transmitting voice over IP. These are: (1) long distance calls at low cost and may be of low quality, (2) cheaper two in one service, (3) Use of PC as a true multimedia terminal, (4) one connection for all services, (5) local exchanges can support telephone with Internet as backbone and without high investment in expensive back bone infrastructure, and (6) use of packetized voice allows voice compression that in turn decreases transmission time and cost. Earlier, telecommunication traffic or telephony connections outnumbered the data traffic. The future is to see the explosion of data traffic. When there will be crossover is debatable, but sooner or later data

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traffic will dominant the telecommunication traffic. “ Consequently, now should be the time for datacom to act as a carrier for telecom.” But Internet, as such cannot be used to carry real time service as it was designed to carry data and as the characteristics of real time services like voice and video are different from data. Table (16) shows the different characteristics and different requirements of voice, video and data The need to deploy Internet for the real time services like voice and video, have lead to redesign some features of Internet. The important two features related to this emerging issue are: (i) redesign of IP datagram format, and (ii) to use RTP (Real Time data transfer Protocol) and IP for carrying voice over conventional IP datagram and Internet. It is believed that with deployment of Ipv6, VoIP will be reached. Table 15: Projected growth of IP telephone (A) As per[12] Voice IP Traffic 1998

310 million minutes

1999

2.7 billion minutes

2004 (expected)

135 billion minutes

(B) As per [16] Year

Average unit (millions per year)

Unit growth rate (%)

Yearly revenues (millions)

Yearly revenue growth rate (%)

2000

3987.2

256

388.75

209

2002

22,386.2

162

1511.07

136

2004

167,896.2

114

8814.55

88

2006

587,636.9

75

22036.38

46

Table 16: Characteristics of different services Voice

LAN data

Transactional Data

Video

Predictability

Constant/On-Off

Bursty

Highly bursty

Constant/Bursty

Bandwidth/ Bit rate

Very Low to Low

Medium to High

Low to Medium

High

Delay/Jitter

Sensitive

Tolerant

Tolerant

Sensitive

Loss

Sensitive/ No recovery

Sensitive but can recover

Sensitive but can recover

Very sensitive/ No recovery

Error/Integrity

Can tolerate

Can not tolerate

Can not tolerate

May tolerate

Technical problems of voice packet transmission over Internet PSTN (Public Switched Telephone Network) based on circuit switching provides voice service with guaranteed quality of service. This is not the case in case of voice service provided by Internet that acts on packet switching. Many technical challenges the voice packet faces while

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in transition over packet switching network like Internet. These include packet loss, packet transfer delay and jittering delay. Voice communication is involved with human interaction. As such, a few losses of the voice packets could be tolerated due to human intelligence and perception involved in recovery. But too much loss of the voice packets may seriously degrade the voice quality. Moreover, PSTN is a reliable voice service provider whereas Internet is not, as because Internet is datagram based. Table 17: End to end voice packet latency delay Delay source

Typical value (end to end or Phone to Phone) in ms.

Recording

10-40

Encoding/Decoding (CODEC)

Each 5-10/Both together 10-20

Compression/Decompression (SPEECH)

Each 5-10/Both together 10-20

Internet Delivery

70-120

Jitter buffer

50-200

Average

150-400

Delay is the more serious issue for real time interactive services like voice. By delay it is meant that the time difference between the time the sender releases the packet to the network and the time at which the receiver receives the packet from the network. Delay refers to: (1) total transfer delay of a packet that includes coding/decoding delay, propagation delay, transmission delay, node processing and queue delay, switching and routing delay; and (2) jittering delay that refers to the phase delay between two successive packets. Typical delay from different sources are as in Table (17)[12]. If the total delay exceeds a certain value, customers may get irritated to the service. A statistic says that a delay up to 80 msec between the caller and callee is acceptable but beyond it causes irritations to the users. The total delay is a variable quantity, and it varies from packet to packet. The jittering delay is very serious issue. If the phase lags between the voice packets at the source and destination varies, the service quality degrades. The phase lag between packets differs from the source end to the destination end because the total transfer delay varies from packet o packet. Due to jittering problem, a sending voice “I shall go home” may be received as “I shall go home”. Compared to the transmitter, the phase delay between “i” and “shall” has increased and that between “shall” and “go” has reduced to zero at the receiver. While the total delay could be limited by increasing the bit rate capacities of the link and by adopting efficient routing technique among others, the jittering effect can not be solved so simply. There are several techniques to reduce the affect of the jittering problem. One such technique is known as accelerating and de accelerating. In fact the jittering problem is due (Di + 1 – Di) which is finite and a variable. Here, Di + 1 and Di are both variable quantities and represent respectively the total transfer delay of (i + 1)th packet and ith packet. To avoid the jittering effect, it is required that Di + 1 – Di = 0. In the accelerating and de accelerating technique, at the receiver end a variable delay (say Wi for ith packet) is caused to each packet such that Di + Wi = K, a constant for all packets (i.e. for i = 0, 1, 2, 3…..) before delivery of the packets to the terminal equipment for play back. By the process, the variable delay caused by the network between two successive packets is made zero as (Di + 1 + Wi + 1) – (Di + Wi) = 0. This ensures that the phase delay between packets at the transmitter remains same at the receiver. The scheme is illustrated in the Table (18). As illustrated in the table, the success of the technique depends on the choice of K.

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Table 18: Illustrating of accelerating and de accelerating technique to cope up with the problems of jittering Instant at which a packet is released at the transmitter (xi) in ms th.

Variable delay with which the packet reaches the receiving node in ms (Di)

Variable delay (Wi) caused at the receiving buffer (100–Di) in ms (K has been chosen as 100 ms)

Delay with which the packet is delivered to the terminal device (xi +100ms)

Packet-1

0

80

20

100

Packet-2

10

70

30

110

Packet-3

15

85

15

115

Packet-4

25

100

0

125

Packet-5

30

110

–10

130

(packet-4 is the marginal case. Packet-5 is the failed case. Both could have been avoided had the constant K been chosen more than 110 ms in this case. So the success of the technique depends on the choice of fixing K) VoIP is going to be a dominant service issue of IP. VoIP has several motivations as we discussed earlier. PSTN supports only toll-quality sound (4 KHz sound), and not suitable for high-fidelity sound. VoIP can support higher grades of sound. This will be another major driving factor for VoIP. But there are several issues that need to be resolved before VoIP is used. Standards are still not finalized, although H.323 of ITU is being projected as a possible standard. H.323 may be under new version 2 be used for interoperability between different service networks like PSTN and Internet to support voice. The standard H.323 is for multimedia or videoconferencing. The audio G.7xx standard of H.323 may be many based on choice of xs. The choice of xs will define the intelligibility of the voice service provided. 3.22.3 Ipv6 for real time services The conventional packet switching is not appropriate to carry real time services. There are many reasons for this. For example HDLC or SDLC packets are variable in size. To synchronize and identify a packet, flags are required to be located. To avoid occurrence of flag byte in the payload, stuffing and de stuffing are done. These cause huge node processing delay, and hence packet transfer delay. ATM was proposed as the replacement of packet switching to support real time services. The problems of conventional packet switching were solved in ATM by making ATM packet, called cell simpler. The simplicity in ATM is in two respects: (1) shorter cell and (2) fixed size cell. This philosophy was extended to design Ipv6 datagram to replace Ipv4 datagram so that IP can carry real time services. IPv6 has a simple and basically fixed header format. The overhead bits of Ipv6 are less than that of Ipv4. The overhead bits in Ipv4 is 12 bytes in the header format of 20 bytes (8 bytes are for address), whereas the overhead bits in Ipv6 is 8 bytes in the header format of 40 bytes (32 bytes are for address). IPv 6 proposes to provide QoS (Quality of Service) service support to real time services like voice and video. The flow level and priority in the header of Ipv6 facilitate the support of real time data. Ipv6 has an efficient header format compared to Ipv4.

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3.22.4 Wireless Internet Two proposals for further development of the Internet are: (i) under sea super speed Internet and (ii) wireless Internet. A proposal for a global optical-fiber under sea cable network called Project Oxygen has significant industry support and financial backing. This project is called “the best of bandwidth on demand” project as per the company release. Experts say “Project Oxygen is the most ambitious communication project in the 20th century…..The Internet and video transmission are the major drivers for the expansion…..a global optical fiber network could erase the boundaries between Internet and the traditional communications, and shift the profit model from voice service to data and video.” Construction of the under sea network began in September’98. In the first phase, the cable shall be stretched over 158000 Km in 74 countries with three major network management centers in USA, Spain and Singapore. The major transatlantic and transpacific links are likely to be operational by 2000. Phase two shall start in 2002 and cover the whole of the world. The speed of cable is projected at 1920 GBPS with minimum capacity of 640 GBPS. It is reported that with under sea Internet, a video-based Internet shall come with over 10,000 video channels. The growth of wireless technology is immense. At the same time Internet traffic is growing exponentially. These have motivated for wireless Internet access research and development. The proliferation of the Internet-enabled wireless devices[56] has also excited the wireless Internet project. The wireless Internet access to mobile subscriber of UMTS (Universal Mobile Telecommunication Systems), GSM and other 3G and 2G technologies; and even with wireless ATM, are studied in literatures[46,54,57]. The modification of RSVP (Resource Reservation Protocol)[58] is directed to implement wireless Internet that can support for different broadband services. The wireless Internet will provide a scalable global mobile system for different services.

4 LOCAL LOOP TRANSPORT TECHNOLOGY 4.1 Fiber-free optical or Optical Wireless Communication and Networks It is often said that science is back to basics. Old science is science of light. Modern science and technologies are now finding and exploring the viable and potential applications of light. The two pillars of the information technology, namely computer and communication see their brighter future in use of light. The future computer technology is heavily dependent upon optical storage because of getting higher density, lower error and higher speed etc. Even the secure transportation of computer data is seeing a high hope with quantum transportation and computing that uses nothing but a single photon characteristics. The communication technology prefers fiber optics as the best choice as transmission medium because of fiber’s several advantages like low noise contamination, and very high bandwidth etc. But the cost of fiber optics is considerable. So if optical communication is made possible without fiber? Grapes the technology and use it. That is what is “fiber-free optical communication” or “free-space optics” (FSO) or “optical wireless”(OW) as known in industries. FSO appears nothing new, but the clever and intelligent application of ancient basic technology of use of light for communication purposes. Remember the early men used to use light signal or smoke signal for messaging with no cost as light travels through air with no cost. FSO technology has the attraction that “it’s cheaper to beam data through the air than to build infrastructure with wires.” Light through free-space or air provides high speed transport over short distance and that too at no cost. This transmission medium may be used with proper transmitter and receiver to realize FSO. Thus for economic advantages the free-space optics

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technology may be used for Gbps (giga bit per second) transport over metropolitan or city distances. The appealing other advantages of FSO are: no cable cost, no cable installation, trenching & digging cost, no cable maintenance cost, and no link failure (virtually link availability is almost 100%!). It is said that ““Free space optics really only provides a very limited application when you consider five 9s of reliability. Some of the free space optics companies will tell you that the five 9s are outdated and that they actually have trials with alternative operators that are just going to three 9s and four 9s” and “Five 9s is probably the greatest myth that exists today in the world of telecom.” Free-space optics is the hybrid of the optical and the wireless technology, presently the two most important carrier technology of communication. FSO offers free-for-all transmission medium. A study says “FSOs also offer lower deployment costs and reduced installation time compared with metro fiber builds. Business cases we have seen start at one-fifth the cost of metro fiber and can be six months faster to install in some metro areas.” As the name implies, FSO uses optical laser technology to transmit data across open spaces and uses the property of straight line propagation of the light beam. The low-power infrared beams that do not harm the eyes are used in FSO technology to transmit data through the open space between transceivers. The transceivers are mounted on rooftops or behind windows (Fig. 15) which are in line of sight with each other over the distances of a few hundred meters to a few kilometers. The part of the electromagnetic spectrum above 300 GHz that includes infrared is unlicensed and available free of cost. The FSO technology then is to ensure only that the radiated power does not exceed the standard defined by the International Committees. Usually the equipment works either at the 850 nm or the 1550 nm laser. Lasers of 850 nm are much cheaper than those of 1550 nm. But the safety regulations permit the lasers of 1550 nm to operate at higher level than that of the 850 nm laser. The FSO with 850 nm laser thus suitable for moderate distance whereas FSO with 1550 nm is favored for distance of kilometer ranges.. Actually 1550 nm has two fold power advantages and five fold distance advantages over 850 nm laser but about ten fold cost disadvantages compared to 850 nm. Table 19 gives a comparative study. A few major applications of FSO are in the areas of metro network extension, last-mile access, enterprise connectivity, dense wave division multiplexing services, SONET ring closures, wireless backhaul, back up, disaster recovery, service acceleration, storage-area network and LAN interconnectivity. FSO may be deployed to extend the existing Fiber Ring of MAN (Metropolitan Area network) by connecting with other networks. This may compete with SONET (Synchronous Optical Network) network. FSO may be deployed in the last-mile access in the sense that it may be used in high speed links that connect Internet service providers or other networks with end users. It is reported that “domestic service providers and foreign carriers are using FSO not only as a broadband backup but also as a viable last-mile technology. For a technology that depends on straight lines, free space optics is taking a circuitous route to espectability.” FSO may be used as redundant back up in lieu of a second fiber link, particularly over short distance communication. This has a clear advantage. Consider the Sept’11 disaster. Had there been FSO, some means alternative communication could have been available in case of fiber failure. A report goes on saying “While FSO will never defy the laws of physics, it can provide a valuable last link between the fiber network and the end user-including as a backup to more conventional methods. A key example was the Sept. 11 tragedy, when carriers learned that having a backup fiber optic network was of little use if both fibers went dark.” AS a backhaul, FSO may be used to carry cellular telephone traffic from towers back to fixed wire PSTN (Public Switched Telephone Network).FSO may further be used to provide immediate or instant service to customers while their fiber link is being laid.

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FSO or OW has another important application in the last mile solution for broadband services. This application is otherwise known as bridging technology. To support broadband services to residential customers, the problem of the last mile made of twisted wire pair exist. The clever utilization of last mile has made the access rates to vary from 128 Kbps to 2.3 Mbps. One important technology of the clever utilization is the DSL (Digital Subscriber Line) technology which provides access rate at 144 Kbps. With OW technology the access rate is believed to increase to Mbps. This a great offer of OW technology. FSO technology is believed to change the optical communication and “Optical networking technology is radically changing the foundation of carrier backbones, boosting Internet bandwidth exponentially while slashing costs dramatically.” But FSO is not free from disadvantages. FSO link may suffer from weather conditions, for example the Fog may hamper the link operation. Till date no standard is available for FSO operation. The vendors have to do a lot to utilize the technology’s viability and consequent products’ marketability. Let us hope for the best for this old technology. It is concluded with a few observations of some industrialists and members of academic: 1. Optics Alliance and chief technical officer for vendor fSONA said: “To have alternate paths using free space optics is getting much more interest from carriers.” 2. “People are realizing that if they have two fibers, they’re not necessary protected if it’s a correlated event and they both go out,” said Steve Mecherle, chair of the Free Space 3. Michael Sabo, senior vice president of sales and marketing for vendor AirFiber, said “FSO is earning a place as more than a fiber backup. Billions of dollars have been spent on longhaul fiber builds out on the trunks. This technology fits the last-mile kinds of applications to fill in all the leaves of those networks.” 4. “Qwest uses FSO in commercial deployments because they are the vast majority of the users of Qwest’s broadband network. We’re pleased with the technology, but we cannot [speculate] about its future deployment in the Qwest market,”Qwest Communications. 5. “Nevertheless, fiber doesn’t go everywhere, and it can’t always be deployed quickly. In all those cases, FSO is a superb alternative,” Werne , CEO, Utfors, A Sweden broadband carrier 6. Ken Corriveau, Tribal’s IT director: “You could rent dark fiber, but that would take forever to figure out in the city. You could rent a T-1 or DS-3, but both of those are 30 to 90 days out.” 7. “In Madrid, 80% of business users are within 500 meters of fiber,” said Paul Kearney, Alua’s (a carrier in Spain)chief technical officer. He further said : “We plan our [FSO] network by using very short ranges to be within the weather limitations.” 8. “In general, the technology has a lot of future for the carrier networks—if it’s marketed well,” said Gartner’s Tratz-Ryan. And therefore to many, ”at least FSO cleared the first hurdle on its circuitous obstacle course” Table 19: Comparison of lasers used in FSO Laser in FSO

Typical Cost

Typical Data Rate

Typical Coverage distance

850 nm

US$ 5000

10-100 Mbps

A few hundred meters

1550 nm

US$50000

Upto Gbps

1-2 Kilometers

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T T

TR

T T

TR

T MAN/LAN TR

TR = Transceivers Free space or air ______ Fiber link

Fig. 15: FSO operation.

4.2 DSL Technologies: ADSL and VDSL Over more than a century, a vast analog telephone network is existing world over. The telephone line in general and the last miles, in particular is made of twisted copper wire pair that is suitable for voice communication. Due to information technology, the need was arisen to transport many diverged types of information. Information relates to many different applications and services, viz. voice, video, data, image, facsimile etc. The information age is motivated by a new culture of value added communication where communication of video, data, image, facsimile and graphics etc has become imperative besides, basic communication of the voice or speech. The characteristic of diverse services, voice, data, video, image, graphics and facsimile etc are quite different from each other. Therefore for each of the services logically there is a requirement to have each one’s nature based communication system. Obviously, such a proposition is not techno-economically viable and sound. The only other economically viable alternative left was to find techniques to use existing vast copper cable of telephone network for the value added services. Even after evolution of fiber optics, a vast copper line is still existing that can be guessed from the following statistics of 1997[35,36] 1. In USA, the unloaded twisted wire pairs up to 18000 ft (between central office and customer - local loop) account around 70% of all loops. 2. In USA, the Loaded loops (>18000ft) account around 15% only of all loops, 3. In USA, the derived loops up to 12000 ft with unloaded twisted pair connected with FTTC / DLC, Fiber - To - The - Curb or Digital - Loop - Carrier, accounts around 15% only,

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4. The world picture in this respect is 600 million unloaded twisted copper wire pair versus 6 million hybrid fiber/coaxial lines, i, e the ratio is 100:1, 5. The annual growth of telephone network in 1990-95 in Africa, Arab States, Latin America and Asia Pacific was respectively 8%, 9%, 10% and 27% 6. Around 1000 million telephone subscribers exist in the world in 2003. Actually the varied services like video conferencing, video on demand, fast access to Internet and interactive multimedia services require higher bandwidth than that of voice. Therefore new technology and signal processing are prime needs if the copper is used to carry these services in the last miles. xDSL (Digital Subscriber Line) is the unique technology that supports more than one services like voice, video and data simultaneously over a shared access line of copper. The DSL is established as a scalable service that provides quality service delivery and at the same time provides a cost effective local loop infrastructure. The DSL appears to be an efficient solution for providing multimedia services. In order to provide value added services, broadband services and multimedia services using existing unloaded telephone lines, over the last few years communication engineers developed a number of techniques. These are: Modem culture and xDSL technology. XDSL technology[38-40] includes: HDSL (Higher-rate Digital Subscriber Line), ADSL (Very-high-rate Digital Subscriber Line), G.lite (splitter less ADSL—this is also called UDSL, Universal DSL), SDSL (Symmetric DSL), VDSL (Very High Rate DSL), IDSL ( ISDN DSL), RADSL (RateAdaptive DSL) etc. 4.2.1 Modem Versus xDSL Using modem, the copper wire provides the data services. For example Internet access with dial up facility is done through the modem. As of today the modem speed is 56 Kbps. The speed of 56 Kbps is not sufficient to support high quality broadband services. Moreover modems occupy the entire 0-4Khz bandwidth allocated to voice, thereby preventing simultaneous services of voice and data over copper of local loop. Within last few years, the slogan of communication technology has become “Speed is the ultimate.” Technology is being developed in pace to serve with the demand of more and more data rate, namely, from bits per second (bps) to Kbps to Mbps to Gbps and finally, to Tbps with WDM, Fiber amplifier, solution and fiber optics communication in hand. High bit rate communication is not possible with copper twisted wire per. Alternative may be to use optical fiber link at loops. This may be a long run solution, but xDSL technology was developed out of this race of more and more fast data communication but using copper cable. The oldest technology of communication digital data through along twisted pair cable of telephone loop, is modem technology. Bit (oldest modem, example is V.21/Bell 103) to as high as 33.6 Kbps (as predicted in V.34 extended standard). With standardization of V.43 modem with 28.8 Kbps, it has been postulated to provide low graded multimedia service to customers through POTS. But there is a big bug in modem technology. A 3 KHz voice line (local analog loop) with 30 dB signal-to-noise ratio can have maximum bit rate of about 30 Kbps as per Shannon theory. Thus using modem technology to carry data over analog telephone line is handicapped by the above speed constraint. This is the reason that modems sometime do not work at the vendors’ advertised speed. There is also report of 56 Kbps modem technology, which can well fit to carry multimedia and Internet services to customers’ promises using telephone lines. But the 56 Kbps technology does not communicate data in between two modems. It communicates data between a modem and a digital ISP (interface signal processor) system which creates a reduced noise like environment. Therefore use of 56 Kbps modem technology to transport high bit rate services to customers, promises may be the limit.

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It was already mentioned that local loops of copper twisted pairs designed for caring voice signals, are not suitable to carry high speed digital data. Local copper loops are primarily designed to carry voice traffic. Voice traffic is relatively short duration and on an average of 3 minutes. Internet traffic is on average of 30 minutes duration. The impulse noise and pulse dispersion of copper loops is the main obstacles in carrying data at high speed. But with growing World Wide Web culture and demand of multimedia services like video-on-demand, boosting of capacity of copper twisted pair local loops, by using some alternative technology of modem, was felt essential. This gave the birth of xDSL technology in general and ADSL technology in particular. It is often said that ADSL is for boosting the capacity of installed copper and fiber optics link. In xDSL technology, special circuits and software called transceiver are used. Transceiver software perform the function of encoding/decoding or modulation/demodulation by which serial binary digital data streams are converted into signal suitable for transmission through analog copper twisted pair link. Transceiver also performs the other functions like equalization, signal shaping and processing, and amplification to compensate for signal attenuation and phase distortion. The other important function performed by transceiver is error detection and correction of data. 4.2.2 ISDN versus xDSL technology ISDN (Integrated Services Digital Network) was developed to provide integrated and simultaneous services of voice, data and low speed video at a basic rate signal of 144 Kbps. The payload of 144Kbps consist of two B channels each 64 Kbps and one D channel 16 Kbps. The DSL signals was first coined to carry 144 Kbps of ISDN over copper loops of 18000 ft or less. This was made with 2BIQ four level line code. The 2BIQ code provides baseband signal spanning from zero to voice frequency band. In this mode of ISDN, voice is served in digital mode using PCM (Pulse Code Modulation) and B channel at the rate of 64 Kbps; but ISDN does not support POTS(Plain Old Telephone Service). Data at the rate of B channel of 64 Kbps (which is much higher than the maximum permissible rate in MODEM culture—about two folds) is served in ISDN. Therefore, why to go for xDSL ? Reasons behind going for xDSL technology are two. First, xDSL technology provides much higher data rate than ISDN. With growing web culture and demand of multimedia services, bit rate of the order of a few Mbps become common. Services like video on demand can not be meet with 64 Kbps or even of 64*2 = 128 Kbps of ISDN technology. Second, ADSL and VDSL are different from ISDN in the respect that unlike ISDN, they retain the service of POTS while providing high rate data service. 4.2.3 ADSL technology ADSL technology has become most appropriate technology out of all xDSL technologies. HDSL is a variant of ISDN technology which provides data communication at the bit rate of about 784 Kbps (T1 carrier) over twisted copper paid loop upto 12000 ft. like ISDN, HDSL uses 2 BIQ line code. ADSL technology was developed mainly to provide multimedia service like video-ondemand service and growing Web service. The characteristics of these two service are quit asymmetric in nature. For Web accessing and / or interactive video two ways communication is essential. Out of the two ways communication downstream (towards the subscriber) communication requires much higher bandwidth then upstream (towards central exchange/

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office) communication. This is because, typically Web surfer is more interested in downloaded on uplink request. ADSL technology[37-43] offers higher data rate of say 6 Mbps for downstream data and lower data payload of say 640 Kbps for uplink data using copper installed loop of telephone. In addition, ADSL provides POTS or conventional voice service. As the service nature is asymmetric, SDSL technology got lost to ADSL technology. Due to the asymmetric nature of ASDL technology, it provides an interesting technological benefit. When many wires are squeezed together in a cable, cross talk is inevitable due signal overlapping. In case of downstream data, signal amplitudes are same because they all originate form the exchange. Due to the same amplitude, there is no effect of destruction of weak signal by strong signal. For uplink data, signal may originate from different customer premises, which are the different locations. Therefore signal reaching through wire pairs of a cable may greatly varies in amplitude. But as the cross talk increases with frequency, problem is tackled by limiting upstream data and keeping it at low end of spectrum. This is exactly what is done in ADSL. ADSL technology increases capacity of installed copper link of telephone to 6 Mbps. In the technology data traffic and voice is carried simultaneously. It carries data in digital form and voice in analog form, unlike ISDN which carries both in digital form. ADSL System POTS splitter/filter preserves the 4 KHz spectrum for POTS service; and prevents hampering of POTS service due to any fault of ADSL equipment. The rest available bandwidth of 10 KHz is used for ADSL data communication at the rate 6 BPS for every hertz of available bandwidth. Fig. (16) portrays the operation of ADSL system. The transceiver software of ADSL uses an advances modulation technique known as discrete multitone (DMT) technology. The ANSI T1E1.4 has standardized DMT as the line code for ADSL. DMT divides bandwidth 10 KHz to 1 MHz in 256 independent subgroups each of 4 KHz width. Each of the sub channels referred to as tone, is QAM modulated on the separated carrier. The carrier frequencies are multiples of basic frequency of 4.3125 KHz. The DMT is used in ADSL technology as because it has the unique ability to overcome typical noise and interrupts in the local loop twisted wire pair cable. The ADSL frequency spectrum is shown in Fig. (17). The available spectrum ranges from about 20 KHz to 1.1 MHz. The low 20 KHz is reserved for voice services under normal POTS. To perform bi directional communication, ADSL modems divide the bandwidth in one of two ways: (1) FDM where non overlapping bands are used separately for upstream and downstream links, (2) echo cancellation where for both the upstream and the downstream the overlapping bands are used but separation is made by local echo cancellation technique. Echo cancellation technique is bandwidth efficient. Advanced forward error correction techniques are used to tolerate error bursts as long as 500 msec. A comparison of different DSL technologies is given in Table (20). ADSL is about 400 times faster than most sophisticated modem and 60 or more times faster than ISDN. However ADSL down stream speeds depend on the loop distance as shown in Table (21). But typical coverage distance is about 4 km. Over distances, the natural degradation of data rate occurs. To provide services to customers beyond 4 km, an embedded rate adaptive mechanism may be used.

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INFORMATION TECHNOLOGY, NETWORK AND INTERNET Customer Premise Telephone

POTS Splitter Copper Local Loop

Computer

ADSL Modem

Local Switching Exchange

Network

Pots Splitter

Processing Circuit

Network

Local Exchange

Fig. 16: ADSL System

Pots band

Guard band Up stream band

Down stream band

4 KHz

138 KHz

30 KHz

1.104 KHz

Fig. 17: Frequency Spectrum of the ADSL

The arrangement may be coupled with growing ATM (Asynchronous Transfer mode) network, which is predicted to be a network for multimedia services. Recent advances in ADSL technology promises to transfer data at the rate as high as 50 Mbps to the customers over a short distance of twisted copper pair from FTTC. This advancement is termed as VDSL. ADSL technology and WDM technology support the predicted cyclic nature of analog digital transmission. Table 20: Comparison of DSL Technologies Service / Network

Data Rate

POTS (Plain Old Telephone System) with Modem ISDN ADSL VDSL HDSL IDSL SDSL

28.8 - 56 kbps 64-128 kbps 1.544-8.448 Mbps for downstream 16-640 kbps for upstream 12.96 - 55.2 Mbps 784, 1544, 2048 kbps 128 kbps 800–2000 kbps for downstream 64 - 200 kbps for upstream

RADSL

1.544–8 Mbps for downstream 64 kbps–1.544 Mbps for up stream

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Table 21: Down stream speed versus distance of ADSL technology Distance in feet

Speed in Mbps

18,000 16,000 12,000 9,000

1.544 (T-1 carrier) 2.048 (E-1 carrier) 6.312 (DS-2) 8.448

The major applications of ADSL technology are: (1) Information highway to wide community, (2) High speed to Internet access, (3) Distance learning by the process of video conferencing etc (4) Video on Demand, (5) Video telephony. ADSL was standardized by the ITU-T in recommendation G.992.1 in 1999. The splitter less ADSL known as ADSL lite was recommended in G.992.2. In the ADSL lite the use of splitter in the customers’ premises are avoided at the cost of lower transfer capacity as 1.5 Mbps and 512 Kbps respectively for downstream and upstream. 4.2.4 VDSL Technology Very high speed or high rate DSL technology is the most recent and important addition to the DSL technologies. The technology is believed to provide the bridge between today’s existing copper infrastructures with near future’s future’s entire fiber infrastructure. VDSL modems [140-43] are placed in the customers’ premises and at the end of fiber installation. The end of fiber installation is the neighborhood or exchange point where the fiber link terminates. With the technology, very high speeds are possible on the copper link spanning about 1.5 km between fiber end and customers’ premises with as high as 15 Mbps total in both directions and over a short distance of 300 m or less with 52 Mbps. VDSL offers about 100 times faster tan normal modems. The proposed VDSL can use up to 30 MHz bandwidth compared to 1.104 MHz of ADSL and 300, 580, 1100 kHz for HDSL. VDSL supports two service classes : Asymmetric known as Class I service and Symmetric known as Class II service. Asymmetric service type is compatible to ADSL technology and primarily aims to meet residential customers. Symmetric service aims to serve business purposes. VDSL is supposed to provide broadband services to both business and residential communities on existing copper infrastructure. Data rates of VDSL is at Table (22). VDSL system VDSL is aimed to be coupled with FTTC (Fiber To The Curb) and FTTB (Fiber To The Building)/FTTH (Fiber To The Home), the technologies that uses fiber in part of the local loop. In that context the VDSL reference model is shown in Fig. (18). Table 13: Typical VDSL data rates Service Class

Upstream data rate in Mbps

Downstream data rate in Mbps

Spanning Distance in m

Asymmetric

6.4

52

300

3.2

26

900

1.3

13

1500

26

26

300

13

13

900

6.5

6.5

1500

Symmetric

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Customer Premises

Copper Link

Fiber Link

Central Office/Exchange

VDSL

VDSL Transceiver at NT

Transceiver at ONU

NT = Network Termination ONU = Optical Network Unit (a) System Reference LT = LINE Termination

Splitter

Splitter Copper wire

PSTN/ISDN

NT = Line termination

Network Interface

PSTN/ISDN

(b) VDSL reference model Fig. 18: VDSL System/Model

Attenuation and Cross Talk The subscriber loop is made of copper wire of different gauges. A number of pairs are grouped together in cable bundles. The attenuation of the signal in copper wire depends on the dielectric used, gauge, type of twisting, and length. But attenuation usually increases with both frequency and the length. That is why the data rates in ADSL and VDSL falls wit length as pointed out earlier as well the distance coverage is lower in VDSL than that of ADSL It may be noted tat NEXT is not attenuated by the line transfer function. That is why NEXT is more harmful than FEXT. In both ADSL and VDSL, by FDM technique the effect of NEXT is made lower. But both cause the data rates to fall with lengths. Both the technologies, ADSL and VDSL are believed to provide wider broadband services to residential and business users using the existing copper link of last miles. However future research directions will aim to tackle the issue of falling rates with length.

5. MULTIMEDIA COMMUNICATION AND CONFERENCING STANDARDS It is said that in the future, multimedia shall be the rule and monomedia shall be the exception. Multimedia is a tele-service concept that provide integrated and simultaneous services of more than one telecommunication service, namely, voice-world, video-world and data-world. Truly, multimedia is supposed to provide such service in real time and in interactive mode. Typical examples of multimedia applications are teleconferencing, videoconferencing, telemedicine, telemarketing, teleshopping etc. Multimedia is fast emerging as an important tool of information technology and as a basic tool of tomorrow’s life. Multimedia proposes to simulate human-like communication and services in an environment of “you see as I see” and “you feel as I feel”. Virtually reality is envisaged in multimedia services. Multimedia transferred your message in your way. Multimedia

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is believed to prosper with the general human trend from “nice to have” to “value to have” to “essential to have”. With multimedia a society with “plug and play”, “look and fell” and “point and feel” and “point and click” shall emerge. In near future, we shall have multimedia cities and centres. It is often said that in near future multimedia shall be the rule and the monomedia shall be the exception. Interactive multimedia is a service, which provides simultaneous access, dissemination, transportation and processing of more than one information service like voice, video and data in the interactive mode and in the real time environment. Multimedia is to integrate three communication worlds, namely, telephone world, data world and video/TV world into a single world communication. multimedia application shall comprise more than one information type, namely the non real time service of data, images, text and graphics, and the real time service of voice and video. Future world of information and communication shall be converged to multimedia application and shall provide comfort, competition, mobility, efficiency and flexibility. As per Fred T. Hofstetter “Multimedia is the use of a computer to present and combine text, graphics, audio and video with links and tools that let the user navigate, interact, create and communication.” Technologically multimedia shall be service of services and nontechnically a community of communities”. Multimedia shall enable people to communicate and access at any time at any where at reasonable costs with acceptable quality with manageability. Location of man, materials and machine resources shall be irrelevant in business in the era of multimedia. It is said that “It makes no sense to ship atoms when you can ship bits.” “Virtual reality with virtual presence in virtual worlds, virtual cities, business enters, virtual schools and virtual rooms will emerge in the next future ……… For example, virtual reality at short notice allows collaboration between changing partners on specific tasks, sitting at virtual writing tables without real offices and addresses other than the network. Transactions in this enhanced telecooperative working environment would be electronic analogies of the normal world.” Faster work flow, comprehensive 24-hour service, remote operation and maintenance, easier trouble shooting, life long and leisure time activities, less travel, less cost and more fun shall be the important attraction of the multimedia world. Multimedia communications provide a chickenegg benefits to information world, and have acceptance at all levels: (1) contact acceptance, viz., service availability, user-interface, (2) economic acceptance, viz., less cost, more benefits, (3 ) content acceptance, viz. quality, and (4) social acceptance, viz., desirability, privacy.

5.1 Standards A great challenge is to standardize broadband services and system for the purpose of deployment. In fact, the deployment of seamless integrated mobile broadband services will greatly benefited from the standardization process[48]. In order to define any standard, the International Telecommunication Union (ITU) usually forms a study group. This study group submits recommendations for standards pertaining to the assigned functions. A list of different study groups along with their assigned functions, made by ITU for 1997-2000, is given in Table (23). SG9 and SG16 respectively deal with television and sound transmission, and multimedia services and systems. The low bit-rate (kilo bits per second-kbps) audio coding standards specified by ITU for multimedia application are listed in Table (24). The standards G71X and G72X are mainly used in different multimedia applications. MPEC-I (removing picture export group) audio coding decoding is applied in H.310 multimedia conferencing standard.

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Table 23: ITU Study Groups SG 1 SG2 SG3 SG4 SG5 SG6 SG7 SG8 SG9 SG10 SG11 SG12 SG13 SG14 SG15

Service definition Network and service operation Tariff and accounting principles, economic and policy issue Telecommunication management network (TMN) and network maintenance issue. Protection and policies against electromagnetic environmental effects. Outside plant Data network/open system intercommunications Features and characteristics of telemetric system TV sound transmission Software aspects of telecommunication systems Signal and Protocol End-to-end transmission performance of network and terminals Network aspects in general Modems and transmission techniques Transport networks systems and equipment’s

SG16

Multimedia services and systems.

Table 24: Standards of low bit rate audio coding for multimedia communication Standard G.723,D G.723,1 G729,A G.729 G.711 (PCM/POTS) G.722 (Broadcast quality) G.723 (Low bit rate POTS) G.726 G.728 MPEG.1 layer (CD audio)

Bit rate In kbps

Frame size in mg/cc

Algorithms delay in m sec

Required RAM size with 16 bit words.

5.3 6.3 8 8 56

30 30 10 10 —

37.5 37.5 15 15 —

2.2k 2.2k 2k 2.7k —

48-64

—

—

—

5-6 32 16

— — —

— — —

— — —

32-256

—

—

—

Different video coding standards for multimedia services are listed in Table (25) along with bit rate and applications. H.26X standards are used for videoconferencing and MPEG-I is used for video-on-demand. H.26X standards are mostly used in multimedia videoconferencing standards like H.320, H.324, H.323 and H.310

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Table 15: Video-coding standards for multimedia applications Standards

Bit rate

Typical Multimedia Application

H.261 H.263 MPEG.1 MPEG.2

64kbps-1.92Mbps 15kbps-34kbps 1.2Mbps–2Mbps 3-15Mbps

Videoconferencing (N-ISDON-64) Low rate videoconferencing Video on demand Temperature-Diagnostic video on demand

Table 16: Multimedia conferencing and terminal standards Standard

Network

Video Coding

Audio Coding

Data Standard

Multiplexing

Control

Remarks Application

H.320 (1990)

N-ISDN

H.261

G.711 G.722 G.728

T.120

H.221

H.242

Multimedia conferencing with G.711

H.324 (1996)

PSTN/ GSTN/ POTS

H.263 H.261

G.732.1 G.729

T120

H.223

H245

Multimedia conferencing with H.263 and G.723.1

H.322 (1996)

LAN internets packet switching

H.261 H263

G.711 G.722 G.728 G.723.1 G.729

T.120

H.225.0

H.245

Multimedia conferencing H.261, G.711

H.322

Isoethernet

H.261

G.711 G.722 G.728

T.120

H.221

H.242

—

H.321

B-ISDN/ ATM

H.261

G.711 G.722 G.728

T.120

H.224

H.242

—

H.310

B-ISDN/ ATM

T.120

H.222.0 H.222.1

H.245

Multimedia conferencing with H.262, MPEG.1, H.222.0

H.262 MPEC.1 MPLG.2 G.711 H.261 G.722 G.728

Table (26) is a comprehensive list of different multimedia standards, their network performs, video coding, audio coding, and data standard multiplexing standard, control standard and applications. The standard H.324 may be used to provide videoconferencing, putting to work the existing telephone network. H.323 may be used for the same over LAN (local area network) and H.320 may be used over N-ISDN using nx64 kbps channel, whereas H.310 may be used using BISDN/ATM. The table also lists the users terminal requirement for different multimedia standards.

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5.1.1 H.320 multimedia conferencing standard H.320 is the narrow band (< 2 Mbps) conferencing standard meant for conferencing over telephone networks such as ISDN with bandwidth typically in the range 384 kbps. H.320 family of standard is to serve video conferencing with any H.320 compatible terminal irrespective of whether it is stand alone video conferencing unit or video telephone or PC based system. H.320 is often treated as a de facto standard of video conferencing. H.261 is the video coding standard has a lot of similarity with H.320. The H.261 standard has a lot of similarity with MPEG technique, and uses the DCT transformation technique with motion compensation and Huffman coding [see-Box 4] to active compression. But unlike MPEG, it has rate control to cope up with variable video bandwidth within the rate of 40kbps to 2Mbps. H.261 standard supports two picture sizes: the larger one is called CIF with pixel size of 352X288 and the smaller one is called QCIF with size of 176 x 144. H.320 terminals are having H.261 video code. The audio coding of H.320 standard can be any one of three: G.711, G,722 and G.728. The G.711 is equivalent to a -low and m-low PCM such coding. It supports 3.1 KHz audio at 64, 56 or 48 kbps. The G.722 provides higher quality audio with 7KHz bandwidth using 64kbps. The G.728 is equivalent to a -law and m-law encoding and supports 3.1 KHz bandwidth of 16kbps. The H.221 is the framing standard. The audio and video bit streams are multiplexed together to create a frame that is to be sent. The H.221 define how the frame is achieved. Each frame is made to the 80bytes of information. Each frame creates 8 sub-channels with each bit within each byte allocate to a sub channel. They are numbered 1 to 8. First seven channels are used to carry video and audio data. The 8th channel is used not only to carry data but to carry other codes also. The other codes are: FAS: Frame alignment signal. BAS: Bit-rate allocation signal. ECS: Encryption control signal. These sub-channels are called service channels. The standard H.230 provides frame synchronization control and audio-video signal indication control. H.242 is for achieving capability exchange, mode switching and frame reinstatement. The H.243 is multipoint control standard. The video conferencing is not only point-topoint but multipoint too. To control multipoint conferencing, multipoint control unit is required (MCO). H.243 is a standard for MCU. The T.120 is for communication of all forms of data between two or more multimedia terminals. The H.233 is security coding. It is used to define a method of encrypting data. Different wireless combination have been investigated in Japan[45] to define multimedia terminals: PDC (Personal Digital Communication System) + PHS ( Personal Handy phone System), PDC + 3G, 3G + 4G, 4G + MMAC ( Multimedia Mobile Access Communication), 3G + 4G + MMAC. Such investigation will definitely lead to wireless + integration to coexist to implement true personal communication.

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BOX 4 The Huffman code is a compression code designed by Daceid A Huffman in 1952. It is a simple improved code over Shannon-Fanon code. In order to illustrate Huffman code, let us say we have an original body of data which reads only source triple as in table to present some message. The probability of occurrence of any source triple in the message is also shown. According to the Huffman coding, the corresponding compressed codes are shown in the table. The average size of the compressed code under Huffman coding becomes: 1 X 0.4 + 2 X 0.2 + 3 X 0.2 + 4 X 0.1 + 4 X 0.1 = 2.2 bits per code. Whereas the code size of the original source code is 3 bits per code. Source Triple

Probability of Occurrence

Corresponding compressed word

000 001 010 011 100 101 110 111

0.25 0.25 0.125 0.125 0.0625 0.0625 0.0625 0.0625

11 10 011 010 0011 0010 0001 0000

There are several disadvantages to Huffman coding. First, to design the code, one must know the probability of occurrence of any code in the original block of data. What shall happen if the probability is not known a priori? And what shall happen if probabilities pattern changes over time? Second, Huffman coding is not unique in nature. The code is also block code. But the redundancy under this code is either minimized or optimized

6. UTN PERSONAL COMMUNICATION To offer UTN services, in USA, a band of 160 MHz near 2 GHz has been allotted. Personal communication shall mature with UTN. Frequency band allocation for different services of personal communication is shown in Table (27). Cellular communication is the early personal communication. Personal communication shall coverage to and merge with total wireless, total service independent and total UTN based communication. If we consider the growth and development of wireless communication at the present rate, total wireless (to a constraint) may be achieved within next 5 years. Total UTN service may need another 5-10 years. Integrated and application oriented communication needs new technology and new integrated terminal which shall be affordable to mass customers. Technology is with us with ATM. Integrated terminal is under development. Computer integrated telephone is the first such device. It may require another 5-10 years to commercially develop an integrated terminal for voice, video and data. Therefore a matured PCN (Personal Communication Network) is expected within next 510 years. As per forecast made in literatures, narrow band personal communication service may cover 750000 km2 and 1500000 km2 in USA in next 5 and 10 years respectively. Indian communication is lagging behind international communication by several years. As per account this lag is a uniform 4 years since 1986. GSM started in Europe in 1988, India adopted the in 1994. ISDN started in Europe in 1990, India has adopted only in 1996. On the ATM, BISDN and UTN aspects, India is yet to open its chapter. Hence it is evident that Indian lag is more

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than 5 years and even may be 7-10 years in PCS/PCN. India is lagging behind its neighbors like Singapore, Taiwan and Hong Kong. Table 27: Frequency bands of different PCN services Service

Frequency Band

Cellular

800-900 MHz Ex : GSM - 890-915MHz 935-960MHz

CT-2

864/944 MHz

Cordless

46/49 MHz

Satellite/VSAT/MSS

C band

Narrowband PCS (FCC)

900-940 MHz

Broadband PCS (FCC)

1850-1890 MHz 1930-1970 MHz 2130-2200 MHz

7. FROM 2G TO 3G 2G (second generation) technology for mobile connection started around 1990s and it was revolved around GSM cellular communication that is mainly for voice communication. 3G were then expected to be deployed around 2000 and were targeted towards: • implementing anywhere and any time mobile connection with low cost and flexible handheld devices • implementing wireless data access particularly with wireless Internet connection. This was motivated by the exponential growth of Internet access. Users are prone to get Internet access anywhere and anytime with hand held devices • implementing high data rates at 2Mbps whereas previous GSM or 2G offered to 10 to 50 Kbps • implementing high speed multimedia or broadband services causing shift from voice oriented services to Internet access (both data and voice particularly with technology of VoIP), Video, Music, Graphics and other multimedia services • use of spectrum around 2 GHz whereas spectrum allocation for 2g was 800/900 MHz • global roaming to support global communication • flexible network to support existing and future changing requirement • a mobile multimedia services that will be able to transmit data, voice, video, image etc over variety of networks like point to point, point to multi point, broadcast, symmetric and asymmetric etc. The key benefits of 3G will be: delivery of broadband information direct to users and global access with a unified single radio interface. Several major challenges are to be overcome to implement 3G: wireless Internet for exponential growing users will be difficult to implement till IPv6 is implemented, global roaming with single number as proposed in PCN, fixed access with technologies like ADSL with high data rates of 12 Mbps has become competitor as that of IEEE 802.11 b WLAN in wireless local data interface, low cost flexible devices are yet to mature.

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7.1 Beyond 3G Mobile comprehensive broadband integrated communication will step forward into 4G (fourth Generation) all mobile services and communication. The 4G technologies will be migration from other generation of mobile services with an aim to overcome limit of boundary and achieving total integration. The evolutionary approach towards a wireless information age proceeds as in Fig. (19)[44,47,59] in comparison with other technologies as progress in pace to pace. The key characteristics of 4G systems will be: higher transmission capacities per user, larger frequency band, higher traffic densities, and integrated services. The technical challenges behind the expected technology lie with the associated different technologies as discussed earlier.

2G GSM, PDC, IS95

IG Analog cellular

3G : UMTS, CDMA

Towards wireless communication society Multimedia Content, High Bit Rate and IP Transport

802.11b WLAN

WLAN

Circuit Switched Networks

Wired Internet

802.11a WLAN

Broadband Internet/ DSL

4G Total Wireless, Seamless coverage & Integration, Anytime & anywhere communication

Wireless/Mobile Local Area Integration

Broadband FTH (Fiber to Home)/Fiber to Business

Fig. 19: PCN Evolution /Migration and other technologies as progress in pace to pace

The motivation behind aiming 4G information society are many: high speed transmission, next generation Internet support (Ipv6, VoIP, Mobile IP), high capacity, seamless integrated services and coverage, utilization of higher frequency, lower system cost, seamless personal mobility (LEO), adoption and integration of fixed and wireless support (ADSL/VDSL/ WILL/FSO), mobile multimedia (Standards), efficient spectrum use, QoS service, flexible and re configurable network and end to end IP systems. The convergence of local fixed wired network including wireless home or local network with broadband fixed and coming up ad hoc wireless networks will shape how we will communicate in next decades that may include[49.60]: Complete unification and integration of all and every services, Single communication number for each and every services, and Freedom to communicate any time any where. All these provisions are required to be meet with simplicity, cost effective, reliability and flexibility. The problems to be solved in achieving the expected results are: lack of bandwidth, lack of standardization, high error probability of wireless links,

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multiplicity of different systems & operators, and cost reduction. The problems are being addressed. The research in tackling the high error probability of wireless links has reached the expected directions[50-53] with BEC (Backward Error Control) technique. The research[55] in this context for optimizing Internet access over IEEE 802.11b has demonstrated with frame level FEC (Forward Error Control) technique.

8. e-BUSINESS AND e-COMMERCEA DURABLE APPLICATION OF IT Noble Laureate of Chemistry, Ilya Prigogine once told: “We can’t predict the future, but we can prepare it.” Certainly the future will be what we make of it with today’s single most technology, IT (Information Technology). It is a technology of the networks, the telecommunications and the computers. IT will make an impact in all aspects of our life. It is believed to bring about a profound change. The success and the effectiveness of the changes will be measured with proper perspectives in future. But it has undoubtedly brought an unprecedented change in business and commerce, by giving time, space and volume continuity. Today business and commerce have no geographic boundary, no volume restriction, and no time limitation. They provide just-in-time and just-on-scale solutions. On the scale of effectiveness, the business is measured by the low of the loss (W) of process cost. If P is the process cost, OPE (Overall Process Efficiency) is the efficiency factor of business, W = P - (P X OPE). With application of IT and its derivative like KM (Knowledge Management), OPE increases causing W to fall. Besides, IT gives the competitive advantages to the business activities. Information technology believes that: Investment + Web technology + Users = Big Profits. In such a scenario, ebusiness and e-commerce have eventually been emerged as the sound strategies for business and commerce. e-business and e-commerce will be facilitated by different technologies like global-reach Internet, WWW (World Wide Web), E-mail, Electronic publishing, Multimedia systems and communications, Interactive video, Image recognition and processing, Voice recognition, MSS (Mobile Satellites Services) and Personal Communication among others. The use of border less Internet is increasing following Moore’s law that estimates the doubling of the performance of silicon every 18 months. Even Internet growth may go well beyond Moore’s law. Gilder’s law may be more accurate estimate for growth of Internet traffic. Gilder’s law predicts the doubling of packet on the network every few months, and that few months may be in the range 4 to 9. It is estimated that the Internet traffic will increase by 1000 fold in the next ten years. As of today, about 50 million users use Internet with about 16 million servers in more than 140 countries. Internet is the best facilitator of electronic mode of business and commerce. Today, e-business and e-commerce refer to the business transaction over Internet. E-Business and e-commerce mean to doing business over wires or over Internet or using Information Technology. They are changing the rules of traditional business pattern, and making new rules and means for fast and border less business. The confusion on the difference between e-business and e-commerce is standing. e-business defined in most of the literatures reflect that there is actually no difference between these two. Yet two things appear to be somewhere and some how different.

8.1 E-Business E-business refers to the operation of the business objectives through and using IT. It may also be defined as business activities over digital infrastructure or doing business over wires. As per Colin, Director of the integration division of CNS, UK the e-business refers to the issue of supply chain integration. “ An ideal scenario is when a customer places an order. All of the

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suppliers and agents involved in the transaction are contacted electronically. Every system involved in the supply and delivery of that product is linked to every other system, hence talk of ‘zero latency transactions’ whereby there is no waiting for someone to do something because everything happens at the speed of light.” A report says, ” E-Business relates to how you and your customers place orders and ensure efficient delivery. E-commerce is the financial aspect of doing business. Both aspects will affect your operations sooner or later.” The economists usually identified four types of e-business: • Business-to-business (B2B). This refers to transaction between one business house to another. For example, the transaction between a large organization and their suppliers falls in the category. B2B is the most common business model. One example of B2B e-business is MetalSite.com • Business-to-customer (B2C). This refers to online retail activities. For example, software, journals and books sold over Internet using web sites. • Customer-to-business (C2B). The example of this is the booking of railway tickets or air tickets on any agent’s computer that has the network or the Internet connection. C2B is just the reverse of B2C. • Customer-to-customer (C2C). Online auction is the best example of this type of transaction. One example is eBay.com Currently e-business is mostly confined to B2B. Other areas of business are of course coming up.

8.2 E-Commerce E-commerce is basically financial transaction via computer networks, between people and organizations. E-commerce is a financial part of e-business.. Harvard Academic, Jeffery Rayport defined e-commerce as “selling real products for real money.” Eddie Rabinovitch observed “ Not surprisingly, the expected pay off of e-commerce projects is, of course, the bottom line: money. However, despite the prevailing notion of access to global markets as the most important competitive advantage enabled by e-commerce, most companies expect of e-commerce ways to reduce spending rather than increase profits. Let’s for a moment think about the rationale of the previous statement, which is also going to answer another e-commerce question: ‘ why is business-to-business (B2B) market considered by many experts several magnitudes more important than business-to-consumer (B2C)?’ Well, it’s probably easier to convince a CEO to spend $100,000 on a solution that will demonstrably ‘save’ $1 million than to spend the save amount on a solution that might ‘make’ $1 million....... Making money on the Internet is still quite dicey. But it’s not too difficult to demonstrate that B2B e-commerce will save money by improving efficiency and therefore reducing expenses for transactions between companies.”

8.3 Problems for e Commerce and e Business Money is the ultimate motive, if not sole of the business. Thus there will not be any compromise on financial transaction. E-business has to deal with flexibility, interoperability, scalability, performance and security of business; e-commerce has to deal firmly with security of the transaction e-commerce improves quality of service and performance. Security of e-commerce is required at two levels: Confidentiality and authenticity. Failure at the security of e-commerce virtually means the failure of the e-commerce itself. Financial transactions are done by several modes: Electronic cash, Electronic cheque, and Electronic transfer and payment advice etc. Security of the transaction in electronic payment system is the key to e-commerce. Public key techniques, Digital Certificate are useful security measures of e-commerce.

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9. KNOWLEDGE AGE AND MANAGEMENT As we move forward and as more and more human-IT interaction plays role in shaping society, an all inclusive knowledge society turns into. Knowledge Management has become a central issue of the knowledge age. What is then Km (knowledge management)? In a theory, “KM is seen as a logical extension society in that its purpose is to cope with explosion of information and capitalize on increased knowledge in workplace.” According to Peter, “The successful companies, in the knowledge management terms are the ones that have looked at the business processes rather than seeing the solution revolving round the company intranet.” According to his research, “the main reasons for using knowledge management techniques are to be competitive. Through globalization, there are a lot more competitors coming into markets quickly. Therefore, you need to do more in order to appear different. Another layer of knowledge is how to integrate things in the organization so that this process makes the organization look different.” His research suggests, “Content management is important.” He mentioned a figure that “ out of 1000 pages of a marketing intranet, 873 pages were not used, the reason being that they were out of date.” KM is not dumping data on the intranet, but for sharing of knowledge and information. British Telecom (BT) is recorded on saying the following reasons for sharing knowledge: (1) knowledge is the basis of services, (2) knowledge helps to cope up with changes, (3) knowledge sharing is the natural next step to information sharing. Alain J Godbout[61] analyzed the concept of KM from views of Peter Drucker, Nonka, Tom Davenport, and the American Quality and Productivity center among others. Following Peter Drucker, he viewed that the KM process “is a question of proper vision, organizational networks, educated decisions and best use of lessons learned as the key to organizational learning.” He further said: “In a sense, knowledge management is a form of application of sound management practices to an object: human resources which are the carrying vector of knowledge.” Tom Davenport and the American Quality and Productivity Centre is believed to emphasize more on explicit knowledge, and “their emphasis is to focus on means of optimizing these holdings (the explicit knowledge of organizations is contained in information holdings), improving the methods of formalization and increasing the use or usability of the available knowledge.” Referring to Nonaka’s model of mental process, Alain said: “ knowledge management is a form of sound management practices to another object: information resources with a different carrying vector of knowledge.” Taylor[62] views knowledge management as a “process of ensuring that the organization’s knowledge needs are met and exploiting the organization’s existing knowledge assets.” DiMattia and Oder [63] defined KM as “ KM involves blending a company’s internal and external information and turning it into actionable knowledge via a technology platform.” It is to note that in the definitions, sometimes the knowledge and the information are used interchangeably. In our eastern philosophy, the knowledge management can be seen in unique terms. As per Bhagbat Gita any action has two components: Karmayog that proceeds along path of action and Sankhyog that proceeds along the path of knowledge. To our philosophy, nature is the best manager. Again the first law of management of nature is the principle of least action and least time that aims to “accomplish most with least effort” and least time. In nature, everything follows the least path of action. Apple falls from a tree to ground on straight line, waters falls down in straight line, and light travels on straight path etc. Business or Organizational Management being a action of man where man is a part of nature, any man as manager desires to accomplish any action with least path and time for which Information Technology is with us today by our own creation. One can see in Fig. (4) how the information technology, which is due to the marriage of Computer with Communication, is tending to be like nature type technology.

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Over time the gap between human axis and technology is reducing. Therefore KM is an action to achieve goals along the path of knowledge with least action both mental and physical, or otherwise to do management so far done absolutely by man by technology in order to go along a path of least action, the path of nature by expanding intelligent technologies like brainy computers and personal communications. Swamiji made a following few comments over nature, man and knowledge: “Nature with its infinite power is only a machine.” “All our knowledge is based upon experience…. All human knowledge proceeds out of experience; we can not know anything except by experience.” “Man is man so long as he is struggling to rise above nature, and this nature is both internal and external.” These observations of Swami Vevekananda imply that man by earns knowledge from experience, and he applies his knowledge to be creator of nature, which is not impossible so long nature is assumed a machine. It will be pertaining to mention here that Tagore told that everything in nature follows a rule. This supplements my views that the KM is a step of human effort where he attempts to be his known creator.

9.1 KM-Conflicts and Confusion Knowledge management appears to be a collection of organizational knowledge in machines whereby the collected knowledge can be shared instantly at anywhere, at any time and by any body for the managerial purposes, be it for the policy decisions or for the routine works. But can the collected knowledge be ever creative? Or the human knowledge, which is ever creative, can be collected? Human knowledge has an ever alive and changing creative dimension. The creativity of human knowledge brings invention and innovation in framing organizational problems and solving organizational problems. Human and computer have their own different merits, and merely are not supplement to each other. Therefore the goal of the KM to capture and keep the knowledge of the employees leaving the organization will how far be successful is not without doubt. The objective of the KM is to share the knowledge at intra organizational and inter organizational level for arriving at a decision. The sharing provides a number of solutions and practices done and/or in operation in the organizations so that some of them could be deployed for the current need. All such sharable solutions and practices are definitely pre-programmed and heuristics in nature. A preprogrammed solution is fitting to a stable environment. A hostile environment may be of innumerable types caused by the wicked environment. How far a pre arranged solution is applicable to processes and problems of wicked environment? One recent example is the fight between ICC (International Cricket Council) and BCCI (Board of Cricket Control of India) over match referee’s report against six Indian cricketers during their second cricket test against South Africa. No pre-programmed knowledge was available for providing a solution. The problem was first of its kind and dragged into a wicked environment. Only human creative empowerment has saved the situation. Yogesh Malhotra[64] authoritatively analyzing the KM in inquiring systems duly highlighted the stated limitation of KM. He showed that out of the four inquiry systems, namely: 1. Leibnizian systems those “are closed systems without access to the external environment: they operate based on given axioms and may fall into competency traps based on diminishing returns from the ‘tried and tested’ heuristics embedded in the inquiry processes”. Example: as per some mathematical models (ESPN ratings, Pepsi ratings ..), the best team and players of test/ one day cricket.

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2. Lockean inquiry systems those “are based on consensual agreement and aim to reduce equivocality embedded in the diverse interpretations of the world view.” Example: Selection board meeting for a cricket team 3. Kantian inquiry systems those “attempt to give multiple explicit views of complementary nature and are best suited for moderate ill-structured problems.” Example: Result of a final match 4. Hegelian inquiry systems those “are based on a synthesis of multiple completely antithetical representations that are characterized by intense conflict because of the contrary underlying assumptions.” Example: Which party is to form government when no party has got majority in any Indian Parliamentary Election! The KM may have the significant role in Lockean and Leibnizian systems as they are “suited for stable and predictable organizational environments”, but the KM will have limitations in applying to other two systems as they “are better suited for wicked environments.” The wicked environments are characterized by discontinuous change, and the information technology has a trend to create wicked environment, it is not yet clear how the KM will suit to information technology driven present and future world. 5. The one of the main features of the KM is sharing of knowledge for improving business process and activities. The expectation and the results from knowledge sharing in many cases, particularly in the environment of competition, however cause havoc. In one final examination the topper of the class and the second topper sat side by side. The topper wanted to share the answer of a problem which he correctly got as, say 60. The topper when asked the second topper, although the second topper got 60 as answer, yet just to confuse the topper he told him that the answer was 50. The topper being confused scrapped out that answer, and tried another; but before its completion the time was out. Consequently, in the result the topper went down to the second position and the second topper moved up to the first position. This shows the possible consequence and counter productive feature of knowledge sharing particularly in competitive business environment. This phenomenon of knowledge sharing may be called “calamity of knowledge sharing.” The calamity may also occur when sub standard knowledge is shared. 6. The more serious conflict of knowledge sharing lies in its very definition. If knowledge is power, if knowledge is saleable, and if knowledge brings prestige, power and authority; why one should share his or her knowledge? The very basics of knowledge do not support the knowledge sharing. This being the case, the KM itself lies under a cover of confusion. Thomas H Davenport described [49] this phenomenon, as “sharing and using knowledge are often unnatural acts.” He felt that “sharing and usage have to be motivated through time-honored techniques-performance evaluation, compensation for example …..Lotus Development, now a division of IBM, devotes 25% of the total performance evaluation of its customer support workers to knowledge sharing. Buckman Laboratories recognizes its 100 top knowledge sharers with an annual conference at a resort. ABB evaluates managers based not only on the result of their decisions, but also on the knowledge and information applied in the decision-making process.” The other type of problem of same nature also exists in the organization. An employee who is an expert in obsolete technology may do not like to share knowledge of expert of new generation due to several reasons like ego, inferiority complex, and fear of being out classed. This phenomenon can be analogically compared with electric circuit as illustrated in Fig. (20). The organization likes to attain at a knowledge level, K. It has a storage capacity, C. But the organization offers a resistance. This

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resistance delays the organization to attain at the knowledge level K. Until and unless the offered resistance is removed by organizational process of transformation, the conflict will exist and resist the implementation of KM. The organization resistance (R) restricts the flow of knowledge.

C >> Storage Capacity of Organization

K

R >> The Organization Resistance (Physical, Mental and Cultrual Resistance)

Fig. 20: An analogy of a conflict

7. KM involves two words: Knowledge and Management. Who is for whom or who will rule to whom is a big question. Does KM mean the management of organization by the knowledge or does it mean the management of knowledge of the organization or a hybrid? This confusion is pictorially illustrated in Fig. (21). 8. Lester C Thurow documented some factual conflict that is existing in the USA: The information technology has been projected as a high productivity in nature. But the Lester C Thurow studies claimed that “ Financial services in the United States have had negative productivity growth for the last ten years. Every year productivity is falling about 1 percent.” His studies on office automation show that offices still use paper in the same ways for the last 500 years. The paper less office or automated office still remains a far cry.

Knowledge

Management

Knowledge

or

Management

OR BOTH ?

Fig. 21: A conflict in picture

Knowledge management is the technology-based management. Therefore its impact and consequences will change with technology and technological trends over time. However it will be not wrong to define KM as a management using computer and communication or for that purpose if we write : KM = MC2 The technology in general and information technology in particular follow a few empirical laws. In that light, we can analyze and predict the future technologies and hence future KM.

9.2 What is there after Knowledge Management Knowledge age has emerged in pace with information technology that innovated information age a few decades ago. The rapid transition is unprecedented in the history of technology

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applications. Thus logical speculation is: what is next to the knowledge age? One incident reported in the Indian may through some light to it. The great Akbar once asked his naba- ratnas: “what moves fast?” When eights of nine ratnas pointed towards Royal Horse, the ninth ratna, Birbal got an edge over others by saying “Our Mind, Sir.” We at least find a technology area where the trend is to achieve something like speed of mind, and this is nothing but communication. From the trend of communication we have no hesitation (and I am sure all will agree to it) to conclude that it is the speed of communication that is growing leaps and bound. We have seen the age of kilobits per second, and mega bits per second, and presently in the age of gigabits per second, and are seeing a tomorrow of tera bits per second. This is an indication that after knowledge age, the next age may be the age of mind or the age of conscious. The universe is made of non-living and living things. Their comparison in terms of level of intelligence, conscious and communication power is made in table (28). S Ranade, a great admire of Aurobinda told [65]: “Knowledge by identity will change current science completely. Particularly physics and biology will see radical changes. The wave-particle duality and the mass-energy equivalence will be seen in the light of the more basic substance of consciousness” and then he defined [65]: “consciousness is awareness, awareness of yourself and of others. In the human being both exist. In the animal, there is only awareness of others, not awareness of itself, it is a more limited awareness. In plants the awareness is even less. In the crystal it is still less, but nevertheless it is there.” If the crystal is having awareness, it is surely possible that “the next century will be the century of consciousness” and “you can focus your body consciousness on a point outside the body.” Will the “Will power” or “Mind Power” of Iswar Patuli depicted by great Bengali Novelist Sarat Chandra prevail upon the society, organization, culture and economy at the fragile end of knowledge age? Mother’s in the historical declaration[66] made on April’24 1956 said: “The manifestation of the supramental upon earth is no more a promise but a living fact, a reality. It is at work here, and one day will come when the most blind, the most unconscious and even the most unwilling shall be obliged to recognize it.” Perhaps that will be in the age of consciousness that is next to knowledge age. The collaborative views on this prediction is one important research found in [68]. Table 28: Comparison of different entities in universe in terms of sense and communication Non-living things Living things

Apparently no sense and no communication. Dr Ranade sees otherwise Plants

Limited sense and no communication

Animals

Low level sense and communication

Human beings

High level sense and communication

10. AGE OF DIGITAL DIVIDE Tagore once told “we have only one country in this universe, and that is world”. Rabindranath Tagore’s such a powerful philosophy may ultimately be realized if to-day’s tenet of “one world one village is implemented in true sense in future. To achieving this, a trend has already been initiated the world over. Privatization, Liberalization and Globalization are replacing liberty, fraternity and equality all over the world including the countries of third world. It does not mean that library and fraternity have no relevance in to-day’s society. They are ever alive and

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their universal appeal shall ever remain for the noble human society, but to day they are not all in all. Privatization and universalization shall be the other social partners with them. This is a wave brought forward by different emerging technologies, which are often interactive, interdependent and diffusive. Information technology, computer, communication, microelectronics, Genetic engineering, Biotechnology, Space technology are a few to name worthy. Developing world in general is far lagging behind the modern technological evolutions and revolutions. Besides the developing countries are hardly having capital to deal with such fast, rapid and perpetual changes. Developing world in general is labor intensive rather than capital intensive. Therefore, debate on the ability, suitability and the acceptability of liberalization is going on and will continue to go on for some more time in the developing countries. Initial mismatch and inertia are parts of life and the fact is that the society never denies mobility. The society ultimately accepts technological changes, which might be off-touch to the society even a few years back. And irony is that delayed such acceptance is done in quite haphazard and irregular ways. What has happened to the deployment of computer in government sectors in India today is anybody’s guess. This is a lesson that the third world always forgets. Consequently the third world continues to lag behind International trend, and losses money as, there is hardly any planning for technological up gradation and applications. We can sight a figure to justify this point. Telecommunications lines of India are 66% digitized; where as figures of Brazil and Hungary are respectively 35.7% and 41%. But the faults’ figures are 218 faults per 100 lines in India and 2 faults per 100 lines in USA and Japan. In Table (29), the percentage share of information technology for America, Europe and Asia, and that of the e-commerce buyers are shown. It is noticed that in both terms, the position of Asia is very poor. Table 29: % share of IT and E-commerce buyers % share of information technology in 1995

% share of E-commerce buyers in 1998

America

45.5

72.57

Europe

30.9

22.8

Asia and pacific

23.7

4.6

Better is not the sole dimension of competitive advantages; faster is equally another important dimension. Thus it will be a sound strategy for the developing country to take part in the globalization with out any further loss of time, but with intelligent, selective, judicious and strategic applications of globalization process, uses of and innovation with few technologies. Analyzing the problems of Third world in depth Dr. Colombo observed “The ability of developing countries to derive all the benefits of the new technologies faces one stumbling block right from the start. Although rapidly and seemingly effortlessly permeating the economic and production systems of the world, these technologies are not available “off the peg”. They have to be absorbed, metabolized, mastered and controlled. Their application calls for a pre-existing capability to insert new ideas, new practices, and new elements into a flexible system. This does not simply exist in the vast majority of the developing countries. Furthermore, it is essential that as the new technologies are introduced into the socio-economic fabric of the third world, they do not impair or destroy existing local cultures—we must equally concern ourselves with safeguarding the richness of the world cultures, mankind’s “cultural genoma”. Despite these problems it is strongly believed that the intelligent application of the new technologies in the developing countries can indeed speed up process of economic growth”.

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10.1 Gap Studies In history of social studies one important component of research deals with the findings reason and cause of growing gap between rich and poor; and for that purpose to suggest measures and steps to reduce the gap. But the fact remains that the gap has not been reduced even after thousands of such studies and the implementation of their recommendations those including those of some noble laureates. A few research findings report[67] 1. If the present growth trends in world population, industrialization, pollution, food production, and resource depletion continue unchanged, the limits to growth on this planet will be reached sometime within the next 100 years. The most probable result will be a sudden and uncontrollable decline in both population and industrial capacity. 2. It is possible to alter these growth trends and to establish a condition of ecological and economic stability that is sustainable far into the future. The state of global equilibrium could be designed so that the basic material needs of each person on earth are satisfied and each person has an equal opportunity to realize his or her individual human potential. 3. If the world’s people decide to strive for this second outcome rather than the first, the sooner they begin working to attain it, the greater will be their chances of success. To us those conclusions spelled put not doom but challenge - how to bring about a society that is materially sufficient, socially equitable, and ecologically sustainable, and one that is more satisfying in human terms than the growth-obsessed society of today.” Whatever gab and whatever challenge is to be met revolves around three factors : i) economic and social gap, ii) education gap and iii) status gap between agriculture and industry.

10.2 Problems of Agriculture Sector The existing economic and social gap between rich and poor is primarily due to two avalanche affects: (a) In agriculture sector, negative avalanches is “produce and perish” and (b) In business sector, positive avalanche is “produce and flourish”. The only solution to bring the balance is that the prices of agriculture produce must be raised at those of business produces by strict control of governments. Education is an investment not only in terms of money but also in terms of time and human resources. Parents have noticed that the boys/ girls after getting school level education become useless/worthless/resource less rather than resourceful in terms of earnings in the family. They neither get job nor by that time skillful for laborious jobs including agricultural jobs. Had these boys not been sent to schools rather been engaged from the childhood in agriculture related sectors; they would be more useful for earnings for the family. This clearly demonstrate that the education till not is sure with guaranteed minimum income to family, the poor family does not like to take risk of spending mainly time and money in education. Mac Bridge Commission report that the farmer and the agriculture producers must have the direct market knowledge to get actual price of the produces. This is believed to be possible only with IT.

10.3 Case of Industries The state of West Bengal in India has achieved a considerable amount of rural economic growth in the last two decades. The average income of the rural people has increased and the social

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security of rural people has been established on the solid footings. The disparity in income among the rural people has decreased considerably. An all around development of rural people and society has been noticed. However this development is due to land reforms and “barga” system sincerely implemented by the Left-front government of W. B. in their 25 years of rule. By the process of land reforms and barga system, the agricultural workers or farmers are given confidence that they will never be thrown out of work and land they do cultivate. This confidence has led to generate among farmers the more sense of belongingness and sincerity in their work. This has reduced the victimization and the injustice meted out to them in terms of payment or no payment earlier by the Land-Lords; which in other ways has caused the agricultural productivity to increase and loss of agricultural working days to decrease as well as the agricultural disputes between labor and owner to lessen. The barga solution is our own and is not something copied from the developed nations. The economic and productivity failures in all sectors namely agricultural, industrial and banking is mainly due to disputes between labors and owners. Thus if such disputes in agriculture sectors are overcome by the barga system; it is logically extensible for other sectors like industrial and banking too. In this paper we propose an “industrial barga” system for Indian Industries. We have achieved something unique by our own system of bargas in agriculture sector. Similarly the industrial barga not prevailing elsewhere dose not mean it is inappropriate in India. In Indian environment where economic disparity is huge and where labor is cheap and for which victimization of labor is easy; the industrial barga will be the right solution. The proposed industrial barga aims to provide share of production and profit of industries with labor, management and owner as in agricultural barga. There may be several means of implementation. The Industrial barga will not be easy to implement. With IT age, the difference gap is easily to meet with. What is need of the hour is the strategy and goodwill for the application in right perspectives.

11. CONCLUSIONS The goals of both the near and the far futures of IT is Fig. (22). In the field of computer, the major challenge of the 21st century will be the designing of bio/brainy computer. The basic science has been searching, since the days of its journey, the design if any behind the universe as well the theory of birth of the universe; and possibly a new “Theory of Everything” as Prof Hawking’s prediction made in 1980 of achieving his famous “Theory of Everything” by the end 20th century is proved wrong. The debate on deterministic vs. probabilistic nature of universe or whether the nature is a machine or not, is oscillating. In such a scenario the debate on possibility of designing brainy computer only be a logical extrapolation; and definitely will take long time to answer. On the other hand, future all wireless, anywhere and any time communication is relatively non-debatable issue and expected to be achieved, although not without overcoming many obstacles. Even small deployment like IEEE 802.11 based WLAN faces many obstacles[69]. Other than systems and standards, two inherent problems of future communication need to be properly addressed: higher error probability of all wireless links and information security. Whereas the error control is basically a technical issue, the security of information has several dimensions. The requirement of security for a durable application of IT, namely e commerce and e business was illustrated earlier. It is reported[70] that “The increasing frequency of malicious computer attacks on government agencies and Internet business has caused severe economic waste and unique social threats.” As per the second law of thermodynamics the open systems cannot bring order without making its surroundings disorder.

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The security measures that are for bringing order to information process, inevitably brings disorder to its surroundings that may again itself be the source of hackers or security breakers. This is a manifestation of chaos and complexity. Of course the men create problems only to solve it afterwards. Is the nature likes to see man dancing between problems and solutions? Are we then leading us to a state of chaos and complexity[71]? This is compounded by the fact that “computer system and network security is increasingly limited by the quality and security of the software running on constituent machines. Researchers estimate that more than half of all vulnerabilities are from buffer overruns, an embarrassingly elementary class of bugs”[72,73]. The steps to go out of chaos and complexity will be the major challenge for investigation in 21st century. High Speed Computing, Autonomous Computing Optical Computing, Quantum Computing Chemical/Bio/Intelligent Computing Seamless Power + Intelligence

In Neat Future Information Age to Knowledge Age Knowledge Society, Knowledge Factory, Knowledge Workers, Knowledge as Wealth

In Far Future Age of conscio usness

3G Mobile to 4G Mobile, Cellular, GSM, PDC, PHS, Paging, UMTS, FSO, xDSL, Next Generation IP and VoIP, Wireless Ethernet-IEEE 802.11, Wireless Home Networking IEEE 802.15.4, Wireless Internet, LEO, Multimedia Standard, Wireless ATM, PCN Seamless Mobility, Coverage, and total Integration

Fig. 22: IT in 21st century

Entering into the knowledge age is the inevitable consequence of the application of networks in the business, organization, government, society and economy. The entry needs to break several hurdles. The issue of the acceptability of knowledge economy with non-material wealth, knowledge along with the new status of human resources as knowledge workers, and the concept of sharing knowledge for organizational benefits are a few areas to be addressed. The quantification of the knowledge and the exchange rules of knowledge for the purpose of sale and business of and with knowledge are the technical challenges and need serious

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investigation in this century. The consciousness as Penrose told “is the phenomenon whereby the Universe’s very existence is made known.” Thus in the age of consciousness, the man’s desire to be the master of nature with which this paper started, may be realized. Will it be really! The constructive and judicious application of IT may lead to overcoming the consequences of “digital divide.” Several studies[74,75] have suggested for the application of IT in Education & Training, Telemedicine and Diagnosis, E-Government, Rural Information Sharing for the purpose of food conservation & sale, and Entertainment among others for deriving maximum benefits in the developing countries. Like digital divide, another negative application of IT is like what happened on 11th September in USA. Analyzing the 11th September issue, a famous research work[76] has reported to examine the issue for developing a system dynamics for positive application of technology. This is a new direction of research in application of technology. The same direction may be extended to remove digital divide.

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74. 75. 76. 77. 78. 79. 80.

APPENDIX-A Edholms Law The following table depicts the growth of data rates under different communication/network technologies. The data rate follows Edholm’s law that states the data rates for all three communications, namely wired, nomadic and wireless are as predictable as Moore’s law. The rates are increasing exponentially and the slower rates trail the faster rates within a predictable time gap. Table: Date rate growth of different Communication/Network Technologies Year

Wired Technology/ Standard

19751984

19851994

19952004

Nomadic Data rate

Technology/ Standard

Wireless Data rate

Technology/ Standard

Data rate

Ethernet

2.94 Mbps

Hayes Modem

110 bps

Wide Area paging

A few hundreds bps

Ethernet

10 Mbps

Modem

9800 bps

Alphanumeric paging

A few Kbps

Ethernet

100 Mbps

Modem

28.8 Kbps

Cellular/GSM

≈ 50 Kbps

Modem

56.6 Kbps

IEEE 802.11 b

11 Mbps

IEEE 802.11 g

108 Mbps

PCN/UMTS

> 2 Mbps

B3G (Beyond 3 G)

12 Mbps

MIMO

200 Mbps

Ethernet

1 Gbps

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2 1.

Network and Internet Technology

HISTORY BEHIND COMPUTER NETWORK

The convergence between two important technologies, namely computer and communication gave the birth of Computer Communication and Network. A useful study of convergence of telecommunication and computing is literature [1]. Computer network means a network of geographically distributed many autonomous system connected [2,3] in such a mode that meaningful transmission and exchange of information become possible among them. Resource sharing and load sharing are the major two objectives of a network. To meet the objectives the networks were evolved over times. In early 1950s, peripheral devices and remote job entry points were connected to the central computer through communication links (Fig. 1). This is to use the resources optimally and economically, as then computers were very high cost machines. By sharing of resources, the cost was distributed over number of users. In 1960s the number of peripherals devices expanded rapidly, the computer power also increased by several folds and time shared computer was evolved. This made the use of separate long haul /distance communication link to each peripheral device. But the solution was most uneconomical and technically unsound. Fig. 2 illustrates the falling cost of communication and computer, but the rate of fall was more in case of computer than in the case of communication links. Moreover the design of a high bit data link is more economic than several low data rate links. For Example the cost of a 1.5 Mbps link is about six times that of a 64 Kbps link but the data rate is 24 times higher. The trade off is low cost per unit data versus high cost per unit data. This leads to the use of remote multiplexers or concentrators to connect a number of remote terminals in the same area and to use a single shared communication link to connect with central computer (Fig. 3). As the number of devices increased, the task of communication became huge. To free central computer from communication task, special processors called Front End Processors were used. The communication is automated in such system but the control of communication still remained central to the central computer. In 1970s, with advent of PCs—low cost, portable but not less capable computer, the problem takes a U turn in several aspects: automated system at remote locations, distributed control etc. This leads to the concept of connecting several geographically distributed computers—some of them doing the computational jobs and others communicational jobs

99

100


(Fig. 4). General purpose network concept then evolves as Communication Sub network + Users’ Sub network. In 1980s and thereafter, several networks of different types were connected through bridges and gateways to develop an interconnected network (Fig. 5). Interconnecting network of different networks is known as internets (note the lower case of “i”). The special such an unique internet is called Internet (note the upper case “I” in Internet). Printer

T

T

Central processor

Terminal T

T

T R. I. F

Falling cost

Fig. 1: One central processor and a separate communication link to each device like terminal, remote job entry (RJE) points, pointer etc.

Communication cost

Computer cost

Year

Fig. 2: Falling cost of Computing and Communication with year (Cross over year 1970)

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NETWORK AND INTERNET TECHNOLOGY

101

T Printer T

T

T

T Multiplexer

Central processor

Front end processor

Printer T T

T

Terminal T controller T

T Multiplexer T

Fig. 3: One central processor but with shared communication links to devices like Terminal, Font End Processor, Terminal Controller and Multiplexer.

(Issue: Front End Processor performs the job of communication, while central processor remains in processing job)

Personal computer Terminal

Node

Node

Communication subnet

CPU

CPU

Fig. 4: General network with two subnets: Communication Subnet and Users’ Subnet.

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BOX 1 How do we see a Computer Node in the Netowork in Terms of Message in and out ?

Model of Nodes The intermediate computer node or terminal may be seen in the different models of queuing. A model of this kind has four things for specifying: the message arrival probability density, the message departure or service probability density, the number of the servers in the node (basically this is the number of the output links or the number of the computers/machines providing the service) and the buffer space in the node. For the purpose of simplicity and analysis in the computer networks, it is often assumed that the nodes are of having infinite buffer space. However such an assumption works fine as usually the analysis is of comparative nature. The different statistics are used to define the arrival and the service probability density functions and these may be: exponential probability density function represented by the symbol, M (Markov); deterministic nature represented by D or general arbitrarily type represented by G. A model of type M/M/1 means it is a node where: The message arrives at the node at the Markov Process or in the exponential probability The message departs the node at the Markov Process or in the exponential probability The number of the server is one The exponential probability density function is most suitable for a large number of independent customers. The large number of independent customers actually comprises a network. Thus it has been observed that the exponential arrival and the exponential service process best suit the computer network. The M/M/1 model is one of the appropriate models for a node. The Markov process is governed by the Poisson Law that states that the probability, P k(t) of arrival of exactly k messages in an interval of time, t is given by: Pk(t) = [(λt)k/k!] e–kt ...(1) where λ = average arrival rate of messages. We define the state of the node by the numbers of the message staying in the buffer (queue) and in the server of the node (Fig. 1). Pk defines the probability that the node is having k messages. Queue/Buffer Message arrives,

Server M

M

M Message departs,

Fig. 1: M/M/1 model

We define m as the average message service rate of the node. If we define C as the capacity of the server’s output link in bps (bits per second), and m as the average size of the message in bits, we have: µ = C/m ...(2) Based on our defined state of the node, the state diagram of the M/M/1 node will look like as in Fig. 2.

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NETWORK AND INTERNET TECHNOLOGY P0

P0

P1

P1

P1

Pk – 1

103

P2

P2

Pk + 1

Pk

Pk

Pk

Pk + 1

Pk + 1

Fig. 2: State diagram of the states/State transition probability.

The node is in equilibrium when from Fig. (2) we have: λP0 = µP1 λP1 = µP2 λP2 = µP3 ………...…. λPk = µPk+1 ………...…. we then have: Pk = (λ/µ)k P0 K=∝

∑ P = ∑ (λ/µ)

and

k

k

...(3)

P0 = 1

K=0

or

P0 = 1/(1 – (λ/µ)) = 1/(1 – ρ) where ρ = λ/µ [ as Σ

(λ/µ)k

= 1/( 1 – (λ/µ)) or

So,

Σ ρk = 1/(1– ρ)

Pk = (1 – ρ)

...(4)]

ρk

If N is the average number messages in the system, we know: K=∝

N=

∑ K.P

k

= (1 – ρ) Σ k . ρk

...(5)

K=0

Differentiating both sides of equation (4) with respect to ρ and then multiplying both sides by ρ, we find that Σk .ρk = ρ/(1 – ρ)2, using which in equation (5), we get: N = ρ/(1 – ρ)

...(6)

If it is assumed that the average time all the messages in the node stay is T, they by famous Little’s formula: N = λT

...(7)

Using equations (6) and (7), we get : T = 1/(µ – λ) Equations (6), (7) and (8) are used to qualitatively analyse the nodes of networks.

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...(8)

104


QUESTIONS 1. ρ is called load intensity. Why ? What is its physical significance ? From eqn. (8), it is seen that for a physical realizable and stable system, µ should be greater than λ i.e. ρ should be less than 1 (Fig. 3). Actually if µ is not greater than λ, the number of the data in the system and its corresponding delay will continue to grow unbounded. Then natural question be asked if µ is greater than λ, where is the question of delay? Actually the parameters, λ and µ are statistically average. At some instant the arrival rate may be higher than the service rate when the data faces delay. =1

T delay

Fig. 3

2. In the network environment two nodes are as in Fig. (4). The nodes are independent to each other. Prove that the probability that the node N1 is having n1 messages and the node N2 is having n2 messages is given by: n P (n1 in N1 and n2 in N2 = ρ1 1 (1– ρ1) ρ2 n2 (1 – ρ2)

N1

1 l1 = total average arrival rate from other nodes/internal/external at node N1

N2

2

l2 = total average arrival rate from other nodes/internal/external at node N2

ρ1 = λ1/µ1 ρ2 = λ2/µ2 Fig. 4: Figure of the given question

SOLVED PROBLEM 1. At a node, packets arrive at an average rate of 120 per hour. The output service link capacity is 8 characters/sec. Find (a) average number of the packets in the system at steady state condition, (b) average delay per packet, (c) average waiting time in queue, (d) average number of packets in queue.

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105

Assume (i) M/M/1 model (ii) one packet = 144 characters. Solution. λ = 120 pkt/hr. = (120/3600) pkt/sec = 0.033 µ = (8/144) pkt/sec. = 0.055 δ λ = (a) N= = 1.46 pkts 1− δ µ − λ (b) T = 1/(µ – λ) = 44.33 sec. (c) W = T – 1/µ = 26.14/sec( W = δ/µ – λ) (d) NQ = λ . W = 0.86 packets. (NQ = δ2/1 – δ) 2. For the problem 1., what is the probability that there is 2 or less packets in the system under steady state condition? Solution. The probability is

2

∑ (1 − δ) δ

k

K =0 2

= 0.4

∑ (0.6)

k

[δ = λ/µ = 0.6]

K=0

= 0.4 + 0.4 x 0.6 + 0.4 x 0.36 = 0.784. BOX 2 How do the Nodes and/or the Computers Communicate over Share Line ? (This is Basicaalu an Issue of Users’ Subnet Communication)

Multidrop/Multipoint Link Control-Polling When a transmission path between two terminals or nodes or connectors is dedicated, the link is termed as point to point. When many terminals or nodes or connectors share a transmission link, the link becomes shared. For a shared link, if transmission between any two users is allowed at any time, the link is called Multidrop / Multipoint. Link. In order to regularize the transmission of a Multidrop / Multipoint Link, a control is required. The main goal of such regulation is to increase the link utilization of the link. Regularized control prevents the interference caused by simultaneous transmission by more than two stations. If more than two stations try to communicate simultaneously, the interference occurs making the communication the garbage. Thus a control is required to avoid the unproductive communication. In Multidrop/point Link, often one node or terminal or connector acts as a primary station. The role of primary station is to monitor the control of link regularization. The other station or nodes or connectors in the link are known as secondary stations. In a Multidrop/point link, communication between primary and any one secondary is allowed at any given point of time. However, the communications between secondary stations are only done though primary. This is in sharp contrast to the other kind of Multidrop/point communication technique used in networks like LANs (Local Area Networks). In terms of accessing methods of shared link, different techniques may be classified as in Fig. (1). Polling is a technique used in Multidrop/ point link control. Two main categories of polling are serial polling and hub polling. A variation of serial polling is known as selective serial polling. In all polling techniques, the stations are assigned some addresses that are known to all stations in the link.

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Access methods

Non-contention techniques (Deterministic)

Centralized (Hence primary station is responsible for control Example : Poling

Contention-technique (Probabilities/Stochastic) Example: CSMA/CD used in LAN

Distributed ( All stations equally take part in control) Example : Token passing used in LAN

Fig. 1

In serial polling (also known as roll-call polling), the primary station sends a poll to each secondary station one at a time and one often another. For Fig. (2), the primary station sends first a poll message to S1. The poll message contains the address of polled station as a header. The secondary stations check the header address, and thereby identifies whether the poll message is meant for the particular secondary or not. However, S1 getting the poll message accordingly identifies the poll. If the poll is meant for it, it will send a positive response followed by data if it has data to send. Along with the message/data, the secondary sends its address also. If it has no data, S1 will send a negative response, which makes a call nonproductive. After the call of S1 and after receiving its response, the primary will poll S2, then S3 and so on until Sn. After that the cycle will repeat.

S1

Primary

Sn

Fig. 2

However, the major disadvantage of roll-call polling is the waste of time due to nonproductive calls. This problem is uniquely tackled in hub polling. In hub polling, the primary initials the polling. It initiates polling by sending a poll message to S1. S1 sends data if it has to send. Other wise, the secondary S1 transfers the poll to S2 and so on. Primary identifies the senders by the appended header address of secondary in the data message. The last secondary Sn returns the poll to primary. There is other way also to implement hub polling. For example, the primary may initiate poll by polling first Sn. Sn after its response, sends poll to Sn – 1 and soon. A poll is terminated when S1 at last transfers the poll to primary. The processing job in secondary stations for hub polling is more than that for serial polling.

Sn

Si

S2

S1

Polling

Fig. 3

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1

S

2

S

;

S

m

S


107

Analytically serial polling and hub polling may be compared. For such comparison, we shall use a parameter known as poll-scan time (PST). PST is the time taken for just polling (and not for sending data) all the secondary stations once in a cycle. For the purpose of analysis a generalized shared link system of Fig. (3) is to be considered. We assume: tp = Processing time required in primary to initiate a poll. We assume it is same for serial polling and hub polling. ts = processing time required in a secondary to process the poll message and to process the response for serial polling. th = Same as ts but for hub polling. tli = Time required both for propagation and transmit for a poll message to be transferred from primary to secondary, Si i tl = Same at tli but for a poll message to be transferred from primary to secondary Si (i = 1 to n, j = 1 to m) We also assume tli and tli for all possible is and js are same for serial and hub polling. Under these assumptions, we can calculate PST as below: For serial polling For secondary S1, poll scan time will be tp + ts + 2tl1. For S2, the same will be tp + ts + 2tl2, and so on for other secondaries. Thus (PST)s =(n + m) (tp + ts) + 2(tl1 + tl2 + …. + tln) + 2 (t11 + tl2 + …. + tlm) ...(1) For hub polling On similar grounds (PST)n = tp + (n + m) tn + 2(tln + tlm) ...(2) We further assume that all stations are equidistant apart in which case tli = tlj = t (constant). Then (PST)s = (n + m) (tp + ts) + 2t (1 + 2 +…..n) + 2t(1 + 2 +….m) as under the assumption of equidistant we have tl2 = tl2 = 2t tl3 = tl3 = 3t ………....... tln = nt and tlm = mt Then, (PST)s = (n + m) (tp + ts) + n (n + 1) t + m (m + 1) t or

(PST)s = n [(tp + ts) + (n+1)t] + m [(tp + ts) + (m + 1)t] Under the same assumption of equidistant nodes, we have (PST)h = tp + (n+m)th + 2(n + m) = tp + n(th + 2t) + m(th + 2t)

...(3)

…(4)

We can make several observations on equations (3) and (4) : 1. Condition that hub polling should be superior to serial polling is that : (PST)h < (PST)s or

tp + n [(th + 2t) + m(th + 2t)] < [n(tp + ts) + (n + 1)t + m(tp + ts) + (m + 1)t]

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th < [{(m + n – 1) tp + (m + n)ts + t (m2 + n2 – m – n)}/(m + n)] ...(5)] 2. If we assume m = 0, then system of Fig. (3) becomes that of Fig. (2) for which: (PST)s = n[(tp + ts) + (n + 1)]t (eqn. (3) with m = 0) (PST)h = tp + n(th + 2t) (eqn. (4) with m = 0) and condition for hub polling being superior will be : th < [{(n – 1)tp + nts + t (n – 1) n}/n] (Inequality (5) with m = 0) or tn < {(1 – 1/n) tp + ts + (n – 1) t } ...(6) or tn < ( tp + ts + nt) when n>>1. 3. If m = 0, n = 1, it is predicted that (PST)s should be equal to (PST)h. This can be verified from eqns. (3) and (4) provided ts = tn. For inequality (6), it is then seen that, tn < ts which is not the case as we known tn > ts. This suggests that hub polling becomes superior to serial polling as number of secondary increases. Selective serial polling is a compromise or hybrid technique. Serial polling is done by roll-call technique where under a poll cycle each secondary is polled once only. But in the selective serial polling, a particular secondary may be polled more than once, while other particular secondary may be called just once. In such technique, a pre-defined sequence table is maintained based on statistical behavior of data of the secondary stations. This reduces the delay due to non-productive call for a cycle of polling. Addressing in polling technique is done by standard protocol. One such standard protocol developed by IBM is known as BISYNC (Binary synchronous Communication). BISYNC message format is : or

SYN

SYN SOH

Where :

Header STX

Data

ETB/ETX Error

Check

SYN = synchronization Word SOH = Start of Header Word STX = Start of Text Word ETB = End of Transmission Block ETX = End of Text. For error check vertical parity is used if ASCII code is in use CRC is used for EBCDIC. However, ETB is used to terminate a block when there will be more transmission of block, whereas ETX is used to terminate the transmission of full text i.e. to indicate the termination of last block. Solved Problems. Compare serial polling with hub polling for a scheme of Fig. (3) with m = 0. Use the parameter of duration that a secondary waits before polling, for such comparison. Find the average range of such delay. Assume secondaries are having no data. For serial polling we have from eqn. (3) (PST)s = n (tp + ts) + n(n + 1)t Thus the average time to poll a secondary, ta = (PST)s/n = (tp + ts) + n(n + 1)t At any instant, a particular secondary may have to wait for polling of (n – 1) stations before a poll is offered to it or may not have to wait to get poll. As the secondaries are having no data to send, therefore the average waiting time before getting a poll is : average waiting = [{(n – 1) + 0}/2} ta

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We can find the second moment of the waiting time which is by definition is the weighted average of the squares of possible polling time. Hence, second moment = 1/n

( n − 1)

∑ ( pt

a)

2

p=0 2

ta (n – 1) (2n – 1) 6

=

n

as we known

=

∑

p2 = n(n + 1) (2n + 1)

p=0

Therefore, variance by definition is: Var = Second moment – (average)2 =

FG H

ta 2 n−1 (n – 1) (2n – 1) – . ta 6 2

n2 − 1 . ta2. 12 By definition average range is given below:

IJ K

2

=

Average ± 2 var Hence average range of delay before a secondary is polled is

n−1 ta ± 2 or, we can say :

upper limit =

and lower limit

n2 − 1 2 . ta 12 n2 − 1 3

1 n−1 t + ta 2 a 2

F GG H

n2 − 1 3

=

ta n − 1+ 2

=

n − 1 ta n − 1− 2 2

F GG H

I JJ K

n2 − 1 3

I JJ K

However if n = 1, we have Upper limit = lower limit = 0 Which is physically justified. For hub polling from equ (4), we have ta = Hence we can write 1. For serial polling

(PST) h t p + th +2t = n n

upper limit =

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FG t H

p

+ ts + (n + 1) t 2

IJ FG n − 1 + K GH

n2 − 1 3

I JJ K

109

110


lower limit = 2. for hub polling

FG t H

p

+ ts + (n + 1) t 2

FG t /n + t H 2 F t /n + t =G H 2

upper limit =

p

n

lower limit

p

n

IJ FG n − 1 − K GH

IJ FG n − 1 + K GH = 2t I F JK GGH n − 1 + = 2t

n2 − 1 3

I JJ K

I JJ K n − 1I J 3 JK n2 − 1 3 2

Multidrop/Multipoint Terminal Control After the discovery of integrated circuit technology, in general the cost of communication system exceeds the cost of communication line. In order to reduce the total cost of communication, therefore a mean of communicating is provided by which a single transmission line is shared by many terminals or users for purpose of communications. Two broad classes of terminal controls are shown in Fig. (4). Terminal controller may be again broadly of two types: multiplexers and concentrators. Concentrator is also known as asynchronous time division multiplexers or statistical time division multiplexers. In multiplexer each input line is provided with a predefined time slot. Input data are outputted hence according to the sequence. No addressing of input data is required. However, the output capacity will be sum of the input capacity of input. So far link capacities is concerned, multiplexer is not a good choice for terminal handling because of “BA” features of the data. As such it may so happen that most of the time for a small fraction time, some of input lines may have data to send; and hence time slot meant for them will be wasted.

Controller Controller

(a) Multidrop/Shared the communication

(b) Point-to-point lines/dedicated line communication

Fig. 4

In concentrator, the output line capacity is made less than the sum of input capacity; and the inputs are not sequentially allowed to send data at a predefined sequence as in multiplexer. Therefore data of inputs must have address for identification. For a concentrator if there are n inputs each with input data rate of r bps, then output link capacity, c in bps is given as: c < nr whereas for multiplexer c = nr As for concentrator c < nr, it may so happen that when all input terminals, try to send data, there will be loss of data. This will be tacked by buffer size at concentrator. However, it will be not a serious problem for data as data is insensitive to time.

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111

However, if a is the mean fraction of time each input terminal is transmitting, then 0= (e + 1). Table 6 D3

D2

D1

D0

C2

C1

C0

Weight

0

0

0

0

0

0

0

0

0

0

0

1

0

1

1

3

0

0

1

0

1

1

0

3

0

0

1

1

1

0

1

4

0

1

0

0

1

1

1

4

0

1

0

1

1

0

0

3

0

1

1

0

0

0

1

3

0

1

1

1

0

1

0

4

1

0

0

0

1

0

1

3

1

0

0

1

1

1

0

4

1

0

1

0

0

1

1

4

1

0

1

1

0

0

0

3

1

1

0

0

0

1

0

3

1

1

0

1

0

0

1

4

1

1

1

0

1

0

0

4

1

1

1

1

1

1

1

7

Thus we can see that Minimum non-zero weight is 3 as evident from the table. Similarly, for correct up to e bits errors per code, dmin >= (2e + 1) ; while for correct up to e1 bit errors and detect up to e2 (e2 > e1) bit errors per code, dmin >= (e1 + e2 + 1). For a code to detect e2 errors and correct e1 errors we need d >= (2e1 + e2 + 1) where e2 > e1 is not required, dmin = 7

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Correct Detect 3 0 2 2 1 4 0 6 However, in general, for any (m,n) code, the coding rate or coding efficiency is defined as R = (n/ m ) × 100 %. ...(12) Coding rate measures the increased % bandwidth required for transmission of a code. For a good code R should be high. However from rule 1 as stated previously one can verify that R increases with n. This is the reason that in CCN/DCN, the higher data size is used.

5.4.6 Cyclic Redundancy Code (CRC) CRC is also a block code but it is a non-linear block code. However, all EDC/ECC ‘s can be classified as : EDC/ECC

Block codes

Liner block codes (7, 4), (13, 8) etc.

Convolution codes

Non-liner block codes (CRC-12, CRC-16, CRC-CCITT)etc.

The EDC that provides much more confidence to the users for detection of error is known as CRC (Cyclic Redundancy Code). The CRC is an extremely powerful error detection code and easily implemented using simple hardware; thereby, providing a cost - effective solution. The code and its circuit are based on a polynomial known as generator polynomial, which is mutually agreeable and known to both the sender and the receiver. The important four versions of generator polynomial commonly used in industry are: CRE–12 = x12 + x11 + x3 + x2 + 1 (Data bit is 6 and check bit is 6) CRC–16 = x16 + x15 + x2 + 1 (x0) CCITT–CRC = x10 + x12 + x5 + 1 (CRC–16 and CCITT–CRC are Used in US and Europe respectively for 8 bit data) CRC–32 = x32 + x26 + x23 + x22 + x16 + x12 + x11 + x10 + x8 + x7 + x5 + x4 x2 + x + 1 (Local Area Networks under IEEE 802 uses this code. One of the hottest LANs, Ethernet uses this CRC 32). Check bits generating Instead of going into theory, we can justify that there is a quite simple circuit which can be used to generate check bits of CRC instantaneously based on serial data input. CRC-16 may be taken up, for example. The generating circuit for check bits is shown in Fig. (2). The location for feedback taps of exclusive OR gate can be simple determined. Subtract each power of x of

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generator polynomial from the number of location of shift registers (which equals number of check bits). The results of all such subtraction (in the example; 16 – 16 = 0, 16 – 15 = 1, 16–1 = 14, 16 – 0 = 16) are the tap locations. Before sending any data, the shift register is to be initialized to all zero. After transmission of data bits using shift, the content of the shift registers will be check bits. You now simply transmit these check bits. At the receiver, same circuit (fig16) will generate the check bits based on received data bits. The difference in any or more bit positions in between the generated check bits at the receiver and the received check bits at the receiver will detect an error. In addition to the simply circuit and lower cost, CRC offers another improvement over parity. For generation of check bits, we do not have to wait for whole of the message data. It is done serially and instantaneously. This is not true in case of systematic parity, the systematic parity has the effect of delaying the message more than CRC. The efficiency of any code is measured by a parameter known as code rate (Cr): Efficiency (Cµ) =

Message size Code size

No way inferior If 256 bytes of data (equals to 2K or 2048 bits) is sent under 32-bits CRC (a typical case used in Ethernet) the efficiency of CRC becomes equals to 98.5 percent. Lowest efficiency in Ethernet scheme is 92 percent. A parity bit per bytes is usually used for systematic parity code. Thus, the efficiency of this simple parity code is usually 89 percent. Thus, in terms of efficiency CRC is no way inferior to the simple parity. The above example also illustrates that in case of CRC while a total of 2080 bits are to be exchanged per 256 bytes in case of simple parity the figure is 2304 (= 2048 data + 256 parity bits) per 256 bytes. This is how transmission time is less in CRC. However, CRC is not free from limitation. 16 bits check can only have 216 = 65536 unique words. Therefore, there may be some extremely rare combination of errors to fool the users. However, the above noted four generator polynomial, G(x) can detect early : • All single bit errors (typically 100 percent error detection). • All double bit errors so long G(x) is a factor with at least three terms (typically 99.8 percent error detection). • Any odd number of errors so long G(x) contains a factor of (x + 1) (typically 99 percent error detection). • Any burst error whose length is less than the number of check bits and most of the larger burst errors.

XOR gate

XOR gate Shift register 1

2

3

4

5

6

7

8

9

10 11 12 13 14 15 16

XOR gate Message data input

Fig. 16 : CRC check bit after transmission of message data.

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Till today more attention was given on single-bit error correction code rather than multiple-bit error correction code. The reasons are many: If the bit error rate of the channel is 10-4, the probability of the single-bit error and the double-bit error per code word of size 7 bit are respectively 7 × 10–4 and 21 × 108. This means that multiple bit error is much less probable than the single bit error.

5.4.7 Basic BEC techniques BEC has three basic techniques. These are: Stop-and-Wait Automatic Repeat Request (S/W), Go-Back-N Automatic Repeat Request (GBN) and Selective Repeat Request (SRQ). In S/W ARQ technique, after transmitting a packet, (a packet is a code appended at both end with flags of start and end, source address and destination addresses and other control bytes), the transmitter waits for an acknowledgement from the receiver before transmitting the next packet. On receiving a packet, the receiver checks, using the error detection technique used in the process, for any error. If no error is found, the receiver sends an acknowledgement, known as positive acknowledgement (ACK), to the transmitter through the feedback path. On the other hand, if any error is detected, the receiver sends a negative acknowledgement (NAK) to the transmitter. On receiving an ACK for the already transmitted packet, the transmitter transmits the next packet. But on receiving a NAK for any transmitted packet, the transmitter retransmits the previously sent packet. In short, until and unless a packet is received correctly by the receiver and it is positively acknowledged, the transmitter will not transmit the next packet. However there remain several questions to such operations. What will happen if ACK or NAK is lost in the feedback path? The transmitter waits for a period known as time out period, which is greater than twice the propagation delay between the transmitter and the receiver, for the acknowledgement. If no acknowledgement is received within the time out period, the transmitter retransmits the previous packet. The receiver understands the received packet as retransmitted one by checking the sequence number of the packet and takes decision accordingly. What shall happen if ACK is changed to NAK or vice-versa during the transmission through the feedback path? The change of ACK to NAK is tackled by the same technique, as that is used in case of loss of acknowledgement. When NAK is changed to ACK, the receiver on checking sequence number only detects the change. By this time the previous transmitted packet for which NAK was changed to ACK is not available with the transmitter. This causes a serious problem. The performance of the techniques is measured by a parameter known as throughput efficiency (n). It is defined as number of the information bits correctly transmitted divided by the total number of bits transmitted for the purpose. If we assume (i) (m, n) code were used in the protocol. (ii) processing time at the transmitter and the receiver for ACK/NAK or packet is negligible, (iii) transmission time of ACK/NAK is negligible and (iv) feedback path is error free; ν(s/w) = n/{(m + RT)E} ...(13) where E = expected number of transmission for successful reception of a packet, R = rate of transmission, T = total round trip delay. When each packet has the same probability that it is received with error, E = 1/β ...(14) where β is the probability that a transmission for a given packet is the last transmission. If P and Pu are the probability that a packet is in error and the probability of the undetected packet error respectively,

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β = 1 – P – Pu = 1 – P, as Pu 5) packet of the block is the first negatively acknowledged packet when up to Nth packet has been transmitted, the transmitter will then discard first to fourth packets from its memory, and now will retransmit all the packets from 5th to Nth. Worst situation in GBN ARQ occurs, when the first packet of the block is negatively acknowledged, the case of when the whole block of N packets requires retransmission. Best situation occurs when none of the packet is negatively acknowledged, thereby successful transmission of N packets involves with minimum two propagation delay rather one packet being involved with two-propagation time as in S/W ARQ. This gives throughput advantage to the GBN ARQ over S/W ARQ. GBN ARQ may be two types: continuous and non continuous. In the continuous scheme, after transmission of a block of N packets, the transmitter does not have to wait for the acknowledgements of these packets before starting the transmission of the next block. In the non-continuous mode, before starting the transmission of the next block, the transmitter has to wait for the acknowledgements for the packets of the previous block. If the transmit time of a packet/acknowledgement is one unit, we have: N >= (1 + 2a) for the continuous scheme and N < (1 + 2a) for the non-continuous scheme. The throughput efficiency for GBN ARQ is given as: ν(gbn) = {n(1 – P)}/{m(1 + 2aP)} for continuous scheme, = (n/m){1 + NP/(1 – P)}–1 where N=1+T/(m/R)=1+2a, [Note that m/r is chosen so as to make N = 2,3,4…. of GBN technique. When T = m/R, N = 2 and transmitter goes back by two blocks.] ν(gbn) = {n . N(1 – P)}/{m(1+2a)(1 – P + NP)} for non-continuous scheme …(17) The through put of GBN ARQ technique is higher than that of the S/W ARQ but still the throughput is a function of propagation delay, a. Selective Repeat Request (SRQ) ARQ further improves the throughput. It operates like that of the GBN ARQ but retransmits only the packet for which negative acknowledgement is received. This means that theoretically infinite buffer is required at the transmitter. It has also two modes of operation, namely continuous and non-continuous.

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The throughput efficiency is given as: ν(srq) = {n(1 – P)}/m for continuous scheme ν(srq) = {n . N(1 – P)}/{m(1 + 2a)} for non-continuous scheme ...(18) The problems of the loss and/or the change of acknowledgements in GBN ARQ and in SRQ ARQ are tackled by the same techniques as in S/W ARQ. It is the trade off between buffer size and throughput that plays the role among the three basic BEC schemes, S/W ARQ, GBN. ARQ and SRQ ARQ. A comparison in terms of buffer size or memory requirements and throughput efficiency of the schemes is shown in Table (7). When the parameter “a” is zero i.e. tp (1 + 2a) : Here calculation of Tr is straightforward as below : Tr = Tt/(1 – p) But Tt = tt as propagation over large n when averaged get zero [that is, Tt = (Ntt + k.2tp)/N where k is number of error out of N. Value of k is very small compared to N]. So, Tr = tt/(1 – p) U = (1 – p) and nt = (n/m).(1 – p) = (n/m).U

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Case II : When N < ! (1 + 2a) : In this case, average number of transmission required for successful transmission of a code is : Nr = 1/N . Summation of {i . pi–1 (1 – p)} where i = 0 to infinity = 1/N(1 – p) and Tt = tt(1 + 2a) Thus, Tr = tt (1 + 2a)/N(1 – p) U = N(1 – p)/(1 + 2a) nt = (n/m).{N(1 – p)/(1 + 2a)} = (n/m).U Observations 1. When N < (1 + 2a), in all the cases it is seen that U and hence nt depend on N. Therefore the cases of N < (1 + 2a) are avoided in design. 2. When N = 1, as predicted go-back-N and selective repeat request become identical to stop and wait ARQ. 3. U and nt are practically measure same thing and directly related to each other through coding rate. For a fixed U, increased coding rate increases throughput (why?). Parameter N is known as window.

SOLVED PROBLEMS 1. For a communication system, a = 10–4 when Eb/N0 = 8.4 db. If data rate is 4800 bps, what is the required received signal level ? Assume effective noise temperature is equal to the room temperature. Solution. We know, N0 = kT where k = Boltzmann’s constant = 1.3803 × 10–23 J/oK T = temperature in oK = 290 oK as given. Thus : Where

Eb/N0 = S.Tb/N0 = (S/C)/kT Tb = bit duration = 1/C where C = bps.

In decibel notation, (Eb/N0)db = Sdbw – 10log10C – 10log10k – 10log10T or or

8.4 = Sdbw – 10log10 (4800) + 228.6dbw – 10log10(290) 8.4 = Sdbw – 36.81 + 228.6 – 24.62

or Sdbw = – 158.766. 2. A 4-bit message is transmitted via FSK and bit energy to noise density ratio (p) at the receiver is 12.32 db. (α =

1 2

e–p/2 )

(a) Compute the probability of single bit error rate. (b) If single parity is encoded, what is the probability of single bit error rate ? Assume equal transmitted power. Solution.

α= =

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1 2

e–(17.06)

10–4

since 12.32 db = 17.06


167

(a) Without coding : P1 = 4C1.10–4.(1 – 10–4)3 = 4 × 10–4 (b) With coding : As equal transmitted power is to be maintained (Eb/N0) is to be changed. (Eb/N0)new = (S/Cnew)/N0 = S/N0 . 4/5 . 1/C [Cnew = 5/4C] = Eb/N0 . 4/5 ∴ αnew = 1/2 . e–(17.06/2 × 4/5) = 0.5 × 10–3 Hence P1 = 5C1 . 5 × 10–4 . (1 – 5 × 10–4)4 = 25 × 10–4. 3. In the problem (2) it is seen that due to coding P1 is increasing. Then how does coding offer benefits. Solution. Due to coding all single bit errors may be detected, which is not the case without coding. In problem (2) if we transmit 25 x 104 bits while in case of no-coding there will be (25/4) = 6 bits undetected one bit error ; in case of coding there will be no undetected error. 4. Calculate the probability of message or word error for problem (2) which can not be detected. Solution. (a) Without coding : Pw = 1 – (1– α)4 = 1 – (1 – 4 × 10–4)4 = 1.6 × 10–3. (b) With coding : Only single bit error can be detected and others cannot be detected. So, Pw = Σ[5Ci (α)newi (1 – αnew)5 –‘i’ ] where ‘i’ = 2…..5 = 10.25 × 10–8 = 2.5 × 10–6 This is the answer to the problem (3) also. 5. A brute code is repetition code. For example in triple-repetition code 0 is coded as 000 while 1 is coded as 111. (a) Find the probability of undetected word error in case of detection only (b) Assume majority rule(that is you are using a majority gate for detection) for correction. Find the probability of undetected error that would result. Solution. (a) For detection only, any word other than 000 or 111 is a detected error. Single and double errors like 001 or 011 (when 000 was transmitted) are detectable. But triple errors like 111(when 000 was transmitted) are undetectable. Hence, Pw = 3C3 . (α)3 . (1 – α)0 = (α)3 where α = BER . (b) Majority rule means assumption that at least two bits out of three received bits are correct. Thus 110 and 001 may be corrected as 111 and 000 respectively. This rule may correct words with single error but double or triple errors would result undetected error.

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Thus undetected error probability : Pw = 3C2 . (α)2 . (1 – α) + 3C3 . (α)3 . (1– α)0 = 3(α)2 (1 – α) + (α)3. 6. In general, a repetition code may be expressed as (2t1+ t2, 1) code. For a double repetition code t2 = 0 and t1 = 1; For a triple repetition code t2 = t1 = 1. However as for even numbered repetition, majority rule is not applicable; it is often taken as that generalized repetition code by (2t + 1, 1). For this code, (a) Plot word correction probability v/s t as well as (b) Plot code efficiency v/s t. Comment on the plots. Assume (i) α = 10 – 4 when t = 0; (ii) α = 0.5 exp (– p/2) . Solution. α = 10–4 that is p = 17 Therefore, (αt = 0.5) exp(– 17/2 . 1/2 t + 1) (a) (2t + 1, 1) can correct up to t-bit errors probability of which is : PC = Summation of [2t + 1Ci . (αt)i . (1 – αt)2t + 1 – i ] where ‘i’ = 0……t Now why ‘i’ = 0, as no error is also corrected. Based on this equation we can plot PC v/s t. (b) The code efficiency ---- nC = 100/(2t + 1) % Based on this equation we can plot nC v/s t. 7. In a data communication system, on average 8 hrs. is used in for full transmission. Find the average daily likely range of bit errors. Line is 2400 bps. α = 10–4 Solution. Average daily transmission of bits over syste = 2400 × 3600 × 8. = 6.912 × 107 Mean = 6.912 × 107 × 10-4 = 6.912 × 103 Variance = 6.912 × 107 × 10–4 (1 – 10–4) = 6.911 × 103 Likely range is Mean ± 2(Var)1/2 = 6.912 × 103 ± 2 × 83.13 = 7078 to 6746 Thus we can expect up to 7078 to 6746 erroneous bits out of transmitted 6.912 × 107 bits. 8. The BCH (Bose, Choudheri, Hocquenghem) codes are for MBR correction. One (m, n) BCH code can correct up to e-tupple bit error where : e = (m – n)/p where p is the integer being related to m as m = 2p – 1. The code has minimum distance as : (2e + 1) = (1 + 2a)

Go-back-N N = Summation of Cim1 where i = 0 to e Moreover there are q bits per column. Then interleaving technique as discussed can correct burst error of length L where L (1 + 2a) as N = 7. Hence, nt = n/m(1 – p) = 973/1023 × (1 – 0.1) = 0.856 = 85.6% Example 1. Binary data is being sent over a channel where BER = 0.1. If bits are independent, what is the probability that a transmitted nibble “1110” is received as “0111”. Solution. Probability = P10 × P11 × P11 × P01 = 0.1 × (1 – 0.1) × (1 – 0.1) × 0.1 = 0.1 × 0.9 × 0.9 × 0.1 where (1 – 0.1) represents the probability that the bit is not in error. = 8.1 × 10–3. Example 2. Data is being sent at 10 MBPS over a link whose propagation delay is 10 micro sec. If the probability that the frame in error is zero, find the size of frame that will give 50 % link utilization for IARQ technique. Solution. Link utilization = (1 – p)/(1 + 2a) = 1/(1 + 2a) × 100 % or 50 = 100/(1 + 2.100/N) or N = 200 bits [since a = tp/tt = 10 × 10–t/(N/10 × 106)] If Pe = 0.1, then 0.1 = 1 – (1 – 0.001)N or 0.999N = 0.9 CASE III.

or If Pe = 0.5, then 0

N = (ln 0.9)/(ln 0.999) = 105.3 .999N = 0.5

or N = (ln 0.5)/(ln 0.999) = 692.869 Example. Data is to be transmitted between a source and destination in the following cases : (a) 1000 meter twisted pair wire having a transmission rate of 1200 bps. (b) 10 km of co-axial cable at 1 MBPS (c) 72000 km of satellite link at 10 MBPS Assume velocity of signal is 2 × 108 m/sec. Determine in each case how many bits, source shall transmit before the first bit arrives at the destination. Why is it so. Solution. CASE A. Tp = 100/(2 × 108) = 5 × 10–7 sec. The first bit shall reach receiver after Tp time from its transmission instant. So by this time, number of bits transmitted is 5 × 10–7 × 1200 = 6 × 10–4 = 0. CASE B.

Tp = (10 × 103)/(2 × 108) = 5 × 10–5 Number of bits = 5 × 10–5 × 106 = 50 bits

CASE C.

Tp = (72000 × 103)/(2 × 108) = 36 × 10–2 Number of bits = 36 × 10–2 × 10 × 106 = 360000 bits.

This is due to Tp . Reasons,

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In the above problem, we have used relations derived on the following assumptions : Except propagation and data transfer time, all other such as acknowledgement signal transfer, processing time, modem turn-around time etc. are zero. But if acknowledgement is of 16 bits, modem turn-around time is 75msec., what shall be the link utilization in case of Half Duplex and Full Duplex. In case of Half Duplex, U = tt(1 – p)/{tt + 2(tp + tmodem turn-around + tack)} = 0.1065(1 – 0.5)/{ 0.1065 + 2(5 × 10–6 + 75 × 10–3 + 16/3600)} In case of Full Duplex : U = tt (1 – p)/{tt + 2(tp + tack)} as modem turn-around will be zero. BOX 6 Error Control in Networks A Short Questions 1. What do you mean by bit error? When a transmitted bit is received erroneously at the receiver, bit error is said to have caused. If the data is binary, bit error occurs if receiver detects a “1”, when transmitter has transmitted actually a “0” and vice-versa. 2. Who causes bit error? Different types of noises are responsible for bit error. They include Channel noise like Gaussian noise, system noise like thermal noise, fading, lighting, man-made noise produced by making On and Off of heavy electrical machines. 3. What are the different types of but error and who cause them? Different types of bit error are : random bit error and burst error. Burst error is also known as correlated error. Random error occurs when errors are distributed randomly over the message or word. Following is an example of a random error: Transmitted bytes : 11001110 Received bytes with random error at third and eight positions from left : 11101111. Random error can be again of different types: SBR (Single bit error), DEBR (Double bit error)… and MBR (Multiple Bit Error). SBR, DBR and NBR respectively mean one bit in error per word, two random buts are in error per word, and multiple random bits are in error per word. The earlier example of random error is an example of DBR. Burst error means a sequence of adjacent bits of the message or the word are in error. Error are clustered together in the message or the word. The following is an example of burst error : Transmitted bytes : 11001110 Received bytes with burst error of burst length 3 : 11010010 (4th, 5th, and 6th bits from left are in error). Random error is usually caused by channel noise like Gaussian white noise. Burst error is caused mainly by system’s component failure, on-off switching of heavy electrical machines like motors, impulse noise produced by lightening, failure of radio transmission system due to rapid fading. But causes of burst may also cause random error and vice-versa.

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4. Define BER (Bit Error Rate) or bit error probability? Bit Error Rate or probability is defined as : BER = Lim ( η / N) N→α

where η is the number of bit in error out of total transmitted N bits. 5. If BER = 10–3, how many bits shall be in error out of the total transmitted bits of 106? BER = 10–3 = 1/103, which in turn means that on average there shall be 1 bit in error out of 103 transmitted bits. Therefore for a total transmitted bits of 106, there shall be 103 bits in error. The generalized formula for calculation expected number of bits in error out of a given number of transmitted bits and for a given BER is : Bits in error = BER × Total Number of transmitted bits. 6. What is the controlling parameter of bit error? Does bit error rate depend on the data or the bit transmission rate? Eb/No which means that the ratio of energy per bit to noise density of the channel is the controlling parameter of bit error. As this ratio increases, bit error rate decreases. For given single power, S, we have Eb = S.Tb = S/C where Tb is the bit duration and C is the transmission bit rate. For a given signal power and given channel, as C increases, Eb/No decreases and this in turn increases bit error rate. We can also explain the same physically. As data transmission rate (C) increases, the spacing between consecutive data decreases, that is overlapping of spectrum of data becomes more. This causes more error. 7. If BER = 10–4 and bits are being transmitted at a rate of 1 MBPS, what shall be the average bit error per second ? Average bit error per second = BER × bit transmission rate = 10–4 × 1MBPS = 100 bits per second. 8. If the average packet size is 2048 bytes per packet and BER = 10–4, what is average bits in error per packet? Average bits in error per packet = BER × Average packet size in bits = 10–4 × (2048 × 8) = 1.638 per packet. 9. What are the different classes of probabilities associated with a received frame in terms of error? What is the probability that a frame is in error? There are three distinct probabilities : (1) frame received without any error (P1), (2) frame is received with detected/corrected one or more bit errors, but with no undetected bit errors (P2), (3) frame is received with one or more undetected bit errors (P3). Here P1 + P2 + P3 = 1 The total probability that a frame is in error = P2 + P3 = 1 – P1 10. What are the different probabilities of question (9) in terms of BER if no means are taken to detect/correct error? Assume that BER is constant and independent of bit position. Frame size is of N bits. Examine the result. P1 = (1 – BER)N ; P2 = 0; P3 = 1 – P1 We see that as frame size (N) increases, probability of receiving frame without any error (P1) decreases; thereby causing the probability of receiving frame with undetected error (P3) to increase.

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11. What is the probability, whether you use any coding or not, in terms of BER, that a frame is in error if the frame size is of N bits? Probability = 1 – (1 – BER)N . But BER of a coded system is slightly higher than that of the uncoded system. 12. Pe is the BER of binary data transmission, what shall be the BER of S-array data transmission? Comment on your answer. BER for S-array data transmission shall be approximately S times of Pe, i.e. SPe. Binary data transmission is less error prone. 13. What is the common unit of Eb/No? It is db. 14. What is appropriate probability model of bit error? Why? Give the model. The most appropriate prob+ability model of bit error from data/computer communication and networking point of view is binomial probability model. Data errors are discrete in nature. Two discrete probability functions are : binomial and Poisson. Discrete variable may be real as well integer in value. In bit error, error must be discrete and integer in value like 02 bits in error and not like ¹.4 bits in error. Binomial function deals with integer valued discrete function. Therefore it is appropriate model of bit error. Probability that there is i errors in a code or message of size m (i n) bits where number of check bits, c = m – n. The code is then called (m, n) code. 17. Classify different types of error correction/ error detection codes with example. Codes can be classify into basic two classes, viz, block codes, and convolution codes. Block codes again can be of two types: linear block code and non-linear block codes. Examples of linear codes are : Parity codes like Systematic and non-systematic parity codes. Hamming (7, 4) single bit error correction codes, Hamming (13, 8) single bit error correction code, Golay codes, BCH codes etc. Examples of non-linear codes are CRCs (Cyclic Redundancy Codes) like CRC-12, CRC-16, CRC-CCITT etc. 18. How the different error correction/detection codes are related to different types of bit errors? Linear block codes are usually used for correcting/detecting random bit errors. They can be used to correct/detect burst error, but in that case either the technique used (example : interleaving technique) shall be quite complex or the code be less efficient.

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Convolution code is used when making blocks out of a message is difficult, viz, satellite communication. 19. How to find the number of check bits of any linear block or parity code? The minimum number of check bits© required to correct dingle, double …. Upto e-tupple bit errors per code is given by :L e

2c > =

∑

n+c

Ci

i=0

Where n + cCi = (n + c) !/(n + c – i) ! i ! and n is the message size. Therefore if n = 4, to correct upto only single bit error per code minimum required check bits, c shall be 3. That is why one single bit error correction code is Hamming (7, 4) code, 7 is coming in adding message size (4) with number (3) of check bits. 20. Why do we always use minimum check bits? As check bits increase, bandwidth required to transfer message increases. The amount (IBW) by which the required band-width increases is directly proportional to the number of check bits: IBW = Required increases band-width – Original required band-width = [BW (n + c)/n] – BW. = BW (c/n). Where BW is the bandwidth required originally to transfer message of n bits without coding, and c is the check bits used in coding. 21. What is coding rate? For any code (m, n) , coding rate or coding efficiency is defined as: Coding rate = n/m = n/n + c where c is the check bit (i . E . m = n + c). Coding rate or coding efficiency is often specified in % and in that case it is (n/m) × 100. Coding rate or coding efficiency is related to the increases bandwidth required to transfer a message with coding that we discussed in question (20). Required increases bandwidth is inversely proportional to coding rate : Required increased bandwidth = BW (1/coding rate). 22. Is there any relation between coding rate and code capability? Yes, there is a relation between them. As check bits increases, the capability of the code increases. But with increases of check bits, the coding rate decreases which in turn requires increased bandwidth for transmission of the code… However, we can illustrate this feature with an example. Say we are using repetition code, (2t + 1, 1). In this case, probability of bit error shall be : 2t + 1

Pe =

∑

2t + 1

C i a i (1 − α) (2 t + 1 − i)

i=t+1

Using this, we can have following table: Code rate (1/2t + 1)

Probability of codeword in error (BER = 10–2)

1 1/3 1/5 1/7

10–2 3 × 10–4 10–6 4 × 10–7

1/9

10–8

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23.

24.

25.

26. 27.

28.

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As number of check bits increases, coding rate decreases but improvement in error performance increases. Why do we prefer to higher block size for coding? We have already defined coding rate or coding efficiency. This rate or efficiency can tend to 1 or 100% only when n tends to infinite (as n tends to infinite, n/(n + c) tends to 1). This is the reason that we prefer to higher size of message or packet or data block. But there is a trade-off. As we increases n, probability that a frame is in error increases. What is the relation among uncoded bit rate, code rate and coded bit rate? Such relation exists if we assume that codeword with m bits must be transmitted within the time required to transmit message with n bits. If Tb and Tc are respectively the bit duration in case uncoded and coded cases, it is required that : mTb = nTc : or fb/fc = n/m or fb/fc = coding rate, where : fb = 1/Tb, is known as uncoded bit rate, and fc = 1/Tc, is known as coded bit rate. What is throughput of a code? It is the probability of code acceptance. Probability of code acceptance is nothing but the probability that a code word is received with no error. If BER is the bit error rate when (m, n) code is being used, the throughput of the code is (1 – BER)m provided bit error is independent of bit position. How can you increase the throughput of a code? One solution is decreasing size of code i.e. decreasing m or n of any (m, n) code. Is the solution suggested in question (26) feasible? Not always. Decreasing m or n will cause false alarm (FA) as then it will increase the probability of one possible code word will be converted into another acceptable code word. FA is a case when code word is received with undetected error. What is coding gain? The coding gain is the reduction in the single to noise ratio or energy per bit to noise density ratio permitted by coding. Say, we achieve some reduced frame error probability using a (m, n) code. If the same reduced frame error probability were achieved, by increasing signal to noise ratio by 3 db in uncoded system, we can call the coding gain of (m, n) code as 3 db.

29. What are Hamming Codes? The Hamming Codes are linear block codes with following properties : Code size = 2P – 1 Message size = 2P – P – 1, and the number of check or parity bits = P where P > = 3. 30. Define minimum Hamming Distance. How dose it determine the capability of a parity code? If wi denotes the weight of the it h code word, minimum Hamming Distance, dmin is defined as: dmin = min(wi) of all possible i. The weight of a binary code word is the number of “1” present in the that code word. For example if a code word is 1110001, the weight of this code word is 4. Smallest non-zero weight of all weights of any code is the Hamming Distance of that code.

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32.

33.

34.

35.


For Hamming (7, 4) code, number of the code words = 24 = 16 and hence number of weights = 16. Out of these 16 weights, smallest non-zero weight is 3 which is the Hamming Distance of this code. Hamming Distance is related to the capability of a code in the following means: 1. For a code to detect upto e bits per code word, required minimum Hamming Distance > = (e + 1). 2. For a code to correct upto e errors per code word, required minimum Hamming Distance > = (2e + 1). 3. For a code to correct upto .1 errors and detect upto e2 (e2 > e1) per code word, required minimum Hamming Distance > = (e1 + e2 + 1). 4. For a code to correct upto e1 bit errors and detect upto e2 bit errors per code word, required minimum Hamming Distance > = (2e1 + e2 + 1). Dose coding increase BER? If so, why do we then use coding? Coding duly increases BER, If the transmitter power is S, this power is shared by n bits of message in case of uncoded system; and the same power is shared by n + c (where c is the number of check bits) bits in case of coded system in the same time interval as of the uncoded system. Naturally energy per bit decreases in case of coded system, and this causes BER to increase. Energy per bit equals to signal power divided by bit rate. We have seen earlier the relation between uncoded bit rate and coded bit rate which tells fb/ fc = coding rate. Energy per bit in case of uncoded system, Eb = S/fb and that in case of coded system, E = S/fc. Hence Ec/Eb = fb/fc = coding rate < 1. This means Ec < Eb. Therefore BER in case of coding is greater than that of the uncoded system. Even then we use coding because by coding we shall be able to detect/correct bit errors to some large extent. Detection/correction rate of bit errors is more than the amount of bit errors caused by increased amount of BER due to coding. So, we use coding. What is repetition code? In repetition code, bit to be transmitted is transmitted more than once. In triple repetition code “0” and “1” are coded respectively as “000” and “111”. At the receiver majority rule is applied to decide about the bit. In general repetition code can be expressed as : (2t1 + t2 , 1) where t2 = 0 and t1 = 1 for a double-repetition code; and t1 = t2 = 1 for a triple repetition code. However, double or even repetition code can not use majority rule for decoding or detection, and that is why repetition code is usually a odd repetition code. A generalized odd repetition code is represented as (2t + 1, 1) where t = 1, 2, 3, ……… What is shortened code? A shortened code is made by deleting any number of message bits from any block code (m, n). If e bits are deleted, the shortened code becomes (m – e, n – e) code. For example by deleting one bit of message, Hamming (7, 4) code could be converted to a (6, 3) code. In (6, 3) code, minimum distance is three, therefore it shall able to correct single bit error like (7, 4) code. Define the capability of (13, 8) Hamming code. (13, 8) Hamming code can correct upto one bit error and can detect upto two bits error per code word. How Hamming (7, 4) code be used to detect upto two bits error per code word. By adding an extra check bits i.e. by a code (8, 4), the same can be achieved. The extra check bit shall be used to provide parity over 7 bit codes of the (7, 4) code.

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36. What is BCH? What is the lower bound on the error correcting capabilities of BCH codes? BCH is Bose-Chaudhuri-Hoequenghem code. It is the generalized parity code. Lower bound (e) on error correcting capabilities of a BCH (m, n) code is given as: e > = (m – n)/log2 (m + 1) = (1 – r)n/log2(n + 1) where r is the coding rate. 37. In general how the error correcting capabilities of BCH code can be defined? Hence define the limits of the distance of the code. A BCH (m, n) code can correct upto e bit errors per code word where: e = (m – n)/p = (1 – r)n/p where p is an integer related to m such that m = 2p – 1 and n >= (m – pe). The code has minimum distance as : (2e + 1) M) bit checksum can have only 2M, where 2M > 2N, different and unique fingerprints. This can be minimized using longer generator polynomials, but then coding rate shall go on decreasing. The probability that there can be single unique fingerprint per individual code word is 2M/2N, and therefore the probability that there shall not be unique fingerprint for individual data or there can be same fingerprint for more than one data is 1 – (2/2N). 60. Why CRC is more efficient than even simple parity? It can be illustrated with a simple. Say, we are using simple one bit parity each for a byte. Therefore its coding efficiency is (8/9) X 100 = 90% and the percentage of extra bits is (1/8) X 100 = 12.5%. We can use only 32 check bits of a CRC-32 code over a data of size 2048 bytes. In this case coding efficiency is { (32 + 2048 X 8) / (2048 X 8) } 100 = 99 % and percentage of extra bits is {32 / (2048 X 8)} X 100 = 1.6%. In addition, capability of CRC32 is far more than simple parity.

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61. How a circuit is made to generate checksum or FCS, for, say CRC 16? Step 1. CRC 16 = X16 + X15 + X2 + X0. Take a shift register of size equals to the highest power of given generator polynomial. In this case, shift register shall be of 16 bits. Step 2. Subtract the powers of X of given generator polynomial from 16 (size of the shift register as find in step 1), and note the results as different positions. In this given case, the positions are 0 (which is due to 16 – 16), 1 (which is due to 16 – 15), 14 (which is due to 16 – 2), and 16 (which is due to 16 – 0). Step 3. Use XOR gates at positions found at Step 2. as shown in the following Fig. (1). Before transmitting data, Clear the shift register. After that transmit the transmit the data as shown. When all data bits are transmitted, what is left in the shift register is nothing but FCS. 1

14

16

Shift register 2

3

4

5

6

7

8

9

10 11 12 13 14 15 16

Checksum

Input data

62. What theory gave the birth of CRC? It is algebra theory. 63. What CRC is used in Local Area Networks of IEEE 802 series? What is the size of check bits or FCS (Frame Check Sequence)? CRC-32 code is used in LANs of IEEE series. Number of check bits or size of FCS of CRC-32 is 32 only. 64. What CRC is used in link level of SS7 signaling, LAP B and LAP D protocols? In all these cases CRC-CCITT code is used. 65. What is concatenated code? Concatenated code is a coding technique which uses more than one code. For example, for a given message block of n bits we can a error correcting code (m, n). Now the code of m bits so generated shall be used as a message block to generate a new code say, (M, m). Concatenated code improves the performance of coding. For example if two CRCs are operated on a message as illustrated above, detection of burst error of length may go upto the sum of lengths of individual CRCs. 66. What shall be the coding rate of the concatenated code? It shall be equal to the product of individual code rates. 67. What is maximum length code? For any positive integer P > 3, a maximum length code is a code that following: Code size m = 2P – 1, Message size = P, and Minimum distance = 2P – 1 These codes use generator polynomial (1 + Xn)/h(X) where h(X) is any primitive polynomial of degree P. 68. What is convolution code? Where is it used? In block code, a fixed message size is required. Coding is done over this message block. There are situations where message come in form of serial bits rather than block, viz,

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70. 71.

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73.

74.

75.


satellite communication. In such situations, convolution coding is used. Convolution coding operates serially on incoming message continuously. An encoder of a binary convolution code with rate 1/m, produces a coded output sequence of length m(L + N) bits corresponding to a N bit message sequence using a finite state machine that consists of 1 shift registers. Coding rate is then, N/m (L + N) bits/symbol. Usually N > > L, hence coding rate is 1/m bits/symbol. Give the advantage and disadvantage of convolution in general. Convolution code in general more effective and simpler than block coding. The code has advantage : (1) codes are applied to channel of fixed bandwidth, (2) in general, code has better performance than block code for equivalent circuit complexity, and (3) decoding algorithm in case of convolution coding can be adopted in compliance with data source statistics. The disadvantage of the convolution coding is that code design with bit error approaching zero fairly depend on a random coding argument rather than on a specific construct as in block codes. Name a few decoding algorithm used in convolution codes. Code tree algorithm, State and Trellis diagram, Viterbi algorithm are to name a few. What for convolution is used? Convolution coding in general used for error correction, where block codes are used in error detection, in general. State a practical concatenated code that makes use of both block code and convolution code. In fact, concatenated or hybrid code that is mostly used in practice, uses both block code and convolution code. The given message is first encoded using block code, the block coding thereby being known as inner code. The resulting block code then is encoded using convolution code, whereby convolution code is outer code. This combination provides an excellent error correction and detection capability. What are the basic two error control techniques? How are they related to error detection/correction coding? Basic two error control techniques are : Forward Error Correction (FEC) technique, and Backward Error Correction (BEC) technique. In FEC technique, error is corrected at the receiver; and therefore Error Correction Code is must for it. In BEC technique if error is detected at the receiver, receiver asks transmitter to retransmit the data: and this way the error correction is made with. Thus, in BEC error detection code is sufficient. What error control technique is used in practice in data/computer communication and networking? Why? We have seen CRC is the best code for networks due to a number of reasons. CRC being a error detection code, BEC is the usual technique of error control in networks. Beside, error correction code requires much more check bits than error detection code which tills the decision in favor of BEC. What are different techniques of BEC technique? Do they require some memory storage at the transmitter, exclusively for error control? Different techniques are : (1) Stop and Wait Automatic Repeat Request (ARQ), (2) GoBack-N Automatic Repeat Request (Go-Back-NARQ) and (3) Selective Repeat Request. ARQ dose not, as such, require any memory for error control. Go-Back-N ARQ requires about N × D (where D is the data/packet size in bytes) bytes of memory locations. Theoretically, Selective Repeat Request requires infinite memory location.

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76. If acknowledgement is lost in different ARQ techniques, how the error control operation shall be successful? Transmitter in ARQ techniques waits for acknowledgement from the receiver before taking subsequent appropriate action. But if the acknowledgement is not reached due to loss etc within a time, known as time out period, after transmission of a frame or a set of frames the transmitter assumes that the transmitted frame or frames have not reached the receiver correctly. The transmitted, then starts retransmission. 77. If the transmitter retransmits frames after a time out period in case of loss a positive acknowledgement (signifying that frame has been correctly received by the receiver) from the receiver, how the receiver shall identify the retransmitted frame as retransmitted frame rather than next frame? The receiver identifies retransmission frame from the next frame by the sequence number of the frames, which is attached by the transmitter as a header to each frame during transmission. 78. At what value of N, both selective repeat and Go-Back-N-ARQ shall reduce to ARQ? When N = 1. 79. What technique is mostly used for error control in networks? Why? Go-Back-N ARQ is mostly used. It requires moderate memory which is much less than that is required in selective repeat request. Besides its throughput efficiency and link utilization are better than those of IARQ, although they are less than those of selective repeat request. 80. Name a few data link protocol where Go-Back-N ARQ is used? SS7 signaling, LAP B, LAP D and X.25 are to name of few. 81. What is the value of N used in Go-Back-N ARQ technique used in ARPANET? It is 127. 82. Define link utilization as used in case of error control? Link utilization measures the efficiency of utilization of available link capacity. It may be defined as the ratio of transmission time required by a transmitter to transmit a frame to the time before the next frame could be transmitted. It can also be defined as a ratio of transmission time of a frame to the actual average time spent in successful transmission of the frame under a particular error control technique. 83. What is the throughput efficiency as defined in case of error control? Throughput efficiency of an error control technique is defined as the product of link utilization of the technique and coding rate of the code used in that error control technique. 84. Compare throughput efficiencies of different error control techniques. Throughput efficiency of the selective repeat request > that of the Go-Back-N ARQ > that of the ARQ. 85. Define parameter “a” commonly used in analysis of link utilization etc? What dose it signify? “a” is defined as the ratio of the propagation time in between source and destination to the transmission time of the frame. Significance is as below: 1. If “a” is less than 1, the round-trip delay between source and destination is mainly due to the transmission delay.

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90.

91. 92. 93. 94.


2. If “a” is equal to 1, the round-trip delay is determined by both the propagation delay and transmission delay. 3. If “a” is grater than 1, the round-trip delay is mainly determined by the propagation delay. When is the maximum link utilization achieved? It is achieved when probability of the frame in error is zero. (A) If the maximum link utilization is attained when probability that the frame is in error is zero, it then natural that utilization shall be 100%. Dose it happen so in all cases of error control technique? NO, it dose not happen in all the cases. It happens in two cases : (1) Go-Back-N ARQ with continuous operation and (¹) selective repeat request with continuous operation. In all other cases, namely, ARQ, Go-Back-N ARQ with discontinuous operation and selective repeat request with discontinuous operation, the utilization is not 100% even in case of zero probability that a frame is in error. (B) What is problem with discontinuous operation? What is the maximum link utilization attainable in case of ARQ? Why is it so? The maximum attainable link utilization in case of ARQ is 1/(1+2a), (provided processing time, acknowledgement time and modem turn-around time etc negligible) which is less than 1 i.e. 100%. The maximum utilization is achieved when the probability that a frame in error is zero. Even then, as per IARQ technique transmitter has to wait for a positive acknowledgement from the receiver for each frame. This makes additional loss of twice propagation time per frame. This contributes to make utilization less than 100% What is piggybacking? This is a technique of sending acknowledgement with data. Receiver sends acknowledgement (positive or negative) by the sequence number of received frame. Instead of sending acknowledgement sequence number separately, receiver sends the same with its piggybacked frame which consists of data field, a field reserved for data sequence number, and a field reserved for acknowledgement sequence number along with others. By this, communication resource is utilized in a better way. What will happen to a piggybacked frame, if a receiver has only data to send and no acknowledgement? The receiver shall send piggybacked frame where the field of acknowledgement sequence number will be filled with previous number. This previous number when reaches at the other end will be simple ignored. What will happen if there is only acknowledgement, and no data to send? Receiver has to send a separate acknowledgement frame. Is piggybacking used in Go-Back-N-ARQ? Yes, it is used duly. In a Go-Back-N-ARQ, if a frame is received out of sequence, what is done? Frame is simple discarded. What is mode of sending acknowledgement sequence number in Go-Back-N-ARQ? If the sequence space is k bits, sequence number can range from 0 to 2k – 1. If k = 3, sequence number ranges from 0 to 7. Sequence numbering is done with modulo 2k operation. This means after sequence number 2k – 1, the next sequence number starts at 0 and proceeds towards 2k – 1.

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95. What is maximum window size in Go-Back-N-ARQ technique? If the sequence space is k bits, the maximum window size is governed by: Window size < = 2k – 1. For k = 3, therefore maximum window size is 7. 96. What can be maximum window size in case of selective repeat request? Maximum window size is half of the sequence space. If sequence space is k bits, maximum window size is 2 k – 1. If k = 3, this maximum is 4. 97. Can we use ARQ in satellite communication? The use of ARQ in long distance communication could be unacceptable. Long distance communication means long propagation delay, and that means a very high value of “a”. We have seen earlier chat the propagation delay or for that purpose the parameter “a” limits the link utilization to a maximum of 1/(1 + 2a) when even probability of frame in error is zero. With increase of “a”, the utilization decreases. Therefore, ARQ is not suitable for long distance communication like satellite communication. 98. If the channel is Gaussian, what shall be the limit of the ratio of energy per bit to noise density, in order to communicate at the full channel capacity under Shannon limit? Eb/No > = – 1.6 bd. 99. What is 74LS636 IC? What is 8004 IC? 74LS636 is a complete error detection and correction chip for (13, 8) Hamming Code. The chip is of Texas Instruments. 8004 IC is a dual CRC-32 SDLC generator and checker. 100. (A) Different types of Errors occur in frame as below : In case of synchronous communication: (1) invalid frame, (2) Abort, (3) Overrun and (4) FCS error; and in case of asynchronous communication : (1) framing error, (2) overrun error, and (3) parity error. By error control, what error of the above do we reduce ? (B) In data/computer communication, we have three important aspects: Security, Accuracy and Privacy, which are controlled by coding. By error coding, which of these we are controlling? (A) By error control, we control FCS error and parity error. (B) By error correction and/or detection coding, we control accuracy. 101.Write a critical note on Turbo Code Turbo Code-An Error Correcting Code for Next Generation The coming up knowledge age like the present age of information will vastly and mainly depend on how reliably, securely and effectively the talking computers (the computers connected in the networks) exchange data, information and knowledge among themselves. The data, information and knowledge the talking computers exchange among them are represented in the form of a string of binary logical bits, 1s and 0s. The reliable transport means the sender’s bits are transported accurately over the network and are correctly received by the receiver. The message may be incorrectly received due to bit errors. The bit error occurs when a transmitted 1 is received as 0 or a transmitted 0 is received as 1. The transformation of bit either from 1 to 0 or from 0 to 1 is caused by noise signals in the channel or in the system. The logical bits are transported over networks as physical signals, the simplest example of which is: logical 1 is represented by a positive pulse of 5 volts or more and logical 0 by negative pulse

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of – 5 volt or less. The noise signals when are contaminated with physical pulses, a positive pulse may convert to a negative pulse and vice versa. This is how bit errors are resulted in. The different methods are used to combat with the bit errors. The well-known two methods are Forward Error Control (FEC) and Backward Error Control (BEC). FEC uses error correction code. Error correction code is capable of correcting error. Thus the error is corrected at the forwarding devices, the receivers. FEC is better suited to the environment where the bit error rate is higher. The environment of wireless communication and network is such an example. For the environment of moderate to low bit error rate the example of which is the current wired based network, the technique of BEC is used. BEC makes use of error detection codes. Error detection codes can only detect errors but can’t correct errors. Thus the receivers can’t correct errors. Once the errors are detected, the receivers ask the transmitters for retransmission of message received but detected in error. This is how BEC corrects error. Both the error correction codes and the error detection codes are based on the philosophy of “redundancy increases reliability.” For example, if you post same letters twice to your friend, you become more confident that at least one of the two letters posted, should reach your friend even if another is lost and/or misplaced or damaged. In the example, one more letter (redundant) increases the probability of your friend getting the letter, that is, the reliability of reaching the letter to your friend increases. But the increased reliability is achieved at the higher cost of post, because the required postal stamps for two letters will be double that of the one letter. A simple example may further clarify the concept. Say you want to send two data, x and y to your friend. In addition to original data x and y, you may send two more redundant data as x + y and x – y. So your code now: x, y, x + y and x – y. You transmit your code to your friend. Say error occurs at y, and due to the error your friend received the code as: x, z, x + y and x – y. Your friend from received x + y and x – y can compute x and y. On comparing computed x with received x and computed y with received z, he can easily detect the occurrence of error. Similarly the error correction codes and the error detection codes use redundant bits, known as check bits. Examples of error detection source codes among others are parity code and CRC-32 code that use respectively 1 and 32 check bits. Examples of error correcting source codes among others are (7,3) Hamming code, Repetition code and BCH codes. The redundant check bits are used to correct or detect error but they result in the increased cost of transmission by consuming more bandwidth like the increased cost of postal mailing of redundant letters. The increased reliability of data transmission due to the application of the correction code or error detection Step 1 :

Data

Encoder 1 (Convolution encoder) Step 2 : Interleaver

1 0 1 0 0 1

Step 4 : (Parity 1) Step 3 : Encoder 2 (Convolution encoder

0 1

Function of Interleaver : An example Transmitter Side

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Puncture. Example : data bits followed (Parity 2) parity bits of encoder 1 and encoder 2


Step 6 : Data received :

Step 7 :

Step 5 :

Data bits as received

Decoder 1 (Convolution decoder)

Parity 1 bits as received

Interleaver

De interleaver Decoder 2 (Convolution decoder)

189

De-puncture

Parity 1 as received

Receiver Side (Turbo Decoder) Fig. 1: Illustration of working principle of turbo codes

code has a trade off with increased consumption of bandwidth for transmission. The error correction code uses more check bits than that of the error detection code. Thus till date the application of the error detection codes has outnumbered the application of the error correction codes. But with the appealing flexibility and other benefits, the coming age of networking has widely titled towards the wireless networking. This has made necessary for a search of powerful error correction code. The famous Turbo Code is ahead in the race in this regard as FEC line code. Turbo code has out scored. As we had seen code is made of original data bits and a set of extra bits called check bits. Turbo code does follow the same concept but with a great innovation. Turbo code is a combination of two simple recursive convolution codes (Fig. 1) in parallel. Thus turbo code may be called PCCC (Parallel Concatenated Convolution Code). Every code has a rate. Rate is the number of the information bits divided by the number of the information bits plus that of the parity bits used in code. In the turbo code, each of the two encoders produce the number of parity bits equal to the number of the information bits. The total parity bits are twice the number of information bits. Thus the turbo code is usually a 1/3 rate coder. The input data is a block of data of n information bits. Actually turbo encoder starts with three copies of input information bits. One copy directly goes to the puncture unit. Second copy goes to first encoder. Third copy goes to the interleaver before it goes to encoder 2. One out of two encoders of the turbo code generator generates parity bits based on input bits. The real innovation in the turbo code is the use of an interleaver. The interleaver allows a permutation of the n information bits before they are input to the second encoder. The second encoder generates parity bits based on the interleaved bits of the original sequence. Even though the individual groups of the parity bits from two respective encoders produce each a weak code, yet the combination becomes a powerful code. Logically the combination produces increased redundant bits thereby increasing reliability. The increased redundancy can be achieved by just duplicating encoder, but that will not produce any increased redundancy in effect. Only the interleaver innovates increase redundancy. How? Let us take a very simple example. If x and y are the two data, we can have an encoder to produce two redundant check data, x + y and x –y. If we duplicate the encoder, we may get one more set of x + y and x – y. But that does not produce any unique redundant data. Had the original data were interleaved as y and x before inputting to the second encoder, the redundant data would have been y + x and y – x. So by interleaving three unique redundant data are produced, x + y, x – y and y – x (note that x + y and y + x are same and one, but x – y and y – x are not same and one so long they are not equal).

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Really, the interleaver makes the turbo code what it is today. The information bits and the two groups of the parity bits from two encoders of the turbo encoder are punctured before releasing to the link. A few examples of puncture are: (a) information bits followed by parity bits of encoder 1 and then parity bits of encoder 2; and (b) first bit of information followed by first bit of parity bits of encoder 1 followed by first bit of parity bits of encoder 2…. the pattern repeats for other bits. In the link the logical bits propagate as physical electromagnetic or optical signal as per the media of the link. The link noise contaminates with the physical signals of the bits resulting bit errors. The received signals are used in the turbo decoder at the receiver to recover the corrected bits. The decoder at the receiver performs the complimentary functions of the turbo encoder as shown in fig(1). In order to finally correct the errors, the two decoders exchange their individually assessed data repeatedly. After a number of exchanges or iterations, around 5 to 10, the decoders may reach an agreed decode data as a fully corrected data. The operation may be illustrated in a very simple way with our previous example of x and y respectively as 7 and 5: Turbo Encoder Turbo Decoder #Data bis = 7, 5 #Encoder 1 produces parity bits = 12, 2 #Interleaved data = 5, 7 #Encoder 2 produces parity bits = 12, –2 #Puncture data = 7, 5, 12, 2, 12, –2 which is released in link

#Say data received as 7, 6, 12, 2, 12, –2 (note that data 5 is in error and is received as 6) #Decoder 1 receives data set as = 7, 6, 12, 2 ; and say it decodes data as 7, 6 assuming wrongly received data 2 as error. Then it sends decoded data 7, 6 to decoder 2 #Decoder 2 receives data set as = 7, 6, 12, – 2 ; and say it decodes data as 7, 5. Then it sends decoded data top decoder 2 #Decoder 1 getting the data 7, 5 from decoder 2 verifies that this data set conforms to its received parity 12, 2. So it agrees to 7, 5 Agreed corrected data = 7, 5

(THIS IS ONLY A TOO SIMPLIFIED ILLUSTRATION. IN ACTUAL CASE, agreed corrected data is achieved after a number of iterations) The existing error correction codes namely Hamming Codes, BCH codes, Reed Solomon code and Convolution codes are widely used. They are very powerful codes too. So what is the great in Turbo code? How is it different from other codes? The problem basically lies with cost and complexity required to decode the data. As was pointed out earlier, the capacity of a code in terms of error correction and/or detection increases by increasing the check bits. As check bits increase, the code size increase that necessarily results in higher data rate. With data rate the bit error rate increase. So the problem and its solution are in trade off and counter challenging to each other as if in conformity to the basic natural law of the Newton that to every action there is an equal and opposite reaction. Thus obvious solution of the trade off problem is to have an infinitely long code word that was duly demonstrated by Shannon, the father of the information theory. But it is only theoretical answer. It was Claude Borrow and Alain Glavieux, two French Professors by their invention known as turbo code implemented practically the idea of Shannon. Two encoders of turbo encoder actually generate two unique codes (in our example considered earlier two codes x, y, x + y, x – y and x, y, x + y, y – x) each of 2n bits (if information is n bits, check bits are n bits) for a single but transmit not 2 × 2n = 4n bits but 3n

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bits. This is the beauty, and it is possible only with interleaver of the turbo code. This is how turbo code solves the cost and the complexity problem of existing codes as data or code size increases. Turbo codes are very powerful FEC. The code has viable application in high bit error rate links like wireless environments and/or in the power limited noisy environments like satellite links, deep space communication, space communication, military hand held satellite radio communication. The turbo code is believed to be effectively applied in Wireless LAN (Local Area Network), WLL (Wireless Local Loop), CDMA (Code Division Multiple Access), 3G/4G (Third Generation/Fourth Generation) and PCS (Personal Communication Services) that require codes to handle bit error rate ranging from as low as 10–5 to as high as 10–2. A bit error rate of 10–2 means on average out of 102 transmitted bits 1 bit will be in error. In the satellite and the deep space communication, the transmitted power is limited due to battery operation. This causes the system to have low signal to noise ratio resulting high bit error rate. The turbo code is suitable for these applications. High data rate is another source of bit error. As data rate increases bit error rate increases. Turbo code effectively maximizes the data rate. The data rate is limited by the famous Shannon’s theory. It is estimated that with turbo code the Shannon’s limit may be achieved. In fact it was Shannon who first gave the idea of the code to tackle errors that plagued all communication links. He also argued that with right code word it is possible to achieve the channel capacity - that is exactly what is being expected now from the turbo code. The multimedia communication is based on high data rate communication in order to provide guaranteed quality services. With turbo code, the high data rate multimedia communication will be easy to operate.

5.5 CONTROL FIELD HDLC or SDLC frames can be of three types, namely information frame (I-frame) that carries valid information in the information field, supervisory frame (S-frame) that is used to transfer supervisory signals, and unnumbered frame (U-frame) that is used for specific services and administration. When the first bit (LSB) of the control field is 0, it is an I-frame. When the first two bits of the control fields are 10 and 11, the frames are respectively S-frame and U-frame (Fig. 17). The three bits N(S) and N(R) field of the control filed respectively indicate the send and receive sequence of the frames. The reason and use of sequence numbers are discussed in the section of error control. The two bit codes of different frames are shown in Fig. (17). Flag

I-Frame

Address

Control

0

P/F N(S)

S-Frame

1 0

N(R) P/F

Code U-Frame

Information

0

N(R) P/F

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Flag

P/F

Pool/Final bit.

N(S)

Sequence number of frame sent.

P/F

Pool/Final bit.

N(R)

Sequence number of next frame expected.

Code

Code for supervisory or unnumbered frame

Code

(a)

DHARM

FCS

192


Only in I-and U- frames

Flag

Address

Control

Information

FCS

Flag

Flag

Address

Control

User information

FCS

Flag

I-frame

Flag

Address

Control

Flag

Address

Control

FCS

Flag

U-frame

FCS

Flag

S-frame

Management information

(b) Different HDLC frame types The four different types of S-frames as : Code 00 01 10 11

Frame type RR REJ RNR SREJ

Frame name Receive Ready Reject Receive Not Ready Selective Reject

(c) S Frames Code Command/Response

Meaning

SNRM

Set normal response mode

SNRME

Set normal response mode (extended)

SABM

Set asynchronous balanced mode

SAMBE

Set asynchronous balanced mode (extended)

UP

Unnumbered poll

UI

Unnumbered information

UA

Unnumbered acknowledgement

RD

Request disconnect

DISC

Disconnect

DM

Disconnect mode

RIM

Request information mode

SIM

Set initialization mode

Command/Response

Meaning

RSET

Reset

XID

Exchange ID

FRMR

Frame reject

(d) U-frame control command and response

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Code 00 11 11 11 00 00 00 10 00 11 11

001 011 100 110 000 110 010 000 100 001 101

10

001

Command

Response

SNRM SNRME SABM SABME UI

DM

DISC SIM UP RSET XID

193

UI UA RD RIM

XID FRMR

(e) U-frame control field in HDLC Fig. 17: HDLC control fields in details with illustration

5.6 OTHER PROTOCOLS In addition to OSI protocol, other established protocols are SNA (Systems Network Architecture) developed by IBM and DNA (Digital Network Architecture) developed by DEC. However there is no single protocol, which is to be used as superior to other in most applications. But the functional similarity among OSI, SNA and DNA is readily observed (Fig. 11). Two networks having different protocols may be connected through a protocol converter. Protocol converter is a complicated system. OSI/ISO protocol is open to any networking system. The initial ISDN (integrated services digital network) was also based on this protocol. If incompatible device and nodes follow the protocol, the incompatibilities among them would not cause any problem in communication. That is why protocol is known as 'open system' i.e. open to any system, whether it is a device or/and network. BOX 7 Determining packet size still may be a research topic ! Switching Techniques: Review and Modifications

Introduction In the non peer-to-peer networks, there is no direct path between every pair of computers and/ or terminals that may wish to communicate. This needs some form of switching within the network. There are different forms of switching [1] used in different applications (Fig. 1): 1. Circuit switching 2. Message switching 3. Packet switching 4. Hybrid switching 5. ATM/Fixed size Cell Switching

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Switching techniques

Circuit switching (Used in telephone networks)

Store-and-nforward switching

Message switching (Used in telegraph networks/e-mail)

Virtual circuit

Packet switching (Used in data networks)

Very fast/fixed size packet switching/ATM cell

Datagram

Fig. 1: Switching Techniques

1. Circuit switching The principle of circuit switching is to form a continuous physical “copper’’ or “wire” path in between the source and the destination by appropriate switching [2] at the intermediate switching centers; and to dedicate the path in between the source and the destination (a single set of users) for entire duration of the transmission without interruption. No other potential user can use the path until released and during this time even if intervals are available. Circuit switching is the technique on which public switched telephone networks (PSTN) work. This technique is appropriate for voice communication as it provides instantaneous (note that voice is time sensitive in nature) and two-way interactive link so important for man-to-man communication. However when circuit switching is used for data, there develop a number of problems: 1) the link utilization is low [3] and hence the system is uneconomical. This is due to the fact that data is very bursty. The data traffic often consists of short bursts of data followed by long intervals of no data (during which dedicated path is not being used). Typically two computers can use only 1% of the time allocated. But the users have to pay for the entire duration of the dedicated link period. A possible alternative might be to realize separate call for each burst of data; but it is likely to be inefficient due to relatively long call set-up time (typically in seconds for analog switching exchanges). 2) data transmission has a quite wide range of transmission rate-typically from hundred bits per second (communication in between a computer and a terminal) to million bits per second (communication in between two computers on o long haul network). Hence switching must provide the maximum transmission rate between all users simultaneously (i.e. exactly same data rate to all users - how is it possible?) to tackle peak demand as data can not be stored or delayed in circuit switching mode and as the two station are in direct communication in circuit switching.

2. Message Switching The message switching technique [4] overcomes most of the limitations of the circuit switching technique when used in data networks. In this technique, instead of switching the centers to establish the link as in circuit switching, the circuit (links) are made permanent and the message is switched around the network to reach the destination from the source i.e. the message is passed from node to node till it reaches the destination. An address is attached as header to

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the beginning of message. Based on the address and routing strategies, the message is forwarded from node to node when link facilities become available. During the intermediate period (when link is not free or when routing is being processed based on address etc.), the message is temporarily stored; and that is why the message switching is also known as “Store and Forward” switching. Message switching is in use for telegrams and E-mails. The technique overcomes the problems of the circuit switching as there is no call set-up delay and two users do not have to communication at the same speed, as they do not communicate directly [5]. The problems of message switching are due to the fact that the whole of the message under transmission is treated as one unit. For some applications message may be vary long (exam: a complete file, a whole of a database etc.) and for some other applications message may be quite short (exam: a database query, a file query etc.). The long message may monopoly a particular link preventing other messages some of which may be of more urgent in nature, to use the link. Moreover, the storage space of an intermediate node may be insufficient for a long message or for a whole set of message. These two problems may lead to slower response time to other users.

3. Packet Switching In the packet switching technique each message is broken into smaller but optimal pieces called “packet” [6] that provide an acceptable compromise in between response time and efficiency. By this, delay is minimized and problem of storage as in message switching is avoided. Thus the packet switching is nothing but a modern version of old concept of message switching. In packet switching a packet as the unit of data is transmitted rather than while message as a data unit as in message switching. Typically a packet is of length from 1000 to a few thousand bits. The packet size may be different for different public switched data network(PSDN). However when such different PSDNs are inter-connected; data packets may have to be combined or split as they pass from one network to another network. Each packet bears a header carrying the address of destination and a sequence number. However packet switching is an attempt to combine the advantage of both the message and the circuit switching (using two concepts of datagram and virtual circuit; and optimal packet length); and at the same time to minimize the problem of both. Packet switching is rather like a convention postal service in which letters are passed from one post office to another till they arrive their destination. Typically the delivery time on modern packet network is about tenth of a second. The problem in packet switching is the proper handling (sequencing, error, flow and congestion control etc.) of the stream of packets. However there are two approaches in packet switching; datagram and virtual circuit. In datagram [7] service, each packet (datagram) is treated independently. They are transmitted from node to node independently till they reach destination. Routing at intermediate nodes is determined based on network’s strategies and availability of link. Thus the packets of a particular message may reach the destination out of order of sequence. Thus it is the destination’s responsibility (host machine’s responsibility) to order the sequence as well as to see loss of or duplicate packets by providing error and flow control. Hence in datagram service error and flow control are done as in the layer-4 (transport layer) in terms of OSI/ISO protocol. In virtual technique, a logical circuit in between a source and a destination is established prior to data transfer; and the circuit is used to route the data. The logical circuit is not dedicated path as in circuit switching, and hence the virtual circuit may be shared by other potential users. The packet may be buffered and queued at each node. The difference of the virtual circuit from the datagram service is that there is no need of routing decision for each packet at

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the intermediate nodes. It is made once for all along the route and the logical circuit remains open till it is disconnected by the users. As all packets follow same path they arrive in order of sequence. Error control and flow control can be done via virtual circuit facility in intermediate nodes. The error and flow control are done at level-2 (OSI/ISD protocol) for each link and at level-3 end-to-end control. If two stations wish to exchange data over a long period of time the virtual circuit approach is preferable to the datagram approach. A comparative study of datagram and virtual circuit is given in table (1). Crucial Issue of Packet Size—A New Look In literatures [8-11], the packet switching is often compared with the message switching in terms of speed only. Based on such comparison, often a decision about the optimal size of packet is arrived at. Undoubtedly, speed is an important parameter of comparing switching techniques. But the actual trade off parameter of the packet switching is speed versus overhead bits. In this section, we propose a comparison of the switching techniques based on the trade off parameter: speed vs. overhead bits. Parameters of comparison A parameter with name power, η is introduced for the comparison of the switching techniques. The power, η, is defined as: η = Coding Efficiency × Speed ...(1) The power η of the switching technique is analogous to the gain-bandwidth product of an amplifier. The coding efficiency takes care of the effect of overhead bits; and thereby, the product of coding efficiency and speed, defined as the power of the switching technique shall measure the quantitative aspect of the switching technique in reference to the trade off parameter: speed versus overhead bits. Source

Destination

1

3

2

4

K+

Nodes are equidistant apart. C = link speed in BPS.

Fig. 2: A typical Network

In order to compare the packet switching with the message switching in terms of the power, h, we assume a network of Fig. (2). A message of M bits is assumed to be transferred from source to destination under the message switching and the packet switching. Under the message switching, the full message of M bits shall be transferred as one unit with negligible overhead bits. Under the packet switching, we assume that N packets each with h overhead bits are required to transfer the whole message of M bits. Hence, neglecting propagation delay, we have:

and

LM 1 OP N K. M/C Q R| U| L O 1 1 =M N M + N. h PQ × S| K(M/N + h) + (N − 1) (M/N + h) V| C T C W

ηmessage = 1 ×

…(2)

ηpacket

...(3)

where speed is measured in messages transferred per sec.

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From eqns. (2) and (3), we derive a relative power as : ηpacket ηr = ηmessage

...(4)

From eqn. (4), the packet-switching shall have maximum gain over the message switching when ηr is maximized with respect to N. Maximizing ηr with respect to N, we get optimal N (as N1) as:

M 2 (K − 1) =0 …(5) h If conventional comparison as done is literatures[8, 9] is done in terms of the speed only, the optimized N (as N2) that could have offered maximum gain to packet switching over message switching is obtained as: N13h + N12 {h(K – 1) + 2M} –

N2 =

M(K − 1) h

...(6)

Results of comparison For different sets of K, M and h, we make a comparison N1 and N2 and relative power in table (2). In table (2) we have :

FG M IJ × RS H M + N . h K T K(M/N F M IJ × RS =G H M + N . h K T K(M/N

ηr(based on N1) = and

ηr(based on N2)

1

2

UV W UV + h) W

...(7)

1

KM + h) + (N 1 − 1) (M/N 1 + h)

...(8)

2

KM + h) + (N 2 − 1) (M/N 2

The value of N1 was obtained by Computer Program based on Newton-Raphson’s method and using equation (5). We find that the optimal N (as N1) derived from the comparison based on η offers better result than that of N (as N2) derived from the comparison based on the speed only. It is concluded that the power defined in the paper may be more appropriate for comparing all switching techniques of store and forward types. Frame bit

A

B

C

A

B

C

A

B

C

A

B

C

A

B

A

(a) STM Flag

2-bytes header

Variable-length

Information field Frame check sequence

(b) Conventional Variable Length Packets 5–byte header

48–information field (c) Fixed size packet/ATM cell

Fig. 3: Illustration of different switching/Packets

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Flag

198


Table 1: Comparison of Virtual and datagram services Datagram Service

Virtual Circuit Service

1. There is no call set-up phase. Good for short period of exchange. Data delivery is quicker.

1. There is call set-up phase. Not good short period of exchange.

2. Communication processing in terms of routing decision based on address and relieving of station etc. is required for each packet at each intermediate node. Not good for extended period of exchange.

2. No routing decision etc. required. Good for extended period of exchange.

3. More flexible. If congestion develops at any node, incoming packets may be routed through other link to avoid congestion.

3. As there is predefined path it is more difficult to deal with congestion.

4. More reliable. If a node is lost, packet may find alternate route.

4. If a node within path fails, no alternate route is there.

5. Service quality is not so high Sequencing is a must at destination along with flow & error control.

5. Service quality is high as there is no need of sequencing at destination. Error & floe control are done at level 1 and level 3.

Table 2: Comparison of optimal N and ηr Set

N1

ηr based on N1

N2

K = 10, M = 1024 bytes, h = 4 bytes

32.64

6.16

48

5.97

K = 10, M = 1024 bytes, h = 8 bytes

22.64

5.16

33.94

4.93

K = 10, M = 1024 bytes, h = 16 bytes

15.55

4.09

24

3.84

K = 10, M = 2048 bytes, h = 4 bytes

46.74

7.046

K = 10, M = 4096 bytes, h = 4 bytes

66.66

7.76

K = 100, M = 1024 bytes, h = 4 bytes

95.81

26.04

159.19

23.43

K = 1000, M = 1024 bytes, h = 4 bytes

195.85

52.61

505.71

37.96

7.88 96

ηr based on N2

6.88 7.64

4. Hybrid switching Now-a days different hybrid forms of circuit and packet switching are possible as computer and communication technology are coming more and more closer together. Fast-connect circuit switching is an approach where call set-up time of telephone switching is expected to be milliseconds or less. However such systems will be expensive [8]. Time division switching is an interesting variant of packet switching. In this technique each node will scan input line in a predefined rotation and each packet is immediately outputted on a correct output line as soon as the header is read. Fixed size packet and strict synchronization are needed. No storage space is required at node.

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5. ATM Cell Unlike conventional packet switching, in ATM cell switching the ATM cell(a typical packet) uses a fixed length and very short cell (Fig. 3). Each cell is 53 bytes in length with 5 bytes headers and 48 bytes data field. Such a fixed and short cell was proposed to make switching technique applicable to integrated service of voice, video and data etc. The time dependent service like voice and video suffer from randomly varying delays and irregular gaps. Data has no such problem but may have a message of 64 kilobytes or more. By breaking the message into short cells and assigning priority to voice, video and other time dependent service over time independent service, the multimedia over a single channel is possible to realize. By allowing several service on a shared physical medium, ATM increase the efficiency of communication network. This make ATM a practical switching for any service. ATM is an extension of fast packet switching. ATM is virtual connection oriented technology. The connection is ensured and determined by 5 bytes of header. ATM is a best choice for an integrated network because of its ability 1) to handle variable bit rates upto tens of megabits per second, 2) to comply with real time services and 3) to handle scalable service; and all these with maximum flexibility. Short cell of ATM ensures integration possibility and high speed network.

Variant ATMAn Efficient Proposal It is seen that ATM differs from traditional packet switching in two respects. First, the ATM cell is fixed in size unlike the variable size of packets in packet switching. Secondly, the size of ATM is short and only 53 bytes; whereas packet size in conventional packet switching may be on average more than 512 bytes. These two departures of ATM cell from conventional packet switching, make ATM appropriate for voice and voice-like communication in addition to traditional applications of data communication[12-17]. By incorporating voice and voice-like services in this manner. ATM may likely to do some unjust to traditional data communication. We shall discuss a concept of multilevel ATM. Multilevel ATM may be an open transport technology for incorporation of existing and future services and application independent networks.

Coding Efficiency and Quality Factor Two parameters those shall be used in comparing ATM, with proposed multilevel ATM are coding efficiency and quality factor. Coding efficiency measures how less extra bits are required in a particular scheme for carrying a given message. It is defined as per conventional definition of coding efficiency used in error correction or detection codes. Quality factor measures the quality of service being provided. It is a factor of a number of things like effective utilization of bandwidth, intelligibility, acceptability (audibility in voice for example), latency and jittering etc. However, we shall assume a simple model for evaluating quality factor. If Q0 is the quality factor of ATM cell for data communication, the quality factor, Q of other packet schemes for data communication shall be measured as: Q = (Q0/48) * (Packet size of the scheme). The model is justified as the quality of data communication usually increases with packet size. The quality measurement can be done in many other ways, but here we have assumed a very simple model. Although the model is simple, yet it is not far from real environment. This is because the quality measurement in the present model is relative.

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Binary ATM In a concept of binary ATM (Fig. 4), there are two ATM cells: one for data and related services, and another for voice and voice-like services. The cells will have 3 extra bits and they shall define the service type as below: 000 111

for time dependent services; and for time independent services.

The simple majority logic will be used by nodes and/or stations to identify the service types based on received 3 extra bits. It is illustrated as below: Received extra bits

Decision

000

Time dependent service

001 010

Time dependent service Time dependent service

011 100

Time independent service Time dependent service

101 110

Time independent service Time independent service

111 Time independent service We can compare binary ATM with conventional ATM in terms of relative efficiency and relative quality factor. Relative coding efficiency of binary ATM divided by that of the conventional ATM and is given by: n=

(1 + (5/48)) [(43/16) ((1/ y1 ) + (1/ y2 ) + 1)]

Similarly relative quality factor is the quality factor of binary ATM divided by that of the conventional ATM. Using the same simple mode as discussed earlier, if Q0 be the quality factor of any service under conventional ATM, the quality factor of time dependent and time independent services under binary ATM shall respectively be:

LM 48Q OP N y Q 0

1

and

LM y . Q OP N 48 Q 2

0

and hence average quality factor of binary ATM shall be : Q = [0.5{ 48Q0/y1 + y2Q0/48} ] and, therefore, relative quality factor shall be : Q/Q0 = 0.5[ 48/y1 + y2/48] The mathematical analysis is given in appendix-I. For a different set of fixed y1 and y2 the variation of relative efficiency and quality factor was calculated as shown in table (3). In binary ATM the data sizes for data and relative services; and for voice and related services have been proposed as 64 and 32 respectively as optimal choice. A three or four levels ATM may be possible with 6 extra bits in header of the cell. The encoding algorithm for service identification at source may be as below:

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First 3 extra bits Next 3 extra bits Service Type 000 000 type–1 000 111 type–2 111 000 type–3 111 111 type–4 Decoding algorithm at the receiving nodes and/or stations shall be simply majority decision based on received two individual groups of 3 bits out of 6 extra bits. However, in general with n times use of 3 bits in groups as above we can accommodate 2n numbers of ATM levels. And in each case majority logic will be used in the decoding. ATM and conventional packets can be compared with a single quanta of fixed frequency and quanta of any frequency respectively of quanta physics. We can have a multi quanta of a few frequency as a compromise i.e. we can suggest multi cell ATM. While suggesting the points to be remembered are: (1) cell size shall not be very high in order to avoid large latency which is unwarranted in time dependent services and (2) levels in multilevel ATM shall not be too high in order that the node processing becomes simpler.

Full Bytes Cell To make proposed ATM cell of full bytes in length, the extra bits may be used for error coding and security purposes. The error control will be an issue to reckon with mobile ATM particularly for data packets. In case of proposed binary ATM cell, three bits are used for service identification. To make the cell of full bytes in length, another five bits are required to be added. These five bits may be used for error control and security/coding purposes. The conventional ATM cannot be taken as an end but a means for achieving better transport technology for application independent network. Fixed (48 + 5) cell ATM may not be suitable for some of the future services - both time dependent and time independent. Therefore, multilevel ATM with proposed service identification encoding and decoding algorithms may be thought of an alternative and needs experimentation. Table 3: Efficiency and Quality factor of Binary ATM for different sets of y1 and y2 y1

y2

8 16 32 48

88 80 64 48

Relative Efficiency 80.7% 91.86% 98.03% 99%

48 bytes For CBR For VBR

Relative quality

5 bytes

Y1 = 48 but fixed

3.92 2.3 1.4 1 Traditional ATM

5 bytes 5 bytes

3 bits 3 bits

CBR = Constant Bit Rate VBR = Variable Bit Rate Fig. 4: Illustration of Multilevel ATM

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U| B | NI V| A || RY W

ATM

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Performance of Switching Techniques Generally, the comparison of techniques based on performance cannot be done because the performance depends on a number of factors, like number of nodes, total traffic, processing speed of nodes, packet size of the network, etc. However, a few observations can be made as: (a) For time sensitive service, till now there is no substitute to circuit switching on the ground of quality (b) for light and/or intermittent load under-interactive traffic, circuit switching is the most efficient (c) for heavy amount of data and efficient link utilization, packet switching is most suitable; (d) store-and-forwarding switching is not applicable to interactive real-time traffic; (e) for fast rate, packet switching may be applicable to interactive and real-time data transfer. (f) For multimedia services, ATM and proposed multilevel ATM is more appropriate based on QoS desired Performance of Proposed ATM in comparison with Conventional ATM In the networks, the techniques and systems are compared usually in terms of throughput and delay. In the above, the comparison was made in terms of quality factor and coding efficiency. In the present section we propose to compare in terms of the delay. Such a comparison depends on a number of factors, like number of nodes, total traffic, processing speed of nodes, packet size of the network, etc. However, we assume same simple network as in Fig. (2). We shall compare average transmission delay, and assume other components of the delay, namely propagation delay, node processing delay etc are negligible. We assume that: Tt = transmission delay of a conventional ATM cell over a link Tt’ = transmission delay of a proposed binary ATM packet (with six bytes header) over a link N = total number of packets constitute the message (the packets are both voice and data packets) Then the average transmission delay under the conventional ATM transport is given by: Dc = [K . Tt + ( N – 1) . Tt]/ N whereas the average delay of voice packets and that of the data packets under binary ATM transport are respectively: Dv = [K . Tt’ + (V – 1) Tt’}/N Dd = [K Tt’ + (N – 1) Tt’]/N

and

where V is the number of voice packets out of the total N packets. We perform numerical evaluation assuming each link capacity of 48 Kbps that gives Tt = 8.8 msec and Tt’ = 9 msec. The result is given in table (4) with corresponding curve at Fig. (5). From the results so obtained we find that : 1. so long the number of voice packets are less compared to the number of data packets, the advantage of sending voice packets at far less average delay is possible with proposed binary mode of transport

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2. when the number of voice packets are insignificant compared to data packets, the gain in terms of less average delay for voice packets is insignificant 3. when all the packets are data packets, the increase in average delay in the proposed binary ATM is very insignificant. Thus an overall conclusion of the suitability of the proposed binary ATM over conventional ATM is apparently established, although the same type of calculation over more generalized network needs to be done for achieving at a conclusive judgment. The advantages achieved at in the binary ATM as mentioned above are at the cost of the 1 byte higher header in the binary ATM. But the increased 1 byte header size in the binary ATM does not significantly reduce the coding efficiency.

Conclusion Behind the choice of 53 bytes size ATM cell, the motivation was to define a packet standard for all services. The same motivation worked behind the proposal of binary or multilevel ATM with the added consideration of reducing the delay for time sensitive packets. Our analysis with a simple network confirms in meeting the objective. No significant disadvantage is found in respect of binary ATM. Thus the binary ATM is the new challenge that needs some more experiments before final acceptance. The work critically analyzes the different switching techniques in terms of applications, link utilization and effectiveness. In addition to these common parameters of comparison, we have introduced a parameter of power making use of overhead bits and speed to compare the techniques. The power based comparison as usual but with clearer terms shows that the size of the packet is the critical to derive maximum gain in applying packet switching for data transport. We have also proposed a multilevel ATM cell switching and showed with analytical derivation that based on quality requirement and coding efficiency, variant ATM could be tried out. Further research is required to support the ideas. We also note that the implementation of binary ATM on the stated logic fails only if two or more bits of the proposed extra header bits for service identification are in error. The probability of two bits in error is: 3C α(1 – α)2 1 where α is the bit error rate. For common maximum bit error rate of 10–3, the probability is not so significant. Table 4: Numerical result in msec V

Dc

Dv

Dd

10

9.5920

1.7100

9.8100

20

9.5920

2.6100

9.8100

30

9.5920

3.5100

9.8100

40

9.5920

4.4100

9.8100

50

9.5920

5.3100

9.8100

60

9.5920

6.2100

9.8100

70

9.5920

7.1100

9.8100

80

9.5920

8.0100

9.8100

90

9.5920

8.9100

9.8100

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INFORMATION TECHNOLOGY, NETWORK AND INTERNET Average delay comparison 12

Average delay in ms

10 Average delay for conventional case

8 6

Average delay for voice packet

4 2 0 1

16

11

16

21

26

31

36

41

46

51

56

61

66

71

76

81

86

No. of voice packets out of 100 total packets

Fig. 5: Delay Comparison

APPENDIX-I We assume a message of p bytes. The message contains 50% VBR services and 50% CBR services.

Under conventional ATM The number of cell required = p/48 (or next higher integer). The coding efficiency (A) is therefore:

p 1 = ( p/48) * 5 + p 1 + (5/48) Under proposed Binary ATM The number of cells required for CBR Services = p/2y1 (or next higher integer ) The number of cells required for VBR Services = p/2y2 ( or next higher integer ) where y1 = information field bytes in proposed binary ATM cell for CBR services y2 = information field bytes in proposed binary ATM cell for VBR services. The coding efficiency (B) =

p ( p/2 y1 + p/ y2 ) (5 + 3/8) + p

Relative efficiency = (B/A) =

[1 + (5/48)] [(43/16) (1/ y1 + 1/ y2 ) + 1]

REFERENCES 1. 2. 3. 4.

Andrew S. Tanenbaum, Computer Networks, Prentice Hall of India. 1988. William L. Schweber, Data Communication, McGraw Hill International, 1988. Torub & Schilling, Principles of Communication Systems, McGraw Hill Pub. Co. 1986. James Martin, Computer Networks and Distributed Processing, Prentice Hall International, 1981.

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5. 6. 7. 8. 9. 10. 11. 12. 13. 14. 15. 16. 17.

205

Lyun A. Denoia, Data Communication, CBS Publishers, 1989. A. Bruce Carbon, Communication Systems, McGraw Hill Pub. Co. 1986. Stalling, Data & Computer Communication, Mac Million Pub. Co. Taub and Schilling, principle of Communication System, Tata McGraw Hill, 1991, Ch. 16. U. D. Black, Data Communication and Distributed Networks, Prentice Hall, 1993, Ch. 7. William Stallings, Data / Computer Communication, Addision Wesley, Ch. 12. V. Ahuja, Design and Analysis of Computer Communication Networks, McGraw Hill, 1982, Ch. 3. William Pugh and Gerlad Bayer, Broadband Access : Comparing alternatives, IEEE Communication Magazine, Aug, 1995, pp.34-46. J.P. Coudruse, General Principles of ATM, L ‘e’cho des Pechrches, English issue, 1992, pp. 5-18. John D. Hunter and William W. Ellington, ISDN: A Customer’s Perspective, IEEE Communication Magazine, Jan, 1996, pp. 20-22. Roger Levy, High Speed and Flexible Switching with ATM. Express Computer, Special issue networking, Sept., 1995, pp. 7-16, India. M. De Prycker et al., BISDN and the OSI Protocol Reference Model, IEEE Network, March, 1993, pp. 10-15. Borko Furht et al., Design issues for interactive television system, IEEE computer, May, 1995, pp. 25-38.

Application layer protocol UDP

TCP Gateway protocol

Address mapping protocols ARP, DCHP

IP and ICMP

Data link layer Physical link layer

(a) The TCP/IP (Internet) Model HTTP

FTP

SMPT

HTTP

TCP

ICMP

FTP

SMPT

TCP

IP Data link layer of IEEE 802.3 (Ethernet) Electro mechanics of IEEE 802.3 (Ethernet)

(b) Layering relationship between protocols

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TCP

Transport

IP

Network

Physical network/ ethernet etc.

Datalink and physical

(c) TCP/IP and OSI relation (TCP/IP may be either 3 layers, when physical network is taken as one layer or 4 layers, when the data link layers and the physical layer of the physical network are taken as two layers, protocol but they are same and one. In analogy with the OSI reference, thus Internet node is made of IP and lower layers whereas the Internet full host and station are made of TCP, IP and other lower layers) Fig. 18: TCP/IP in different angles

5.7 TCP/IP PROTOCOLS TCP/IP (Fig. 18) is actually made of two protocols - TCP and IP. That is why TCP/IP is often known as the Internet protocol suit. TCP and IP are the two protocols in the suite. Although, TCP/IP is a four-layered protocol. Layer 1 is the network interface services layer. It corresponds to layers 1 and 2 of the OSI/ISO protocol. It provides services relating to MAC (medium access controls), device drivers, physical medium, physical attachment and physical signals. In the layer, datagrams are packaged into frames. Layer 2 is the internet protocol (IP) layer. This layer corresponds to layer 3 of the OSI/ISO protocol. It provides routing of the datagrams as units of data. Logically, they are packets. The service is connectionless. Thus switching is a datagram service rather than virtual circuit switching. IP provides the basic service of getting the datagram to their destinations. It provides this service in best effort protocol. IP receives the TCP packets from the upper layer TCP, and then forms it own packet known as IP packets. Each IP packet is associated with several IP headers. Data pack 1

Data pack 2

Data pack 3

(a) Say a three bytes original message is fragmented into three data pack each of one byte ↓ One Original Data Pack is sent to TCP layer from higher layers TCP headers ↓ TCP headers TCP Packet (= Data Pack + TCP headers added by TCP layer) is formed by the TCP layer and then it is sent to IP layer.

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TCP headers

207

IP Headers

IP Packet (= TCP packet + IP headers added by IP layer) is formed by the IP layer and then it is released to the Physical Layer. (b) Illustration of IP packet formation. Three such IP packets are done to send the whole message under the illustration. 0 3,4 7,8 15,16 31 (bit) Version (4)

Length(4)

Type of service (8)

Total length (16)

Identification (16) Lifetime (8)

Flags (3)

Protocol (8)

Fragment offset (13)

Header checksum (16)

IP source address (32) IP destination address (32) Options (variable)

……

…….

Padding (variable)

(c) IPv4 Headers Fig. 19: TCP/IP pack

5.7.1 IP Header Description The 4-bit version field is used to indicate IP version that is being used in the IP datagram. The current version is IPv4 (IP version 4). IPv6 (IP version 6) is emerging as a standard for next generation IP datagrams. The 4-bit length field, which is also known as Internet Header Length (IHL) contains information about the number of 32-bit words in the header of the IP datagram. The minimum number of headers in any IP datagram is five 32-bits words or 20 bytes, and this happens when there is no option and padding fields. In this case, the IHL field contains 0101. As the IHL field can have maximum value 1111 (=15), the maximum number of headers in any IP datagram will be fifteen 32-bit words or 60 bytes. As the number of headers will be in multiple of 32-bit words or 4 bytes words, the headers that an IP datagram can carry will be one of the following sizes: 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 60. The 8-bit type of service or TOS field indicates if any special type of service the IP datagram should receive. The indication of special service offers priority or precedence given to the particular packet or packets over the other packets, and enables the routers to choose appropriate path for the transport of the packet or packets. In past, the field was largely not used. The TOS field splits into two sub-fields: 5-bits D/T/R/C/R (low Delay/high Throughput/ high Reliability/low Cost/Reserved) sub-field and 3-bit precedence sub-field (Fig. 20). The precedence is absolute and not relative. An IP datagram for a regular packet with no special and no priority will have D/T/R/C/ sub-field set as 0000 and precedence sub-field set as 000 Consider a multimedia packet which may need a higher throughput and highest level precedence. For this multimedia packet, IP datagram will set D/T/C/R sub field as 0100 and precedence sub field as 111. The TOS field is used by routers to choose routing path. If a regular datagram that needs a normal service reaches a router, the router selects a normal

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route as per outing strategy. But when say a datagram requiring minimum cost reaches a router, the router will select a path with minimum number of hops. Similarly for a datagram with higher reliability, the router will select a more reliable (lower BER) path. Likewise for a datagram with higher throughput, a path with higher link capacity needs to be selected. The details of the TOS field are available with RFC-2474, RFC-2475 and RFC-2430. A few examples of use of TOS field for the different services are given in the table (VIII). Precedence bit

Precedence bit

Precedence bit

Delay bit

Throughput bit

Reliability bit

Cost bit

Reserved bit

0

1

2

3

4

5

6

7

Fig. 20: Bits of TOS field

Table 8: TOS field for different services Service

Delay

Throughput

Reliability

Cost

bit

bit

bit

bit

Remarks

FTP data

0

1

0

0

Service requires high throughput only

FTP control

1

0

0

0

Service requires low delay only

SNMP

0

0

1

0

Service requires high reliability

SMTP data

0

1

0

0

Service requires high throughput

SMTP Command

1

0

0

0

Service requires low delay

DNS TCP query

0

0

0

0

Service is normal type

DNS UDP query

1

0

0

0


ICMP query/error

0

0

0

0

Normal service

Remote Login/Telnet

1

0

0

0


DNS zone transfer

0

1

0

0

Service requires high throughput

TFTP

1

0

0

0


BOOTP

0

0

0

0

Normal service

The 16-bit total length (TL) indicates the total number of bytes that the datagram carries including the headers. Thus the maximum number of bytes that a datagram carries is 1111111111111111 or 65,535 (= 216–1) bytes. The IHL counts only the header fields in units of 32-bits or 4-bytes words. But the total length counts the entire packets in unit of bytes. The

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entire IP datagram is a variable one and so also header fields. By IHL, the receiver identifies the number of header bytes in the packet. The receiver by subtracting the header bytes from the total bytes identifies the beginning and the end of data, which is actually the TCP pack. The following table (9) illustrates this for maximum data (TCP pack) sizes under valid sizes of IHL. As per RFC-791, the IP end stations must be capable of handling 576-byte datagram. Table 9: Valid maximum sizes of data(payload) in IP datagram 4-bits IHL field

Valid IHL in bytes (multiple of 4)

Maximum data or TCP pack size = (65535 – IHL) bytes

16-bits TL field

0101

20 (minimum)

65,515 (maximum)

1111111111111111

0110

24

65,511

1111111111111111

0111

28

65,507

1111111111111111

1000

32

65,503

1111111111111111

1001

36

65,499

1111111111111111

1010

40

65,495

1111111111111111

1011

44

65,491

1111111111111111

1100

48

65,487

1111111111111111

1101

52

65,483

1111111111111111

1110

56

65,479

1111111111111111

1111

60 (maximum)

65,475 (minimum)

1111111111111111

BOX 8

SOLVED PROBLEMS 1. As per RFC 791, the minimum TL size is 576 bytes. For this what will be size of possible payloads? Answer is given in table (1): Table 1 4-bits IHL field

Valid IHL in bytes (multiple of 4)

Maximum data or TCP pack size = (576 – IHL) bytes

0101

20 (minimum)

556 (maximum)

0110

24

552

0111

28

548

1000

32

544

1001

36

540

1010

40

536

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1011

44

532

1100

48

528

1101

52

524

1110

56

520

1111

60 (maximum)

516 (minimum)

2. Let us define the header coding efficiency of IP datagram as [payload/(payload + header)]. Find coding efficiency for maximum and minimum possible/permitted payload. Answer: The maximum permissible payload is 65515 bytes when the headers are 20 bytes. In this case the coding efficiency = 65515/ (65515 +20) = 0.9996 The minimum permissible payload is 516 bytes when the headers are 60 bytes. In this case the coding efficiency is = 516/ (516 + 60) = 0.8958 ≅ 0.9 3. What possible inference can you draw from the results of question (2) above. Possibly, the RFC aimed to keep the coding efficiency >0.9 4. List possible maximum and minimum size of the data (payload) in the IP datagram when TL field is made of all 15, 14, 13, 12, 11 and 10 left most bits are 1s. What does happen when 9 left most bits are all 1s? Answer is given in the Table 2 TL field

Bytes in whole IP datagram

0111111111111111

32767

0011111111111111

16383

0001111111111111

8191

0000111111111111

4095

0000011111111111

2047

000001111111111

1023

Maximum and minimum payload respectively correspond to 20 bytes and 60 bytes headers

Payload in bytes

Maximum

32767– 20 = 32747

Minimum

32767 – 60 = 32707

Maximum

16383 – 20 = 16363

Minimum

16383 – 60 = 16323

Maximum

8191 – 20 = 8171

Minimum

8191 – 60 = 8131

Maximum

4095 – 20 = 4075

Minimum

4095 – 60 = 4035

Maximum

2047 – 20 = 2027

Minimum

2047 – 60 = 1987

Maximum

1023 – 20 = 1003

Minimum

1023 – 60 = 963

When left most bits are all 1, the total bytes in the IP datagram will be = 511. But as per RFC 791, the minimum size is 576 bytes. Hence it will correspond to an invalid IP datagram

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The identification field is a 16-bit header. Internet connects the different types of networks. The different networks will have the different packet size. For example Ethernet packets may be in the rage of 64 to 1518 bytes, a MAN may have maximum packet size of 1500 bytes, an FDDI packet can be of 4472 bytes and a token ring packet may be typically 17,800 bytes. When a packet of, say 1500 bytes arrives at a node or router and the same is required to be transferred or routed to a network say having maximum packet size of 512 bytes, it is required that the arriving packet of 1500 bytes is to be fragmented into several smaller packets each having a size of 512 bytes or less. Each part or fragmented packet of a particular original must be given some identification so that the receiver can recombine all the related fragments in packets. Same identification to all fragments of a packet ensures that the wrong fragments are not combined to reconstruct the original packet in the receiver or otherwise speaking the reassembling of the correct fragments is done to reconstruct the original packet at the receiver. The identification field assigns a single group number to all the fragments of a packet being fragmented. The 3-bits flags are used to control and convey information in regard to fragmentation. The use of the flags bits is shown in the table (10).The left most bit is reserved. The middle bit known as DF (Don’t Fragment) provides a flexibility to the sender. If the sender likes that the packet must not be fragmented even if it does not reach the destination, the sender can construct IP datagram with DF bit set to 1. When DF bit is set to zero, the fragmentation is allowed. When a node or routers breaks a packet into several fragments, MF (More fragment) bit of all the fragments except the last one is set to 1. The MF bit is set to 0 in the last fragment. By this receiver is made to known about the last fragment of a packet. Table 10: Illustration of fragmentation Bit position

Flag name

Function

Bit 0 or the left most bit

Reserved

Set to 0 on transmit, and ignored on receive

Bit 1 or middle bit

DF (Don’t Fragment)

Set to 0 while allowed to fragment and to 1 when not allowed to fragment

Bit 2 or right most bit

M (More fragment)

Set to 1 for all fragments but set to 0 for the last fragment

The 13-bits fragment offset indicates the how to insert fragments in the receiver’s buffer in order to reconstruct the original datagram. The field is measured in unit of 8 bytes. But why? There are only 213 offset values but the datagram can be as big as 216 bytes. So each fragment must be multiple of 8 bytes long as 216/213 = 8. The requirement applies only to the data field (or TCP pack) of the original packet as because the header fields of original IP by its right will also be the header fields of each fragment, and one can not expect the header fields be divisible by 8 always. So a fragment’s data field will always be multiple of 8 bytes long, but the total length of fragment (total length of the fragment = original packet header length + data length of fragment) will be multiple of 4 bytes long. However the total length of fragment may be made multiple of 8-bytes long by padding for the purpose of storage at the receiver. The multiple of 8-bytes long is only applicable to data field and not to the whole datagram. In the worst case, the data field may be just 8-bytes long. So for the permissible

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header lengths, the worst case fragmented packet sizes would be 28, 32, 36, 40, 44, 48, 52, 56, 60, 64 and 68 bytes long. But IP does not run over the networks that have MTU (Maximum Transmission Unit) smaller than 68 bytes (see it is the highest fragment size in worst case). Thus for IP datagram with 20-bytes IHL, the worst case refers to fragments having data field multiple of (68-20 =) 48 bytes long. 48-bytes are multiple of 8-bytes. Consider an original IP packet with the 20-bytes minimum IHL and the maximum TL as (65535-20 = ) 65515-bytes long. In worst case such an original IP packet may be fragmented into (65515/48=) 1364 fragments with remaining 43 bytes left over as 1365th fragment. The 1365th fragment will be made 68 bytes long by padding. Problem: Assume that an original IP datagram with TL of 1500 bytes and IHL of 20 bytes requires to be transported over a network that supports MTU of 512 bytes (Fig. 21). Propose a fragmentation scheme for router R.

IP with 1500 bytes

Router

Network with MTU 512 bytes

Fig. 21

The original IP packet with 1500 bytes has a data field of 1500 – 20 = 1480 bytes. Network with MTU of 512 bytes will carry fragments each with data field of maximum 512 – 20 = 492 bytes. This is because each fragment must have 20 bytes header field as in original IP packet. But 492 is not divisible by 8. Nearest lower number of 492 that is divisible by 8 is 488. Thus each of the fragment will have a data field of 488, and number of fragments will be 1480/ 488 = 3 with remainder of 16 bytes that will need one more fragment. Thus total of four fragments will be needed with three fragments each of size 488 + 20 = 508 bytes and fourth fragment of 16 + 20= 36 bytes. However each fragment will now be made multiple of 8-bytes long by padding of 4 bytes in which case each of the three larger fragments become 508 + 4 = 512 bytes and last fragment becomes 40-bytes long. Now the fragments will have the fragmentation characteristics as shown in Fig. (22). OS stands for fragment Off Set field. The first fragment bears OS field as 0, the second fragment has OS field as 64 (this means total 64 numbers each of 8-bytes fields are there in the previous fragment), the third fragment bears OS field as 128 (which means previous fragments has total 128 numbers each 8-bytes fields) and so on. Such a numbering of offset field gives advantage in storage at receiver. Fragments move in the networks just like datagram. So the fragments may reach at the receiver out of sequence. If offset number was given like serial sequence number, then looking into the fragment number reassembling might have been done, but storing of the fragments at the receiver would have not been so easy. Consider the arrival of the fragments at the receiver as in Fig. 23). We find: Fragment with off set field 128 arrives FIRST Fragment with off set field 0 arrives SECOND Fragment with off set field 192 arrives THIRD Fragment with off set field 64 arrives LAST.

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The receiver based on the offset value places the fragments in the buffer location as in Fig. (23). The offset value refers to the first location of buffer for the storage. By the process, the fragments are arranged or stored in order in the buffer. This is called pigeon holding. Fragment 1 ID = xx M=1 OS = 0 TL = 512

Fragment 2 ID = xx M=1 OS = 64 TL= 512

Fragment 3 ID = xx M=1 OS = 128 TL = 512

Fragment 4 ID = xx M=0 OS = 192 TL = 40

Fig. 22

Fragment 3 reaches first with OS as 128 and M as 1

Fragment 1 reaches second with OS as 0 and M as 1

Fragment 4 reaches third with OS as 192 and M as 0

Fragment 3 reaches last with OS as 64 and M as 1

Say each location is of 8 bytes in the receivers buffer Location 0 to 63 Location 64 to 127 Location 128 to 191 Location 192 onwards

Fig. 23

The 8-bits time to live (TTL) field is used to ensure that an IP packet does not persist in the Internet forever. If packets persist in the network forever, the congestion is bound to occur. So TTL in its own right provides a mechanism to avoid congestion in the Internet. The maximum TTL field is 255. Early days it was taken as 255 seconds. Presently the TTL field is taken a hop counter. Thus an IP packet can move the maximum 255 hops in the Internet. The transmitting station sets the TTL field of the datagrams. When the datagram passes through routers or nodes, each router or node decreases the TTL field by default value set by the router or network administrator; the minimum value of default being one. If a router or node receives a datagram with TTL field as 0 and if the datagram not then reaches its final destination, the datagram is discarded by that router or node. The 8-bits protocol field defines the protocol in which the data is encapsulated. A list of a few permissible protocol is given in the table (11). For example when the IP datagram carries a protocol field with 6, it means the data is a TCP pack. Table 11: Protocol as defined in the protocol version field in the IP header Protocol

Protocol Version Field in the IP header

ICMP (Internet Control Message Protocol)

1

IGMP (Internet Gateway Message Protocol)

2

GGP (Gateway-to-Gateway Protocol)

3

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IP in IP (encapsulation)

4

TCP (Transmission Control Protocol)

6

EGP (Exterior Gateway Protocol)

8

IGP (any private Interior Gateway Protocol)

9

UDP (Users’ Datagram Protocol)

17

ISO TP4 ( ISO Transport Protocol class 4)

29

IPv6 (Internet Protocol version 6)

41

IPv6-Routing Header

43

IPv6- Fragmented Header

44

IDRP (Inter-Domain Routing Protocol)

45

RSVP (Reservation Protocol)

46

GRE (General Routing Encapsulation)

47

MHRP (Mobile Host Routing Protocol)

48

ESP (Encapsulation Security Payload for IPv6)

50

AH (Authentication Header for IPv6)

51

SWIPE (IP with Encryption)

53

NARP (NBMA Address Resolution Protocol)

54

MOBILE (Mobile IP)

55

ICMP for IPv6

58

No Next Header for IPv6

59

Destination Options for IPv6

60

DGP (Dissimilar Gateway Protocol)

81

IGRP (Interior Gateway Routing Protocol)

88

OSPF (Open Shortest Path First)

89

PIM (Protocol Independent Multicast)

103

IPX in IP

111

Unassigned

130-254

Reserved

255

(RFC-1700 or its successor provides a complete list of protocol field) The header check sum field is 16-bits. The field is to check error on the headers fields of the IP datagram. The check sum is calculated using 1’s complement addition rule, which was discussed at length previously at the section of error control. The checksum is computed by taking 16-bits words from header fields. The checksum field ensures checking of error only at link level error on header fields. TCP as we will see later will perform end to end error control, which will take care of error on data field. That is why at the IP level error checking at the data

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field is not done. The IP header checksum is computed at each router/node or on hop-to-hop basis since each router/node changes the TTL field as well during fragmentation the IP headers (like identification, M flag, off set fields etc) are changed. IP header has two address fields: source IP address and destination IP address. These are used to route the datagrams. IP uses 4 modes of addressing for this purpose. These are known as A- class, B-class, C-class and D-class. The difference in classes is due to how many bits are used for network identification and how many bits are used for host identification. The different address formats under different classes are illustrated in Fig. (24). The IP addresses are 4-bytes and they are usually mentioned as four sets of dot-separated octets. IP address first-octet ranges are shown in table (12). For A class, most significant bit of first octet is always 0. Therefore next seven bits are used for network identification, and this means that only 27 = 128 networks can be of A-class. The hosts may be 224 . For class-B address, first two bits of first octet must be 10, and this makes first octet to be 128 to 191. 127 is kept reserved for special purposes. Total networks and hosts under class B are respectively only 214 and 216. For class-C address, first three bits of first octet is 110. This means the first octet will be from 192 to 255. Class-C address has only 221 networks and only 28 hosts. The minimum hosts under different addressing mode are: Class A = 27 × 224 = 231 Class B = 214 × 216 = 230 Class C = 221 × 28 = 229 The address scheme under class-A to class-C is for unicast communication. The other addressing schemes are for multicast communication. Class-D and Class-E belong to this scheme. In class-D, all the four octets are used to identify the group of nodes designated to receive a multicast. Class-D addresses do not specify the network. Class-D addresses are in the ranges of 224.0.0.0 to 239.255.255. Class-E addresses range from 240 to 255 in the first octet, and are used for experimentation. 1

2

3

Class A

0

Class B

1

0

Netid

Class C

1

1

0

Class D

1

1

1

0

Class E

1

1

1

1

4

8

Netid

16

24

Hosted Hosted Hosted

Netid Multicast address 0

Reserved for future use

(a) Structure of IPv4 Addressing (netid = network identification, hostid = host identification) Different classes

Starting address

Last address

Class-A

0.0.0.0

127.255.255.255

Class-B

128.0.0.0

191.255.255.255

Class-C

192.0.0.0

223.255.255.255

Class-D

224.0.0.0

239.255.255.255

Class-E

240.0.0.0

247.255.255.255

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216


(b) Address space under different classes. (Bold space is for netid and rest is for hostid; except in case for class-E. In case of class-E, the whole of the address space id undefined) Class

Range

A B C D E

1.0.0.0 to 126.254.254.254 128.0.0.0 to 191.254.254.254 192.0.0.0 to 223.254.254.254 224.0.0.0 to 239.0.0.0 240.0.0.0 to 255.0.0.0 127.0.0.0 to 127.254.254.254 255.255.255.255

Number of Networks 126 16,000 2,000,000 NA NA NA NA

Number of Hosts per Network 16,000,000 64,000 265 Multicast Test Loop back Broadcast

(c): Details in different addresses Fig. 24: IPv4 addressing scheme

Table 12: IP Addressing range under different classes

Class A Class B Class C

Range of first octet in IP address

Maximum number of network that can be addressed

Maximum number of hosts or nodes per network that can be addressed

Maximum number of total nodes or hosts that can be addressed under the addressing class

1-126 128-191 192-223

126 16384 2097152

16777214 65534 254

2113928964 1073709056 532676608

Maximum number of hosts or nodes per network is in Class A and it is about 256 times of that of the Class B and about 216 times of that of the Class C. Maximum number of hosts or nodes per network in Class B is about 256 times of the that of Class C. Maximum number of total hosts or nodes in Class A is about two folds of that of Class B and four folds of Class C; and that of Class B is about two folds of that of the Class C. The logical conclusion is that Class A addressing is for Larger networks, Class B for Medium networks and Class C for Small networks. So if your network is large, you can apply for Class A address, and for medium network, Class B and for small network, Class C address. There is a trade off among the different address classes. The trade off is: the number of network versus the number of hosts or node per network. Class A has least number of network address but highest number of host/node address per network, class C has highest number of network address but least number of host/node address per network. Class B falls in between these two extremes. The special addressing schemes are as below : 1. The address 0.0.0.0 is used to mean this host on this network. It actually refers to Internet itself.

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2. 255.255.255.255 is used as a broadcast packet for all networks. Network ID with any but host IDs with all 1’s or 255 is used for directed broadcast to the network. For example 10.255.255.255 is used for broadcast message to the class A network with ID 10. Broadcast addresses are destination addresses. 3. Packets with first octet as 127 are used for network testing. Address with network ID as 127 and host ID as any is used for internal host loop back address. Thus loop back addresses are 127.0.0.0 to 127.254.254.254 4. An entire network is specified by providing only the network identification and with 0s in all other octets such as: 124.0.0.0 for a class A network, 129.155.0.0 for a class B network and 200.127.110 for a class C network. 5. A specific host on this network is specified by network ID as 0s and with required host ID. This is used as a source address. Question. The different classes of addresses may have been identified by first two bits of the 32 bits as below: 00 for class A 01 for class B 10 for class C 11 for class D Compare the advantages and disadvantages of the proposed scheme of class identification with that of the IPv4 addressing.

Unregistered or Private Address Space For having global Internet connectivity, the hosts and networks on which hosts are connected must have distinct IP addresses. Till 1999, the IP addresses were assigned by IANA (Internet Assigned Numbers Authority) and now the job is being carried by ICANN (Internet Corporation for Assigned Names and Numbers). If any network is not connected to the Internet, the network even then can use some address space kept reserved for the purpose in RFC 1597. This address space is known as unregistered or private address space, which is listed in the table 13. When the network needs to connect to the Internet, the network can continue with the unregistered address. There are systems like NAT (Network Address Translator) gateway that will translate private unregistered addresses to public registered addresses. Table 13: Private unregistered address space Address class

Private unregistered address space

Number of networks

Class A

10.x.x.x to 10.x.x.x.

1

Class B

172.16.x.x to 172.31.x.x

16

Class C

192.168.0.x to 192.168.255.x

256

Subnet Mask Subnet mask is a special addressing scheme. It is used for two purposes: to show the class of addressing in use, and to divide a network into different sub networks to control traffic. For first purpose, a subnet mask determines which part is for network ID and which part is for hosts ID. A subnet mask for A class network is: 11111111.00000000.00000000.00000000

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(=255.0.0.0). All 1s indicate Network ID and all 0s indicate host ID. This mask is full mask or default mask or implicit mask. However for the purpose of masking it is not always that full or default mask be used. The Host ID part that is in the hand of the organization may be used for masking also (This is illustrated in the example of Fig. 25). For second purposes, subnet mask is used to divide the network within network administrator. For example: the entire third octet of class-B may be designated for subnet ID that would be 11111111.11111111.11111111.00000000 (=255.255.255.0). Another example: of subnet ID that under class - B could be: 11111111.11111111.11110000.00000000 (=255.255.240.0). In this example 4 left hand side bits of the third octet are used for subnet networks ID, whereas other 4 bits plus last octet is kept for host ID. The possible class full subnetting in class A, B and C network is listed in table (14). Table 14: Possible classful subnetting Possible Classful Subnetting breakdown under Class “A” addressing Mask size

Number of

Number of Hosts

Decimal Mask in Bits Network

8 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30

16,000,000 4,000,000 2,000,000 1,000,000 512,000 250,000 128,000 64,000 32,000 16,000 8,000 4,000 2,000 1,000 510 254 126 62 30 14 6 2

1 4 6 14 30 62 126 254 510 1,000 2,000 4,000 8,000 16,000 32,000 64,000 128,000 256,000 512,000 1,000,000 2,000,000 4,000,000

255.0.0.0 255.192.0.0 255.224.0.0 255.240.0.0 255.248.0.0 255.252.0.0 255.254.0.0 255.255.0.0 255.255.128.0 255.255.192.0 255.255.224.0 255.255.240.0 255.255.248.0 255.255.252.0 255.255.254.0 255.255.255.0 255.255.255.128 255.255.255.192 255.255.255.224 255.255.255.240 255.255.255.248 255.255.255.252

Possible Classful Subnetting Breakdown under class “B” addressing Mask size

Number of

Number of Hosts

16 17 18

64,000 32,000 16,000

1 2 4

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Decimal Mask in Bits Network 255.255.0.0 255.255.128.0 255.255.192.0


19 20 21 22 23 24 25 26 27 28 29 30

8,000 4,000 2,000 1,000 510 254 126 62 30 14 6 2

6 14 30 62 126 254 510 1,000 2,000 4,000 8,000 16,000

219

255.255.224.0 255.255.240.0 255.255.248.0 255.255.252.0 255.255.254.0 255.255.255.0 255.255.255.128 255.255.255.192 255.255.255.224 255.255.255.240 255.255.255.248 255.255.255.252

Possible Classful Subnetting breakdown under class “C” addressing Mask size 24 25 26 27 28 29 30

Number of 254 126 62 30 14 6 2

Number of Hosts 1 2 4 6 14 30 62

Decimal Mask in Bits Network 255.255.255.0 255.255.255.128 255.255.255.192 255.255.255.224 255.255.255.240 255.255.255.248 255.255.255.252

The use and operation of mask in IP addressing and routing could be further explained with an example network topology of Fig. (25). In the example, three Ethernets (I, II, and III) are connected using routers. There are two hosts on Ethernet I, and one each on Ethernets, I and II. Assume the IP addresses of the networks, Ethernet–I = 139.39.1.0/24 Ethernet–II = 139..39.2.0/24 Ethernet–III = 139.39.3.0/24 Thus all the interfaces and hosts connected to Ethernet–I will have IP address with first 24 bits as 139.39.1; similarly all the interfaces and hosts connected to Ethernet-II and EthernetIII will have IP address with first 24 bits as 139.39.2 and 139.39.3 respectively. 24 refers to mask. The mask of network is 24 numbers of 1 and followed by 0: 11111111 11111111 11111111 00000000. In the figures, addresses of the hosts and the router interfaces as shown are: Router-I interface of Ethernet-I = 139.39.1.1 Host-A = 139.39.1.2 Host-B = 139.39.1.3 Router-I interface of Ethernet-II = 139.39.2.3 Host-C = 139.39.2.1 Router-2 interface of Ethernet-II = 139.39.2 .2 Router-2 interface of Ethernet-III = 139.39.3.1 Host-D = 139.39.3.2

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The issue, now, is how a host will send the packets to other hosts. Hosts connected in a same Ethernet has the same 24 leading bits in address, whereas the hosts connected in different Ethernet has different 24 leading bits in address. When a node has a packet to send to a destination, it first checks whether the destination is in same network or not. This is done by mask, as illustrated bellow: Case-I. A to B communication: (Source) Host A's address: Host A's mask (24):

10001011 11111111

00100111 11111111

00000001 11111111

00000010 00000000

AND operation(X): 10001011 (Destination) Host B's address: 10001011

00100111 00100111

00000001 00000001

00000000 00000011

Host A's mask (24): AND operation(Y):

11111111 00100111

11111111 0000000

00000000 00000000

11111111 10001011

As X and Y are same, A shall understand that destination is on its network. Case-II. A to C communication: (Source) Host A's address: Host A's mask (24):

10001011 11111111

00100111 11111111

00000001 11111111

00000010 00000000

AND operation(X): 10001011 (Destination) Host C's address: 10001011

00100111 00100111

00000001 00000010

00000000 00000010

Host A's mask (24): AND operation(Y):

11111111 00100111

11111111 00000010

00000000 00000000

11111111 10001011

X and Y are not same. So A will understand that C is on different network. Therefore, A will send data to router. 139.39.1.2 Ethernet-1 139.39.1.0/24 139.39.1.1

Host-A

Host-B

Router-I

139.39.1.3

139.39.2.3 Ethernet-II 139.39.3.0/24

139.39.3.1

Router-2

139.39.3.2

Ethernet-II 139.39.2.0/24

Host-C

Host-D

Fig. 25: example network for subnet masking

The subnet masking actually eases out the routing problems of the network to the extent that within the organization the routing is internal and not on the Internet routing.

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Classless Addressing and Routing In Ipv4, the address classes are A, B and C. Any organization may opt for any one of these classes based on the size. Say an organization is having 2000 hosts under it’s a local network. Thus organization can not opt for class C address, as because under one network address of class C, maximum 28 = 128 hosts can be addressed. The optimal option is then for class b when under a network address of class B, maximum 216 = 64 k hosts can be addressed. Here lies the problem. If the organization is opting for a network address of class, the organization will not be able to use full host address space; and the 64 k–2000 host address space will remain unutilized. To solve the problem, RFC 1519 proposed the concept of classless addressing. Unlike the fixed numbers of bits in the network address as in class a, b or c; the class less addressing proposes to have any bits in the network addressing. Thus a classless address for a network may be in decimal dotted of the form a.b.c.d/n where n indicates the number of leading bits of 32 bits address that constitutes the network address bits. The address space must always be in the block of 2n. Thus in the present example for the 2000 hosts, a block of 211 = 2048 address space is required. The 11 bits will identify the hosts. So the 21 bits (as 32-11 = 21) may be used for network addressing of the organization in the form of a.b.c.d/21. Again the organization may use 11 bits for subnetting. The organization’s network address in this example may be: 110010000.00010111.00011000.00000000 (bold and underlines bits for network address and remaining 11 bits for hosts) → 200.23.24.0/21 The classless addressing has unique application in ISP (Internet Service Providers). Say one ISP has four class C addresses as 192.16.88.0, 192.16.89.0, 192.16.90.0 and 192.16.91.0. Instead of using four addresses for the ISP, one address ISP can use for Internet routing. In the present example, the address 192.16.88.0 can be used as the address and other addresses can be thought of a group of this address and can be recognized for Internet routing by the subnet mask 255.255.252.0. How? When a packet with the address of 192.16.90.xxxxxxxx reaches, the AND operation between 255.255.252.0 and 192.16.90.xxxxxxxx will result to 192.16.88.0 that is the address of ISP. This makes the Internet routing simpler. The question remains how the mask was selected? The two things to be considered for such cases are:: (1) as we pointed out earlier, the address block must be in the poser of 2 and (2) the staring address of the group must be evenly divided by the block size (why? This is because then only the block will be of in the power of 2). We assumed block size of 1024 (10 bits). Thus, the network address bit is 22, and the mask is 11111111.11111111.11111100.0000000000 (=255.255.252.0). BOX 9

QUESTIONS 1. What shall be binding rules for using subnetting/subnet mask? The following conditions must necessarily be meet with: (1) Each and every host/ node must have unique IP address, (2) Each and every network must have unique classful IP address, (3) Each and every network segment must use unique IP subnet address space, (4) All hosts/ nodes on a network segment must use same subnet mask, and (5) All hosts/ nodes on the same IP original network must use same subnet mask 2. Give a pictorial illustration of benefit of classless Inter Domain Routing (CIDR) compared to conventional routing. The Internet users and hosts are growing exponentially. To cope with such growing demand, the Internet routers are supposed to enhance at the same space but that was not practically feasible. Therefore to solve the problem of routing, the CIDR was proposed.

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The following Fig. (1) illustrates the benefit of CIDR in application of routers. In the class full routing the routers is supposed to have routing information/ mesh of users of an Internet Service Provider (Fig. 1a), where as with CIDR routing the class full addresses of all users of the service provider are integrated into one aggregate class less address as 195.65.0.0/16. the notation/16 or prefix /16 denotes the number of the mask bits. 195.65.70.0

195.65.70.0

195.65.70.1 195.65.70.254

Internet service provide

195.65.70.255

195.65.70.1 195.65.70.254

Internet global router

195.65.70.255 This block must be in routing table

(a) Routing with class full address 195.65.70.0 195.65.70.1 195.65.70.254

Internet service provide

195.65.0.0/16


A single integrated address for routing of all hosts under service provider

195.65.70.255

(b) Routing with class full address with explicit mask of 16 bits (notation/16) i.e mask is 255.255.0.0 Fig. 1: Illustration of CIDR

3. (a) How is an IP address broken into several subnets/segments? (b) In CIDR, the prefix can be of any length unlike fixed 8, 16 or 24 in classful address namely for class A, B or C respectively. What is its use? The following table (1) illustrates how an IP address is broken into number of segments. Table 1 The given class B address is 150.150.0.0

Network Segment as class C

Mask

150.150.0.0

255.255.255.0

Segment-0

150.150.1.0

255.255.255.0

Segment-1

150.150.2.0

255.255.255.0

Segment-2

150.150.3.0

255.255.255.0

Segment-3

150.150.4.0

255.255.255.0

Segment-4

150.150.5.0

255.255.255.0

Segment-5

150.150.6.0

255.255.255.0

Segment-6

150.150.7.0

255.255.255.0

Segment-7

Note that the all eight subnets are identified by a single address 150.150.0.0; and the segments/subnets are identified by third octet in the address notation.

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(b) The prefix can be of any length. For example it may be/22. In above table (1), we have converted a class B address to several class C segments. But what is then its use? Say an organization has about 1000 hosts. So a single class C IP address is insufficient. On the other hand a single class B IP address is supposed to provide IP addresses to 216 = 32 K hosts. Thus allocation of class B address to the organization will lead to poor utilization of address space. In such a situation, with CIDR the organization may be assigned four class C segments with prefix of/22. Thus not only routing simplification, but also the increased address space utilization is the major application of CIDR scheme. In this case subnet mask will be 255.255.252.0. Say the registered address is 193.193.0. x for the organization. Now all the four segments are identified with one address as 193.193.0, but segments are identified as 193.193.0.0; 193.193.1.0; 193.193.2.0 and 193.193.3.0. 4. Consider a INTERNET GLOBAL router as in Fig. (2). It receives one aggregate address as 195.65.0.0/16 from one neighboring service provider and another aggregate address 195.65.25.0/24 from another neighboring. Now two packets one with destination address 195.65.25.38 and another with 196.65.1.35 reach the router. How will the router route the packets? 195.65.0.0/16

195.65.25.0/24


Fig. 2: For question (5)

CIDR uses the principle of longest match for routing. The packet with destination address 195.65.25.38 will be routed as per routing entry of /24 as its match is longest. The packet with destination address 195.65.1.35 will be routed as per routing of /16. 5. Enumerate the rules for special addressing. The special addresses basically uses the following rules: • Internal host loop back address for a host refers to Net Id= 127, and host Id. (Example address = 127.45.78.90 is a loop back address for a host number 45.78.90) • This host on this network or Internet itself refers to Net Id = all 0s and Host Id = all 0s. (Address = 0.0.0.0) • A particular node on this network refers to Net Id= all 0s and Host Id. This is for a source address. (Example address = 0.0.0.34 for a class C network with host address as 34) • Broadcasting to all nodes on this network (Internet) refers to Net Id = all 1s and Host Id = all 1s. This is only for destination address.( Address= 255.255.255.255) • Broadcast to hosts of a particular network refers to Net Id and Host id = all 1s. This is for destination address only. (Example address = 86.255.255.255 for a class A address with Net Id = 86) • Broadcast to a particular subnetwork refers to Net Id, Subnetwork Number and Host Id = all 1s. This is for destination address only. (Example address= 150.150.5.255 all hosts of segment number 5 of class B network with address 150.150) • Broadcast to all subnets and their hosts refers to Net Id, Segment number = all 1s and Host Id = all 1s. This is for destination address. (Example address= 150.150.255.255) 6. Write a flowchart that will determine the class of a given IP address.

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Option fields of IP The option field in the IP header is optional. The options field is made of option type field followed by option length field and then the option data bytes as shown in the Fig. (26). Option type byte (1 byte)

Length byte (1 byte)

Option data field

Copied flag

Option classes

Option number

Fig. 26: Option field format

The option type is the first byte of option header and it has three fields: 1 bit copied flag, 2-bits class field and 5-bits number field. When the copied flag bit is “0”, the option field is not copied in the fragments of the IP datagram. But when the copied flag is set to “1”, the option field is copied in all the fragments of the IP datagram. The different option classes are as under: 00 → Control 01 → Reserved 10 → Debugging 11 → Reserved whereas the numbers are assigned as shown in table (XV): Table 15 Number

Descriptions

Bytes that follow

0

End of option list

Followed by 0 bytes, Total 1 byte

1

No operation

Followed by 0 byte, Total 1 byte

2

Security

Followed by 10 bytes, Total =11 bytes

3

Loose source and record routing (LSRR)

Followed by variable bytes, MAXIMUM total = 255 bytes

4

Internet Timestamp

Followed by variable bytes, Maximum total = 255 bytes

7

Record Route (RR)

Followed by variable bytes, Maximum total = 255 bytes

8

Stream Identification

Followed by 3 bytes. Total = 4 bytes

9

Strict Source and Record

Followed by variable bytes,

Routing(SSRR)

Maximum total = 255 bytes

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The end of option list is of type 0 as its option type byte has a value 0 (00000000). It belongs to class “00” This option field is used as the end of all options but not the end of each option. However it does not necessarily coincide with the end of all IP headers as per IP header length field. This is due to the fact of requiring padding to make IP headers a multiple of 32 bits words. The end of option field may be copied, introduced or deleted on fragmentation. The no operation option field is used between options for the purpose of alignment for 32 bits words. The field has a type equals to 1 (00000001) and belongs to control class “00”. The field may be copied, introduced or deleted on fragmentation. The Security option field looks as shown in Fig. (27). The option belongs to control class. The field has a type 130 as its option type field carries a value of 130 (10000010). The length field carries a value 11 (00001011) as its has total 11 bytes. The other fields are as below: 10000010

00001011

SSS .... SSS (2 bytes)

Option type byte Length byte (1 byte) (1 byte)

CCC .... CCC (2 bytes)

HHH .... HHH (2 bytes)

TCC .... TCC (2 bytes)

Actual option data bytes

Fig. 27: Security option field of type 130.

SSS Field: This is 2-bytes security field that specifies different levels of security as illusrated in table (16) for a few. Table 16: Different codes for SSS field Code

Description

Code

Description

00000000 00000000

Unclassified

00110101 11100010

Reserved

11110001 00110101

Confidential

01001101 01111000

Reserved

10101111 00010011

Restricted

00010011 01011110

Reserved

01101011 11000101

Top secret

11000100 11010110

Reserved

10111100 01001101

MMMM

10011010 11110001

Reserved

01111000 10011010

EFTO

00100100 10111101

Reserved

01011110 00100110

PROG

10001001 10101111

Reserved

11010111 10001000

Secret

11100010 01101011

Reserved

CCC Field: This is also a 2-bytes field. It is called compartment field. The field contains all 0s when data is not compartmented. The other codes are with Defense Intelligence Agency. HHH: This a 2-bytes field known as Handling Restrictions field This code of the field is available from defense agency. TCC Field: The field is known as Transmission Control Code and is of 3-bytes in length. It is used to segregate data and to define control. It is also defense controlled. The field must be copied on fragmentation. The Loose Source and Record Routing (LSRR) has the options field as shown in Fig. (28). The LSRR allows the source of an IP datagram to define routing information to be used by

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routers/gateways to forward the datagram to its destination. This is also used to record the routing information. Thus the options field provides the alternative routes. The routing is specified by recording the IP addresses of the routers and gateways in the data area of the options field. The routers and the gateways may use the information provided in the route data to route datagram and may not use their own routing tables for the same. In the Fig. (28), it is seen that its copied flag is 1 and it belongs to control class with type number 2. Alternatively it may be called type 131 option field as the type bears a value 131 (10000011). The length header indicates the number of bytes in the options field. The pointer field indicates the bytes after which the beginning of the next IP address to be processed into the route data. As the IP address is 4 bytes, the pointer minimum value is 4. The pointer value is basically increases by 4 at each router/gateway. The route data is made of a number of IP addresses of specified routers/gateways. The pointer value can be the maximum number of bytes in the route data. When the pointer is greater than the length of the route data, the router/gateways will consider their routing information for further transporting the datagram. This is because at this instant the routing information provided in the route data is exhausted. The options field is called loose because the routers and gateways are permitted to use any routing information either the routing information specified in the route data or routers/gateways’ own routing information. 10000011 (1 byte)

Length (1 byte)

Pointer (1 byte)

Route data (variable bytes)

Option type header

Fig. 28: LSRR options field

The Internet Timestamp options field has the format as shown in Fig. (29). The option type header belongs to copied flag as “0”, debugging class (10) with number 4 (00100). The option field is, therefore, type 68 as the type header has a value of 68 (01000100).The 1 byte length header specifies the number of bytes in the options field but a maximum value of 40. The 1-byte pointer header indicates the byte at which the time stamp is to be recorded. The minimum value of pointer is 5 as the first time stamp record must start 5 bytes after the pointer (1 byte for overflow + flags, 4 bytes for Internet address).. The 4-bits over flow header is used to hold the number of IP addresses (that may be routers, intermediate nodes, gateways etc) that cannot record the timestamp for want of space. The 4 bits lags are used to define different modes of recording timestamps, and these are: 0 for storing only the time stamps as 4 bytes word 1 for IP address followed by time stamp 3 for adding time stamps if only the gateway IP address is found in a specified table. In the mode, the IP addresses that can record the time stamps are prior specified in a table. Time stamp is a 4 bytes or 32 bits word that records value for the number of milliseconds since midnight under universal time frame. If non standard time record is made, any time can be recorded but in that case highest order bit of the time stamp is set to 1. The Internet address refers to the IP address of the source that has started the option. The source must reserve the sufficient bytes for records of time stamps. The size of the option does not change as the datagram traverses over the Internet. If the space is exhausted, an ICMP parameter Problem may be sent to the source.

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01000100 (1 byte)

Length (1 byte)

Pointer (1 byte)

Overflow (4bits)

227

Flags (4bits)

Internet Address (4 bytes) Time Stamps (4 bytes) ……… ……… Fig. 29: Internet Time stamps

Layer 3 or the transmission control protocol layer corresponds to layer 4 of the OSI/ISO protocol. It is transport protocol. The layer serves in connection oriented mode and guarantees the delivery of data. TCP provides the service of breaking messages into datagrams at the source end, retransmitting any datagram, which has been lost or acknowledged negatively, and sequencing the datagrams, etc. Service in the layer is guaranteed by acknowledgement of receipt of data from the receiving side. If the acknowledgement is negative, the data is retransmitted, and if the acknowledgement is not received with in a time period known as time out period, retransmission of the packet is done. This layer also performs multiplexing and demultiplexing functions if required. As we mentioned earlier, the TCP layer makes TCP pack using the data received from the application or higher layers. Each TCP pack is made of TCP headers and fragmented data (Fig. 30). A TCP pack has a number of header fields as shown in Fig. (31). Original data or payload data Data is converted into several fragments Data fragment 1

Data fragment 2

Data fragment 3

Data fragments are converted into TCP packs TCP header

Data fragment 1

TCP pack 1 TCP header

Data fragment 2


Data fragment 3

TCP PACK and TCP SEGMENT ate same and one


Data fragment 4

TCP pack 4

Fig. 30: Conversion of original data into TCP packs

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Data fragment 4

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Source Port (16 bits)

Destination Port (16 bits) Sequence number (32 bits) Acknowledgement number (32 bits)

Data offset (4 bits)

Reserved (6 bits)

U R G

A C K

P S H

R S T

S Y N

F I N

Checksum (16 bits) (TCP Header + Data)

Window (16 bits)

Urgent Pointer (16 bits)

Options (Variable)

Padding (Variable) Data (Variable) Fig. 31: TCP headers

5.7.2 Description of TCP headers Like IP, the TCP header is of minimum 20-bytes in length. The TCP header has the following fields: The 16-bits source port number of sending device. The port refers to a virtual circuit between two end-stations communicating parties. The port is also called socket or session. The concept of ports or sockets implies that more than one process can communicate over a session at any time between end-stations or nodes. TCP ports are documented in RFC 1700. A few examples are given in table (17). The 16-bits destination port refers to the port of the receiving device for communication of an application process. Table 17: TCP ports Port Number

Function

1

Multiplexing

5

RJE (Remote Job Entry) applications

9

Transmission discard

15

Status of the network

20

FTP data21FTP commands

22

SSH(Secure Shell)

23

TELNET applications

25

SMTP e-mail applications

37

Time transactions

43

Who is Protocol

53

DNS server applications

79

Finger protocol / Find active users’ protocol

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80

HTTP (Hyper Text Transfer Protocol)

93

Device Controls

102

SAP (Service Access Point)

103

Standardized e-mail services

104

Standardized e-mail exchanges

110

Post Office Protocol

119

Network News Transfer Protocol

139

NetBIOS applications/WINS

443

HTTPS secure WWW server

512

Berkeley commands

513

Login

543

Klogin (Kerberos Login)

544

Kshell (Kerberos Shell)

750

Kerberos Server

751

Kpasswd (Kerberos Password)

2105

eklogin (encrypted Kerberos login)

2049

NFS (Network File System)

229

The 32-bits sequence number( 0 to 4,294,967,295) field is used to assign a sequence number to each TCP pack. It is actually assigned to the first byte of the TCP message. The transmitter assigns the sequence number. The receiving on reading the sequence number of the packs ensures about whether all packs are received. Using sequence number, the packs received out of order are also placed back in order. Using sequence number the receiver also identifies the duplicate copy of the packs, if any received and accordingly takes the corrective measure. While IP sends the datagrams these may be routed through deferent links. This in turn may deliver the datagrams out of sequence at the receiver end. The datagrams are put in order by using sequence numbers assigned to them during transmission. This job is performed by the TCP layer. The sequence number and the acknowledgement number together ensure the reliable transport of TCP. TCP is reliable transport, whereas IP is unreliable transport. The 32-bits acknowledgement field is used to send the acknowledgement of the pack received correctly. The receiver performs this function. The receiver checks the sequence number of a received pack. If the pack is received without error, the receiver send an acknowledgement number using the sequence number of the pack to inform the sender that the pack has been received correctly. The 4-bits offset field is used to indicate the number of header fields in units of 32-bits words. The offset field is also known as header length or HLEN. If there is no padding and option fields in the header, the offset field will be 0101. This means there are five 32-bits words or 5 × 4 = 20 bytes header. This information, the receiver uses to determine the start of the data field so that data will be derived from the received pack.

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The flags, window and urgent pointer fields are used to connect and manage the TCP connection. The SYN (Synchronization), ACK (Acknowledgement) and FIN (Finish) flags are each a 1-bit flag. These are used to establish a TCP connection. TCP is a reliable and connection oriented protocol, the TCP session needs a virtual connection. This connection is established by a process called TCP handshake (Fig. 32). To establish a connection, the initiator or sender sends a TCP pack with the SYN flag set (i.e. SYN=1) and ACK flag set to 0. The calling program sends a TCP message through a port that has been allocated to it, like 21 for a FTP connection. The initial connection request TCP pack chooses a random number as the sequence number. Actually the connection requesting message is a message of one byte. Thus the random number provides the number to the calling sequence. This sequence number is called ISS (Initial Send Sequence number) for the receiver. If the random number is 200, this value is assigned to calling sequence. If and when the receiver receives this pack correctly, it responds by sending a pack to the sender with both the SYN and ACK flags set (i.e. SYN=1 and ACK=1). The receiver choose a sequence number by its random generator, and response message contains this sequence number. If the chosen number is 300 (IRS Initial Receive Sequence number for the sender), the reply message contains 300 in the sequence number field. The acknowledge number field contains a value 201 (as the received sequence number is 200). If the sender receives correctly the pack from the receiver, the sender sends another pack with ACK flag set (i.e. ACK=1), sequence number field set to 201, acknowledge number field set to 301, and the connection is then established. After the connection is established, the communication parties can perform transmission in full-duplex mode. The connection can be terminated by any station, either sender or receiver by sending a pack with FIN flag set (i.e. FIN=1). For example, after transmitting all data, the sender can send a pack with FIN flag set. When this pack is acknowledged by the receiver, the connection is terminated. SYN = 1 SYN = 1 and ACK = 1 Sender

ACK = 1

Receiver

Connection

(a)TCP handshake for establishing connection FIN = 1, say with sequence number = 100 ACK = 1 with say sequence number = 200 and acknowledgment number 101 Sender A

FIN = 1 with acknowledgment number = 101 and sequence number = 201

Receiver B

ACK = 1 with say sequence number = 202 and sequence number 101

The sender, A terminates by first two communications, the receiver terminates by last two communications; As if four way termination (b) TCP connection termination Fig. 32: TCP connection establishment / termination

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Initial Sequence Number(ISN) generation During connection request phase, the initial sequence number should be so chosen that the previous connections (sockets) are not confused with new connection requests (sockets). This typically happens when a host application crashes and before the other side times out, reestablishes the crashed connection quickly by recognizing that the connection request is for old socket. The ISN selection is made by a 32 bit random generator created during connection request phase. The number is generated by a 32 bit clock. The clock has a ISN cycle of 4.55 hours ( clock is incremented approximately every 4 ms), thereby providing unique ISN number within a period of 4.55 hours. BOX 10

QUESTIONS 1. As the sequence numbers, ISS and IRS are used for synchronization, where is then the need of handshaking for connection phase? The sequence numbers are local and not global. Thus the handshaking for establishing the connection is required by the process of the setting of the flags 2. How is it ensured that the sequence number is not duplicated? The host has to wait a maximum segment lifetime, known as MSL before re transmitting segments after the connection. 3. Sometimes the sequence number is addressed as the number allotted to each bytes of TCP message. Sometimes it refers to the first byte of TCP message. Again sometimes it refers to TCP message as a whole. What is the correct position? In practice each TCP pack is sent with a sequence number. But theoretically each byte is assigned a sequence number. When the sequence number is assigned to the first byte in the segment, that sequence number becomes the segment sequence number(SSN). That segment is transmitted with SSN. 4. How is the TCP reliable transport? The TCP is reliable transport as because: (a) the sequence number and the acknowledgement number together take care of the loss identification and recovery, and (b) advertisement of window by which the receiver announces the number of bytes may be accepted by it. To further illustrate the reliable transport of TCP, We shall look into the flow control mechanism used in TCP in the next section. The 16-bits window flag indicates the number of bytes that the sender can transmit without receiving acknowledgements from the receiver. This field is basically used by the receiver to inform the sender about the availability of the buffer in the receiver. Based on the information the sender control the flow. The flow control is actually implemented by the window field. The mechanism of control is discussed in details under the window flow control. The urgent pointer field is used to alert the receiver about coming of some urgent data and indicates to the end of the urgent data within the sequence of the transmission of packs. The value in the urgent pointer is valid provided the URG (URGENT) flag is set (i.e. URG=1). The urgent pointer value defines the end of urgent data and the start of normal data in bytes. For example if urgent pointer is 0000000010000000, the receiver will understand that next 128 bytes data will be urgent.

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The Reserved bits (6 bits) are for the future use. All the flags, namely URG, ACK, SYN, FIN, RST and PSH together are known as control field of the TCP. We have already discussed the important applications of URG, SYN, ACK and FIN flags. The PSH flag is set when the receiver is told to pass the data immediately to the application. Otherwise the receiver will buffer the segments until sufficient storage of data is made. RST flag is used to abort the TCP connection. When this flag is set, the receiver is advised to terminate the connection for some abnormal condition. We earlier discussed about the purpose of the options field in IP data gram. Nearly for that kind of purpose the options header is used in TCP pack. The options header can be maximum of 40 bytes, and always be a multiple of 4 bytes (32 bits words). To make it, a multiple of 4 bytes, padding may be used, if required. A few important options of TCP pack are: (a) MSS (Maximum Segment Size) option: this is used to indicate the maximum size of the segment that the sender can accept. This option is used during connection establishment phase, whereby the sender specifies the MSS. This is 16 bits option field. So the largest block that the MSS can specifies is 216 – 1 = 65,535 bytes. So, what is the largest data size that an IP/TCP pack can carry? A TCP pack or segment has minimum 20 bytes header and so is the case for IP header. Thus the largest block size of data that a TCP segment or IP datagram can carry = (65,535 – 40) bytes = 65,495 bytes. The declaration of MSS during connection set up time is made to overrule the TCP default MSS. The TCP default MSS is 536 bytes. (b) WSO (Window Scale Option): TCP pack has a header of 16 bits window. Hence the maximum permissible advertised window size = 216 – 1 = 65,535 bytes. With WSO, the larger advertised window size may be used for which upward scaling upto 214 is permissible. Thus, with WSO, the scaled advertised window size may go upto 214 × 216 – 1 = 1.073,725,440 bytes. (c) Times stamp option: This is used for round trip delay calculation and high speed connection. The round trip calculation is used for calculating time out period so useful for flow control. After discussion on sliding window protocol in the next section, we shall discuss about finding the time out period. 5.7.2.1 Sliding Window Protocol The window protocol has been discussed in connection with ARQ protocols of error control. The matter has further been discussed in chapter 3. 5.7.2.2 Time out Period The concept is used to determine the maximum size of packet.

SOLVED PROBLEMS 1. Compare the coding/bandwidth efficiency of IP datagram with the TCP default MSS with that of the largest MSS possible. Comment on the results. Solution. The TCP default MSS = 536 bytes. The minimum IP + TCP headers = 40 bytes. Thus the coding efficiency for the default MSS = {536/(536 + 40)} = 0.930. The largest block of data in MSS size = 65,495 bytes. Thus the coding efficiency in this case = {65495/(65495 + 40)} = 0.999

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Comment : We find that the coding efficiency is much higher in largest possible MSS. This may be one of the reasons of having the option included. 2. Is local IP authorized to fragment the data block under defined MSS? Solution. No. Actually once a MSS is defined by a sending process, the MTU is defined ( MTU = Maximum Transfer Unit in bytes = MSS + 40 bytes). With MTU size the data is transferred without fragmentation by the local IP. There are two variations of layer 3. These are UDP (user datagram protocol) and ICMP (Internet control message protocol). UDP is used when no sequence number is used when no sequence number is required, and a simple example of this situation would be when a message can put into one data gram. ICMP is simpler than UDP. In ICMP, the message is put into one datagram only. Besides, the message is for the network, and hence no addressing is required. ICMP is intended for TCP/IP software only. TCP divides the data into manageable packets that are easier to deal with and thereby provides a guaranteed error free data communication. Figs. (30 & 31) illustrate this. TCP breaks a NT service pack into several packets. If an error occurs, a small packet needs retransmission rather than whole pack. TCP uses a header (Fig. 31) to establish connection, to ensure successful transmission of a packet and completion of transmission of whole message. The source and the destination port fields keep the trace of which packet belongs to which application (table 16 shows application-level Internet Protocols and their assigned port number) so that running of e-mail, on-line chart and multiple browses can be allowed at the same time. Sequence number is used to reassemble the packets at the receiver. Receiver on getting a packets performs check-sum operation to detect error. If packet is received error free, a positive acknowledgement is sent. If packet is received erroneously a negative acknowledgement is sent. Transmitter either getting a negative acknowledgement within a time known as time out period, retransmits the earlier packet. This is ARQ (Automatic Repeat Request Protocol) [812] for error control. Flags, window and urgent fields are used to manage TCP connection. Flags are used to connect and terminate connection. Window field tells transmitter how fast to send packets. When receiver buffer is full, the window is set to zero. This is the means of regulating throughput by the transmitter by the receiver. Urgent field is used to send urgent information. When a receiver gets a packet with urgent field, it processes the packet with priority and acts accordingly like aborting a transmission. RFC 1240 refers to the UDP datagram. UDP datagram are used to carry the time sensitive data like voice and video. UDP works in connection less mode. UDP has very less overhead bits and overhead fields are very simple. The format of the UDP pack is shown in Fig. (33). Source Port (16-bits)

Destination Port (16-bits)

Length (16-bits) Checksum

(16-bits) Data

(Variable) Fig. 33: The format of UDP pack

5.7.3 UDP headers The 16-bits source port field gives the address of the process at the sender, whereas the process means the individual process that is in communication with the same process at the receiver.

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The 16-bits destination port refers to the address of the port at the receiver that is in communication with the same process at the sender. The 16-bits length field refers to the length of the whole of UDP pack in bytes. How does it then separates out the data field from the pack? This is simple as because the total of the header fields is fixed, and it is 8-bytes only. The 16-bits checksum field is used to check error in the frame. The checksum measure is taken over the whole of the frame. Problem. A UDP frame is given in Fig. (34). Calculate the check sum that will be filled in at the transmitter. Now if the same frame is received by the receiver correctly how the receiver will know about it? However if the frame is received as in Fig. (35), how the receiver will detect error? 0000111100001111

0101010101010101

0000000000001100

Checksum (16-bits)

0011001100110011

0011001100110011 Fig. 34

0000111100001111

0101010101010101

0000000000001100

Checksum (16-bits)

0011001100110011

1111001100110011 Fig. 35

The checksum will be calculated at the transmitter by adding 16-bits words in the UDP frames using 1’s complement addition rule as below: 0000111100001111 0101010101010101 -----------------------0110010001100100 0110010001100100 0000000000001100 -------------------------0110010001110000 0110010001110000 0011001100110011 ------------------------1001011110100011 1001011110100011 0011001100110011 ------------------------→ Final sum 1100101011010010→

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The checksum will be 1’s complement of the final sum so obtained. Thus in this case checksum field will be: 0011010100101101. The receiver calculates the sum as is done at the transmitter end, but it does so over the received bits. If there is no error in the received bits, the receiver will get the final sum as exactly it was obtained at the transmitter end. And in the present example, the receiver will get the final sum as 1100101011010010. With this final sum, the receiver will add the received checksum field as: below in the present case: 1100101011010010 0011010100101101 ------------------------1111111111111111 Now the receiver finds that there is all 1’s in the sum, and then it will conclude that the frame is received correctly. When the receiver gets the UDP frame as in Fig. (35), the calculated check sum will be different one from that of the transmitter. Hence when receiver’s calculated checksum will be added with the received transmitter checksum, the result will not be all 1’s. Hence the receiver will detect the error. The Commonly used port for UDP applications are as in table (18). Table 18: Commonly Used UDP Ports Port Number

Service

49 53 67 68 69 137 138 123 161 1645 1646

TACACS authentication server DNS (Domain Name Server) BOOTP server BOOTP client TFTP NetBIOS name server NetBIOS datagram service NTP (Network Time Protocol) SNMP (Simple Network Management Protocol) RADIUS authentication server RADIUS accounting server

2049

NFS (Network File System)

Layer 4 is the application and service layer; and this corresponds to the remaining higher layer of the OSI/ISO protocol. The services of the layer include FTP (file transfer protocol), remote login, computer mail and program to program communications using socket programming interface and RPC (remote procedure calls). FTP allows a user, a computer or a terminal to get (or to send) files from/to another computer. FTP is a tool of the Internet. Remote login is based on TELNET (network terminal protocol) which is another tool of the Internet. It allows any user to log in any computer in the network. Computer mail allows users to send mail to any other computer in the network, and one classic example of this is the E-mail. Each of the layers, while performing its function, adds a header to the message/datagram at the transmitting side ; and these headers are removed by the corresponding layer at the receiving side. Layer to layer function is a peer process.

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TCP/IP is the internetworking protocol. The Internet is an interconnected network of wide varieties of networks. Therefore the question may arise: How does the TCP/IP actually interconnect incompatible networks and provide compatible services? Interconnection among different networks is done by several methods: 1. By repeaters which cover only the physical layer, an example of which would be the connection of Ethernet to Ethernet. 2. By bridges which cover physical and data link layers, corresponding to the OSI/ISO protocol, an example of which would be Ethernet to token bus connection. 3. By routers which cover physical, data link and network layers (complete node coverage) corresponding to the OSI/ISO protocol, an example of which could be any IEEE802 series LAN to any X.25 WAN connection as shown in Fig. 36 Corresponding to TCP/IP, routers can cover up to the IP layer. 4. By gateways which are used to connection networks of different protocols, an example of which would be a connection between the OSI/ISO network and a DNA network. Repeaters, bridges and routers are used when interconnecting network are of the same protocol. Gateways, repeaters, bridge and routes are all made of computing resources. In fact, they are all computers. The complexity of the systems increases as one moves from repeaters to gateways. Bridges and routers can be defined as simple gateways. With the help of the Fig.36 we may illustrate the operation of TCP/IP in internetworking. Consider a situation that A wants to transfer some data to station B. TCP of station A sends datagrams with the necessary information of source and destination address to IP of station A. The IP layer attaches a global header to the datagrams. The header includes the internet global address. The global address has two parts-network identifier and station identifier. Till date, IP version 4 is being used wherever 32-bit addressing is used. IP version 6 is under consideration. In IP version 6, the addressing size is proposed to be higher. IP of station A now finds that the datagrams have a destination of another subnetwork, and therefore searches the routing table to route the datagrams to the corresponding router. In Fig. (37), we have shown only one router but in practice there may be more than one router to connect different subnetworks. In our example, IP of station A sends the datagrams to router (1) through lower layers. On receiving them, the IP of router (1) sends the datagrams to router (2). But to send to router (2), it has to utilize the X.25 network. Therefore ties among them would not cause any problem in communication. So we see that this ‘open system’ is really open. LAN-1

B

LAN-1

B

B G

Network LAN-2

B

LAN-4

B = bridge

G = gateway

(a)

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Long haul or wan


Higher layer

Higher layer

Network

Network

LLC

LLC

Data link

MAC

MAC

MAC

Physical

Physical

Physical

Data link

Bridge

LLC = Logical Link Control MAC = Medium Access Control Higher layer

Higher layer

Network

Network

LLC

LLC

Data link

MAC

MAC

MAC

Physical

Physical

Physical

Data link

Gateway

(b) Source

Destination Bridge

Packet from higher layer

MAC

802.3 Packet

Physical

802.3 Packet

Packet to higher layer

802 Packet

802.4 Packet

802.4 Packet

802.3 Packet 802.4 Packet

802.4 Packet

802.4 Packet 802.3 Packet

CSMA/CD LAN

TOKEN BUS LAN

Network A to internet

Network A

Network B to internet Buffer

Internet to network A

Internet to network B A full gateway

A full gateway

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Network B

237

238


Network A

Network A to internet

Network B to internet

Internet to network A

Internet to network B

Network B

Two halfways T

T T

802.5 LAN T T

G

G X.25 LAN

T

802.4 LAN

G T

T

T

T

G

T T

HG T

X.75 HG

B IMP T

IMP

T 802.3 LAN

X.25 T

HOST

G = Gateway HG = Half gateway T = Terminal / Node / Host.

B = Bridge IMP = Interface message Processor. (d) Fig. 36

BOX 11

A Write Up on Internetworking Internetworking Communication between any two entities may be direct or indirect. The point-to-point link or the multipoint link provides direct communication link; a switched network provides indirect

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communication facility. The case of more indirect communication arises when two entities which do not share the same network but may be connected through two or more networks desire to exchange information; and for this sort of indirect communication internetworking is required. The internetworking effectively creates a single very large loosely coupled network, which is often known as Internet. The internet in terms of internetting LAN (Local Area Network) and WAN (Wide Area Network) may be basically of three types (Fig. 1). Two or more LANs are internetted. Such Network is termed as extended LAN. The LANs in any extended LAN may be homogeneous or heterogeneous. LANs and WANs are internetted. Two or more only WANs are internetted. Such Internetted network is known as catenet.

Objectives The objectives of internetworking are many. The maximum number of stations attached to a single isolated network may be limited due to technical, legal or performance grounds. Internetworking effectively overcomes this limitation of individual network. Internetworking provides the ability to a user to share the resource of other networks. The geographic coverage of any isolated network is limited due to technical, economic or legal reasons. Internetworking overcomes such limitations of isolated networks. Internetworking may provide global coverage. Now naturally one can ask: why don’t we then design a single global network? The first answer to such a question will be that a global network is not viable due to administrative and political factors, which differ from country to country. The second answer is a technical one. A single network is not technically sound so far as operation, maintenance, reliability and flexibility are concerned. Moreover, until we have a standardized communication and computer systems and protocols (which is widely optimistic due to existing abundant networks of different types and the worldwide very high competitiveness of vendors in order to meet the interest of customers), any step to implement any global network will be an exercise in futility. The concept of global network for all purposes (Integrated Services Digital Network) is based on typical internetworking.

Gateways and their Classification As internetworking is to connect together usually the many different networks which are having different technologies, protocol and standards, it is necessary to use a system (a set of hardware and software) which will be able to remove the incompatibility of multiple networks in an internet. Such as a system is known as gateway in general. There are basically two factors that determine the key issues in designing a gateway, thereby providing as many as four types of gateways namely protocol translator. Internet protocol X.75 standard and Bridge. The two deciding factors are interfacing level and the nature of transmission services. The interfacing level may be of two types-station (DTE–Data Terminating Equipment) level and node (DCE-Data Circuit Terminating Equipment) Level (Fig. 1). As summing that the networks are packet networks, the nature of service may be again of two types-datagram service (connectionless service) which provides end-to-end service and virtual circuit service

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(connection oriented) which provides network by network connection. Hence 02 × 02 = 4 types of gateway systems exit (Fig. 2). The gateway are also known as data relay. Relays may be bilateral (connecting only two networks) or multi-lateral (connecting more than two networks). The simple gateway is a bridge used in internetworking LANs at the data-link layer of OSI (Open System Interconnection) model. The repeater may also be used in implementing extended LAN. But the repeaters at the physical link layer of OSI model copy the individual bits of a LAN and amplify and then transfer them to other LANs. Bridge only accept the frames from one LAN if these are to be transferred to other LANs; and operate in a store and forward mode. The bridge interconnects the similar networks, basically LANs. The Internet Protocol (IP) operates the above network layer. This was originally developed by the defense department of the US. Protocol converters operate at higher levels. Gateway are used in internetworking dissimilar networks; and operate at higher levels of OSI model in store and forward mode. While internetworking different networks, the networks may be widely separated. Then the question may arise: who will have the right to own and operate and maintain the gateways? For this each network may have a half-gateway. Similarly half-bridge is used in extended LAN.

LAN 802.3

G1

LAN 802.3

G1

G2

X.25 WAN

G2 Extended LAN

LAN 802.3

One type of catenet

LAN 802.5

G2

G3 Station level interfacing

Snanet wan

Dec net wan

• = Network Node. O = Network Station (DTE) G1 = Bridge G2 = Gateways (X.75) Standard Typically G3 = Protocol Translator or Internet Protocol Except A, All Other Are Of Node Level Internetwork Interfacing. Fig. 1: Internet in terms of Internetting LAN and WAN

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Service

241

End-to-end (usually used in LANs0)

Network-by-network (Usually used in WANs)

Station Level

Internet Protocol (IP)

Protocol Translator

Node Level

Bridge (mostly used in extended LAN)

Gateway (Example : X.75 Standard gateway functions for X.25 network)

Interfacing

← Connectionless →

← Connection

Gateway

oriented gateway

Fig. 2: Gateway System

5.8 ADDRESS RESOLUTION PROTOCOL (ARP) IP addresses often known as protocol addresses are unique logical addresses each assigned to a physical machine or system. IP layer releases the IP packet with IP logical addresses to the physical layer for the purpose of transport. The physical layer treats the whole of IP packet as a data pack and encapsulates this pack into the transport packet commensurate to the physical network. For example, if the transport network is an Ethernet LAN, the IP packet as a data pack is encapsulated in the Ethernet packet. Hence making the Ethernet packet, the destination address required in the Ethernet packet must be the physical address of the destination node. The physical address is the address provided by the NIC of the node. And it is only the physical address that transport network can understand for transporting the packet to the desired destination. So before making the Ethernet packet, the physical destination address is required to be known. Actually the software in a node determines the next node or hop based on the IP addresses in the IP packet. To transfer the IP packet to next node or hop, the corresponding physical address is required to be known. The address resolution refers to finding the physical address for a given IP address. Problem. The Internet task forces assign IP addresses and the IEEE committees develop link MAC (Medium Access Control) addresses of LAN. Why so many addresses? We could have just one address for networking. Solution. Note that different addresses are to serve different purposes. IP addresses for TCP/IP internetworking and MAC addresses are for linking devices in a network. Of course technically it is right that one address could have served the purpose. But in this complex world, that just did not happen. The address resolution is always local to the network. A computer can resolve the physical address of another computer from its IP address provided that computer is attached to the same network. A computer can not resolve the address of a computer attached to another network. For example consider the network in Fig. (37). In the Fig. (37), the physical networks, namely network 1, network 2 and network 3 are connected by routers R1, R2 and R3 . The Computer A can resolve the physical address of the computer B once its IP or protocol address is known as they are on the same physical network. But the computer A will not be able to resolve the physical addresses of the computers C, D , E and F as they are on the different physical networks. Now if the computer would like to send data to say F, the software of

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computer A will find that the computer F is not on its network, and hence will resolve the physical address of default router. See there are two routers, R1 and R3 on the network1. But assume that R1 is the default router. So the computer A will send IP packet meant for the computer F to the R1. The R1 will determine that the packets must go through R2, and accordingly resolve the physical address of R2, and send the packets to R2. The router R2 then will resolve the physical address of the computer F, and send the packets to the computer F. A

C

E

R2

R1

Network 1 B

Network 2 Network 3

D

D

R3

Fig. 37: Address resolution is local

Three techniques (Fig. 38) are employed for address resolution. The techniques are: Table look up, Closed form computation and Message passing/exchange. Address resolution techniques

Table look up (Used in WAN)

Closed form computation (Used in configurable networks)

Message passing/exchange (Used in LAN)

Fig. 38: address resolution techniques

In the table look up technique, a table known as ARP table or binding table is used for the address resolution. The table contains an array of data, each array is having protocol address and the corresponding hardware address. The table (18) is a typical ARP table. For each separate physical network, separate ARP table or binding table is constructed. Hence, the IP address entries in the table will bear the same network ID for all nodes. The table (19) is an ARP table for a class C network with network ID 230.120.92. As separate ARP table is required for separate physical network, the network ID (prefix of IP address) being the same for all nodes may be omitted from the ARP table in order to save memory space. The main advantage of the table look up is the simplicity in operation. Once an IP address is known, a search of the table will resolve the physical hardware address of the corresponding node or host or computer. When a network has less than a dozen hosts or nodes, a sequential search will suffice. For networks having higher number of hosts or nodes or computers, hashing or direct index search would be better solutions. The table look up technique is basically used in the WAN.

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Table 19: ARP table for a class C network having network ID 230.120.92 IP or Protocol Address

Hardware Address (Ethernet Address in this example)

230.120.92.2

00-80-C4-45-E3-87

230.120.92.3

00-80-E6-45-F4-4E

230.120.92.4

00-80-B3-89-56-7E

230.120.92.5

00-78-E8-45-12-E4

The closed form computation technique of address resolution is meant for the networks that use the technologies of configurable addressing. In the configurable addressing, the physical addresses are not static. In the closed form computation, a computational or mathematical or relation function exists between the IP address and the corresponding hardware address. For example, suppose a class B network with address 155.45.x.x is a configurable network. As and when computers are added to this network, the physical addresses to the computers may be assigned such that the 1’s complement of host’s ID in IP address becomes the corresponding physical address. Thus the host with ID in IP address 155.45.0.1 will have the physical address FF.FE or 11111111.11111110 (the host’s ID in the IP address 155.45.0.1 is 0.1 which actually means 00000000.00000001. The 1’s complement of this ID is 11111111.1111110). Thus the mathematical rule for resolving address is to get 1’s complement of host’s ID in the IP address as the physical or hardware address of the host.

QUESTIONS 1.

2.

In a closed form computation technique, the conversion formula is physical address = IP address ^ 00EF Find the physical address for a host with IP as 203.67.56.39 In a closed form computation technique, it is desired that host ID will be the physical address of host. Find the conversion formula.

The address resolution protocol (ARP) is basically used in the message passing technique of address resolution. The specifications of the ARP are documented in the RFC 826. As pointed out earlier, the message passing technique of address resolving is mostly used in LAN. The nodes in the LAN may work with the look up tables also. In that case the look up tables are known as the ARP cache. To resolve any address, the nodes first search their respective ARP cache. When the ARP cache fails to supply the necessary resolution, the message passing is evoked. The node then sends a broadcasted ARP request to all other stations of its network to find the link address of the target or destination IP address. Any station that recognizes the target IP address sends a reply to the inquiring node. The reply contains the physical address of the target IP address. The technique is illustrated in the Fig. (39). In the Fig. (39) , the node “A” sends an ARP request and the node “D” sends the ARP reply.

A

B

Inquiring Node 130.24.45.90 00-89-FE-10-79-06

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D

Destination or Target Node 130.24.78.45 00-45-D4-A0-89-78

E

F

244


Broadcasted ARP request from inquiring node, A ARP request

Source Hardware Address (00-89-FE-10-79-06)

Source Protocol Address (130.24.45.90)

Target Hardware Address (FF-FF-FFFF-FF-FF)

Target/Destination Protocol Address (130.24.78.45)

A → All (broadcasted/ FF-FF-FF-FF-FF-FF) Here is the request. What is the physical address of this target protocol address ? ARP reply from the node, D that has recognized the target or destination protocol or IP address ARP request

Sending Hardware Address (00-45-D4-A0-89-78)

Sending Protocol Address (130.24.78.45)

Target/ Destination Hardware Address (00-FE-10-

Target/Destination Protocol Address (130.24.45.90)

79-06)

89-78.

D → A Here is the reply. Well the physical address of your target /destination is 00-45-D4-A0Fig. 39: Illustration of ARP request/reply

The ARP message format is shown in Fig. (40). The destination address and the source address refer to respectively the Ethernet (if the network is Ethernet) destination and source MAC address. Ethertype field is very important. This field actually identifies what the type of the data in the frame. The Ether type assignments for ARP and RARP (Reverse ARP discussed latter) are as below: Assignment

Decimal number

Hex number

ARP

2054

0806

RARP

32821

8035

The hardware address space is 2 bytes. This is set to a value of 1 for Ethernet. The protocol address field is 2-bytes. For IP this field is set to a value of 2048. The hardware length field defines the length of the MAC address in bytes, which is typically 6 bytes. The protocol length field defines the length of the protocol address in bytes, which for IP is 4 bytes. The opcode field contains a value of 1 for ARP request and a value of 2 for ARP reply. The next four fields will be what we have described in the previous Fig. (39) in illustrating the ARP request/ reply.

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Hardware Type Hardware Length

245

Protocol Type Protocol Length

Operation Request 1, Reply 2

Sender hardware address Sender Protocol address Target hardware address (It is not filled in a request) Target protocol address

(a) ARP Packet Destination address (6 bytes) Source address (6 bytes) Ether type (2 bytes) This ethernet data packet

Hardware address space (2 bytes) Protocol address space (2 bytes) Hardware address length (1 byte) Protocol address length (1 byte) Op code ( 2 bytes) Sending hardware address (6 bytes)

This actually ARP DATA field as shown in Fig. (39)

Sending protocol address ( 4 bytes) Target/destination hardware address (6 bytes) Target/destination protocol address (4 bytes)

(b) ARP Message Format C B

F LAN 1 Router A

E LAN 2 D

(c) Proxy ARP Fig. 40: ARP Illustration

5.8.1 Proxy ARP The ARP techniques illustrated in section (5.8) are based on address mapping procedure. Proxy ARP is a flexible technique in resolving addresses. The proxy ARP works as in Fig. (40 C). In the figure, it is seen that two local networks are connected by a router. The router hides the

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networks connected in it, from each other. If the host A likes to send data to the host F, the host A will issue an ARP request to get the physical address of the host F. The technique does not allow the ARP request to reach the host F. The router intercepts the ARP request, and sends a reply of ARP request with the router’s own physical address. The host A on receiving the reply, processes it and sends the data. Thus the router in the example acts as a proxy ARP server. This technique will work only the network has the installation of the proxy ARP.

5.9 REVERSE ADDRESS RESOLUTION PROTOCOL (RARP) RARP is just reverse of ARP. The RARP protocol is used to find the protocol or IP address of a station whose link or physical address is known. The RARP is not so commonly used as ARP as because most stations or computers known their IP address as they have hard disk to store the same. The RARP is required for the stations having no hard disk like terminals or diskless workstations. . The RARP works like ARP, but to run a RARP service a RARP server is a must in the network. The RARP server maintains a mapping table like a look up table that was discussed earlier. In the ARP all nodes are at par. But in the RAR, the service is client–server oriented. The specifications for RARP are defined in RFC 903. The inquiring node sends a broadcast RARP request. The RARP server recognizes the request and searches its data base to find the protocol address for the given physical address of the inquiring node. The server then sends a RARP reply to the inquiring node. The operation is illustrated in the Fig. (41). When the node A is sending RARP request, the destination hardware address will be the hardware address of the RARP server, but the destination protocol address will be broadcast which is net ID plus all 1s in the field of host ID. In the example the network being a class C IP network, the last byte is thus made 255. The message format as in ARP is used in the RARP except that the opcode now changes. The op code is 3 for RARP request and 4 for RARP reply.

A

B

C

00-Ad-EE-67-45-89 00-67-89-09-56-AC/230.89.67.59 RARP SERVER

The data base of server Physical address

Protocol Address

00-45-EF-B4-D6-89 00-56-78-90-A4-BD 00-AD-EE-67-45-89

230.89.67.56 230.89.67.57 230.89.67.58

00-D8-C8-78-99-65

230.89.67.60

A is sending RARP request RARP request

Source Hardware Address (00-AD-EE 67-45-89)

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Source Protocol Address (???????)

Destination Hardware Address (00-67-89-09-56-AC)

Destination Protocol Address (230.89.67.255)


247

RARP server is sending reply to A RARP reply

Source Hardware Address (00-67-89

Source Protocol Address

Destination Hardware Address

Destination Protocol Address

90-56-AC)

(230.89.67.59)

(00-AD-EE-67-45-89)

(230.89.67.58)

Fig. 41: Illustration of RARP

BOX 12

QUESTIONS 1.

2. 3. 4. 5. 6.

One of the reasons for using leading bits to classify IP addresses into different classes instead of using a range of value is that the use of bit can decrease the computational time. Hence a table of first four bits may be used to compute the classes of address. Design table. Is it advisable to redesign IP address with higher bit size (> 32 bits) to eliminate the address classes? Give a comparative table of different ARP technique in term of your defined parameters. It is often argued that “message passing ARP is hopelessly inefficient” ----------------- why? Suggest a solution to overcome the problem. ARP is called local. Can it be remote? If more than one different ARP replies with different hardware addresses are received for a single ARP request; how will it be processed?

Hints for Question (1) First Four Bits of Address

Table Index (in decimal)

Class of Address

0000 0001 0010 0011 0100 0101 0110 0111 1000 1001 1010 1011 1100 1101 1110 1111

0 1 02 3 4 5 6 7 8 9 10 11 12 13 14 15

A A A A A A A A A A A A A A A A

Table that can be used to compute the class of an address. The first four bits of an address are extracted and used as an index into the table.

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5.10 IPV4 TO IPV6 5.10.1 Ipv4 addressing review IP provides the basic service of getting the datagram to their destinations. It provides this service in best effort protocol. IP on receiving the TCP packets, forms it own packet known as IP packets. Each IP packet is associated with IP header. IP header has source IP address and destination IP address. These addresses are used to route the datagrams. IP addresses are universal machine identifiers, which are shared by all the machines in the Internet. IP uses 3 modes of addressing[2-6] for the purpose of routing packets. These are known as class A, class B, and class C. An IP address is a 4 bytes or 32 bits binary number. IP address has a Network Address Field (often known as Network ID) and Local Address Field (often known as Node or Host ID). Network address ID identifies the network to which a node or host is attached, and host ID identifies the individual node or host attached to the network already identified by the network ID. The difference in classes is due to how many bits are used for network identification and how many bits are used for host identification. The different address formats under different classes and the address space under different classes were illustrated in earlier sections. The different classes are identified by the left most bits of the address: for class A left most bit is 0, for class B left most two consecutive bits are 1 & 0 and for class C left most three consecutive bits are 1, 1 & 0. The 4-bytes of an IP address are usually mentioned as four sets of dot-separated octets in decimal notation. For example 10.0.16.9 is a valid IP address, and in binary notation this is: 00001010.00000000.00001000.00001001. The address in the example refers to class A address as the left most bit is 0. This address points to a network number 10 (the decimal value of first octet), and to the host or node number 4105 (the decimal value of second, third and fourth octets taken together as a binary number) attached to network number 10. For class A, most significant bit of first octet is always 0. Therefore next seven bits are used for network identification, and this means that only 27 = 128 networks can be of class A. As address 0.0.0.0 is used for system initialization, the class A has network address field from 00000001 to 01111111, that means from 1 to 127. But the packets with first octet as 127 is used for network testing. As an example: the address 127.0.01 is used for loop back address. Thus address for the network 127 is not open to use. Therefore for addressing purposes, class A uses network addresses from 1 to 126. Each of the network under class A address will have node or host address field of 3 bytes or 24 bits. Each network may have maximum number of nodes or hosts equal to 224 . The identification of a network under any class is made by filling the local part or node/ host field of address with zeros. Therefore the following addresses are reserved (x is a binary bit and may be either 0 or 1): Class A: 0xxxxxxx.00000000.00000000.00000000 = Network ID.0.0.0 Class B: 10xxxxxx.xxxxxxxx.00000000.00000000 = Network ID.0.0 Class C: 110xxxxx.xxxxxxxx.xxxxxxxx.00000000 = Network ID.0 Again the broadcast, under a specific network, is done by filling local part or host/node field of address with ones. Therefore the following addresses are reserved: Class A: 0xxxxxxx.11111111.11111111.11111111 = Network ID.255.255.255 Class B: 10xxxxxx.xxxxxxxx.11111111.11111111 = Network ID.255.255 Class C: 110xxxxx.xxxxxxxx.xxxxxxxx.11111111 = Network ID.255 These mean that for identification of networks and for internal broadcasting in networks, two numbers of host or node addresses under each class are kept reserved. Thus the host or node addresses available under class A = 224 – 2 = 16777214 – 6 = 16777214

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For class-B address, first two bits of first octet must be 1 & 0, and these make first octet to be from 10000000 to 10111111 i.e. from 128 to 191. The second octet will be from 00000000 to 11111111 i.e. from 0 to 255. The first and second octet in combination provide Class B network addresses to range from 1000000000.00000000 (or 128.0) to 10111111.11111111 (or 191.255). Total networks and hosts/nodes under a network of address class B are respectively 214 and 65534 (=216 – 2). For class-C address, first three bits of first octet is 110. This means the first octet will be form 11000000 to 11011111 i.e. from 192 to 255. Class C network address will range from 11000000.00000000.00000000 to 110111111.11111111 i.e. from 192.0.0 to 223.255.255. Class-C address has 221 networks and 254 (=28 – 2) hosts or nodes per network. The total maximum hosts under different addressing modes are: Class A = (27 – 2 ) × (224 – 2) = 126 × 16777214 = 2113928964 Class B = 214 × (216 – 2) = 16384 × 65534 = 1073709056 Class C = 221 × (28 – 2) = 2097152 × 254 = 532677708 The address scheme under class-A to class-C is for unicast communication. The other addressing schemes are for multicast communication. Class-D belongs to the multicast scheme. Multicasting allows a stream of data to be sent simultaneously to a designated subset of network users. This is a very effective way of transmitting data to many receivers. This is contrast to uni-casting and broadcasting. In uni-casting, separate data packet is sent to each receiver. In broadcasting, all packets are sent to everyone. Class D addresses are identified by four left most consecutive bits as 1110. In class-D, all the four octets are used to identify the group of nodes designated to receive a multicast. Class-D address does not specify the network. Class-D addresses are in the range of 11100000.00000000.00000000.00000000 to 11101111.11111111.11111111.11111111 i.e. 224.0.0.0 to 239.255.255.255. Class-E addresses are identified by the consecutive four left most bits as 1111 and next bit as 0, a n d t h e r e f o re the addresses range from 11110000.000000.00000000.00000000 to 11110111.11111111.11111111.11111111 i.e. from 240.0.0.0 to 255.255.255.255. But 247.255.255.255 is used for broadcast for all networks. Class E address is still under experimentation. The special addressing schemes are as below: 1. 255.255.255.255 is used as a broadcast packet for all networks. 2. For broadcast in a specific network, local part or Node/Host field of the address is set to 255. 3. IP address 0.0.0.0 is used for system initialization. IP address 0.0.0.0 is called “This Host, This Network” address. 4. Packets with first octet as 127 are used for network testing. 5. IP address 127.0.0.1 is the Loop back Address. 6. An entire network is specified by providing only the network identification and with 0s in all other octets such as: 124.0.0.0 for class A, 129.155.0.0 for class B and 200.127.11.0 for class C. 7. Subnet mask is special addressing scheme. It is used for two purposes: to show the class of addressing in use, and to divide a network into different sub networks to control traffic. For first purpose, a subnet mask determines which part is for network ID and which part is for hosts ID. A subnet mask for class A network is: 11111111.00000000.00000000.00000000 (=255.0.0.0). All 1s indicate network ID and all 0s indicate host ID. For second purposes, subnet mask is used to divide the network within network administrator. Always a network number is given to an organization.

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If an organization has applied for a Class B address, the organization may be given a number 140.80.X.X. Now 140.80 is what that bothers to Internet regulator. What your organization does with host/node field of address (X.X) that is up to your organization. For example: when you are given a number 140.80.X.X; you may use the entire third octet of this address space for subnet. In that case your mask for subnet identification will be 11111111.11111111.11111111.00000000 (=255.255.255.0). Another example: of subnet ID mask under class B could be: 11111111.11111111.11110000.00000000 (= 255.255.240.0). In this example 4 left hand side bits of the third octet are used for subnet networks ID, whereas other 4 bits plus last octet is kept for host ID. The use of subnet ID is to expand your organizational network. Natural question may then arise: Why Subnet and Why not different network ID. Network space is limited, therefore it is not desirable to allow your organization, say 10 to 20 network address-space. Second allowing a number of network addresses to a particular organization creates complexity in routing table. But with subnet concept, as the subnet is under your network control, it will not hamper the main routing table of the Internet.

5.10.2 Ipv6 Addresses The address scheme defined so far is up to IP version 4 (Ipv4). Ipv4 is facing some serious problems. First, Ipv4 has 32 bits address scheme. But due to exponential growth of Internet users[11], literally this address space is running out of addresses. In IP Next generation (Ipng) or IP version 6 (Ipv6), 128 bits address space (in place of 32 bits of Ipv4) and 40-byte IP header (in place of 20 byte of Ipv4) have been proposed to cope up with the increased demands. IP addressing with 128 bits or 16 bytes is called IP version 6 because of the reason illustrated below: 2 (or 4 bytes) bytes address space refers to IP version 4, then 23 bytes address space may refer to IP version 5 if any one there, and therefore 24 bytes address space would refer to IP version 6. Second, due to huge growth of Internet users, organizations are being assigned C addresses. This is causing exploration in routing table of Internet. Third, Ipv4 is for best effort delivery of packet. It does not guarantee packet sequence integrity and consistent latency in delivery, and hence inherently unsuitable for real time services. Thus with Ipv4, voice, video and multimedia services will not be possible at the required level of quality. On the other hand, growth in voice IP traffic is tremendous. Ipv6 is being explored to avoid the problems being faced by Ipv4 to support real time services like voice, video and multimedia services. Ipv6 is proposed to have the following features: 1. 128 bits address size. This means that a total of 2128 = 256 × 2(10 × 12) = 256 × 10(3 × 12) = 256 × 1036 different addresses would be available in the address space. The inefficiency in the allocation and administration of the address space is measured by the H factor. The H factor is defined as the ratio between the log (number of addresses) and the number of bits in addresses. H factor is usually 0.22 to 0.26. Taking into account the H factor, Ipv6 is believed to support 1015 or quadrillion of networks and 1012 or trillion of networks.

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2. The notion for writing Ipv6 address is each double bytes of the 16 bytes field are separated by colon. The bytes are written in hexadecimal symbols. Thus one example of the address could be: ABCD:23D5:7893:C07E:3425:9BAC:6754:CED6. The proposed notation has the provision to skip leading zeros so that 0000 can be written as 0 or 0056 can be written as 56. It has also the provision of removing a 0 leaving the colons by the technique of double colon notation. For example, an address like ABCD:0:0:0:0:0:7896:DE45 can be written as ABCD::7896:DE45. The double colon notation can be used at the beginning and at the end of the address but only once. In the transition period Ipv4 address would be converted to Ipv6 address by pretending 12 bytes of 0s. Thus an Ipv4 address 102.23.78.10 would be written an Ipv6 address as 0:0:0: 0:0:0:0:0:0:0:0:0:102.23.78.10 or ::102.23.78.10. Ipv6 has a provision of a single address associated with multiple interfaces, 3. Ipv6 has auto configuration facility, 4. Ipv6 provide QoS (Quality of Service) service support to real time services like voice and video, support to mobility, 5. The flow level and priority in the header of Ipv6 facilitate the support of real time data, 6. Ipv6 has an efficient header format. In Fig. (42) we find overhead bits of Ipv6 are less than that of Ipv4. The overhead bits in Ipv4 is 12 bytes in the header format of 20 bytes, whereas the overhead bits in Ipv6 is 8 bytes in the header format of 40 bytes. The less header in Ipv6 helps in achieving higher data rates required for voice and multimedia services [13]. 7. The header of Ipv6 is different from that of the Ipv4 in many aspects: (a) Ipv6 has a fixed header size unlike Ipv4. Options and padding are the variable fields in Ipv4. These are removed in the Ipv6. This makes Ipv6 to act like ATM (Asynchronous Transfer Mode) cell comparison to Ipv4 which acts as a conventional packet. (b) There is no header checksum in Ipv6. This modification is made based on experience that there is no need of checksum at each intermediate node. (c) there is no hop by hop segmentation procedure in Ipv6. Therefore there is no fragment offset, flags and identification fields in the Ipv6. There is also no need of header length field as with next header field the chaining of IP headers is possible in Ipv6 (Fig. 43). (d) the TOS (Type Of Service) field of Ipv4 is removed in Ipv6. This removal is also based on experience. Hardly the TOS field is used in IPV4. 8. Ipv6 defines six extension headers (Fig. 43): Hop by hop options header, Routing header, Fragment header, Encrypted security payload header, Authentication header, and Destination options header. 9. Ipv6 has different options, security, and easy transition facility. We can compare Ipv6 with Ipv4 in terms of two important parameters, namely address space and IP header coding efficiency among others. Like gain bandwidth product of the amplifiers, the address space (AS) and header coding efficiency (CE) relative to address space only, could be used for comparing the versions. The comparison is shown in Table (20): Table 20: Comparison of Ipv6 with Ipv4 in terms of AS X CE Versions

AS (a rough estimate) 109

Ipv4

232 = 4 ×

Ipv6

2128 = 256 × 1036

The achieved benefit is in the ratio of 1:128 × 1027.

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CE

AS × CE

8/20

1.6 × 109

32/40

204.8 × 1036

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5.10.3 How Did IPv6 Come Up Although a few people are projecting the death of the Internet due to increased traffic, yet Internet access is growing exponentially over time and is providing the better services. Forget about the death of the Internet, a next generation gigabit Internet was proposed in 1996 and is being experimented now. Internet is on no-stop move from kilo bits per second to giga bit per second/tera bits per second. The world may enter into the age of the Internet2 in future. Internet2 shall be next generation of the Internet, when the present Internet may have to be designated as Internet1. It is often said that Quality X Quantity = Constant , meaning that if quantity increases, the quality is bound to fall and vice-versa. But this theory has failed in case of the Internet. As per research firm International Data Corporation of USA “the number of people accessing the World Wide Web will hit almost 100 million by the year-end 1998 and 320 million by 2002.” John S Quarteman, President of Matrix Information and Directory Services said ” The quality of Internet is actually getting better – not worse ……… the number of servers supplying information increased rapidly and massively to cope with the demand for the information. In this way Internet is even better than traditional media” Craig Partridge, Chief Scientist, BBN Technologies, Cambridge said” Recent years have seen a tremendous focus on improving access to the Internet. We’ve seen a push to make the Net easier to use, and to improve data rates with 56-kb modems and with new access technologies like cable modems and asynchronous digital subscriber loop.” Commenting over Internet business in Jauary’99, IDC Senior Vice President John Gantz said “ Within five years, every dollar of Internet investment in the United States will be paying back $1.50.” Internet is going to have several changes and version like IPv4 to IPv6, Internet2, Wireless Internet and Cable Internet. But most immediate is from IPv4 to IPv6. Presently, Internet works on IPv4 (Internet Protocol Version 4) as defined in RFC791. By the middle of 1990s, by the time of which the IPv4 became about 15 years old, it was recognized that there are several limitations in the IPv4. Table (XXI) lists the major studies on the run up of IPv4. Two important limitations are the inadequate address space available with 32-bit address space of IPv4 and inability of the IPv4 to support real time services or time-sensitive services. The 32-bit address space is not sufficient to cope up with the growing Internet users. Since it is estimated that the Internet has been growing by a factor of two every year, the underlying principles and assumptions based on which IPv4 was designed are going to be invalid. What was duly sufficient for a few million users or a few thousands of networks will no longer can support a world with tens of billions of nodes and hundreds of millions of networks. Inability of IPv4 to support real time services was the stumbling block to realize Internet telephone. IPng (Internet Protocol Next Generation) initiative (RFC 1752) was then, started by the Internet Engineering Task Force (IETF). By 1996, the IETF proposes IPv6 (Internet Protocol Version 6) under IPng initiatives, which is supposed to solve the problems of IPv4 including the two major limitations mentioned above. IPv6 is therefore the future replacement of IPv4. From the experience over IPv4, it was felt that new version should take care of: More addresses, Reduced overhead, Better routing, Support for address renumbering, Improved header processing, Reasonable security and Support for mobility. Under the IPng initiatives the main techniques investigated were: • TUBA that refers to TCP (Transmission Control Protocol) and UDP (Users’ Datagram Protocol) with bigger addresses • CATNIP that means common architecture for the Internet. The main idea is to define a common packet format that will be compatible to IP, CLNP (Connectionless Network Protocol) and IPX (Internet work Packet Exchange). CLNP has been proposed by OSI (Open System Interconnection) as a new protocol to replace IP, but never been adopted because of its inefficiency

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• SIPP (Simple Internet Protocol Plus) that proposes to increase of the number of address bits from 32 to 64, and to get rid of unused fields of IPv4 header As none of the above three was seen to be suitable. As such, a mixture of all these three along with other modifications was suggested in RFC 1883. The RFC 1883 suggested the modifications as below: • Expanded Addressing in suggesting 128 bits for address that may allow more levels of address hierarchy, increased address space and simpler auto configurable addressing • Improved IP header format by dropping the least used options • Improved support for Extensions that will bring flexibility in operations • Flow Label that will make the real time services possible over Internet Based on the experience gained in operation of IPv4 over about 20 years, the design of IPv6 has considered four major simplifications: • assigning fixed format to each header. This ensures the removal of header length field that is essential in IPv4 • removing header checksum. The main advantage in removing header checksum is to diminish the cost and the time delay in header processing. This may cause the data to get misrouted. But experience has shown that the risk is minimal as most of data pack is encapsulated by the packet checksum at other layers like MAC (Media Access Control) procedure in IEEE 802.X and in adoption layer of ATM (Asynchronous Transfer Mode) etc. • removing the hop by hop segmentation procedure • removing TOS (Type Of Service) field that IPv4 provides, since experience has shown that this field has ever been set by applications. On the other hand, IPv6 has considered two new fields, flow label and priority. These are included to facilitate the handling of real time services like voice, video and high quality multimedia etc.

5.10.4 IPv6 in Details Thus the IPv6 was finally come up with packet format as in the Fig. (42). The final specifications of IPv6 was produced in RFC 1883. The new features of IPv6 are: • A fixed and streamlined 40-byte header: IPv6 is having fixed header bytes like that in ATM (Asynchronous Transfer Mode) cell. This makes the node processing delay to minimize, and thereby becomes more suitable for real time services like voice, video and multimedia. • Expanded addressing capabilities: A 128-bit address space in IPv6 instead of 32 bit as in IPv4, is believed to ensure that the world won’t run out of IP addresses. The 128 bit address size gives rise to a total of 256 × 10 36 different addresses. It is expected the Internet under IPv6 to support 10 15 (quadrillion) hosts and 10 12 (trillion) networks. The Internet under IPv4 can support maximum 2 32 hosts. Therefore the IPv 6 address space is about 64 × 10 9 times more that that of IPv4. This is why it is expected that future and exponential growing demand for Internet connection be met with IPv6. • New Address Class: Besides unicast and multicast, IPv6 has the provision of anycast addressing. Anycast address allows a packet addressed to an anycast address to be delivered to any one of a group of hosts.

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• A single address associated with multiple interfaces • Address auto configuration and CIDR (Classless Inter-domain Routing) addressing • Provision of extension header by which special needs like checksum, security options may be introduced. • Flow labeling and priority: Flow level and priority headers are used to comfortably support the real time services. By assigning higher priority to the real time packets, the necessity of time sensitiveness is restored. Data packets and for that purpose time insensitive packets are assigned low priority and serviced by the best effort approach. As per RFC 1752 and RFC 2460, this new feature allows “ labeling of packets belonging to particular flows for which the sender requests special handling, such as a non-default quality of service or real-time service.” Hence video and audio may be treated as flows whereas traditional data, file transfer and e-mail may not be treated as flows. • Support for real time services • Security support that could be eventually seen as the biggest advantage of IPv6. Today, billion dollars business is done over Internet. To keep the business secure, public crypto system has emerged out as one of the important tools. IPv6 with its ancillary security protocol has provided a better communication tool for transacting business over Internet • Enhanced routing capability including support for mobile hosts. IPv6 as such is not simple extension of IPv4, but a definite improvement over IPv4 in order to meet growing demand of Internet connectivity and the services of real time communication via Internet. Version (4 bits)

Priority (4 bits)


Flow label (24 bits) Next Header (8 bits)

Hop Limit (8 bits)

Source Address (128 bits) Destination Address (128 bits) Variable length TCP pack/ UDP pack (which is TCP header + Payload or UDP header + Payload)…….. Fig. 42: IPv6 Packet Format

The functions of IPv6 headers that is of base headers of fixed 40 bytes are: • Version field (4 bits): It contains the version number. Versions are 4 and 6. For version 6, this field is 6 (i.e. 0110). The various assigned values for IP version label are shown in table (XXI). But it must be remembered that just putting a number “6” or “4” does not make the corresponding IP packet. For the corresponding IP packet the proper format is required to be made. • Priority (4 bits): The bits in the field indicate the priority of the datagram. The priority levels are 16 from 0 to 15. The first 8 priority levels (from 0 to 7) are for the services that provide congestion control. If the congestion occurs, the traffic is backed off. These are suitable for non-real time services like data. The different priority levels under the first 8 levels are: 0 that defines no priority, 1 that defines background

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traffic like Netnews, 2 that defines unattended transfer like e-mail, 3 remains reserved, 4 that defines attended bulk transfer like FTP (File Transfer Protocol), NFS, 5 remains reserved, 6 that defines interactive traffic such as Telnet, X-windows, and 7 that defines control traffic such as SNMP (Simple Network Management Protocol) and routing protocols. The higher 8 priority levels (from 8 to 15) are used for services that will not back off in response to congestion. Real time traffics are examples of such services. The lowest priority level of this group 8 refers to traffic that most willing to be discarded on congestion and the highest priority level 15 is for traffic that is least willing to be discarded. • Flow level (24 bits): It is proposed to be used to identify different data flow characteristics, which will be assigned by the source and can be used to label packets. The packet labels may be required to provide special handling of packet by IPv6 routers, such as defined quality of service (QoS) or real time services. The combination of the sender IP address and the flow label creates a unique path identifier that can be used to route the datagrams more efficiently. The field is still being experimented. Flow is actually a sequence of packets coming from a particular source and destined for a particular destination. A flow may require a special handling by routers. Each flow is uniquely defined by the combination of the source address and a non-zero flow label. The flow label can be from (000001)H to (FFFFFF)H in hex. The packets having no flow label are given a zero label. All packets in the same flow must have same flow label, same source and destination addresses and same priority level. The initial flow label is obtained by the source by pseudo random generator, and the subsequent flow numbers are obtained sequentially. • Payload length (16 bits): The field indicates the total size of the payload of the IP data gram that excludes header fields. It can define up to 65,536 bytes of payload. • Next header (8 bits): The field indicates which header follows the IP header. The next header can be either one of the optional extension headers used by IP or the header for an upper layer protocols such as UDP or TCP. The field defines the type of extension header. For example 0 defines IP information, 1 defines ICMP (Internet Control Message Protocol) information, 6 define TCP information, 44 defines fragmentation header, 51 defines authentication header and 80 defines ISO (International Standard Organization) /IP information. Each extension header again contains an Extension Header Field and a Header Length Field (Fig. 43). When there is no other extension header, the next header will be TCP and hence the next header field will contain 6. The length of the base header is fixed 40 bytes. The extension header gives the functional flexibility to the IPv6 datagram. Maximum six extension headers can be used. The extension headers may be source routing, fragmentation, authentication and security etc. IPv6 currently defines six extension headers: (1) Hop by hop option header, (2) Routing header, (3) Fragment header, (4) Authentication header, (5) Encrypted security payload header and (6) Destination options header. If one or more extension headers are used, they must be in order in which they are presented above. For example, if Authentication header and routing extension header are to be used, the extension header fields must follow as: (1) main IPv6 header, (2) routing extension header (3) Authentication header and (4) TCP header with data. Each extension header must have one 8-bit next header field. For all extension headers except the fragment header (as in case of fragment header the flags and offset is 16 bits fixed), the next header field is immediately followed by a 8-bit extension header length that indicates

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the length of current extension header in multiple of 8 bytes. In the last extension header the next header field contains the value 59. The example that we considered earlier, the next header in main IPv6 packet will contain the routing extension header, the next header field in the routing header will show the authentication extension header, and the next header field of the authentication header will contain the value 59 (A comprehensive list is shown in Fig. (43). By specifying the next header field as TCP, the last of the extension header can be indicated. Why then the need of the number “59?” This is because IP does not only carry TCP segment, it carries UDP pack, ICMP pack etc. To standardized, the end of all the value “59” has been recommended. • Hop Limit (8 bits): This field indicates the maximum number of hops that the datagram is allowed to traverse in the network before it reaches its destination. If after traversing this maximum number of hops the datagram does not reach the destination, the datagram is discarded from the network. The field is used to avoid the congestion that may be caused by the datagram. Each router decreases the hop limit by 1 while releasing the datagram to the network. When the hop limit reaches 0, it is deleted. The hop limit of IPv6 is exactly what is called Time To Live in IPv4. The new name of Hop Limit has been given as the name suits better to its function. • Source Address and Destination Address (Each 128 bits): Both the addresses can be called IP address and are described in RFC 2373. IP address that defines the original source of datagram is called source address. The IP address that defines the final destination of the datagram is called the destination address. The three main groups of IP addresses are: unicast, multicast and anycast. Unicast address defines a particular host. A unicast packet is identified by its unique single address for a single interface NIC (Network Interface Card), and is transmitted point-to-point. A multicast address defines all the hosts of a particular group to receive the datagram. The anycast address will be addressed to a number of interfaces on a single multicast address. The anycast packet therefore goes to the closer interface and does not attempt to reach the other interfaces with the same address. A multicast packet, like anycast packet has a destination address that is associated with number of interfaces, but unlike the anycast packet, it is destined to each interfaces with that address. Unlike IPv4, IPv6 addresses do not have classes. But the address space of IPv6 is subdivided in various ways for the purpose of use. The sub division is done based on leading bits of addresses. The present division of IPv6 address space is as shown in table (22). The IPv6 address space is huge enough. So a portion of the IPv6 is reserved for computer system using Novell’s Internet Packet Exchange (IPX) network layer protocol, as well as the Connection Less Network Protocol (CLNP). Details on Use of Extension Headers Hop by hop extension header: The payload length header field of IPv6 defines the maximum payload size in bytes; and that is 65,535 bytes or 64 K bytes. Some applications like multimedia services or for use of super computers, may require the larger payloads than 65,535 bytes. By using hop by hop extension headers, the size of the payloads may be increased to 219 bytes (why not any size? The header length of extension header specifies the data in extension in bytes, but must be in multiple of 8 bytes. Therefore each extension header can accommodate about 28 bytes. Maximum 6 extension headers are allowed. So the total bytes that can be accommodated = 6 × 28 by the extension headers plus 216 for the payload field ≅ 219 bytes).

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However the large packet can be constructed with the jumbo packet extension header (see Fig. 43 d), where a value 194 specifies the jumbo packet. The option length field identifies the jumbo or large packet. This indicates the jumbo payload length field in bytes. Finally the 32 bits payload length field specifies the payload size that could be maximum 4,294,967,295 bytes,. Routing extension header: The use of the extension header is typically that of the LSRR (Loose Source and Record Routing) option field of the IPv4. This extension header is used to provide router addresses in order that the IPv6 packets may follow to reach the destination node. This is useful for sending packets by the defined routing paths without any variation. This is also useful when the default routing link or router is out of order, in which case the forwarding node defines/ adds the new routing address in the extension headers so as to make the packet to reach the destination. The illustration is given in Fig. (43 f) Fragment extension header: It typically allows fragmentation of the packets for the purposes as in IPv4. The default minimum IPv6 packet size is 1280 bytes. When the sender discovers that the receiving node is on a network that has MTU (Maximum Transfer Unit) is less than 1280 bytes, the sender fragments the packets to make it possible for transfer over the receiving network. The fragmentation information is conveyed to the receiving node by the fragment extension header. For example when the packet is fragmented, each part of that packet is assigned the same identification number. The identification number is of 32 bits data in the variable extension header field. Illustration is given in Fig. (43 e). Is there any difference between the fragmentation in IPv4 and IPv6? Yes, there is. In IPv4, sender or any intermediate node/router may fragment the packet if the network over which the packet to travel has lower MTU size than the packet size. But in IPv6, it is only the sender or the original source can fragment the packet if so required. Actually there is a technique known as path MTU discovery technique by which the source can find the smallest MTU on the path. Using the information so obtained the source may fragment the packet. However if the path MTU discovery is not run, the source may fragment the packet to smallest MTU size of 576 bytes that is the minimum size the networks support those are connected in the Internet. Authentication extension header: This is used to authenticate the receiving node about the received IPv6 packet, meaning that the original packet as was sent by the sender is received in tact. The extension header if included, the variable extension header fields will include the authenticated code (may be MD5 digest or digest by hash function- see chapter of data security) of all headers, except the headers that change and the payload. Note that the Hop Count field will change at each hop. That is why for digest, the hop count field is not included. Or otherwise the authenticated code for hop count is taken as 0. Illustration of this header is given in Fig. (43 g) Encapsulating security payload(ESP) extension header: This is used to provide data security to the payload only. This extension header supports the secret key encryption namely DES. If the extension header is included, the sender provides the security information in the variable extension header fields. The receiver using the information in the variable extension header fields, decrypts the payload. The ESP uses a header and tailor as in Fig. (43h). Very often, the authentication and the encapsulating security extensions are used together. Whereas authentication ensures the integrity of the packet as a whole, the security ensures the confidentiality of the payload. Illustration is given in Fig. (43 i) It is found that several fields present in IPv4 are no longer present in the IPv6; and notably among them are:

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• Checksum field: The main issue of designing IPv6 was the fast processing of packets. This results in designing with fixed header fields and removing the redundant fields. The error check is done at upper layers namely TCP/UDP. As such the check sum field further at IP layer was assumed as redundant and accordingly it was removed from IPv6. Again with check sum at IPv4 packet, the error checking at every node was essential. It was a very time consuming and costly thing and duly unwanted at IPv6. • Options field: Dropping of options field has made the IPV6 a fixed header packet. Of course if required the IPv6 packet may use next header field for the purpose of header extension. • Fragmentation: The IPv6 version has dropped the fragmentation and reassembly feature at intermediate routers. The data is fragmented for packetization at the source only. The reassembly is done at destination only. If an IP packet received by any intermediate router is found as too large to be forwarded on the outgoing link, the router simply drops the packet; and in turn send a ICMP error message of “Packet Too Big” to the sender. Sender on receiving the ICMP error message of “Packet Too Big”, retransmit the data with smaller packet size. Actually the fragmentation and the reassembling the datagram at routers is a time consuming matter; and removing these from routers’ functions to end users’ functions, makes the network to speed up.

5.11 ICMP (INTERNET CONTROL MESSAGE PROTOCOL) ICMP for version IPv4 is used by hosts, nodes, routers and gateways to communicate network layer information to each other. ICMP is specified in RFC 792. ICMP information is carried as IP payload like TCP or UDP information. ICMP messages are basically used for error reporting among others (Table 24). An ICMP message is made of a type field and a code field and also the first eight bytes of the IP datagram for which the ICMP message is to be generated in the first place so that the sender can know the packet that caused the error. The ICMP message is sent as IP datagram (Fig. 44) with TOS field set to 0 and protocol field set to 1 that defines an ICMP pack. After the headers, ICMP message is included as payload. The ICMP message format (Fig. 44) has four parts: Type: 1 byte field specifies type of the message as in table (24) Code: 1 byte code field is specified as per table (24) Checksum: 2 bytes field is used to provide check sum (1’s complement addition) of all 16 bits headers. The field may be filled in all 0s if required. The additional information field depends upon the header of ICMP messages: • For timestamp request, the additional information is made of 2 bytes identifier, 2 bytes sequence number and 4 bytes originating timestamp • For echo request and reply, the additional information is made of 1 byte identifier, 1 byte sequence number and the original IP header • For destination not reachable etc., the additional information contains 4 bytes unused fields followed by the original IP header. A new version of ICMP defined (Fig. 44d) for IPv6 in RFC 2463. The new ICMP has the reorganized existing types and codes as well as added new types and codes. The added new ICNP type includes “ Packet Too Big”, and “unrecognized IPV6 options” among others.

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The different error reporting ICMP messages are as below: • Destination unreachable: When due to some reason (table 23), a node or a host is unable to deliver data to the destination, the node or host send this type of the ICMP to the source host. The typical problem that may arise is that when a router/node receives a data gram with DF (don’t fragment) bit set to 1, but network does not support size of the received datagram, ICMP message of type = 3 and code = 4 (see table 24) is sent. • Source quench message: The source host does not have any mechanism to know whether a transmitted datagram has actually reached the desired destination or not. This is because IP is a connection less protocol. A node/router or host may discard a data when there is congestion in the network or a condition of buffer full arises. In that case the node/router or host that is discarding the datagram sends the source quench (source slowdown) ICMP message to the sender. The sender on receiving the source quench message understand that there is a congestion in the network and accordingly slows down the transmission. • Time exceeded: A router/node will send the time exceeded ICMP message to the sender under any of the following conditions : (a) when a data gram reaches the router/node with TTL field set to 0 yet the datagram has not reached the destination, and (b) when all the fragments of a datagram are not reached within the time limit. • Parameter problem: Any error or ambiguity (semantic or syntactic) in the IP header creates problem in routing or forwarding the datagram. In that case, the datagram is discarded and the ICMP parameter problem message is sent to the sender. • Redirect message: Both routers/nodes and hosts use routing table in order to route the datagram. The routing table in routers and huge and the tables are constantly updated. The routing table of hosts are limited and static in nature. So the hosts may not route the data appropriately. The routers may suggest redirect ICMP message to inform better routing path (see question 5 latter in this section) The different ICVMP messages for management query are as follows: • Echo request and reply: This is used to check whether a communication is possible between entities ( routers/nodes or hosts). Echo request is obliged by echo reply. Both use identification and sequence numbers for correct acknowledgement and synchronization. • Time stamp request and reply: This is used to measure the delay in transfer of the datagram. • Address mask request and reply: When sub networks are in use, this is used to exchange subnet mask. • Router request and advertisement: This is used to add capability to the host to known the routing information. This is also used to check whether a router is alive / working or not. The router advertisement ICMP message takes the shape as in Fig. (43 c-7). It is as per RFC 1256 and includes the fields of (a) number addresses—the number of router addresses advertised in the current message, (b) address entry size—the number of 32 bits addresses for each router address, which is 2, (c) lifetime—The maximum time in seconds that the router advertisement remains valid, the default value of which is 1800 seconds, (d) router address i (1

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