B.Sc.

Computer Technology

3SA DATA COMMUNICATION & NETWORKS

INTRODUCTION TO COMMUNICATIONS AND NETWORKING
Introduction to communications and Networking : Introduction –
Fundamental concepts – Data communications – Protocols- standards -
Standards organizations – Signal propagations- Analog and Digital signals-
Unit I
Bandwidth of a signal and a medium – Fourier analysis and the concept of
bandwidth of a signal - The data transmission rate and the bandwidth.
Information encoding: Introduction – Representing different symbols
Minimizing errors- Multimedia – Multimedia and Data compression.
ANALOG AND DIGITAL TRANSMISSION METHODS
Analog and digital transmission methods: Introduction - Analog signal,
Analog transmission - Digital signal, Digital transmission - Digital signal ,
Analog transmission - Baud rate and bits per second - Analog signal, Digital
(Storage and) transmission – Nyquist Theorem. Modes of data transmission
Unit II and Multiplexing: Introduction – Parallel and Serial communication -
Asynchronous, Synchronous and Isochronous communication - Simplex,
Half-duplex and Full-duplex communication – Multiplexing - Types of
Multiplexing – FDM versus TDM. Transmission Errors: Detection and
correction : Introduction – Error classification – Types of Errors – Error
detection.
Unit:3 TRANSMISSION MEDIA
Transmission media: Introduction - Guided media - Un Guided media –
Shannon capacity.
Network topologies, switching and routing algorithms: Introduction –
Unit III
Mesh topology – Star topology - Tree topology - Ring topology - Bus
topology - Hybrid topology - Switching basics- Circuit switching – Packet
switching - Message switching - Router and Routing – Factors affecting
routing algorithms - Routing algorithm -Approaches to routing.
Unit:4 NETWORKING PROTOCOLS AND OSI MODEL
Unit IV Networking protocols and OSI model: Introduction – Protocols in computer
communications - The OSI model - OSI layer functions.
Unit:5 INTEGRATED SERVICES DIGITAL NETWORKING (ISDN):
Integrated services digital networking (ISDN): Introduction – Background
of ISDN – ISDN architecture – ISDN interfaces - Functional grouping –
Unit V
Reference points - ISDN protocol architecture – Broadband ISDN (B-ISDN).
of ATM – Packet size – Virtual circuits in ATM –
ATM cells – Switching – ATM layers – Miscellaneous Topics
Text Book(s)
1 Data Communications and Networks, Achyut. S. Godbole, Tata McGraw-
Hill Publishing Company, 2007
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Data Communication and Networks- Unit I

Introduction
 The main objective of data communication and networking is to enable
seamless exchange of data between any two points in the world.
 This exchange of data takes place over a computer network
Data & Information
 Data refers to the raw facts that are collected while information refers
to processed data that enables us to take decisions.
 Ex. When result of a particular test is declared it contains data of all
students, when you find the marks you have scored you have the
information that lets you know whether you have passed or failed.
 The word data refers to any information which is presented in a form
that is agreed and accepted upon by is creators and users
Data Communication
 Data Communication is a process of exchanging data or information
 In case of computer networks this exchange is done between two
devices over a transmission medium.
 This process involves a communication system which is made up of
hardware and software.
 The hardware part involves the sender and receiver devices and the
intermediate devices through which the data passes.
 The software part involves certain rules which specify what is to be
communicated, how it is to be communicated and when. It is also
called as a Protocol.
 The following sections describes the fundamental characteristics that
are important for the effective working of data communication process
and is followed by the components that make up a data
communications system.
Characteristics of Data Communication
 The effectiveness of any data communications system depends upon
the following four fundamental characteristics:
1. Delivery: The data should be delivered to the correct destination
and correct user.
2. Accuracy: The communication system should deliver the data
accurately, without introducing any errors. The data may get
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Data Communication and Networks- Unit I

corrupted during transmission affecting the accuracy of the

delivered data.
3. Timeliness: Audio and Video data has to be delivered in a timely
manner without any delay; such a data delivery is called real time
transmission of data.
4. Jitter: It is the variation in the packet arrival time. Uneven Jitter may
affect the timeliness of data being transmitted.
Components of Data Communication
1. Message
Message is the information to be communicated by the sender to the
receiver.
2. Sender
The sender is any device that is capable of sending the data (message).
3. Receiver
The receiver is a device that the sender wants to communicate the data
(message).
4. Transmission Medium
It is the path by which the message travels from sender to receiver. It
can be wired or wireless and many subtypes in both.
5. Protocol
 It is an agreed upon set or rules used by the sender and receiver to
communicate data.
 A protocol is a set of rules that governs data communication. A
Protocol is a necessity in data communications without which the
communicating entities are like two persons trying to talk to each other
in a different language without know the other language.
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Elements of a Protocol
There are three key elements of a protocol:
Syntax
 It means the structure or format of the data.
 It is the arrangement of data in a particular order.
Semantics
 It tells the meaning of each section of bits and indicates the
interpretation of each section.
 It also tells what action/decision is to be taken based on the
interpretation.
Timing
 It tells the sender about the readiness of the receiver to receive the data
 It tells the sender at what rate the data should be sent to the receiver to
avoid overwhelming the receiver.
Standards In Networking
 Standards are necessary in networking to ensure interconnectivity and
interoperability between various networking hardware and software
components.
 Without standards we would have proprietary products creating
isolated islands of users which cannot interconnect.
Concept of Standard
 Standards provide guidelines to product manufacturers and vendors
to ensure national and international interconnectivity.
 Data communications standards are classified into two categories:
1.De facto Standard
 These are the standards that have been traditionally used and mean by
fact or by convention
 These standards are not approved by any organized body but are
adopted by widespread use.
2.De jure standard
 It means by law or by regulation.
 These standards are legislated and approved by an body that is
officially recognized.
Standard Organizations in field of Networking
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 Standards are created by standards creation committees, forums, and

government regulatory agencies.
 Examples of Standard Creation Committees :
1.International Organization for Standardization(ISO)
2.International Telecommunications Union –
Telecommunications Standard (ITU-T)
3.American National Standards Institute (ANSI)
4.Institute of Electrical & Electronics Engineers (IEEE)
5.Electronic Industries Associates (EIA)
 Examples of Forums
1.ATM Forum
2.MPLS Forum
3.Frame Relay Forum
 Examples of Regulatory Agencies:
1.Federal Communications Committee (FCC)
Analog and Digital Signals
 Analog data refers to information that is continuous varying signal
similar to a sinusoidal waveform; ex. sounds made by a human voice
 Digital data refers to information that has discrete states. Digital data
take on discrete values.
 For example, data are stored in computer memory in the form of Os
and 1s
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Data Communication and Networks- Unit I

Bandwidth of a Signal and a Medium

 Bandwidth can be defined as the portion of the electromagnetic
spectrum occupied by the signal
 It may also be defined as the frequency range over which a signal is
transmitted.
 Different types of signals have different bandwidth. Ex. Voice signal,
music signal, etc..
 Bandwidth of analog and digital signals are calculated in separate
ways; analog signal bandwidth is measured in terms of its frequency
(hz) but digital signal bandwidth is measured in terms of bit rate (bits
per second, bps)
 Bandwidth of signal is different from bandwidth of the
medium/channel. If the signal bandwidth is high, the data
transmission speed will be higher. So, the transmission medium used
should be capable of transmitting the data without decreasing its
quality.
A sine wave is characterized by three parameters:
1.Peak Amplitude
2.Frequency
3.Phase
Characteristics of an Analog Signal
Peak Amplitude
 The amplitude of a signal is the absolute value of its intensity at time t
 The peak amplitude of a signal is the absolute value of the highest
intensity.
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 The amplitude of a signal is proportional to the energy carried by the

signal
 Amplitude is measured in volts ,amperes or watts depending on the
type of the signal

Frequency
 Frequency refers to the number of cycles completed by the wave in one
second.
 Period refers to the time taken by the wave to complete one second.
 Frequency is measured is hertz(Hz)

 Signals with frequency 0 to ∞

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Phase
 Phase describes the position of the waveform with respect to time
(specifically relative to time O).
 Phase indicates the forward or backward shift of the waveform from
the axis
 It is measured in degrees or radian
 The figure above shows the sine waves with same amplitude and
frequency but different phases
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Fourier Analysis and the Concept of Bandwidth of a Signal

 In the 19th century, French mathematician Jean-Baptiste Fourier
proved that any composite signal is a combination of simple sine
waves with different frequencies, amplitudes, and phases.
 Any signal can be decomposed into different sinusoidal signals. This
process is called Fourier Analysis.
 A Composite signal can be periodic as well as non periodic.
 A periodic composite signal when decomposed gives a series of simple
sine waves with discrete frequencies i.e. frequencies that have integer
values (1, 2, 3, etc).
 A non-periodic composite signal when decomposed gives a
combination of an infinite number of simple sine waves with
continuous frequencies i.e. frequencies that have real values.
 Fourier Analysis Illustrated
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Bandwidth of a signal in time domain and frequency domain

 The signal shown in the diagram is an composite analog signal with
many component signals.
 It has a minimum frequency of F1 = 30Hz and maximum frequency of
F2 = 90Hz.
 Hence the bandwidth is given by F2 – F1 = 90 – 30 = 60 Hz
Bandwidth of a digital signal
 It is defined as the maximum bit rate of the signal to be transmitted.
 It is measured in bits per second.
 A digital signal of 1 HZ

 A digital signal with infinite bandwidth

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Data Transmission Rate and Bandwidth

Bandwidth of a Channel
 A channel is the medium through which the signal carrying
information will be passed.
 In terms of analog signal, bandwidth of the channel is the range of
frequencies that the channel can carry.
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 In terms of digital signal, bandwidth of the channel is the maximum

bit rate supported by the channel. i.e. the maximum amount of data
that the channel can carry per second.
 The bandwidth of the medium should always be greater than the
bandwidth of the signal to be transmitted else the transmitted signal
will be either attenuated or distorted or both leading in loss of
information.
 The channel bandwidth determines the type of signal to be transmitted
i.e. analog or digital.
Bit Rate
 It is the number of bits transmitted in one second. It is expressed as bits
per second (bps).
 Relation between bit rate and bit interval can be as follows Bit rate = 1
/ Bit interval
Baud Rate
 It is the rate of Signal Speed, i.e the rate at which the signal changes.
 A digital signal with two levels ‗0‘ & ‗1‘ will have the same baud rate
and bit rate & bit rate.
 The diagram below shows three signal of period (T) 1 second
a) Signal with a bit rate of 8 bits/ sec and baud rate of 8 baud/sec
b) Signal with a bit rate of 16 bits/ sec and baud rate of 8 baud/sec
c) Signal with a bit rate of 16 bits/ sec and baud rate of 4 baud/sec
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The Maximum Data Rate Of A Channel

Data rate depends on three factors:
1.The bandwidth available
2.The level of the signals we use
3.The quality of the channel (the level of noise)
The quality of the channel indicates two types:
a) A Noiseless or Perfect Channel
 An ideal channel with no noise.
 The Nyquist Bit rate derived by Henry Nyquist gives the bit rate for a
Noiseless Channel.
b) A Noisy Channel
 A realistic channel that has some noise.
 The Shannon Capacity formulated by Claude Shannon gives the bit
rate for a Noisy Channel
 Where,
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 Bitrate is the bitrate of the channel in bits per second Bandwidth is the
bandwidth of the channel
 L is the number of signal levels.
 Example
 What is the maximum bit rate of a noiseless channel with a bandwidth
of 5000 Hz transmitting a signal with two signal levels.
 Solution:
 The bit rate for a noiseless channel according to Nyquist Bit rate can be
calculated as follows:
BitRate = 2 x Bandwidth x Log2 L
= 2 x 5000 x log2 2 =10000 bps

Shannon Capacity
 The Shannon Capacity defines the theoretical maximum bit rate for a
noisy channel
Nyquist Bit Rate
 The Nyquist bit rate formula defines the theoretical maximum bit rate
for a noiseless channel
 Where,
 Capacity is the capacity of the channel in bits per second
 Bandwidth is the bandwidth of the channel SNR is the Signal to Noise
Ratio
 Shannon Capacity for calculating the maximum bit rate for a noisy
channel does not consider the number of levels of the signals being
transmitted as done in the Nyquist bit rate.
 Example:
 Calculate the bit rate for a noisy channel with SNR 300 and bandwidth
of 3000Hz
 Solution:
 The bit rate for a noisy channel according to Shannon Capacity can be
calculated as follows:
Capacity=bandwidth X log2 (1 +SNR)
= 3000 x log2 (1 + 300)
= 3000 x log2 ( 301)
= 3000 x 8.23
= 24,690bps
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Information Encoding
 Introduction
 In communications and information processing, encoding is the
process by which information from a source is converted into symbols
to be communicated.
 In computer system we have to use encoding to represent the
Information in the format that computer understand i.e. ―binary‖
language.
 Encoding enables us to improve the communication in places where
written languages is difficult to use or impossible.
 Representing Different Symbols
 In information theory and computer science, a code is usually
considered as an algorithm which uniquely represents symbols from
some source alphabet, by encoded strings, which may be in some other
target alphabet.
 The purpose of the symbols is to communicate the idea or meaning.
We use the different symbols to represent the Information in computer
understandable format.
 There are different symbols available e.g. in English language we have
26 capital letters from ―A to Z‖ and same way we have small letters
from ―a to z‖, we have numeric symbols like (0,1,…9) and special
symbols like (!,@,#,$,%,^,&,*, etc.)
 Minimizing Errors
 One way to represent the information is to use sound beep. The
different sound intensities can be utilized to represent the around 162
different symbols.
 Practically we won‘t be able to distinguish between all 162 sound
levels. So it will lead to errors in identifying a symbol correctly. We
should follow the representation that minimizes the errors.
 So another way of doing work is just use two states OFF or ON. With
sound beep we get some sound or none at all. When it is no sound then
the value must be 0. When we hear some sound then value must be 1.
 With this beep and no-beep we can represent only two symbols
correctly. We call this system as binary system.
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 We can have the following table to represent our symbols:

Sound Represented Code

symbols
No-beep A 0
Beep B 1
 We must note that the system which is explained above, even though
the error in representation of symbols is minimized, is restricted with
two symbols. So let us see how to represent more symbols using the
binary system.
Representing more symbols:
 Now we see what happens when we use two sound devices. As shown
in the following table we get the different combinations like 00, 01, 10
and 11. And now we can represent four different symbols.

Sound Sound Represented Code
device device 2 symbols
1
No-beep No-beep A 00
No-beep Beep B 01
Beep No-beep C 10
Beep Beep D 11

 Thus with the pair of sound devices, we get four ON/OFF

combinations and hence we can represent four symbols.
 If we extend the same technique with three sound devices then we can
get eight different codes to represent our symbols like
(000,001,010,011,100,101,110 and 111)
 The generalized we can say that with N number of sound devices we
can have 2N symbols to be represented.
 Number of states : 2N
 Where N is number of sound devices.
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 Multimedia:
 Now a day the computers that we use have additional facilities such
as:
1. Drawing, capturing, storing and viewing pictures of different
formats.
2. Recording, storing the sound/songs and playing them back.
3. Capturing, editing, storing the video information and playing
them back.
 Since video, pictures are not made of alphabets; we cannot represent
them with the help of the character codes like ASCII (American
Standard Code for Information Interchange) or EBCDIC (Extended
Binary Coded Decimal Interchange Code) etc.
 So we use the concept of multimedia in order to code the picture or
videos.
 With the help of multimedia technique it‘s possible to use the
computer system to store, play and process the audio, video and
picture information along with the textual data.
 Pictures/Images:
 We can represent the data in the form of Images / Pictures. Images are
represented by the pixels i.e. the smallest element in the picture.
 The basic idea is shown in the following figure:

(a) (b)

 Figure (a) shows the computer screen made up of number of dots

(pixels). Figure (b) shows the letter W by illuminating specific dots.
 We can use a large number of pixels for better resolution.
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 Higher resolution gives better quality to the picture.

 When we divide a picture into pixels, each pixel is represented by a
unique pattern.
 If Image is black and white then we can use only one bit per pattern.
Bit 1 is used to represent white and bit 0 is used to represent black.
 If the image is having gray shade then we can use 2 bit pattern to
represent each pixel. So 00 is black, 01 is dark gray, 10 is light gray and
11 will represent white.
 If the image is colored then it is to be represented by the pixels that are
constituted of the three primary colors namely red, green and blue
(RGB).
 It means that any picture can be drawn on screen by illuminating or
darkening specific pixels.
 We know that computer can understand on binary values (0 and 1). So
it is important that the way in which we make computer understand
the concept of pixels.
 We can consider the coding scheme where illuminated pixel is
considered to be binary 1 and darkened pixel is considered to be binary
0. We can imagine that any picture that we draw on screen can be
mapped first to a series of pixels. Which in turn, get mapped to a series
of zeros and ones.
 We can show this idea in following figure:

01010111101010

10111010110110

01101010111101

10000111010101

01110101101011

11010101011101

01101100110101
 Computer screen Computer memory
 Video: 0111101100
 Animation is used as a basic technique for creating videos.
 We get the animation if we show set of pictures rapidly, the human
eye gets an illusion that the picture is in motion.
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 The above pictures show the idea behind the video. It shows the
movements of the horse and his rider. If you were shown these four
pictures one after the other very fast, you would believe that the horse
is actually running.
 We use the same principle to store pictures in the disk / memory of
the computer in their binary form and show them rapidly at the rate
of 24 such pictures or images per second on the screen.
 Sound:
 A sound wave in its most basic form continuous in nature. It is
continuous in two aspects:
 First, the strength (the amplitude) and time.
 A typical sound signal takes the form of sine wave as shown in
following figure:

 The sound needs to be converted into the digital form in order to store
it in the computer system. Thus if we have to show the zeros and ones
inside a computer‘s memory graphically, then we get the following
figure:
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 In order to convert the sound into digital form, we have carry out the
following processes on the analog sound signal:
 Sampling
 Quantization
 Encoding
 These processes are collectively called as Pulse Code Modulation
(PCM). The following figure shows all the processes:

 Sampling:
 When we transfer the signal using pulse code modulation and digital
modulation the signal must be in the discrete time form.
 If the message is generated from the computer system or any other
digital source then it is in the proper form for processing by the digital
communication system.
 But in real life the signal can be of analog type (e.g. voice). In such a
case it has to be first converted into discrete time signal.
 For this we use ―Sampling‖ method. Thus using the sampling process
we convert the continuous time signal (analog) into the discrete time
signal (digital).
 The sampling process should satisfy the following requirements:
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 Sampled signal should represent the original signal.

 It should be possible to reconstruct the original signal from its sampled
form.
 The following figure explains the sampling process: amplitude


 Quantization:
 Quantization is the process in which we assign the numbers to the
discrete values depending upon their amplitude values.
 Quantizer converts the sampled signal into an approximate quantized
signal which consists of only finite number of predecided voltage
levels.
 Each sampled value at the input of the quantizer is approximated or
rounded to the nearest standard predecided voltage level
(Quantization levels).
 The following figure shows the overview of the process:
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 We can show the sampling and quantization process as follows:

 Multimedia and Data Compression

 When we store the information in the form of pictures, videos and
sounds there would be lot of repetition that can be observed.
 Storing such repeated information would cause the wastage of
computer memory.
 We must find the better schemes that eliminate the duplication of the
data or redundant information.
 There are various multimedia file formats available which allow
storage of multimedia files in more efficient manner by getting rid of
duplication/ redundancy by means of the process called as Data
Compression.
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Data Communication and Networks- Unit II

ANALOG AND DIGITAL TRANSMISSION METHODS

INTRODUCTION :
There are four types:
1. Analog Signal, Analog Transmission
2. Digital Signal , Digital Transmission
3. Digital Signal, Analog Transmission
4. Analog Signal , Digital Transmission

Analog Signal, Analog Transmission

 The term analog is very common and has been used for decades in the field of telephony.
 The human voice generates an analog signal which is transmitted as an analog signal over
the medium. The signal suffers attenuation.
 Amplifiers amplify noise along with the original signal.

Digital Signal, Digital Transmission

 Information coming out of a computer is in the form of digital signals.
 Digital signal has a infinite bandwidth whereas many medium has only a limited
bandwidth.
 The signal is generated and enters the medium at the point of entry only limited
frequencies are permissible on the medium depending upon its bandwidth. The resultant
signal would not be the same as the digital signal.
 The signal is distorted from the original digital signal. It traverse over the medium , noise
adds further distortion.
 Therefore hardware equipment called regenerative repeater or repeater is used to
regenerate the digital signal.
 Example : Three points A ,B and C. At point A the signal is in its original digital form. It
gets distorted at point B, the signal as 0100101. The repeater recognizes the bits and
outputs the signal in its original form at point C.

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 The input to the regenerative repeater is a signal which looks like a digital signal. The
repeater measures the signal values at regular intervals to recognize the 0 and 1 in the
signal and regenerates them.
 Any line with repeaters placed at the appropriate distance is called a digital line.Such a
line can reproduce the original digital signal at the other end.
 The digital line is called a T1 line,which can carry a data rate of 1,54,400 bps (1.54
Mbps)

Digital Signal , Analog Transmission

Modem
 How to send digital signals over an analog network. Some codification techniques was
necessary to convert the digital signals over an analog one.
 The modem is derived from two components : Modulator and Demodulator
 Modulator basically uses some convention or a coding scheme and converts a digital
signal into an analog than can be transmitted through that channel and a demodulator
converts the analog signal back into the digital signal.
 The digital signals originating from the computer go through the modem where they are
converted in to analog signals whose bandwidth is <4000Hz.
 This is because the channel for telephone conversation requires a bandwidth of 4000Hz.
The analog signal traverses the telephone line.

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 At the other end this signal is fed to another modem where it is converted back into
original digital signal.
Modulation Techniques
 Modulation basically uses a coding scheme or a convention. This coding can be achieved
using the three properties of a signal Amplitude ,Frequency and Phase.
 There are three techniques
1. Amplitude Shift Keying (ASK)
2. Frequency Shift Keying (FSK)
3. Phase Shift Keying (PSK)
4. Another basic combination technique Quadrate Amplitude Modulation (QAM)
Amplitude Shift Keying (ASK)
 In Amplitude Shift Keying (ASK) or amplitude modulation we do not alter the frequency
or phase of the carrier signal. We specify different amplitudes that is we shift the
amplitude values to represent a binary 1 or binary 0.
 Example : A bit string 10011010011 modulated as an analog signal , only amplitude of
the signal changes as per the values 0 and 1 , the signal can be sent over the telephone
lines

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 The receiver end , measures the amplitude at regular intervals to decode them as 0 and 1
and then generates a digital signal .The binary bits then can be stored at the destination
node
 ASK encoding techniques, noise is the major problem.
Frequency Shift Keying (FSK)
 In Frequency Shift Keying (FSK) or frequency modulation techniques the amplitude and
the phase of the carrier signals unaltered. Assign a certain frequency f1 to denote 1 and f2
denote 0.
 Both f1 and f2 must be in the bandwidth of the channel that is between 0 to 4000 Hz,
which can be easily carried by the telephone wires
 The signal component with slower cycle is f1 and the signal component that show the
rapid cycle portions is f2.The two represents 1 and 0 of the input digital signal
respectively.
 Example The bit string 10011010011 sent using frequency shift keying . The phase and
amplitude are same , only frequency varies. But both f1 and f2 are between 0 and
4000Hz.
 This technique is less error than ASK.

Phase Shift Keying (PSK)

 In PSK the amplitude and the frequency of the carrier signal unchanged and only change
the phase denote 0 and 1.
 A Phase of 0 degrees to represent binary 0 and the change the phase to 180 degree to
represent binary 1.
 In PSK chage the timing of the carrier wave immediately to encode data. After a phase
shift happens the carrier wave still continous to oscillate , but it immediately jump to a
new point in its cycle
 The 0 bit indicates no phase change at all, if the preceding bit is either 0 or 1.
 This techniques is more noise resistant
 The main limitation of PSK is the inability of the hardware equipment to small
differences in terms of phase changes.
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Data Communication and Networks- Unit II

Quadrate Amplitude Modulation (QAM)

 In ASK,FSK, and PSK alter only on characteristic of the carrier wave.
 The bandwidth of the transmission medium is a major limitation, cannot combine FSK
with anything else. The only possibility is to combine ASK and PSK.
 X variations in phase and y variations in amplitude this called QAM.
 QAM makes higher data rates possible.
Baud Rate and Bits Per Second
Baud Rate:
 The baud rate is the number of times the signal level changes in a channel per second.
 The bandwidth of a transmission medium defines the maximum and minimum frequency
that it allows for the carrier wave.
 One bit with each change in the signal level. By associating more than one bit for each
signal level, one can achieve a higher data rate. That is bit rate will be higher than the
baud rate.
 In traditional case where in bit rate=baud rate. Consider a FSK with a base frequency
=1700 Hz. Two frequencies 1200Hz to represent 0 and 2200Hz to represent 1.

 The bit coming from the computer is examined by the modem and depending upon
whether it is 0 or 1, the carrier signal is modulated at 1200Hz or 2200Hz frequency,
keeping the amplitude and the phase same.

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Analog Signal, Digital (Storage and ) Transmission

Pulse Code Modulation (PCM)
 To represent an analog signal into digital bit and then transmit it as a digital signal

The basic steps in PCM:

 We require an equipment called codec (Coder/Decoder) at both the source and the
destination .It is also as A/D (Analog to Digital) converter and D/A (Digital to Analog)
converter
 The discrete values of the amplitude shown are : 0.21,0.49,0.83,1.04,1.07,0.68 and 0.78

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 To represent these values to binary number. We approximate these values to the nearest
number such as 0.2,0.5,0.8,1.0,1.1,0.7 and 0.8. We multiply each value by 10 to get the
number 2,5,8,10,11,7 and 8.
 To convert into binary values as 0010,0101,1000,1010,1011,0111 and 1000
 Send these bits as a long bit stream as 0010010110001010101101111000
 At the destination the equipment at that end can now split the received bit stream into
chunks of four bits each , find out its decimal value 2,5,8,10,11,7 and 8 to get the values
0.2,0.5,0.8,1.0,1.1,0.7 and 0.8 by dividing these by 10 and generate an analog signal with
specific values at the time interval t at times 0t,1t,2t,3t,4t and 5t.
 This signal is slightly different than the original signal due to the approximation which
we carried out at the source. This difference is known as quantization noise or
quantization error.
 PCM would be reduce the quantization noise to a negligible level without increasing the
load on the network.
 The current PCM standard assumes eight bits/sample
 There are two variations of the basic PCM method:
(1) Adaptive Differential Pulse Code Modulation (ADPCM)
 In this model outputting the absolute value of the digitized amplitude. The output
values is the difference between the current amplitude value and the previous
amplitude value
(2) Delta Modulation
 Only one bit is reserved for the difference between two successive readings.
But the signal is sliced very rapidly to have a number of samples.
 If the next reading is more than the previous one a bit 1 is output. If it is less a
bit 0 is output.

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Nyquist Theorem
 We choose the time interval for sampling or slicing the analog signal , the equipment at
both ends has to be capable of handling that high speed of sampling and reconstructing
the analog signal.
 At higher sampling speeds there is less of missing the ups and downs than at lower
speeds.
 A signal with very low frequency which changes very slowly and smoothly
 A signal with high frequency which changes rapidly
 The sampling of both the signals at time interval t.
 At low frequency the sampling at that speed is good enough.
 At high frequency the sampling rate is low. It misses many ups and downs.
 For instance refer to points x,y,z in the figure. If we know the reading at points x and y
there are a number of ways in which these points could be connected.
 The sampling speed is related to the highest frequency in a signal. To consist of a number
of sinusoidal signals leading to the concept of bandwidth of a signal.
 Nyquist showed that the sampling speed should be 2*fmax where fmax represents the
highest frequency in that signal resulting out of Fourier analysi . This is called Nyquist
theorem.
 This theorem proves to reproduce the analog signal into its equivalent digital form with
minimum loss.

 Video signal have a higher bandwidth with signals at very high frequencies than voice
signals.
 The sampling for digitizing video signals has to be done at higher rate than for voice
signals. This lead to a demand for higher data rate and storage volume for video signals.
 The human voice has various frequency components in the range of 0 to 20000 HZ.

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 The frequency range of 300 -3300 Hz is sufficient to recognize the voice in the telephone
conversation. Telephone company provide for 0-4000 Hz.
 The frequencies 0-300 Hz and 3300-4000 Hz act as guard bands, so multiplexing of
many signals in a single wire is possible.

 The following equation for the bandwidth required of the telephone lines to carry
digitized human voice:

MODES OF DATA TRANSMISSION AND MULTIPLEXING

Digital data transmitted in many ways

1. Parallel and Serial Communication
2. Asynchronous ,Synchronous and Isochronous Communication
3. Simplex, Half-duplex and Full-duplex Communication
Parallel and Serial Communication:
 To send digital data stored as bytes of 8 bits each and in words may be 16 or 32 bits
depending upon the word length
Parallel Communication
 In parallel communication transfer a word or a byte at a time.
 Need many wires parallel to each other, each carrying a single bit. Where eight wires are
transmitting a byte at a time.
 Example : The byte contains : 10011001
 This is the fast method of transmitting data from one place to another.
 It is very expensive method because it requires several wires in sending as well as
receiving.
 It demands accuracy, which cannot be guaranteed over long distance.
 The digital pulses may not traverse at the same speed. This gives to a problem of Skew.
 When a message is skewed the bits 10011001 are sent form the source to the destination
but they traverse at different speeds. The destination the measurement of signal values to
determine whether it was a bit 0 or 1 have to be done at the same time for all the bits.The
result is inaccurate.

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 To avoid this problem, it is used only for a very short distance. This method is used for
data transmission within the computer system, such as cpu register to the memory or data
bus.

Serial Communication

 Over long distance serial communication is used.

 While sending the data serially, characters or bytes have to be separated and sent bit by
bit. Some hardware is needed to convert the data from parallel to serial.
 At the destination the measurement of the signal values is done in the middle of the bit
duration. This is because if the values are taken at the edge or a point where the bit value
changes the reading will be indeterminate.

 At the destination all the bits are collected, measured and puts together as bytes in the
memory of the destination .This requires conversion from serial to parallel.
 In serial data transmission we have to identify where the character starts, also identify the
middle position of each bit interval

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 The transmitter and the receiver have two different clocks. The point is to synchronize
the clock of the receiver exactly with that of the transmitter, so the correct readings will
result and the bit values are understood correctly.
 This is the problem of Synchronization.
 Example: To send 8 bit sb0 to b7.Two arms A1 and A2 at point X and Y rotate in clock
wise directions. When both A1 and A2 point to the bit b0 at X and Y respectively.
 Two arms A1 and A2 have to be perfectly synchronized and have to rotate at the same
speed in order to succeed in sending/receiving all the bits accurately.

 A Line adapter or interface generally achieve this task of synchronization

 Electronic devices called Universal Asynchronous Receiver Transmitter(UART) and
Universal Synchronous Asynchronous Receiver Transmitter(USART) are example of
such interface
Asynchronous , Synchronous an Isochronous Communication
Asynchronous Communication:
 In asynchronous communication the time when character will be sent cannot be
predicted.
 Where the data is transmitted as character and at an unpredictable pace, we have to
synchronize the source and destination for each character. This is called asynchronous
communication
 Each character is preceded with a start bit and succeeded with 1,1.5 or 2 stop bits
 NRZ-L signaling is used for Asynchronous transmission
 This convention a negative voltage denotes a binary 1 and positive voltage denotes a
binary 0.
 When the line is idle that is when no character is being sent over the line, a constant
negative voltage signifying a binary 1 is generated.
 When the character is to be sent, bit 0 is sent first .This positive voltage is called the start
bit.
 A parity bit if used then follows
 In the end 1,1.5 or 2 stop bits are added

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Synchronous communication:
 In synchronous communication the whole block of data bits is transferred at once, instead
of one character at a time.
 The block of bits may or may not consist of different characters.
 We need a bit oriented protocol between the sender and the receiver.
 If the block consist of different characters the receiver needs to known the beginning of
the first bit of the first character

 It needs a byte oriented protocol between the sender and the receiver
 To perform this synchronous each data block is preceded with a unique synchronizing bit
pattern. Use the SYN transmission control character for this.
 SYN has a bit pattern of 0010110. This SYN character can be reserved for indicating the
start of the block.
 Problem in this scheme:
 The transmitter will not send a SYN character as a part of the data. The bit pattern of two
characters could be such that if sent one after the other, they can constitute a SYN
character thereby fooling the receiver.

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
 If the ASCII character b and a are sent one after the other, a bit combination of 4 bits
from the character b and 3 bit from the character a to constitute the SYN character
 At the receiving end , if the receiver is looking for SYN character , it can get fooled.
 In this reason two SYN bytes are sent consecutively. The bit combination of two SYN
bytes that is 00101100010110 cannot be obtained by concatenating any character.
 When the clock starts measuring the bit values the counter within a receiver is
incremented for every bit received and measured and pushed into the character
assembler.

 After a character assembler the character is moved to separate buffer and the bit counter
is set to 0 to prepare for the next character.
 Synchronous communication is used when large amount of data is to be sent from one
place to the other.
Asynchronous Communication
 Assume that for each character of 8 bits , 1 start bit and 1 stop bit are sent. At 2 bits
/character as overhead , to send an overhead of

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Synchronous Communication
 Assume that the file of 100,000 characters is broken into blocks of 1200 characters =9600
bits.
 Assume that 48 overhead bits are sent along with each block
 For SYN – Synchronize, STX- start of transmission , ETX-end of transmission
 Overhead is now computed as :

Isochronous Communication
 This method combines the approaches of asynchronous and synchronous
communications.
 In the asynchronous method each character has both the start and stop bits. However the
idle period between the two characters is not random.
 All idle periods of no transmission consist of an exact multiple of one character time
interval. If the time to transmit character is t the time interval between character cannot
be random as in the asynchronous method. It is also not 0 as in synchronous method. It
has to be t,2t,3t,…,nt where n is a positive integer in isochronous communication
 The isochronous communication to asynchronous method transmission speed is high.
 In asynchronous communication data rate is 2,400 bits per second but in isochronous
communication the data rate up to 19,200 bits per second.

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Simplex ,Half-Duplex and Full-Duplex Communication

 This classification of data transmission is based on which of the communicating devices
can send data and a what point of time.
 Three ways
1.Simplex
2. Half-Duplex
3.Full-Duplex
Simplex Communication
 It is unidirectional. One of the communicating devices can only send data whereas the
other can only receive it.
 Example : Radio, TV or Keyboard

Half-Duplex Communication
 In the half duplex mode both devices can transmit data though not at the same time.
When one device is sending data, the other must only receive it and vice versa
 This requires a definite turnaround time during which the device change from the
receiving mode to the transmitting modes. Due to this delay, it is slower than simplex
communication.

 It is implemented by using a two-wire circuit : one for data and one for ground.
 Example : Conversations over walkie-talkie

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Full-Duplex Communication:
 In this mode both the devices can transmit data at the same time.It means that both
devices are capable of sending as well as receiving data at the same time.
 Example: Telephone conversation
 This can be done using a two-wire circuit or a four –wire circuit.
 In two wire circuit one wire is used for data and one for ground as in half duplex
 In four wire circuit , there are two wires for data and two for ground.

Multiplexing:
 Multiplexing divides the physical line or a medium into logical segment called channels
 In multiplexing different channels carry data simultaneously over the same physical
medium
 Hardware equipment called multiplexer or mux combine the inputs from different
sources and load them on different channels of a medium.
 At the destination a de-multiplexer separates the signals and sends them to their different
destinations

Types of Multiplexing
There are two types
1. Frequency Division Multiplexing (FDM)
2. Time Division Multiplexing(TDM)

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Frequency Division Multiplexing (FDM):

 FDM is a common technique used in public telephone and in the cable TV System,
where a single cable carries multiple video signals from different channels or stations to
TV set.
 In FDM the medium is divided into a number of channels , each with a frequency
bandwidth and data rate.
 The composite signal carried by the medium is analog, the input signals can be analog or
digital
 If the input signals are analog , only multiplexers at both the end are sufficient
 If the input signals are digital, modems is used to convert the digital signal into analog
and convert them back into digital signal

FDM and the Analog Telephone System :

 FDM is very important in the context of the analog telephone system.
 Analog telephone system voice signals are carried over twisted copper wire pairs as
analog signals from home to the nearest exchange called last mile.

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 The frequencies of the human voice that are carried by the telephone system are in the
range of 0 to 4000 Hz. To carry a single voice conversation over telephone a bandwidth
of 4 KHz is sufficient.
 The capacity of the medium such as twisted wire pair or coaxial cable is higher.
 At an exchange, signals from multiple sources can be combined or multiplexing and then
sent to a higher or different exchange. At the other end they are de-multiplexed and sent
to the appropriate destinations.

Analog Telephone Hierarchy:

 First Level : 12 voice channel , each having a bandwidth of 4KHz are multiplexed to
form a Group
 At next level : 5 such groups are multiplexed to form a Super-group
 At the third level: 10 Super-groups are multiplexed onto a channel of higher bandwidth
called Master-group
 At the last level 6 Master group channel are combined to form a Jumbo-group. The
bandwidth of Jumbo-group should be 6X2.52 MHz=15.12MHz

Time Division Multiplexing (TDM)

 TDM is a technique used for digital transmission only.
 We divide the transmission time into a number of time slices.
 Allocate each time slice to different source nodes, each of which wants to sends some
data.
 Node means a computer, a terminal or any device
 During the time slice that a source node is sending some data, the entire bandwidth
belongs to that source node.
 There are two ways to allocating the time slices to various source nodes. There are
1. Synchronous TDM also known as TDM
2. Statistical TDM

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Synchronous TDM or TDM:

 In this technique called synchronous TDM or TDM, the time slice is allocated to a source
node regardless of whether it wants to send some data or not.
 Every source node has a fixed time slot, the position of the data within the data frame
specifies its origin.
 It can be wasteful scheme, because the time slot is allotted to source node even if it has
nothing to send.
 A Small buffer memory is associated with every source node. At any time , not all node
may want to send some data. The timing device in the multiplexer allocates some time for
each node to transmit the data from its buffer and then repeats this cycle example A-B-C-
D- A-B-C-D etc.

Statistical Time Division Multiplexing (STDM)

 STDM is a variation of the TDM technique. This technique is more intelligent. It
monitors which machine or terminal is sending data more frequently and in greater
quantity and allocates the time slices more often to those nodes.
 A completely idle computer / terminal may not get any time slice at all.
 Example: Node B does not send any data at all, while node C is more active and hence is
polled more frequently
 In STDM connect more number of terminals because not all terminals are active and
wanting to send data all the time.
 In STDM a time slot could be allocated to any of the nodes depending upon its past
activity and the pending current requests.
 STEM along with the chunk of the data some control information is sent in the data frame
which contains the addresses of the sender and the receiver as well as the number of bytes
sent etc.

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TDM and The Digital telephone system

 TDM helps in the creation of a digital system hierarchy. This is called Digital Signal
(DS) Service

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FDM Vs TDM:

FDM TDM
Analog Transmission Technique Digital transmission Technique
It is not flexible , because intermix terminals It is more flexible because intermix terminal
and synchronous method is not possible with different speed and synchronous method
FDM is simpler system to implement, cost is Implementation cost is more
minimum
FDM is analog transmission not suitable for all It is more popular because digital transmission
types of information for all types of information

Transmission Errors: Detection and Correction

Errors can be classified into 3 categories
1. Delay Distortion
2. Attenuation
3. Noise
Delay Distortion
 It is caused because the signals of varying frequencies travel at different speeds along the
medium
 Any complex signal can be decomposed into different sinusoidal signals of different
frequencies resulting in a frequency bandwidth for every signal.
 Signal propagation is that the speed of travel of the frequency is the highest at the center
of this bandwidth and lower at both ends.
 At the receiving end signals with different frequencies in a given bandwidth will arrive at
different times.
 If the signals received are measured at a specific time they will not measure up to the
original signal resulting in its misinterpretation
Attenuation
 Attenuation is another form of distortion. A signal travels through any medium its
strength decreases.
 The voice becomes weak over a distance and loses its contents beyond a certain distance
 Attenuation is very small at short distances. Attenuation increases with distance. It is
higher at higher frequencies.

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Noise
 Noise is another component that poses a problem in receiving the signal accurately.
 A signal travels as an electromagnetic signal through any medium.
 Electromagnetic energy that gets inserted during transmission is called noise.
Types of Errors:

 Single bit error a bit value of 0 changes to 1 or vice versa.

 Single bit errors a single bit o f the data unit changes example 0 bit changes to 1 or a bit 1
changes to 0.It is occur in Parallel transmission
 Burst error change at least two bits during data transmissions because of errors. Burst
error can change any two or more bits in a transmission. Burst errors are more likely in
serial transmission
Error Detection
 There are number of techniques used for transmission error detection and correction
Vertical Redundancy Check (VRC) or Parity Check
 It is known as parity check. It is the least expensive techniques.
 The sender appends a single additional bit called the parity bit to the message before
transmitting it.
 There are two schemes: 1. Odd parity 2. Even parity
 Example: A message string 1100011 that need s to be transmitted, assume the even parity
scheme.
 Step1: The sender examine this message string and notes that the number of bits
containing a value 1 in this message string is 4. It adds extra 0 to the end of this message.
This extra bit is called parity bit
 Step2 : The sender sends the original bits 1100011 and the additional parity bit 0 together
to the receiver
 Step 3: The receiver separates the parity bit from the original bits and it also examines
the original bits
 Step 4 : The receive now computes the parity bit again and compares this computed
parity bit with the 0 parity bit received from the sender , it notes that they are equal and
accepts the bit string a correct

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Longitudinal Redundancy Check (LRC):

 A block of bits is organized in the form of a list in the Longitudinal Redundancy check .
 If we want to send 32 bits, we arrange them into a list of four rows.

Cyclic Redundancy Check (CRC)

 A mathematical algorithm is used on the data block to be sent to arrive at the CRC a
small block of bits which are appended to the data block and sent by the sender
 At the destination the receiver separates the data block ,re-computes the CRC using the
same algorithm and matches the received CRC with the computed CRC.A mismatch
indicates an error
 Main Features of CRC
 CRC is a very study and better error detection method compared to others. To compute
the CRC is chosen that given the length of the data block in bits there are only a few and
finite number of permutations and combinations for which the CRC is the same.
 CRC is normally implemented in hardware rather than in software. Two simple hardware
components are used : an XOR gate and a shift register.

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 The data to be transmitted is divided into a number of blocks consisting of several bits
each. It is then divided by a prime number and the remainder is treated as CRC.
\

Recovery from Errors:

 There are multiple ways of the recovery from error once detected.
 Positive Acknowledgement (ACK) - back to the sender if every thing was OK that is
after checking that the CRC was matching and correct
 Negative Acknowledgement (NAK) – if there was any error found it will send a
negative acknowledgement to the sender
Stop-and-Wait :
 This is a very simple method wherein the senders sends one frame of data and necessarily
waits for an acknowledgement(ACK) from the receiver before sending the next frame
 The transmission always takes the form DATA-ACK-DATA-ACK –etc.
 Data frames are sent by the sender and the ACK frames are send by the receiver back to
the sender
 It is simple to implement , every frame must be individually acknowleged before the next
frame can be transmitted, it makes transmission very slow.
 The sender sets a timer for every frame that it sends. If it does not receive an
acknowledgement from the receiver before the timer expires it sends the same frame
again with a new timer.
 In this case the receiver gets two or more instances of the same frame .

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 There are schemes which improves the efficiency of the transmisison by sending multiple
frames say 1-8 at time. If it is OK the receiver can send an ACK in one shot of all the
frames.


Go-back-n:
 Go-back-n which is used when more than one frame is sent at a time
 The sender starts the retransmission with the last unacknowledged frame even if the
subsequent frames have arrived correctly at the receiver.
 The sender send 5 frame to the receiver , the frames 0,1,2 arrive correctly at the receiver
but frame 3 is an error. The receiver sends NAK for frame 3 . By the time NAK reaches
the sender the sender may have already sent frames 4 and 5
 After receiving the NAK the sender now retransmits frame and all frames transmitted
after frame 3 ie frame 4 and frame 5 to the receiver
 This retransmission is successful the receiver acknowledge this correct transmission of
frames 3,4 and 5 after discarding the duplicates for frames 4 and 5
 Another situtation where the sender has transmitted all its frames and is actually waiting
for an acknowledgement that has been lost on the way.
 The sender waits for some time and then retransmits the unacknowledged frames. The
receiver detects this duplication sends another acknowledgement and discards the
redundant data
 Another possibilty is when a damaged frame is received. Frame 0 and 1 arrive correclty,
but the receiver dose not immediately acknowledge them. It receives frame 2 which is
erro.It returns NAK2. This informs the sender that frames 0 and 1 have been received
correclty but that frame2 must be resent.
 In this case the receiver keeps accepting new frames and does not reject them.In this
scheme the receiver needs lot of memory and processing logic to keep a track of all the
frames.

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Sliding Window
 The sliding window technique is a variation of the Go-back-n technique
 The sender must send a frame wait for its acknowledge and only after it receives that
acknowledgement , send the next frame in stop and wait.
 To improve the efficiency would be to send multiple frame at a time, check CRC of all
the frames one by one and send the acknowledgement for all and request for the next set
of frames. The sliding window technique is based on this technique
 A Sliding window o size of 8 frames that is the sender can send eight frames before it
must wait to receive an acknowledgement from the receiver.

 The underlying transmission mechanism defines as imaginary window consisting of

maximum n frames to be sent at a time.
 The transmission mechanism allows the data to be transmitted at a time only up to the
size of the window. The window defines how much data can be sent before an
acknowledgement is received .The term sliding window is used because the data window
slides over the data buffer to be sent
 Bothe the sender and the receiver maintain their own sliding window. The sender sends
the number of frames that is allowed to by its sliding window and then waits for an
acknowledgement from the receiver.
 When the receiver sends an acknowledgement back to the sender it includes the number
of the next frame that it expects to receive
 I f the sender has sent frames numbered 1 to 3 to the receiver assuming that the receiver
has received them correctly the receiver sends back an acknowledgment the includes the
number 4 .
 The sender knows that the receiver has correctly received frames 1 to 3 and proceeds to
send frame number 4.

 Transmission mechanism uses two buffers and on window to control the flow of data.

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 The sender has a buffer for storing data coming from the sending application program.
 The application program creates the data to be sent and writes it to this buffer. The sender
imposes a window on this buffer, and sends frames till all the frames have been sent.
 The receiver also has a buffer. The receiver receives the data, checks it for any errors and
stores the correct ones in this buffer.
 The application program at the receivers end then pick up data from this receive buffer

 Assuming that the sender’s window size is 8 ,if frames 1 to 4 are sent and no
acknowledgement has been received , the sender’s window shall contain four frames
numbered 5 to 8.
 After an acknowledgement number 5 arrives, the sender knows that the receiver has
correctly received frames 1 to 4. The sender ‘s window in now expanded to include the
next four frame making the window consist of framed 5,6,7,8,1,2,3,4.The sender window
shrinks form the left when it sends data frame s and expand to the right when it receives
acknowledgement from the receiver
 The receiver window shrinks from the left when frames are received and expand to the
right when it sends acknowledgement to the receiver

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Data Communication and Networks – Unit III

TRANSMISSION MEDIA
 Transmission media are the physical infrastructure components that
carry data from one computer to another

 There are two categories of transmission media used in computer

communications.
1. GUIDED MEDIA
2. UNGUIDED MEDIA
GUIDED MEDIA:
 Guided media are the physical links through which signals are
confined to narrow path.
 It is also called wired media
 Three common types of guided media are used of the data
transmission. These are
1. Twisted Pairs Cable
2. Coaxial Cable
3. Fiber Optics Cable
Twisted Pair Cable
There are two classed of twisted pair cables
1. Unshielded Twisted Pair(UTP)
2. Shielded Twisted Pair (STP)
1. Unshielded twisted pair (UTP)
 UTP is more common because of its usage in the telephone system.
 This cable can carry both voice as well as data.
 It consists of two conductors. In the beginning the wires used to be
kept parallel. This results in greater levels of noise.
 The copper conductors are covered by PVC or other insulator.
 UTP is flexible ,cheap and easy to install.

1
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 The Electronic Industries Association (EIA) has developed standards

for UTP cables.
There are five categories of UTP

Characteristics of UTP Advantages of UTP

• Low cost • Easy installation
• Easy to install • Capable of high speed for LAN
• High speed capacity • Low cost
• High attenuation Disadvantages of UTP
• Effective to EMI • Short distance due to attenuation
•100 meter limit

2.Shielded twisted pair (STP)

 The twisted wire pair itself is covered by metal shields and finally by
the plastic cover.
 The metal shield prevents of electromagnetic noise.
 It also helps to eliminate crosstalk. The shield prevents such
unwanted sounds.

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Characteristics of STP Advantages of STP:

• Medium cost • Shielded
• Easy to install • Faster than UTP and coaxial
• Higher capacity than UTP Disadvantages of STP:
• Higher attenuation, but same as UTP • More expensive than UTP and coaxial
• Medium immunity from EMI • More difficult installation
• 100 meter limit • High attenuation rate
COAXIAL CABLE:
 Coaxial cable is also called as coax
 It has an inner central conductor surrounded by an insulating sheath
which in turn is enclosed in the outer conductor.
 A second conductor for completing the circuit but also acts as a
shield against noise. This outer conductor is covered by a plastic
cover.
 Coaxial cable is more expensive, less flexible and more difficult to
install in a building. It is much more reliable and can carry far higher
data rates.
 The various coaxial cable standards are RG-8,RG-9,RG-11,RG-58 and
RG-59

Characteristics Of Coaxial Cable Advantages Coaxial Cable

 Low cost  Inexpensive
 Easy to install  Easy to wire
 Up to 10 Mbps capacity  Easy to expand
 Medium immunity form EMI  Moderate level of EMI immunity
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 Medium of attenuation Disadvantage Coaxial Cable
 Single cable failure can take down an
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OPTICAL FIBER
 Optical fibers use light instead of electrical signals as a means of
signal propagation. It is made of glass fibers that are enclosed in a
plastic jacket.
 A laser at one device sends pulse of light through this cable to other
device. These pulses translated into “1’s” and “0’s” at the other end.
 The transmitter at the senders end of the optical fiber sends a light
emitting diode (LED) or laser to send pulses of light across the fiber .
 A receiver end makes use of a light sensitive transistor to detect the
absence or presence of light to indicate a 0 or 1
 In the center of fiber cable is a glass stand or core. The light from the
laser moves through this glass to the other device around the
internal core is a reflective material known as CLADDING.
 The outer jacket can be made of Teflon ,plastic or metal
 Optical fiber has bandwidth more than 2 gbps (Gigabytes per Second)

Characteristics Of Fiber Optic Cable: Advantages Of Fiber Optic Cable:

 Expensive • Resistance to noise
 Very hard to install • Huge Bandwidth
 Capable of extremely high • Higher signal carrying capacity
speed Disadvantages Fiber Optics:
 Extremely low attenuation • Fragility
 No EMI interference • Cost
• Maintenance overhead

Reflection and Refraction

 When the medium through which the light passes changes its speed
and the direction also changes suddenly this is called refraction
 If we change the direction of the incoming light, the direction of the
refracted light also changes in fig a,b,c.
 We reach a stage where the refracted light becomes horizontal fig d,
after this stage if the angle still changes ,the light is not refracted
instead it is reflected

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Propagation Modes
 There are different modes of propagation , depending upon the
physical characteristics of the fiber and the light source

Multimode
 LED is mostly used as a light source.
 Multiple beams pass through the core in different path
Step Index
 The core has one density and the cladding has another.
 Therefore at the interface there is a sudden change. Multiple beams
take different paths on reflection.
 The beam that strikes the core at a smaller angle has to be reflected
many more times than the beam that shifted the core at a larger
angle to reach the other end.
 At the destination all beams do not reach simultaneously creating
diffusion and confusion in terms of interpretation

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Graded Index
 The core itself is made of a material of varying densities .The density
is the highest at the core and gradually decrease towards the edge
 A beam goes through gradual refraction giving rise to a curve, except
that the horizontal beam travels unchanged
 Different beams result in different curves or waveforms

Single Mode
 This uses a highly focused light beam travels more or less
horizontally.
 The fiber core diameter in this case is smaller than multimode and
lower density
 This decrease results in critical angle close to 90 degree to make the
propagation of different beams very close to horizontal

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UNGUIDED MEDIA
 It is also called wireless communication , transport electromagnetic
waves without using physical conductor

Radio Communication

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Microwave Communication
 Microwaves use the line of sight method of propagation , as the
signals do not travel along the surface of the earth.
 Two antennas must be in a straight line, able to look at each other
without any obstacle in between
 The antennas are positioned on mountain tops to avoid obstacles.
Microwave signal travel only in one direction at a time
 This means that for two-way communication such as telephony two
frequencies need to be allocated.
 At both ends a transceiver is used which is a combination of a
transmitter and a receiver operating at the two respective
frequencies
 Repeaters are used along with the antennas to enhance the signal.
The data rates offered are 1Mbps- 10 Gbps
 Microwave is also relatively inexpensive

Satellite Communication
Introduction
 Satellite communication is similar to the terrestrial microwave except
that the satellite acts as of the stations
 It’s function like an antenna and repeater together
 Ground station A can send the information to ground station B via
the satellite
 If the earth along with its ground stations is revolving and the
satellite is stationery the sending and receiving earth station and the
satellite can be out of sync over time
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 Normally Geosynchronous satellites are used which move at the

same Revolutions Per Minute (RPM) as that of the earth in the same
direction
 Both earth and the satellite complete one revolution exactly in the
same time the relative position of the ground station with respect ot
the satellite never changes
 The movement of the earth does not matter for the communication
nodes based on the earth
 Minimum of three satellite is needed to cover the earth’s surface
 SHF which covers the frequency range of 3 GHz to 30 GHz is used for
satellite communication
 UPLINK : The signal from the earth to the satellite
 DOWNLINK : The signal from the satellite to the earth
Access Modes
 Satellite communication is based on modulation techniques
 Three primary modulation techniques are
1. Frequency Division Multiple Access(FDMA)
2. Time Division Multiple Access (TDMA)
3. Code Division Multiple Access (CDMA)
 To splits the transmissions based on that access methods
1. FDMA puts each transmission on a separate frequency
2. TDMA assigns each transmission a certain portion of time on a
designated frequency
3. CDMA gives a unique code to each transmission and spreads it
over the available set of frequencies
4. Multiple Access means that more than on user can use each cell

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Frequency Division Multiple Access (FDMA)

 FDMA splits the total bandwidth in to multiple channels
 Each ground station on the earth is allocated a particular frequency
group
 Each group ,the ground station can allocate different frequencies to
individual channels , which are used by different stations connected
to that ground station
 The transmission begins , the transmitting ground station looks for an
empty channel within the frequency range that is allocated to it and
once it finds an empty channel , it allocates it to the particular
transmitting station
 Most popular method of communication using satellite – TV
transmission works

Time Division Multiple Access (TDMA)

 In TDMA each transmitter is allocated a predefined time slot

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 Each transmitter receives the time slot in turn and it is allowed to

transmit data for the duration of the time slot
 In TDMA there are no modulation frequencies. It is a digital form of
data transmission
 Transmit data in the form of packets of data. These data packet
arrive at the satellite one by one form of data transmission
 Bit rates of 10-100Mbps. This can be translated in to roughly 1800
simultaneous voice calls using 64 Kbps PCM

Code Division Multiple Access (CDMA)

 CDMA allows any transmitter to transmit in any frequency and at any
time
 CDMA uses coding that is unique to a particular transmission and
allows the receiver to disregard other transmissions on the same
frequency
 The coding scheme is a unique frequency with which the original
signal is modulated to form the codified signal that is to be
transmitted
 CDMA is the newest and the least used access method using satellite

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Cellular (Mobile) Telephones

Introduction
 First mobile telephone in 1946 – in USA. It was difficult to operate
and had a single channel for both sending and receiving.
 To push a button to enable the transmitter and disable the receiver.
 This half duplex system known as push-to-talk system in 1950
 Second development in 1960. This was called Improved Mobile
Telephone System (IMTS)

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 It used two frequencies, one frequency used for sending and the
other for receiving. This full duplex system had 23 channels.
 The third development was Advanced Mobile Phone
System(AMPS).In England It is called TACS and in Japan MCS-L1
 The area covered is conceptually divided in small regions known as
cells called cellular phones
 Each cell has an antenna and a cell office to control the cell.
 A Mobile Telephone Switching office (MTSO) controls various cell
office and coordinates the communication between them and the
Telephone Central Office (TCO) or a telephone exchange.
Bands in Cellular Telephony
 Analog transmission is used for cellular telephony
 Frequency modulation is used for communication between the
mobile phone and cell office.
 Two frequency bands are allocated , one for the communication by
the mobile phone and other for the land phone. The two bands are
typically 824-849 MHz and 869-894 MHz
 Each channel requires a full duplex dialog. For preventing
interference adjacent channels are rarely allocated
 The same frequency band can be used for multiple non-adjacent cells
 In USA channel available for each cell is 40.

Calls using Mobile Phones

 The mobile phone by entering a 7,8 or 10 digit phone number , the
mobile itself scans the band and seek a channel for setting up the call

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 After this, it send this number to the closet cell office which in turn
sends it’s to MTSO in turn sends it to the CTO. MTSO allocates an
empty voice channel to the cell to establish the connection
 Paging: When a land phone places a call to mobile, the telephone
central office sends the number to the MTSO. The MTSO performs a
look up to see where the mobile is currently placed by sending
appropriate query signals t all the cells. This process is called paging
Transmitting /Receiving/Handoff Operations
 During the conversation, if the mobile phone crosses the cell, the
signal can become weak.
 The MTSO constantly checks the signal level and if it finds it low it
immediately seeks a new cell. The MTSO then changes the cell
carrying channel.
 The process of handling the signal off from the old channel to the
new one is called handoff.
Step 1: Cell A senses that the user of cell 50 is moving to cell B and that its
signal is becoming weaker. So it alerts the MTSO.

Step2: The MTSO wants to check if any of the adjacent units can take up
the responsibility of unit 50
Cell C responds by saying that it cannot do so, as unit 50 is weak in cell C,
too.

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Step 3: The MTSO redirects unit 50 from cell A to cell B

New Developments
Three developments
1. Digital cellular telephone
2. Integration of cellular phones with satellite communication
3. Integration of the mobile telephony with the PC. This is called Mobile
Phone Communication

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Shannon Capacity
 Claude Shannon introducing a formula that determines the maximum
data rate of a channel .This is called as the Shannon capacity
 Determine the maximum data rate of a transmission medium the
formula reads:

 C is the Shannon capacity in bps, B is the bandwidth of the

transmission medium and S/N is the signal-to –noise ratio of the
transmission medium

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Network Topologies , Switching and Routing Algorithms

 Network topology defines how various computers or nodes are
connected to one another.
 There are 6 basic topologies

Mesh Topology
 Mesh topology is also called complete topology
 Each node is connected to every other node by direct links
 For m node there would be m(m-1)/2 physical links .This means that
every node must have (m-1) I/O port
 Mesh topology does not have traffic congestion problems there are
dedicated link.
 The link are not shared a special Media access Control(MAC) protocol
a part of the data link layer of the OSI model is not needed to decide
who should communicate to whom and for how long.

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 Merits :
1. If one link is down the rest of the network can still continue.
2. Fault identification also easy.
3. Robust
 The main demerit is the cable length.
 Example : 1000 nodes one will require 1000(1000-
1)/2=1000X999/2=499500 cable or links
Star Topology
 Star topology have a central node often called a hub
 If a node wants to send some data to another node it send it to this
hub.
 The hub sends it to the appropriate node
 Merits :
1. Cheaper than mesh topology.
2. If one link goes bad all other node and that link continue to
function
3. Easier to install, maintain and reconfigure.
4. Robust
 Demerit : If the hub goes down , the entire network become defunct

Tree Topology
 It can be derived from the star topology.
 Tree has a hierarchy of various hubs like branches in tree.

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 Every node is connected to some hub.

 Only a few nodes are connected directly to the central hub
 The central hub contains a repeater, which looks at the incoming bits
and regenerates the signal for 0 or 1 as required
 This allows the digital signals to traverse over longer distance
 The central hub is also called active hub
 The tree topology also contains may secondary hub which may be
active hubs or passive hub

Ring Topology
 Each node is directly connected to only its two adjacent neighbours
 If a node wants to send something to a distant node on a ring, it has
to go through many intermediate nodes which act like repeaters
,reproducing the incoming bit stream with full signals on the
outgoing line.
 Merit :
1. Easy to reconfigure and install
2. A node not receiving any signal for a long time indicates a fault
 Demerit :
1. If a node in a ring fails the whole ring cannot function
2. The traffic is only in one direction

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Bus Topology
 Bus topology uses multipoint philosophy
 A Long cable bus forms the backbone to all the nodes
 A node wanting to send some data to some other node pushes the
data on the bus which carries it to the other node.
 A Tap is connector that connects that node with the metallic core of
the bus via drop line
 Merit :
Easy to install and uses less cable
 Demerit:
1. Difficult to add new nodes to a bus because of a node changes
number of tap and the average distance between them.
2. It more inflexible
3. Fault isolation is very difficult
4. If a portion of the bus breaks down the whole bus cannot function

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Hybrid Topology
 Hybrid topology is one that uses two or more of the topologies
together
 In this case the bus, star and ring topologies are used to create this
hybrid topology
 Topology refers to the logical arrangement of the nodes and not the
physical appearance

Switching Basics
 Switching refers to the technique of connecting computers to a
central node called a switch which can then be used to connect to
other nodes
 A switched network is made up of a number of interlinked nodes
called switches
 A switch is a hardware as well as software device that allows a
connection to be established between two or more devices which
are linked to it
 Example : Switched Network

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Switching Methods
1. Circuit Switching
2. Packet Switching
3. Message Switching
Circuit Switching
 A direct physical connection path is established between two
computers.
 Example: Telephone call
 It is more suitable for human communications
 When a computer wants to communicate with another computer a
dedicated connection is established between them over the
switches.
 The computer can then communicate using the connection , when
the communication is over either computer cab send a request for
terminating this connection and only at that stage would the
connection be released.
 The two computers communicate over this dedicated connection no
other computer can use this portion of the connection.
 The total transmission line is exclusively reserved for the two
computers using that connection
 Computer A wants to send some data to computer D .The dark line
shows the dedicated channel between computers A and D.
 A circuit switch is a device with m inputs and n outputs where m and
n need not be equal.
 It creates a temporary dedicated connection between an input
device and output device

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Packet Switching
 Packet Switching has emerged switching technology for computer-to-
computer communications
 Most of the protocols such as x.25,TCP/IP ,Frame Relay ,ATM etc use
packet switching
 In Packet switching data to be sent is divided into discrete blocks
called packets which are variable length
 The maximum size of data called packet size or packet length
 Each packet contains data to be transferred and also control
information such as sender address and destination address
 Packet switching can be classified into two types

Datagram Approach
 Each packet or datagram is considered as a totally independent
packet from all others
 When there are multiple packets sent by the same source to the
same destination for the same message each packet is independent
of all other packets can take different routes or path
 Computer A is sending four packets to another computer D
 These four packets belong to the same original message but travel
via different routes and also can arrive at the destination D in a
different order than how the source A sent them
 The destination node needs to have a buffer memory to store all the
packets and at the end to re-sequence them to form the original
message
 Each packet must have a header containing at different routes is that
the routing decisions are taken for every packet ,each time at every
node as the packets move from one node to the next.

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 The network condition and congestion at different nodes/links differs

every second ,different packets may choose different routes based
on the situation at that time.

Virtual Circuit Approach

 All the packets belonging to the same message take the same route
from the source to the destination
 At the beginning a single route is chosen between the source and the
destination before the actual data transfer takes place
 A unique Virtual Circuit Number (VC number) is allocated to that path
between the source and the destination
 Each node on that path maintains just two entries : VC number and
the address of the next node to which the packet must be forwarded
in order to reach the destination
 There are three phase
1. Connection Establishment Phase
2. Data Transfer Phase
3. Connection Release Phase
 A single route is chosen between the source and the destination
before the actual data transfer takes place.

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 All the packets reach the destination in sequence only

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Message Switching
 It is known as store-and-forward approach for an entire message
 A Computer receives a message stores it on its disk until the
appropriate route is free, and then sends it along the route
 There is no direct link between the source and destination
 In message switching the computer stores the packet to be
forwarded on its route in its main memory until it can be forwarded
 The message switching approach is now replaced by message queue
technologies such as Microsoft Message Queue (MSMQ) and IBM
MQ series

Router and Routing

 A router is a device that connects two or more computer networks
together. This allows two or more different computer networks to
send data to each other
 A router connecting to two network via: A (Token Ring) and B
(Ethernet ) at points X and Y respectively
 This means that the router must have two interfaces and also two
network interface card (NIC) one to interact with network A at point
X and the other to interact with network B at point Y.
 Routing algorithms determine the routing decisions that how a
forward a packet on to its next journey

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Factors Affecting Routing Algorithms

1. Least - cost Routing
 The routers to select the shortest possible path when sending
packet forward
 When a packet is routed its routing cost is calculated , the
value is assigned to each links
 The routing cost is then the aggregate value of all these links
for the route taken by a particular packet
 The routing cost need not depend on the number of routes
encountered alone
2. Distributed Routing
 Once a path from the source to the destination is selected the
routers sends all the packets for that destination via that same
route as in the case of virtual circuit philosophy
 Other routing protocols use a technique called distributed
routing

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 where in a router may as a result of the changes in the lengths

of the links , choose a new route for each packet , even if the
packets belong to the same original message
3. Packet Lifetime
 Once a router decides the path to be taken by a packet, it
forwards the packets to the next router
 In datagram approach the entire route is not pre-decided. It
can change for each packet and for each packet at a given
router the subsequent path may be recomputed and update its
routing table
 This means that the packet may not reach the ultimate
destination at all.
 To resolve this additional information is added to the header
of a packet, which dictates how long a packet can live called as
packet lifetime.
 This field contains the number of hops that are allowed for
that packet before a packet is considered to be lost and
therefore destroyed at its current location
Routing Algorithms

There are two main categories of routing algorithms used to calculate the
shortest path between two routers:
1. Distance Vector Routing
2. Link State Routing
Distance Vector Routing
Introduction

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How Does Distance Vector Routing Work?

 The actual working of distance vector routing , consider a sample
network of networks called an internet
 There are seven networks numbered 1 to 7 connected to each other
by six routers A through F
 Each router is connected at least two networks but it may also be
connected to more than two networks
 Distance vector routing assumes that the cost of every link is 1
 The transmission efficiency depends only on the number of links
required to reach the destination
 Distance vector routing depends only on hop count and not on any
other factors such as congestion level at every link or even the cost
of every link
 Each router sends the information about the internet only to its
immediate neighbours
 Example Router A sends the information to neighbouring routers B,F
and E

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Routing Table
 For storing information about the network and keeping it updated
each router uses a routing table
 The routing table has at least 3 columns
1. Network id : This is the final destination of the packet
2. Cost : This is the number of hops that a packet must take to reach
the final destination
3. Next router : This is the router to which the packet must be delivered
on its way to a particular final destination .The network id of the
destination is mentioned in the network id field of the table

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 The initial entries in the routing tables of the various routers in

network. The three columns correspond to the three columns of the
table
 The routing table of router A tells that it can send any packet to
networks 1,6, or 5 and that the cost for each packet forwarding
process is 1.
 Note that the third column that is the next router is empty, because
at this stage the routers have only the information about the
networks to which they connect directly
 Router A sends its routing table to neighbouring routers B,F and E.
Similarly router B send its routing table to its neighbouring routers A
and C and so on

Initial entries in the routing tables of various routers

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Routing Table Updation

 Router A takes the routing table of B and adds 1 to the next hop field.
This is because whatever is accessible to B directly is not accessible to
A directly
 For instance, network 2 is not directly accessible to A. But now A can
access network2 with the help of B with one extra hop

 Created the temporary routing table router A now adds the contents
of this temporary routing table to its own routing table and sorts it
on Destination network id to get a combined sorted table

 Router A removes the duplicate entries from the combined table to

get the most optimized results, it select the shortest route when
there are multiple ways of reaching a final destination

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 The routing tables of all the routers get updated with the information

Link State Routing

Introduction

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Packet Cost
The cost is not directly based on the hop count .It is weighted based on a
number of factors such as security levels, traffic and the state of the link
How Does Link State Routing Work?
 The Link state routing works with the help of the same internet. Each
router sends its knowledge only about its neighbours to every other
router in the internet
 Link state packet
 Each router sends a very small greeting packets to each of its
neighbours and expects a response in return
 If the neighbor replies the original router considers that the neighbor
is up and running and determine the cost based on the factors
 Using this information the original router then sends information
about all its neighbours to the entire internet in a process called
flooding

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 It sends special packet called as Link State Packet (LSP) to all other
routers via its neighbours
 General Format of the LSP

 Advertiser – it is the id of the router which is sending the LSP packet

 Network – this is the id of the destination network to which this
packet should go
 Cost – the cost calculated based on different fackors
 Neighbour – the id of the neighbouring routers about which this
information is being sent in the LSP

Link State Database

 Every router receives every LSP packet and uses it to create a local
database called link state database
 A link state database is the collection of all LSP.
 Every router stores such database on its disk and uses it for routing
packets

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Dijkstra Algorithm
 The Link state database each router executes an algorithm called as
dijkstra algorithm to create its routing table
 This algorithm considers the internet as a graph and find the distance
along a shortest path from a single node of the graph to all other
nodes in the graph
 A routing table is created to compute the shortest path
Approaches to Routing
S.No Static Routing Dynamic Routing
1 Static routing manually sets up the The dynamic routing uses dynamic
optimal paths between the source and protocols to update the routing table
the destination and to find the optimal path between
the source and the destination
2 Static routing algorithm do not have Dynamic routers can sense a faulty
any controlling mechanism if any faults router in the network
in the routing paths
3 Static routing is suitable for very small Dynamic routing is used for larger
networks networks
4 The static routing is the simplest way of dynamic routing uses complex
routing the data packets algorithms for routing the data packets
5 Static routing requires minimal memory Dynamic routing a few memory
overheads
6 Network administrator finds out the The algorithm and the protocol is
optimal path and makes the changes in responsible for routing the packets and
the routing table making the changes accordingly in the
routing table
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Physical Layer

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Network Protocol and OSI Model

Introduction
 Protocol is nothing but a convention
 A message is a block of statements or sentences. A message could
also consist of only one word OK or Yes denoting a positive
acknowledgement (ACK) of what has been received
 A message could also mean a negative acknowledgement (NAK) or
request for repeating such as come again ,please etc.

 A sentence is like a packet in computer. The previous statement was

not received properly, repeats the sentences. A sliding window
would mean speaking and acknowledge multiple sentences
simultaneously

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 The speed mismatch between the speaker and the listener, error
control and flow control take cake of this issues
 The Data link layer is responsible for error control and flow control
 This layer also decides by a convention , who is going to speak when ,
who has a control of the medium. This is called media access control.
 Exactly at the same time the other party also can start speaking ,
thinking that you want the other party to speak. This results in a
collision
 The electrical signals in the telephone wires change .This is a physical
layer
Protocols in Computer Communications

 Each compute is called node. In distributed processing different parts

of databases /files can and normally do reside on different nodes
 Connecting every node to every other node need huge amount of
wiring
 This is the reason that the concept of Store and Forward is used in
computer networks
 A memory buffer to store the file
 Some software which controls the queuing of different messages and
then transmitting them to the next nodes
 When the file/message is transmitted both the nodes as well as all
the intermediate nodes protocol at the physical level is called
physical layer.
 It deals with what are bits 0 and 1 the communication modes
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 The next node find out whether the file or message was received
correctly or not
The OSI Model
Introduction
 Host X wants to send a message to another host Y. This message
would travel via a number of intermediate nodes.
 These intermediate nodes as well as X and Y are concerned with the
three lowermost OSI layers ie physical , data link and network
 The other four layers are used by the sender X and the receiver Y
only, they are called end-to-end layers

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 Between X and Y the communication appears to be taking place

between the layers at the same level. This is called as virtual
communication or virtual path between X and Y.
 When node A want to send a message to node G node A sends its to
switch RA .After it gets through a specific route to router RF and then
it reaches the node G
 Switches in a Network

OSI Layer Functions

1.Physical Layer
 It is concerned with sending raw bits between the source and
destination nodes
 Physical Layer Factors:

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Data Link Layer

 The data link layer is responsible for transmitting a group of bits
between the adjacent nodes. The group of bits is called frames or
packet.
 The network layer passes a data unit to the data link layer. The data
link layer adds the header and trailer information
 The header contain the addresses and other control information. The
addresses at this level refer to the physical addresses of the adjacent
node in the network

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 Example Node A want to send a packet to node D using the

datagram approach , the logical address (ie IP address)of nodes A nd
D say ADDL(A) and ADDL(D) are the source and destination
addresses.
 The data unit passed by the network layer to the data link layer at
node A will contain these addresses . The data unit DN will looks

 When this data unit (DN) is passes from the network layer at node A
to the data link layer at node A the following happens
1. The routing table is consulted, which mentions the next node to
which the frame sent for a specific destination node
2. The data link layer at node A form a data unit DD

3. Using the physical address of adjacent nodes A and F the packet

move form node A to node F after performing the control functions
4. The routing algorithm is performed at node F using ADDR(D) as the
final destination
5. The network layer at node F passes DN to the data link layer at node F

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6. This continues until the data unit at data link layer DD reaches node
D. The physical addresses are removed to get the original DN which is
passes on to the network layer at node D

 The data link layer also perform the flow control function, based on
the speeds of the CPU, transmission ,buffer size and congestion
control
 If the connection is a multipoint type then the problem of who
should send how much data and at what times has to be solved. This
problem arises in Local Area Networks(LAN) and is solved by the
Media Access Control(MAC) protocol.
In LAN the data link layer is split into two sub layers

 LLC – takes care of normal data link layer functions such as error
control and flow control etc.

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Network Layer
 It is responsible for routing a packet within the subnet that is from the
source to the destination nodes across multiple nodes in the same
network or across multiple networks.
 This layer is ensures the successful delivery of a packet to the
destination node.
 This layer is also responsible for tackling the congestion problem at a
node
 This layer has to carry out the accounting function to facilitate this
billing based on how many packets are routed, when etc.
 A router can connect two networks with different protocols ,packet
length and formats

Transport Layer
 The transport layer is the first end-to-end layer . All the lower layers were
the protocols between the adjacent nodes

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 The transport layer ensures that the complete message arrives at the
destination .
 The transport layer receives data from the session layer on the source
computer which need to be send across to the other computer.
 The transport layer on the source computer breaks the data into smaller
packet and gives them to the lower layer. It adds sequence number to
the packets.
 It also establish a logical connection between the source and destination
The connection consist of three phases : Establishment , Data transfer
and Connection release

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Session Layer
 The main function of the session layer is to establish, maintain and
synchronize the interaction between two communicating hosts.
 Example: To send a big document consisting of 1000 pages to
another user on a different computer. Suppose that after the first
105 pages have been sent the connection between two hosts is
broken for some reason. These issues are the concerns of the session
layer
 To avoid complete retransmissions from the first page the session
layer between the two hosts could create sub-session. After each
sub-session is over a checkpoint can be taken.
 The connection breaks after the first 105 pages haven been
transmitted after the connection is restored the transmission would
start at the 101st page. This is because the last checkpoint would
have been taken after the 100th page was transmitted

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Presentation Layer
 When two host are communicating with each other , they might be
using different coding standards and character sets for
representation data internally
 One host using ASCII code for character representation the other
host using EBDCDIC
 The presentation layer has to take care of such differences
 It is also responsible for data encryption and decryption for security
and data compression and decompression for more efficiency in data
transmission

Application Layer
 It is the topmost layer in the OSI model, enables a user to access the
network.
 The application programs using the network services also reside at
this layer

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 This layer provides user interface for network applications such as

remote log in (TELNET),World Wide Web(WWW), File Transfer
Protocol(FTP), Electronic Mail, remote database access etc.

Integrated Services Digital Network (ISDN)

Background of ISDN
Analog Communication
 The telecommunication network existed everywhere , there was no
choice for the technologies but to reuse the analog telephone lines
for data transmission as well

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 Modems are used to modulate the digital data arriving from the
computer at the senders end into analog signal which are then
carried over by the traditional analog telephone network (Central
Office or CO) to the receiver’s end.

 To increase data rates over the traditional copper wires techniques

such as Quadrate Amplitude Modulation (QAM)
 This technologies can utilize the bandwidth of the copper wire
beyond the 4 kHz limit and provide higher data rates. ISDN can
provide higher data rates over the existing copper wires
 Analog transmission suits voice traffic .The analog signals coming
from a telephone do not require any kind of transformation before
they can travel across a telephone network.

Digital Communication
 In digital communication telephone companies started to provide
digital communication backbones.
 This means that the communication between the Central Office(CO)
and the telephone network became digital

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 To achieve this the incoming analog signal at the CO were

transformed into digital pulses by sampling techniques such as PCM.
 Huge investments in analog transmission between the CO and the
end users (Called Last Mile)

Integrated Service Digital Network (ISDN)

 ISDN aims to integrate voice and non-voice services together in
digital form.
 The main service provided by ISDN will be in terms of voice traffic.
 ISDN can provide features in Customer Relationship
Management(CRM) based on features such as caller identification
 ISDN will use this telephone number of the caller as the key to
automatically read the customer’s information from a database
 Other voice services such as call forwarding, conference calls etc.
 The non-voice applications ,online burglar and smoke detector
alarms can automatically call the police ,fire brigade etc and provide
address details using ISDN

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ISDN Architecture
Digital Bit Pipe
 The fundamental concept in ISDN is the digital bit pipe.
 This is a conceptual pipe through which bits flow between the end
user and the CO.
 The bit pipe is bi-directional .Bits can originate at the customer
premises or at the ISDN exchange
 The digital bit pipe supports time division multiplexing technique to
provide for multiple independent channels
 There are two primary sets of users of ISDN services
1. Home users – low bandwidth standard
2. Business users – high bandwidth standard, need multiple
channels provided by a single business bit pipe
ISDN Channel Types
 ISDN offers three separate digital channels

Bearer (B) channels

 Each Bearer (B) channel carries digitized voice, data or compressed
video up to the maximum rate of 64Kbps.
 It is the basic user channel for carrying any kind of digital
information in full duplex mode with a maximum transmission rate
of 64Kbps.
 Voice Signal digitized using PCM. B channel to be replacement of
analog phone system called Plain Old Telephone System(POTS)
 One B channel is destined to a single recipient only
Data (D) channels
 The Data (D) channel is used for signaling /controlling the B channels
and it operates at a data rate of 16/64Kbps
 Controlling signals such as establishing a call, ringing, call interrupt ,
synchronization etc.

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 In the traditional system the same interface carries data as well as

controlling information called as in-band signaling
 ISDN is different in separating these controlling tasks from the actual
data transmission and a separate channel to carry the control signals
called as out-band signaling
 There is a new protocol called as Signaling System Number 7(SS7)
 This can be used over various digital circuit switched networks to
provide for out-band signaling also called Communication Channel
Signaling
 D Channel is similar to a telephone operator. D channel itself can be
used to carry data such as videotext, teletex, emergency services
,alarm system etc
Hybrid(H) channels
 Used for applications such as video conferencing which require
higher bandwidth H channels provides data rates of 384/1536/1920
Kbps.
 H channel can be used to sub divide the channels as per the user’s
own TDM scheme or high speed lines for applications such as faxes
with high speed , high speed data/audio etc.
ISDN Interfaces
There are two interfaces
1. Basic Rate Interface(BRI) – used by home user
2. Primary Rate Interface(PRI)- used by corporate customers
Basic Rate Interface
 It specifies a digital bit pipe that contains two B channels and one D
Channel.
 The 2B+D channels are called as Basic Rate Interface in ISDN
terminology
 ISDN uses a modified form of time division multiplexing technique
 Two B channels require 64Kbps each. The D channels demands 16
Kbps.
 A BRI channel is suitable for residential and small office subscribers. A
home user could use one B channel for browsing the Internet.

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Primary Rate Interface (PRI)

 The Basic Rate Interface(BRI) the Primary Rate Interface(PRI) has
huge capacity.
 A PRI has 23B channels instead of just two B channels. It also has the
D channel for control signals. A PRI service is a very big digital pipe
consisting of 23B channels and 1D channel.
 The D Channel also is of 64Kbps. This gives a total bandwidth of 1.536
Mbps.
 The PRI service itself needs another 8 Kbps for its overheads. The
capacity of a PRI is 1.544 Mbps.
 PRI service contains 23B channels, it means that it supports up to 23
full duplex connections.
 A PRI works with a T-1 line in the US. In Europe the PRI is changed to
have a 30B+D service. The European telephone standard of 2.048
Mbps as used by an E-1 line.

Functional Grouping
Introduction
 The device used to access ISDN are grouped to form functional
grouping.
 A functional grouping clubs various device, depending on their
functionalities.

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 Some modern digital devices such as digital phone are ISDN

compatible.
 Our normal telephone instruments or computers donot understand
ISDN signaling and protocols on their own.
 The functional groupings used at the user’s premises are of three
major types:
 Network Terminations (NT1 and NT2)
 Terminal Equipment(TE1 and TE2)
 Terminal Adapters(TA)
 Terminal Equipments
1. Terminal Equipment 1(TE1)

2. Terminal Equipment 2 (TE2)

3. Network Terminations : Network Termination 1(NT1)

Network Termination 2 (NT2)

Terminal Adapter (TA)

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Reference Points
A reference points defines the interface between two functional groupings.

ISDN Protocol Architecture

ISDN is classified into three protocol blocks or planes

 The user protocol block (or user plane)
 Control protocol block (or control plane)
 Management protocol block (or management plane)

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Physical Layer
 ISDN Protocol use two interfaces at the physical layer : Basic Rate
Interface (BRI) and Primary Rate Interface(PRI)
Physical Layer – BRI
 It consist of 2B channels and 1D channels (2B+D).
 A subscriber can connect to a BRI interface using the R,S,U reference
points.

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BRI frame format

 The B channel consists of two sub channels,B1 and B2. Each B1 and
B2 is 8 bit and occurs twice, making a total of 16 bit B1 and 16 bit B2.
Overall the B channel contains 32 bits. The D channel contains 4 bits.
There are 12 bits reserved as overhead bits.
 There are two overhead bits before the first B1 etc. A BRI frame
consist totally 48 bits.
Physical Layer – PRI
 The PRI interface supports 23B channels and 1D channel. The PRI
frame format is identical to the T-1 frame format.

 The bit size of each PRI frame is 193 bits. PRI frames are sampled at a
rate of 8000 per second.
 An effective transmission rate of 193 bits per frame X 8000 samples
per second = 1.544Mbps
Data Link Layer
 The B channel consists of the Frame Relay and LAPB protocols .The D
channel consists of the Link Access Procedure for the D Channel
(LAPD) protocol.

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LAPD Frame Format consist of two bytes:

 The first six byte contains six bits of Service Access Point
Identifier(SAPI), one bit indicate Command (C), or Reponse( R) and
the last bit to indicate whether the address is complete or continues
in the next byte.
 The second byte contains a field called as Terminal Equipment
Identifier (TE1)
Network Layer
 The B channel uses the X.25 Protocol , the D channel also supports
X.25 protocol , it also works with the Q.391 Protocol.
 One of the fields in the LAPD frame is information . This information
field actually encapsulates the network layer packet.
 The ISDN packet is called a message. A message field in the network
layer is transformed into the information field at the data link layer.

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ISDN Addressing
 ISDN is based on the concept of the telephone network ,its
numbering (ie addressing) scheme is also based on that of the
worldwide telephone network

Broadband ISDN (B-ISDN)

 ISDN war primarily developed for higher bandwidths to the home
and business users.
 BRI and PRI maximum transmission rate of 128 Kbps and 1.544 Mbps
were not good enough for some high end application such TV, High
Definition Television (HDTV).
 The basic BRI and PRI not sufficient to carry parallel transmission.

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 The development of the next generation of ISDN service called

Broadband ISDN (B-ISDN).It provide the data rate of up to 600 Mbps.
 B-ISDN technology is closely related with the Asynchronous Transfer
Mode(ATM) technology .
 The original plain vanilla version of ISDN is now called as Narrowband
ISDN (N-ISDN).

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Asynchronous Transfer Mode (ATM)

 ATM is an extremely ambitious transmission network that is expected to replace
most of the existing networking protocols
 ATM uses small packets called cells for transmission. It is based on the
asynchronous Time Division Multiplexing (TDM) technique and works best when
the underlying transmission medium is optical fiber
 Differentiates asynchronous and synchronous TDM
 Three input lines A, B, and C which are multiplexed to produce output on a single
output line. In both cases lines A and B have three packets to send ,line C has
two packets to send
 In Synchronous TDM the empty slot for line C is kept empty in the output slot ,
the input line does not have any data to send the multiplexer still allocates it a
time slot which is wasted
 In Asynchronous TDM the empty slot for line C in not kept empty in the output
slot, the next available packet from another input line is selected for
transmission. Thus no slot is wasted.
 Synchronous TDM does not provide flexibility in its switching approach to meet
the needs of different application .These different applications can and do
demand different transmission rates
 B-ISDN to support multiple high transmission rates, B-ISDN application demand
constant transmission rates. In case of N-ISDN the transmissions rate is 64Kbps.

Overview of ATM
 ATM is a packet network
 ATM support multiplexing of multiple logical connections over a single physical
channel.
 The information over a logical ATM connection flows in the form of cells

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 A cell is a fixed-size packet, ATM itself is sometimes called relay.ATM does not
provide any flow control or error control at the data link layer
 ATM is a modern WAN switching technique devised by telephone companies to
deliver voice, data and even video
 ATM network uses switches .Multiple hosts connect to each switch. In central
ATM switch to create a star topology

 If a host on the ATM network goes down or the connection between a host and
the switch breaks, only the host is disabled without affecting the other hosts on
the ATM network
 Each ATM switch connects to many other switches or hosts
 ATM network is highly scalable , more and more hosts are added to a network
 ATM is designed to offer extremely high data rates. ATM connection between a
host and switch can offer a data rate of 155 Mbps
 ATM uses optical fiber, rather than copper wire
 ATM is designed to be a full duplex data transmission network it uses two optical
fibers one for each direction of data transmission
 Challenges for ATM
 1. It is devised to carry computer voice ,audio and video data

Packet Size
 The key issue in ATM networks is the packet size

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1. Large Packet Size

 Each packet contains a header that contains information as the source
and the destination addresses ,CRC etc
 If a smaller packet to include the additional overhead of packet header
 Each packet sizes is few kilobytes, in digitizing analog voice signals
samples the incoming voice signal every 125 microseconds
 Each sample consisting of 8 bits requiring a speed of 8000 samples X 8
bits per sample =64Kbps
 Such voice signal are accumulated ,send them together this means that
the transmission would be delayed
2. Variable Length Packets
To avoid the problems associated with large packet sizes , some networks use
variable length packets
This means that voice traffic can be sent in smaller packets whereas other data
can be sent in the form of bigger packets
This approach also has one drawback : complexity

Small packets must wait for large packets in case of variable length packets
 A Multiplexer that accepts packets from many source routes and sends it
to a single outbound route
 A Large packet P arrives before three small packets X,Y and Z arrive at the
multiplexer from the other route packets x,y and z must wait until the
whole of packet P is transmitted
The ATM Solution
 The packet size cannot be too high or variable in ATM networks
 ATM uses a fixed packet size of 53 octates
 Each 53 octate ATM packet is called as cell.

 Each cell contains a header consisting of 5 octates and data consisting of 48

octates
 The US network 64 octates cells, Europeas and Janpanese wanted 32 octate cells
because it was optimal for them

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 In ATM if the input is received in the form of large or variable length packets all
these packets are converted into equal cells each of 53 octates.
 Using Time Division Multiplexing (TDM) these packets would be delivered in turn
 The large packet P is split up into three smaller packets called A,B and C before
transmission

 Different applications can specify their requirements by using a Quality of

Service (QoS) flag
 Example Voice and Video transmission use the service class of Constant Bit
Rate(CBR), Data use the service class of Available Bit Rate(ABR)
Virtual Circuits in ATM
 The advantage of using small fixed size cells can be lost quickly if the cells are
not routed efficiently
 If the cells travels via different path from source to destination and arrive via
different paths, re-sequence them and check for missing or duplicate cells,
the packet overhead increased
 To avoid this ATM networks use the concept of virtual circuits
 This approach all the cells in a single message would travel via a fixed, pre-
defined route
 The cells would arrive at the destination in the same order
 The header contains the route identification of the route chosen. At every
switch, a table is maintained that provides the next hop the cell should be route
to for a given route. This table is generated during the connection establishment
phase at all the switches on the chosen route and deleted on releasing the
connection .This reduce the header size because no longer need to mention the
full source and destination address
 In ATM , once a message is split up into 48 octate cells and the header is added
to each one , each cell travel via the same route from the source and destination
 Each cell header contain a field called Virtual Path Identifier(VPI)
 The VPI identifies a virtual path uniquely. It is 12 bit number that is generated
when the connection is established

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Virtual circuit approach works

 There is another identifier used called Virtual Circuit Identifier(VCI).Whereas a

VPI identifies a path, the VCI identifies the individual circuit or connection on the
path.VCI is required in addition to VPI because thee could be multiple circuits
,connection or channels multiplexed on the same path.VCI consist of 16 bits
 Connection Identifier = VPI+VCI=12 bits + 16 bits = 28 bits
 Each cell would carry this 28 bit connection identifier

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 The virtual circuit can be pre configured in the hardware , it is called Permanent
Virtual Circuit (PVC) or they can be stored in RAM and updated as and when
required in an approach called Switched Virtual Circuit (SVC)
 PVC and SVC both use packet switching but set up a connection –oriented path
through the network before data is sent
 PVC is provisioned which means that someone places the configuration
information on non-volatile storage, it takes a long time to set up, the PVC
survives power failures
 When using PVC the VPI values are set up in the table maintained by ATM
switches
 An SVC is created on demand – a computer requests an SVC ,sends data, and
then terminates the SVC
 The establish SVC, ATM uses higher layer protocol such as B-ISDN.
 These protocols creates an SVC between the two end points on demand , and
AM uses that SVC for transmission during that session
ATM Cells
ATM works with two cell formats
1. User-Network Interface (UNI)
2. Network-Network Interface(NNI)
 The user access devices connect to the ATM switches through a UNI.
 The switches are connected with other switches using NNI
 UNI and NNI uses 53 octate cells
 In case of UNI the first four bits are reserved for GFC , and the remaining 8 bits
are reserved for VPI
 In case of NNI, the first 12 bits collectively denote the VPI
UNI and NNI

UNI Cell Format

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NNI Cell Format

Field Long Form Meaning /Purpose

GFC Generic Flow Control Receive inputs from a single line that
multiplexes voice ,video and data
components GFC able to detect which of
these is being received
VPI Virtual Path Identifier Virtual path used for the given transmission
VCI Virtual Channel Identifier Virtual channel within a virtual path used for
the given transmission
PT Payload Type If the ATM cell contains user or network
information
R Reserved Not used Currently
CLP Cell Loss Priority This field guides the ATM network when
congestion occurs.If the field contains 0 , it
means that this cell has a very high priority , If
the field contains 1 , it means that in the
event of heavy congestion
HEC Header Error Control This 8 bit field contains error control
information
User Data This field contains the actual data to be
transfered

Switching
There are two types of switches :
1. VPI switch – that routes packets based only on the VPI values
2. VPC switch – that routes packet based on VPI as well as VCI values
VP Switch
 ATM VP switch that has six incoming interfaces and six outgoing interfaces. One
incoming interface (number 2) and two outgoing interfaces (number 10 and 12)
 The incoming interface (number 2) has received two packets with VPI values =78
and 121
 To determine that a packet arriving at interface 2 with VPI=78 should be routed
to interface 10 with VPI=37

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ATM Routing Concept using a VP Switch

VPC Switch

ATM Routing Concept Using a VP Switch

 The incoming interface (number 2) has received two packets with (VPI=78
,VCI=10) and (VPI=121 ,VCI=11)
 The ATM VPC switch maintains a table to determine that a packet arriving at
interface 2 with (VPI=78 ,VCI=10) should be routed to interface 10 with (VPI=37
,VCI=91) and so on
 ATM VPC switch effects forwarding of packets from the incoming interface to the
appropriate outgoing interfaces
ATM Layers
The ATM protocol consists of three layers
1. Physical Layer
2. ATM Layer
3. Application Adaptation Layer(AAL)

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Physical Layer
 It deals with issues related to transmission media,bit transmission
,electrical/optical interface and encoding techniques
 This enables transformation of ATM cells into raw bits which can be transmitted
across a transmission medium
 transmission medium is optical fiber although it can also be twisted-pair or
coaxial cable
ATM Layer
 It deals with cell routing , switching ,multiplexing and traffic management
 The core area of the ATM layer is the definition of the ATM cells which is a 53
octates unit
 The ATM cell is defined in the ATM Layer
 The ATM layer accepts 48 byte AAL segments from the AAL layer and adds the 5
byte cell header to transform the segment into a 53 byte ATM cell that can be
delivered to the physical layer

Application Adaptation Layer (AAL)

 AAL facilitates an interaction between the existing networks and ATM
 AT the incoming end , the AAL layer accept data from the higher layer and
divides it into 48 bytes segments and hand over to ATM layer which adds a 5
byte header to form a 53 byte ATM cell
 The incoming frames can be circuit/packet switched can have fixed /variable
data rated
 Outgoing end the AAL transforms fixed size ATM cells into appropriate protocol
frames
Data Types and Categories
1. Constant bit rate (CBR) – Telephone calls
2. Variable Bit Rate(VBR) – VBR data are compressed data ,voice and video

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3. Connection oriented packet data – Computer Generated data carried by virtual

circuit
4. Connection less packet data – data coming from connection less packet
networks such as IP protocol
5. Simple and Efficient Adaptation Layer(SEAL) – generic data type to point –to –
point and never multipoint or internetwork based transmissions
 To handle these five data types the AAL defines five different categories called
AAL1 through AAL5
 Each category contains two sub layers based on functionality
 The two sub layers are called as Convergence Sub-layer (CS) and Segmentation
and Reassembly(SAR)

AAL1:
 AAL1 supports applications that transfer information at a constant bit rate
 Example : data transmission are voice and video
 The CS sub-layer divides the incoming bit stream into 47 byte segments and
passes them to the SAR sub layer
 The SAR sub layer adds one byte of header making this a 48 byte data unit
AAL1 operations

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AAL1 Header Sub-fields

Data type Description
Convergence Sub Layer Identifier (CSI) This bit is used for signaling purpose
Sequence Counter (SC) Used for end-to-end flow control and error
control. It defines the modulo 8 sequence
number of the segment
Cyclic Redundancy Check(CRC) CRC field serves the same error-detection
purpose
Parity(P) The parity field serves the same error-
detection purpose

AAL2
 It supports variable bit rates such as compressed voice, video and data.
 CS sub layer divides the incoming bit stream into 45 byte segments and passes
them to the SAR sub-layer.
 The SAR sub layer adds one byte of header and two byte of trailer , making this a
48 byte data unit
AAL2 operation

AAL2 Header and Trailer Sub-Fields

Header/Trailer field Description
Convergence Sub Layer Identifier (CSI) This bit is used for signaling purpose
Sequence Counter (SC) Used for end-to-end flow control and error
control. It defines the modulo 8 sequence
number of the segment
Information Type (IT) This field identifies if this is the start
,middle or end of data
Length Indicator(LI) This field is useful for the last segment of a
message to indicate how much portion of

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this segment is data and how much is the

padding
Cyclic Redundancy Check(CRC) CRC field serves the same error-detection
purpose

AAL3/4 Operations
 AAL3 and AAL4 supports connection-oriented and connectionless data.
 The CS sub layer accepts a data packet with a maximum length of 65536 bytes
from an upper layer protocol such as Frame Relay or SMDS
 It adds a header and trailer to indicate the start and the end of the message and
only for the last frame, how much is the data and how much is the padding.
 It split the packets into 44 byte segments and hands them over to the SAR layer
 The CS does not append the header and trailer to every 44 segment , it adds
them to the beginning and end of only the original segment
 The SAR sub layer accepts 44 byte segment from the CS sub layer and appends a
two byte header and two –byte trailers to each frames and passes such 48 byte
frames.
AAL3 / 4 Header and Trailer Sub fields:

Header/Trailer field Description

Type (T) It is old category and is not used here
Begin Tag (BT) One byte field indicates the beginning of data for the
receivers synchronization purpose
Buffer Allocation(BA) This two byte filed informs the receiver how big its buffer
should be for receiving data
Alignment (AL) This one byte field is not used for any special purposes
End Tag(ET) This one byte field indicates the ending of data for the
receivers synchronization purposes
Length(L) This two byte field indicates the length of the CS data unit
Segment Type (ST) This field identifies if this is the start, middle or end of data
Convergence Sub Layer This bit is used for signaling purpose
Identifier (CSI)
Sequence Counter (SC) Used for end-to-end flow control and error control. It defines
the modulo 8 sequence number of the segment
Multiplexing Identification This 10 bit field identifies the cells coming from different
(MID) sources ,which are multiplexed on the same virtual circuit
Length Indicator(LI) This field is useful for the last segment of a message to
indicate how much portion of this segment is data and how
much is the padding
Cyclic Redundancy CRC field serves the same error-detection purpose
Check(CRC)

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AAL5
 The AAL3/4 provides services for flow control and error control that are quite
comprehensive. However all applications do not demand these services
 In such situations the AAL5 is useful. It assumes that all the cells belonging to a
single transmission travel sequentially in order and that the upper layers provide
the necessary services for flow control and error control.
 AAL5 does not provide any addressing, ordering or other header information
either with the CS or SAR. It is name Simple and Efficient Adaptation Layer (SEAL)
AAL5 Operation
Trailer field Description
User to User ID (UU) This one –byte field is not used currently
Type (T) This one –byte field is not used currently
Length(L) This two byte field indicates the length of the CS data unit
Cyclic Redundancy CRC field serves the same error-detection purpose
Check(CRC)

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Miscellaneous Topics
1. Service Classes
Four types of Service Classes :
1. Constant Bit Rate (CBR)
2. Variable Bit Rate (VBR)
3. Available Bit Rate (ABR)
4. Unspecified Bit Rate (UBR)
ATM Service Classes

Service Class Description

CBR This is useful for real time audio or video services
VBR Classified into VBR Real Time (VBR-NT) and VBR Non Real Time
(VBR-NRT) this class is used for applications that use compression
techniques
ABR Suitable for applications that are bursty in nature
UBR UBR does not guarantee anything , it is best effort delivery
mechanism

2. Quality of Service (QoS)

The Quality of Service (QoS) defines parameters related to the performance of
an ATM network.
Each service class is associated with one or more attributes. The attributes can
be classified into two categories :
1. User-related attributes
2. Network –related attributes

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User –related QoS Attributes

Attributes Description
Sustained cell Rate (SCR) This is the average cell rate over a period
of time
Peak Cell Rate(PCR) This is the maximum transmission rate at
a point of time
Minimum Cell Rate(MCR) This is the minimum cell rate that the
network guarantees a user
Cell Variation Delay Tolerance (CVDT) This is a unit that measures the changes
in cell transmission times
Network-related QoS Attributes

Attributes Description
Cell Loss Ratio (CLR) This attributes defines the fraction of the
cells lost/delivered too late during
transmission
Cell Transfer Delay (CTD) This is the average time required for a
cell to travel from source to the
destination
Cell Delay Variation(CDV) This is the difference between the
maximum and minimum values of CTD
Cell Error Ratio (CER) This parameters defines the fraction of
cells that contained errors
3. Characteristics of ATM

Characteristics Description
Asynchronous ATM is an asynchronous network.This means that
the source and the destination do not have to be
coordinated before transmission begins.This also
means that each cell in transmission can travel
independently without regard to other cells in the
same transmission
Virtual Circuit Approach ATM uses the virtual circuit approach the
transmission becomes connection-oriented
Order of Cells Because all cells in a single transmission take the
same route, they always arrive in the same order
Equal Delays Even if there are network problems the delays
apply to all the transmission equally
Automatic error detection and ATM does not provide elaborate retransmission
correction mechanisms
a. Transmission media are very reliable
b. ATM uses small 53 octate cells.

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