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Cos 242 Data Comm and Networks

The document discusses data communication and networking. It defines key terms like data, information, and networks. It also describes the basic components of a data communication system including the message, sender, receiver, transmission medium, and protocols.
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0% found this document useful (0 votes)
36 views71 pages

Cos 242 Data Comm and Networks

The document discusses data communication and networking. It defines key terms like data, information, and networks. It also describes the basic components of a data communication system including the message, sender, receiver, transmission medium, and protocols.
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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1

COS 242 - DATA COMMUICATIONS AND NETWORKING (2 Units)

COURSE CONTENTS:

Introduction, data and information, data processing system, data communication systems, signals:
Introduction to analog and digital, A/D conversions, Time and frequency domain concepts, Fourier
transform, Fourier series, measure of communication, channel characteristics, Nyquest and Shannon
information capacity, transmission media, noise and distortions, modulation and demodulation,
Multiplexing and demultiplexing, synchronous and asynchronous data transmission, Error detection
and control techniques.

INTRODUCTION

OVERVIEW OF DATA AND INFORMATION

In everyday usage, the two terms: information and data are always used interchangeably. There is,
however, a difference between the two in computer studies. The digits “19896206” constitute data, but
they convey no information. They could be interpreted as a student registration number or as a library
catalogue number.
Data, therefore is the name given to basic facts such as names and numbers. Examples are
times, dates, weights, prices, costs, numbers of items sold, employee names, product names, addresses,
test scores etc. Some sources of data may include:
Questionnaire survey, Face – to face interview, Published document review, Telephone interview,
Experimental recordings, Group techniques such as workshops, focus/ discussion groups, Public
opinion pools, Library or database.
On the other hand information is data which has been converted into a more useful or
intelligible form. Examples are printed documents, pay slips, text, reports, etc. So a set of words would
be data but text would be information e.g. Obinna, Okafor are data. “Obinna Okafor scored the highest
examination mark” is information.
Different types of questions/problems require different sources of information. Some sources of
information include: Observations, people, speeches, documents such as, pictures, organizations,
websites, etc. Let us take a look at the differences between data and information as shown in the Table
below.
Differences between data and information
Data Information
Data refers to names given to basic facts Information is data, which has been
such as names, numbers. Examples are converted into a more meaningful form.
dates, addresses, times, prices, costs, etc
Data is facts and statistics that are Information is processed data that make
collected in raw form. meaning to someone.
Data is normally arranged into numbers, Information is normally represented in
blocks and charts. prose form.
Data is the lowest level of knowledge. Information is the second level of
knowledge.
Data by itself is not significant without Information is by itself significant.
processing.
Observation and recording is done to Analysis is done to obtain information.
obtain data.

Note that we use the term “communication” for transfer of information and the term “transmission”
for the transfer of data.
2
DATA PROCESSING SYSTEM
Data processing (DP) is the term given to the process of collecting data together and converting them
into information. The objective of data processing, therefore, is to organize data into meaningful
information.
A data processing system is made up of people, equipment and procedures that process data. There are
basically five steps by which data is expressed, processed and returned to managers or other
individuals as update and useful information. These are:
Step1: Preparation of Source Documents;
Step2: Input of Data;
Step3: Manipulation of Data;
Step4: Output of Information
Step5: Storage of Data.

Data Transmission and Processing Methods


In the last twenty years, there have been many developments that have increased the processing speed
of computer system. The speeds of CPU and I/O devices have increased tremendously because of
latest technological developments; direct access devices have also been developed and improved. In a
data processing cycle, source data must be collected before process, and processed data must be
distributed before it can be used. Thus, even if processing time is reduced, there can still be significant
delays in the system due to distribution. Data communication systems are designed to reduce the
delays in collection and distribution of data.

What is Data Communication?


Data communications means the exchange of data between two devices via some form of
transmission medium such as a wire cable.
For data communications to occur, the communicating devices must be part of a communication
system made up of a combination of hardware (physical equipment) and software (programs).
Characteristics of Data Communications:
The effectiveness of a data communications system depends on four fundamental characteristics:
delivery, accuracy, timeliness, and jitter.
1. Delivery: The system must deliver data to the correct destination. Data must be received by the
intended device or user and only by that device or user.
2. Accuracy: The system must deliver the data accurately. Data that have been altered in transmission
and left uncorrected are unusable.
3. Timeliness: The system must deliver data in a timely manner. Data delivered late are useless. In
the case of video and audio, timely delivery means delivering data as they are produced, in the same
order that they are produced, and without significant delay. This kind of delivery is called real-time
transmission.
4. Jitter: Jitter refers to the variation in the packet arrival time. It is the uneven delay in the delivery
of audio or video packets.
Components of Data Communication
The different components of Data communication are shown in the following figure.
3

1. Message:
The message is the information (data) to be communicated. Popular forms of information include text,
numbers, pictures, audio, and video.
2. Sender:
The sender is the device that sends the data message. It can be a computer, workstation, telephone
handset, video camera, and so on.
3. Receiver:
The receiver is the device that receives the message. It can be a computer, workstation, telephone
handset, television, and so on.
4. Transmission medium:
The transmission medium is the physical path by which a message travels from sender to receiver.
Some examples of transmission media include twisted-pair wire, coaxial cable, fiber-optic cable, and
radio waves.
5. Protocol:
A protocol is a set of rules that govern data communications. It represents an agreement between the
communicating devices. Without a protocol, two devices may be connected but not communicating,
just as a person speaking French cannot be understood by a person who speaks only Japanese.
What is Network?
A network is a set of devices (often referred to as nodes) connected by communication links. A node
can be a computer, printer, or any other device capable of sending and/or receiving data generated by
other nodes on the network.
Characteristics of Network?
A network must be able to meet a certain number of criteria. The most important of these are
performance, reliability, and security.
Performance:
Performance can be measured in many ways, including transit time and response time.
Transit time is the amount of time required for a message to travel from one device to another.
Response time is the elapsed time between an inquiry and a response.

The performance of a network depends on a number of factors, including the number of users, the
type of transmission medium, the capabilities of the connected hardware, and the efficiency of the
software.
4
Reliability
In addition to accuracy of delivery, network reliability is measured by the frequency of failure, the
time it takes a link to recover from a failure, and the network's robustness in a catastrophe.
Security
Network security issues include protecting data from unauthorized access, protecting data from
damage and development, and implementing policies and procedures for recovery from breaches and
data losses.
TYPES OF DATA COMMUNICATION SYSTEMS
Although there are many forms of data-communication systems, they can be broken into four
categories:
1) Offline;
2) Online batch-processing;
3) Online real-time and
4) Time-sharing system.
Let us examine each in turn:
1) OFFLINE SYSTEMS
Offline means that the transmission of data is not directly to or from the computer. Essentially,
the system consists of terminal and communication lines: one terminal at the sending end and another
at the receiving end. The terminals could be card reader at the sending end and card punch, tape drives
or printers at the receiving end.
Offline system is simply a means of eliminating the delays in sending data between two geographical
points.
For example, if a report is to be printed for branch office management, data can be read by tape
drive in the home office and printed on a printed in the branch office management.

2) ONLINE BATCH-PROCESSING SYSTEMS


Online means that data is transmitted directly to the computer. Batch-processing is an
important method of manual and computerized data processing characterized by the following:
a) The accumulation of transactions into batches;
b) There is some degree of delay;
c) The transactions are sorted and then processed;
d) The results of processing a particular data item are not known until the whole
batch as been processed.
Examples of this system are: a punch tape is prepared from customer orders and used as input to the
computer through phone lines. The computer performs editing run while writing the transactions on
the magnetic tape. After the transactions from all the branches are edited and written, the tapes can be
merged and processed.
Advantages of Batch Processing
1) There is no need for the user to be present since there is no interaction between him and
the computer while the job is being run.
2) Preparing the work and operating the computer are done by trained people and not by
users.
3) Less expensive than time sharing on large systems.

Disadvantages of Batch Processing


1) There is always a delay before work is processed and returned.
2) The user cannot take action if anything is wrong; he has to re-input the job and place it
on a different batch.
3) Batching usually involves an expensive computer and a large number of staff.
5
3) ONLINE REAL-TIME SYSTEMS
A real time system is a computer system which is capable of processing data without
significant delay, so that the results are available to influence the activity currently taking place.
In practical terms we could say that a real-time system is a batch processing system with a
single transaction which is processed immediately on demand.
Although real-time system can be used for basic business applications such as order writing,
inventory and payroll, they are more likely to be used for specialized applications as in:
a) Transport- Airline booking systems;
b) Data retrieval systems used by the police, fire services and hospitals;
c) Systems used in the control of processes such as automated process and production
control in factories;
In the Airline booking system for instance, handling customer’s inquiries can be a major
problem for the booking department. Here, records of seat availability on all its planes will be kept by
an airline on a central computer. The computer is linked through terminals to a world-wide system of
agents. Each agent can gain access to the flight records and within seconds make a reservation in
respect of a particular flight. This reservation is recorded immediately so the next enquiry for that
flight finds that particular seat or seats reserved. Notice the computer records reflect on accurate
picture of the airline’s seating load at all times because there is no time lag worth mentioning. The
computer would then output information for the production of the customers’ ticket and flight
instructions, on confirmation of the booking by the customer.

Advantages of Real-Time Systems


1.) Fast response
Disadvantages
1.) The cost of a real-time system is higher than that of batch system.
2.) A computer being used for a real-time application often cannot be used for anything
else.
3) Real-time system involves hardware programming and systems design costs that have
no equivalent in batch systems.
4) TIME SHARING SYSTEMS
A time sharing system is one in which the processor time of a central computer is divided into
time slices which are shared between various interactive users appropriately. In such a system, a clock
is used to divide up processor time into “time slices” and these time-slices are shared between various
users in any appropriate way.
At each pulse of the clock the user program currently being executed has its execution
suspended to make way for next user program. The operating system continually checks the status of
each user, in strict sequence, to see whether the user requires processor time.
When it is confirmed that a user requires processor time that user’s program is allowed to
continue execution for the duration of the time slice.
At any given time the user may have access to the computer and because the computer
switches from one user terminal to the other at a fantastic speed, each person using it may not realize
that another person is using it at the same time. The number of users of any one system is limited, to
ensure a high speed of response. Each has a typewriter terminal or VDU installed and linked to the
computer system by telephone line. Each user has his own password and this has to be transmitted
when access is required.
The usual dialog involves required for a time sharing system takes the following steps:
i) User dials the computer system telephone number.
ii) System asks for the password (or code number)
iii) User keys in his code number and is connected (“signed on”) to the computer.
iv) Now by means of the keyboard he will get the system to do whatever he requires.
v) To do this the computer allocates the incoming user an area on disk, does the necessary
processing in the CPU and sends back the results through telephone line to the users
terminal.
6
vi) At the end of his “session” the user “signs off” and replaces the telephone receiver.

WAVES/SIGNALS

A wave is a disturbance, which travels through a medium and transfers energy from one point to
another without causing any permanent displacement of the medium itself.
For example, waves could be observed on the surface of water if a stone is dropped into a pond
of water. The water transfers energy (information) from one point to another.

TYPES OF WAVES
There are two major types of waves namely:
1) Mechanical and
2) Electromagnetic waves.
Mechanical waves usually require a material medium for their propagation e.g. water waves, sound
waves etc.
On the other hand, Electromagnetic waves do not require a material medium for propagation.
Examples are radio waves, light rays etc. These are also called microwaves.
Representation of waves
Waves whether mechanical or electromagnetic may generally be described as ripples, which alternate
between positive and negative values in a sinusoidal manner as shown in figure 1 (below).
V
λ(m)
crest

T(s)

a t

0 V(t) =
aSin 2πft trough
λ(m)
Or T(s)

Definitions:
1) Amplitude (a): As the wave propagates, the particles of the medium vibrate about a mean
position. The maximum displacement of particles from their mean position is called the
amplitude, a, of the wave. It is measured in metres. Peak amplitude is the maximum value or
strength of the signal over time.
2) Period (T): Is the time required for a particle to complete one cycle of the wave. Period (T) is
1
also the amount of time it takes for one repetition; T  . It is measured in seconds.
f
3) Frequency (ƒ): The number of cycles which the wave completes in one second is called its
frequency. Frequency is measured in hertz (Hz). Frequency- Is the rate (in cycles per sec) at
which the signal repeats. (Note that 1 kilohertz (KHz) = 103 Hz and 1 MHz = 106 Hz).
4) Wavelength (λ): The distance along the x -axis between successive crests or successive
troughs is called the wavelength. It is measured in metres.
5) Wave Speed (v): Is the distance which the wave travels in one second.
x
V  m / s , where
t
x- Is the distance traveled in metres;
t- Is the time taken to cover the distance x;
7
6) Phase ( ( ) : Is a measure of the relative position in time within a single period of a signal =
t
.
T
7) Spectrum: Is a range of frequencies that a signal contains S (t )  ASin(2ft   ) .

RELATIONSHIP BETWEEN T, ƒ, λ AND V


By definitions, T seconds is needed to complete in 1 cycle;
1
1sec ” ” ” ” cycle
T
1
Thus: f 
T

By definitions:


V  = ƒλ
T

Thus: V = ƒ λ

Solved example:
A Radio station broadcasts at a frequency of 200 KHz. If the speed of the wave is 3x108 m/s,
calculate:
a) the period;
b) the wavelength of the wave;

Solution
5
a) Frequency ƒ = 200 KHz = 2x10 Hz
T= 1/f = 1/ (2*105) = 0.5 * 10-5 sec

3 x108
b) Wavelength λ = v/f = =
2 x105
λ = 1.5*103m

Fundamentals of Data and Signals


The major function of the physical layer is to move data in the form of electromagnetic signals across
a transmission medium. Whether the data may be numerical statistics from another computer, sending
animated pictures from a design workstation, or causing a bell to ring at a distant control center, you
are working with the transmission of data across network connections.

Analog and Digital Data

Data can be analog or digital. The term analog data refers to information that is continuous and take
continuous values. Digital data refers to information that has discrete states and take discrete values.
For example, an analog clock that has hour, minute, and second hands gives information in a
continuous form, the movements of the hands are continuous. On the other hand, a digital clock that
reports the hours and the minutes will change suddenly from 8:05 to 8:06.
8
Analog and Digital Signals:

An analog signal has infinitely many levels of intensity over a period of time. As the wave moves
from value A to value B, it passes through and includes an infinite number of values along its path. A
digital signal, on the other hand, can have only a limited number of defined values. Although each
value can be any number, it is often as simple as 1 and 0.
The following program illustrates an analog signal and a digital signal. The curve representing the
analog signal passes through an infinite number of points. The vertical lines of the digital signal,
however, demonstrate the sudden jump that the signal makes from value to value.

Periodic and Non-periodic Signals:

Both analog and digital signals can be periodic or non-periodic

Periodic Signal: A periodic signal completes a pattern within a measurable time frame, called a
period, and repeats that pattern over subsequent identical periods. The completion of one full pattern is
called a cycle.
Non-periodic signal: A non-periodic signal changes without exhibiting a pattern or cycle that repeats
over time.
In data communications we commonly use periodic analog signals (because they need less bandwidth)
and non-periodic digital signals (because they can represent variation in data).
Periodic Analog Signal:
Periodic analog signals can be classified as simple or composite. A simple periodic analog signal, a
sine wave, cannot be decomposed into simpler signals. A composite periodic analog signal is
composed of multiple sine waves.
The sine wave is the most fundamental form of a periodic analog signal. When we visualize it as a
simple oscillating curve, its change over the course of a cycle is smooth and consistent, a continuous,
rolling flow. The following figure shows a sine wave. Each cycle consists of a single arc above the
time axis followed by a single arc below it.
9

A sine wave can be represented by three parameters: the peak amplitude, the frequency, and the phase.
Peak Amplitude:
The peak amplitude of a signal is the absolute value of its highest intensity, proportional to the energy
it carries. For electric signals, peak amplitude is normally measured in volts. The following Figure
shows two signals and their peak amplitudes.

Period and Frequency: Period refers to the amount of time, in seconds, a signal needs to complete 1
cycle. Frequency refers to the number of periods in I s. Note that period and frequency are just one
characteristic defined in two ways. Period is the inverse of frequency, and frequency is the inverse of
period, as the following formulas show.
f= 1/T and t= 1/F
Phase: The term phase describes the position of the waveform relative to time O. If we think of the
wave as something that can be shifted backward or forward along the time axis, phase describes the
amount of that shift. It indicates the status of the first cycle.
10

Wavelength:
Wavelength is another characteristic of a signal traveling through a transmission medium. Wavelength
binds the period or the frequency of a simple sine wave to the propagation speed of the medium.
While the frequency of a signal is independent of the medium, the wavelength depends on both the
frequency and the medium. Wavelength is a property of any type of signal. In data communications,
we often use wavelength to describe the transmission of light in an optical fiber. The wavelength is the
distance a simple signal can travel in one period.
CHARACTERISTICS OF SIGNALS/WAVES
The characteristics of a signal may be one of a broad range of shapes, amplitudes, time durations and
perhaps other physical properties, such as statistical and probabilistic.
In general, we shall examine the following characteristics, namely; periodical, symmetrical and
continuity.
1) Periodical
If a signal is periodic, then it is described by the equation:
S (t) = S (t + KT), K= 0, 1, 2, 3 … (1)
Where,    t  +  and T – is the period of the signal.
For instance, the sine wave sin t is periodic with period T = 2π.

X(t)
Sin t
1

π 2π
0  t
2

T = 2π

The square wave (see figure below) is another example of a periodic signal.
11

0 T 2T 3T 4T

-1`

x x
X(t) = XSinwt  3 Sin3wt  5 sin 5wt  ...

Other signals such as the rectangular pulse, the saw-tooth wave may be
considered “periodic” with an infinite period.

0
t

T
The saw tooth wave:
x x
X (t) = XSinwt  Sin 2wt  sin 3wt
2 3

2) Symmetrical (or symmetrical properties)


We shall use the words even and odd to verify whether a signal is symmetrical or not.
Any even function usually obeys the relation:

S (-t) = S (t) . . . (2a)


For example:
cos ( )  cos .
Thus, Cos  is an even function.
Any odd function usually obeys the relation:
S (t) = - S (t) . . . 2(b)
For example,
Sin(   )  Sin , thus a sin  function is an odd function.

Any signal or wave S(t) can be resolved into an even component Se(t) and an odd component So (t);
such that

S (t) = Se(t) + So(t) . . . . .2(c)


Or
S(-t) = Se(-t) + So(-t) = Se (t) – So (t) . . . .(2d)
Consequently;
Se(t) = 1/2 [S(t) + S(-t)]
(2e)
So(t) = 1/2 [S(t) + S(-t)]

Equation 2(e) is the decomposition into even and odd components.

Example: Decompose the following signals into even and odd components?
12
DIAGRAM
Se(t)
1/2 EVEN part
1 S(t)

-1 0 -1 0 1

Note that the even function is symmetrical along the y-axis whereas the odd function is symmetrical
along the x-axis.
A signal S(t) is continuous of lim S (t )  S (a) for all a.
t a

3) Continuity Property

Consider the signal shown below:


f(t)

Signal with
A discontinuity

-T
T+
0 t
T

Observed that at t = T, the signal is discontinuous. The height of the discontinuity is


f Tt   f (T )  A ... (3)

Where
ƒ(T+) = Lim f (T   )
 0 ... (4)

and and  is a real positive quantity.


ƒ(T-) = Lim f (T   )
 0

In particular, we are concerned with discontinuous in the neighborhood of t = 0 (i.e. in the


neighborhood of the origin).
For equation (4) putting T = 0, the points ƒ(0+) and ƒ(0-) are:
ƒ(0+) = lim : f ( )
 0

ƒ(0-) lim : f (  )
 0

A signal f (t) is continuous if lim f (t ) = f (a) for all a.


t a
13
FOURIER SERIES AND ANALYSIS

One of the famous characteristics of signals is periodicity. As shown above, if a signal is periodical,
then it is described by the equation:
ƒ (t) = ƒ (t ± KT), where K = 0, 1, 2 . . . integers and T- period.

This equation is called general periodic function.

Following Euler’s observations in the 18th century, that vibrating strings produce sinusoidal motion, on
21 December, 1807, Jean Batiste Joseph Fourier in a historic session of the French Academy in Paris,
announced a thesis that opened a remarkable chapter in the history of mathematical analysis and its
engineering applications.
Joseph Fourier proffered that any periodic signal or function can be represented in terms of an
infinite sum of sine and cosine functions or trigonometric series that are themselves periodical. Thus
we obtained:

A0
ƒ (t) =  A1 cos wt  A2 cos 2wt  ...B1 sin wt  B2 sin 2wt  ... (1)
2
2
Where w  is called the radian frequency, which is w  2f and is measured in
T
radian/sec.
nw  with n = 2, 3, … is the nth harmonic.

The trigonometric series (equation 1) above is generally referred to as the FOURIER SERIES.
The first term in equ (1) above is a constant component or zero harmonic of a wave.
The terms with A1 and B1 constitute the first harmonic with w1 radian frequency, while the terms with
A2 and B2 constitute the second harmonics of the wave with w2, etc.
A0, A1 . . .An and B1, B2 . . . Bn are constants.
In a more compact form, the Fourier series equation (1) can be expressed as follows:

A0
 n 1 ( An cos nwt  Bn sin nwt)

ƒ(t) = ... (2)
2

A0  B
=   An (Cosnwt  n sin nwt)
2 n1 An

A0 
=   An(Cosnwt  tann Sinnwt)
2 n1

A0  An
=  Cos(nwt  n )
2 n1 Cosn

A0 
=   C n Cos(nwt  n ) , Where
2 n1

Bn
n  tan1 ( )
An
... (3)
Cn  A B
2 2
n n
14

For a function to be Fourier series transformable, it must satisfy the DIRICHLET conditions, which
ensure mathematical sufficiency, but not necessity. The Dirichlet conditions require that within a
period:
i) Only a finite number of maximums and minimums can be present.
ii) The number of discontinuities must be finite.
iii) The discontinuities must be bounded. That is, the function must be absolutely
integrable, which requires that
T

0
/ f (t ) / dt  
The Fourier Series (equation 1) can be described completely in terms of the coefficients of its
harmonic terms of A0, A1, A2. . ., B1, B . . . etc.
These coefficients constitute a frequency domain description of the signal (or wave).
These Fourier coefficients can be determined from the following equations:

2 T
T 0
A0 = f (t ) dt . . . 4.1)

2 T
T 0
An = f (t ) Cos nwt dt ... (4.2)

2 T
T 0
Bn = f (t ) sin nwt dt ... (4.3)

If we integrate between the limit (0;π) i.e.


T =π, we get;
2 
A0 =
 
0
f (t ) dt ... (5.1)

2 
An =
 
0
f (t ) Cos nwt dt ... (5.2)
2 


Bn = f (t ) sin nwt dt ... (5.3)
0

FOURIER ANALYSIS
Fourier analysis is concerned with determining the Fourier coefficients for a given signal and the
corresponding Fourier series.
Solved Examples
1) Determine the Fourier coefficient and the Fourier series for the sine function shown below:
f (t)
A

-T T π 2π 3π t
2 2

Solution: Here, the period is T = π, Hence; from

w = 2 π = 2π = 2
T π

w = 2
15
Hence, the signal is given as:

f (t )  A / sin t / .

Using equation 4.3, we have;

2  2A 
Bn =
 0
f (t ) sin 2nt dt =
 
0
sin t sin 2nt dt

2A 
Bn =
 
0
sin0 sin0 dt = 0

Since Sin 0 = 0, cos 0 = 1, sin 180 = 0,

Using equation 4.1, we have;


2A 2A 
A0 =
  0
(-1) [cost ]0
sin t dt =
2A 2A
= (1)[cos   cos 0]  (1)[1  1]
 
2A 4A
= .(1).( 2) =
 

Using equation 4.2, we have;


2  2A 
  0
An = f (t ) cos 2 dt = sin t cos 2 nt dt=
 0

1 4A
=( ).
1  4n 2

Thus, substituting the values of the coefficients obtained so far in equation (2), the Fourier series of
the above sine wave is as follows:
A0 
f(t) =  (An cos nwt + Bn sin nwt)
2 n 1
4A
A  cos 2 nt
=    (1  4n 2 )
2 n 1

A 4A  1

f(t) = 
2  n1 1  4n 2
cos 2 nt

Example 2: Find the Fourier series for the function shown below:

f(θ)

0
-2π -π 0 π 2π 3π 4π

A
16
Solution:

The function is symmetrical along the X- axis, therefore it is an odd function.

So, Ak = 0

The Fourier series takes the form given by:



f(θ) = B
k 1
k sin kθ, where the coefficient

2 


Bk = f ( ) sin Kθ dθ Since, θ = wt =2π
0

If t = π, therefore
2t 2
Using the interval of [0, π]; θ= 
T 2
f(θ) = A; where 0 ≤ θ π see (figure)
[0;π]
Thus, by substitution,

2A 
Bk 
 0
sin kd
 2A   2A
= ,[ CosK  ] = (CosK  1)
K 0 K
Bk = 0 for k even numbers and
4A
Bk = , for k- odd numbers.
K
Since f (θ) is an odd function, we take the k- for odd numbers and ignore k- for even numbers.

Therefore,

4A SinK
f(θ) = 
 k 1 K
,
4A 1 1
f(θ) = ( Sin  sin 3  sin 5  ...)
 3 5

Putting k =1,3,5,. . . etc, odd values.


4A 1
=
  4n  1 Sin (2n+1)θ, which is the Fourier series.
n 0

SOME APPLICATIONS OF COMPLEX FOURIER SERIES

It has been shown that

x = a + jb - Is a complex form of a signal.

This signal can be represented diagrammatically as follows:


17

+j +

X Sin α
α a
+1
X Cos α

Since e = cosα + j sinα . .. . . . .(1)

X ejα = X Cos α + j X Sin α . . . . . . (2)

α β

ejα = cos   sin  )


2 2
where = 1 is a unit vector.
Multiplying both sides of (1) by Im, we get

Im e jα =Im Cosα+ j Im Sinα (3)

If we change,   wt  

Where φ- is the phase-shift of the signal, we get

( wt  )
i  Ime j  ImCos(wt   )  j Imsin(wt   ) . . . . . . . (4)

L.H.S = Im ej(wt +φ) = Im ejφ.ejwt is called the PEAK value of the current

i in complex form.. equation (5)

The value Im = Im ejφ . . . . . . . . (6)

is called complex amplitude of current (i)

SOLVED EXAMPLES:

(1) If i = 2 cos (100t - 350). Find Im?


Solution
i = 2 cos (100t – 350) is of the form:
i = Im Cos (wt + φ), where Im = 2 and φ = - 350
Therefore, putting these values in equation (6), we have;
Im = 2 e- j35°
(2) If the complex voltage is Um = 100 ej60, Find the peak or original voltage?
Solution:

e j ( wt  )  100 e j ( wt 60
0

 m
)
18
 100cos(wt  60 )  j100sin(wt  60 )
0 0

(3) If the complex amplitude of a current is given by


Im = 3 + j4 [A]. What is the peak value of the current (i)?
Solution:

Im = √(a2 + b2) = √(9 + 16) = 5;

b 4
tan   
a 3

  530

53) 0
I m  5e jwt  5e ( jwt

I  5 cos(wt  530 )

(4) Given that i = 2 sin (100t - 300). Find Im and rewrite in the form Im = a + jb.

Solution:

I = 2 cos (100t -1200) (since sin 300 = - cos 1200)

 j120 0
I m  2e  2 cos(1200 )  j sin(120) 

3
= 2(1 2)  j 2.
2

Im = -(1 + j 3)

(5) Giving the complex amplitude Um = - 3 + j4. Rewrite in the form Um = Um e jφ

 m  (3) 2  4 2  5

b 4
tgφ = 
a 3

  1300

  m  5e j130
0
19
MODEL OF A DIGITAL COMMUNICATIONS SYSTEM

Communication at its simplest level, involves the symbolic representation of thoughts, ideas
quantities, and events we wish to record for later retrieval or transmit for reception at a distant point.
Operationally, this involves the transformation of one set of quantities (thoughts, ideas, etc.)
into others (symbols) that are somehow more suited for transmission or recording over a degrading
medium and the recovery of estimates of the original quantities at the receiving point.
The goal of communication therefore, is to achieve the maximum information throughput
across channel with fixed capacity.

Figure below shows the elements of a generic digital communication system.


NOISE

Informa Source Chan Channe Source Destina


Chan
tion coder nel l decode tion
nel
source coder decode r
rr
Digital
Demodulator
modulator

Fig ( ) The components of a generic digital communication system.

The information originates as a signal from a source, either as continuous or discrete. The process of
encoding this information for transmission onto, and later retrieval from, a channel involves two
conceptually distinct processes.
First, the information stream from the source must be transformed into a set of symbols- this is
called source coding.
Source coding maps the source information into a set of symbols from a finite alphabet.
Then this information must be impressed on the physical channel properties. The overall
requirement for source coding is that the process must be reversible; that is; the original information
must be uniquely recoverable from its coded transcription
Encoding an information source into as few symbols as possible results in a more efficient and
economical utilization of finite channel resource such as time, bandwidth, and energy.
The sequence of binary digits from the source encoder is to be transmitted through a channel
to the intended receiver. For example, the real channel may be either a pair of wires, a coaxial cable,
and optical fiber channel, a radio channels, a satellite channel, or some combination of these media.
Such channels are basically waveform channels and, hence, they cannot be used to transmit directly
the sequence of binary digits from the source. What is required is a device that converts the digital
information sequence into waveforms that are compatible with the characteristics of the channel. Such
a device is called a digital modulator or channel encoder.
In general, no real channel is ideal. There are noise disturbances and other interference that
corrupt the signal transmitted through the channel.
In order to overcome such noise and interference and thus, increase the reliability of the data
transmitted through the channel, it is often necessary to introduce in controlled manner some
redundancy in the binary sequence from the source.
1) The redundancy introduced at the transmitter aids the receiver in decoding the desired
information bearing sequence. For example, a form of encoding binary information sequence is simply
to repeat each binary digit m times, where m is a positive integer.
2) Another method involves taking k information bits at a time and mapping each k-bit sequence
into a unique n-bit sequence, called a code word. The amount of redundancy introduced by encoding
20
n
the data in this manner is measured by the . In this case, the channel bandwidth must also be
k
increased by this ratio to accommodate the added redundancy in the stream.
1 K
The reciprocal of this ratio, namely  is called the rate of the code or the code rate.
n n
K

A digital signal is a sequence of discrete, discontinuous voltage pulses. Each pulse is a signal element.
Binary data are transmitted by encoding each data bit into signal elements.

First, the receiver must know the timing of each bit, i.e. when a bit begins and ends.
Second, the receiver must determine whether the signal level for each bit position is high (1) or low
(0). This is done by sampling each bit position in the middle of the interval and comparing the value to
a threshold.
R R
D 
K Log2 M

Where;
D = modulation rate or baud
R = data rat, bps
M = no of different signal element 2K
K = no. of bits per signal element.
To elaborate on the function performed by the modulator: suppose the information is to be
transmitted 1 bit at a time at some uniform rate R bits/s.

The modulator may transmit k information bits at a time by using M = 2K


distinct waveforms.

S1(t), i = 1, 2, . . . M, that is, one waveform for each of the 2K possible k-bit sequences. This is called
M- ary modulation. The modulation rate is called baud. Baud refers to the rate at which the signal
level is changed.
We note that a new k-bit sequence enters the modulator every k/R seconds. Hence, the amount
of time available to transmit one of the M waveforms corresponding to k-bit sequence is k times the
time period in a system which uses binary modulation.
At the receiving end, the digital demodulator processes the channel- corrupted transmitted
waveform and reduces each waveform to a single number that represents an estimate of the
transmitted data symbol (binary or M-ary).

TYPES OF COMMUNICATION SYSTEMS


We can identify three types of communications systems. These are:
1) Simplex
2) Half Duplex
3) Full Duplex.
Simplex: This is the simplest type of communication system. It allows transmission of data in one
direction only.
Half duplex: Transmission can be in both directions, but not at the same time. This means that data is
transmitted only in one direction at a time.
Full duplex: In this type of transmission, data can be transmitted simultaneous in both directions.
21
These are illustrated in Fig ( ) below
Transmitter Receiver T R
(T) (R)

SIMPLEX OR

T/R T/R R T
HALF
DUPLEX
FULL
DUPLEX Where T/R – stands for transmitter/Receiver

A half duplex system has a special electronic device that switches the direction of flow of data,
whereas, in a full duplex there is a device that controls the data flow.

MEASURING INFORMATION
Two of the central tasks of information theory are:
i) The systematic representation of information with a suitable set of symbols and
ii) The reversible conversion from one specific representation to another.

The term information may be defined as a measure of the number of equiprobable choices between
several possible alternatives. Thus, information is measured by the logarithm of the number of such
alternatives.
Information implies the ability to resolve uncertainty, or a choice between several possible
alternatives. The simplest uncertainty is that which is completely resolved by an answer to a YES or
NO question. This corresponds to one bit of information when the anticipated answer of either YES or
NO is given.
A system’s capacity for storing information is fully described by a count of its distinguishable states.
Each state of a physical system is a different configuration of the system.
Some of the properties of an information capacity measure are:
1) A measure of information capacity should increase monotonically with the number of
system states;
2) Information should be additive: the aggregate information capacity of two separate
systems should be the sum of each system’s capacity;
3) The amount of information associated with a system having only one state should be
zero (i.e., Log21 = 0).
If Ω is the total number of distinguishable states in a system, then the system’s information capacity or
amount of information is given by:
C = log2 Ω . . . . . . . . . . . . . . . . .. . . . . . . . . . . . . . . . . . .. . . . . . . . . . . . (1)

Where the base 2, denotes the two states of a binary information.


Equation (1) is called Hartley’s measure of information capacity.
The use of writing to represent information is an excellent example of a sequence or string of
symbols, namely letters, numbers, and other typographical symbols, to represent information.
For example: The information capacity of a 450-page book, assuming 500 words per page with
each word containing five symbols chosen at random from a 37-ary alphabet (i.e. 26 letters, 10 digits,
and a blank space) is given by:
22
6
C = 450 x 500 x 5 x log237 = 5.9x10 bits.

Here, log237 = N
2n = 37
N ln2 = ln 37

ln 37 1.5682
N   5.227
ln 2 0.30103

Thus, C = 450 x 500 x 5 x 5.227 bits.


This is the capacity needed to store the information contained in a representative book of that size.

LOGARITHMIC MEASURE OF INFORMATION


Let X and Y be two discrete random variables with possible outcomes Xi, (i = 1, 2, . . . n) and Yi, (i =
1, 2, . . .m), respectively.
Suppose we observe some outcome Y = yj and we wish to determine quantitatively, the amount of
information that the occurrence of the event Y = yj provides about the event X = xi. The problem is to
select an appropriate measure for information.
Notice that when X and Y are statistically independent, the occurrence of Y =yj provides no
information about the occurrence of the event X = xi.
On the other hand, when X and Y are fully dependent such that the occurrence of Y = yj
determines the occurrence of X = xi, the information content is simply that provided by the event X
=xi.
A suitable measure that satisfies these conditions is the logarithm of the ratio of the conditional
probability:
P (X = xi/ Y = yj ) ≡ P ( xi/yj) divided by the probability P( X  xi )  P( x1 ).
That is, the information content provided by the occurrence of the event Y = yj about the event X = xi,
is defined as:
. .. . . . . . . . . . . . . . . . . . . . . . . . . . . . .(2)
P( xi / yj)
I ( xi : y j )  log
P ( xi )

I(xi: yj) is called the mutual information between xi and yj.


The units of I(xi: yj) are determined by the base of the logarithm, which is usually selected as either 2
or e.
When the base is 2, the units of I(x: yj) are “bits”, and when the base is e, the units of I(xi; yj) are
called “nats” (natural units).
x
When the random variables X and Y are statistically independent, P( i )  P( xi ) and hence of I(xi;
yj
yj)=0.
On the other hand, when the occurrence of the event Y = yj unique determines the occurrence of the
event X = xi, the conditional probability in the numerator of equation (2) is unity
1
and I ( xi , yi )  log   log P( xi ) . . . . . . .. . . . . . . . . . . . . . .(3)
P ( xi )
Because (3) is just the information of the event X = xi, it is called the “self-information” of the event X
1
= xi and is denoted as I ( xi )  log   log P( xi ) . . . . (4)
P( x)
Note that a high-probability event conveys less information than a low-probability event
If there is only a single event x with probability P(x) = 1, then
I (x) = log 1/ P (x) = - log 1 = 0
23

The mutual information I (xi: yj) between two discrete random variables x and y has the following
properties:
i) The mutual information between x and y is symmetric; that is I(xi; yj)= I(xi; yj)
ii) The mutual information between x and y is always non-negative; that is, I ( x : y)  0 .
This property states that we cannot lose information on the average by observing the
system output y.
iii) The mutual information between x and y may be expressed in terms of the entropy of y
as I ( x : y)  H ( y)  H ( y / x) where H(y/x) is a conditional entropy.

Example1: Suppose we have a discrete information source that emits a binary digit either 0 or 1,
with equal probability every ts second. The information content of each output from

the source is:

I (xi) = - log2 P(xi). (xi = 0, 1,)

1 1
=  log 2 where P(xi) =
2 2

= 1 bit

We define the average self-information, denoted by H(X), as


n n
H(X)=  P( x ) . I(xi) =   P( x ) . log P(xi)
i 1
i
i 1
i . . . . . . . . .. . . . . . . . . . . . . . . . . . .(5)

When X represents the alphabet of possible output letters from a source, H(x) represents the average
self information per source letter, and equation (5) is called the entropy of the source or source
entropy per code word.
If the letters from the source (special case) are equally probable, P(xi) = 1/n for all i and hence H(X) =
n
1 1
-  log( )  log n .
i 1 n n
In general, H(x)  logn, for any given set of source letter probabilities.
The entropy H(x) is a measure of the average amount of information covered per message.

In other words, the entropy of a discrete source is a maximum when the output letters are equally
probable. The average conditional self-information is called the conditional entropy and is defined as:

n m
1
H(X/Y) =  P( x ; y
i 1 j 1
i j ) log
P ( xi y j )
. . . .. . . . . . . . . . . . . . . . . . . . . . . . . . . . (6)

which is the uncertainty in X when Y is observed.

Example2: Consider a source that emits a sequence of statistically independent letters, where each
output letter is either 0 with probability q or 1 with probability 1-q. The entropy of this source is:
24
H(X) ≡ H (q) = -q log q - (1-q) log(1-q). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (7)
H(q)

1
Entropy in bits/letter

Binary entropy
function

0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9


Probability q
CHANNEL CHARACTERISTICS

Information is communicated by transmitting over a communication channel. Being a physical system,


communication channels impose certain limitations on the rate at which information can be
transmitted. Quantities like the number of channel states, the state transition probabilities, bandwidth,
and noise level can be related to channel physical characteristics.
A channel consisting of M input symbols x1, x2, . . . . .xm , N output symbols y1, y2, . . . yN , and with
input-output transition probabilities completely specified by the first -order conditional probabilities:
{P(yj,/xi); i = 1,2, . . . M; j = 1,2, . . .N} is called a discrete memoryless channel.

Let us consider the simplest model for a discrete channel called the BINARY SYMMETRIC
CHANNEL (BSC.). The BSC has identical input and output symbol or alphabets, namely, the binary
alphabet x: {0,1}, and the binary alphabet y: {0,1}.

The BSC is characterized by symmetrical transition probabilities for the probability that the symbol
that is received is the same as the symbol that was transmitted. If the transition probability (0 → 1 or,
since the channel is symmetric, 1 → 0) is P, then
prob {Y = 0/ X = 0} = 1 - po
prob { Y = 0/ X =1} = p1
prob { Y = 1/ X = 0} = po
prob { Y = 1/ X= 1} = 1 - po
A convenient graphical representation of the BSC channel transition properties is shown in fig. ( )
below:

1- P0
x=0 y=0

P
P
0
1- P1 0
x=1 y=1
Fig ( ): The binary symmetric channel
25
Example: Suppose that X and Y are binary-valued {0, 1} random variables that represent the input
and output of a binary-input, binary-output channel. The input symbols are equally likely and the
output symbols depend on the input according to the conditional probabilities:

P{Y = 0/ X = 0} = 1- P0
P { Y = 1/ X =0} = P0
P { Y = 1/ X = 1} = 1- P1
P { Y = 0/ X = 1} = P1

Let us determine the mutual information about the occurrence of the events X = 0 and X =1, given that
Y = 0.
From the probabilities given previously, we obtain:
P(Y  0)  P(Y  0 / X  0) P( X  0)  P(Y  0 / X  1) P( X  1) =
(Applying the rule that P( X  0)  P( X  1)  1 )
2

1-P0
x=0 y=0

P0

1-P1
x=1 y=1
P1
1  P0 P1 1
P(Y  0)  (1  P0 ). 1  P1 / 2    (1  P0  P1 )
2 2 2 2

P(Y  1)  P(Y  1/ x  0).P( x  0)  P(Y  1/ x  1) P( x  1)  1 (1  P  P0 ) .


2

Thus, the mutual information about the occurrence of the event x = 0, given that y = 0 is observed:

Similarly, given that Y=0 is observed, the mutual information about the occurrence of the event x = 1

P( y  0 / x  0) 2(1  P0 )
I ( x1 ; y1 )  I (0;0)  log2  log2
P( y  0) 1  P0  P1

is:

P(Y  0 / x  1)
log2
2 P1 P(Y  0)
I ( x2 ; y1 )  (1;0)  log2
(1  P0  P1 ) 2 P1
 log2
(1  P0  P1 )

If P0 = P1 = 0 , the channel is called noiseless and I(0;0) = log2 2 =1 bit.

Observed that the output specifies the input with certainty.


26
On the other hand, if P0 = P1 =1/2, the channel is useless because

I(0;0) = log2 1 = 0.

Also by substituting P(0) = P(1) = ½ and the appropriate channel transition probabilities in the
expression for mutual information I(x,y), we obtain the following expression for the capacity of the
binary symmetric channel:
C(P) = 1 + P logP + (1-P) log (1-p). . . . . . . . . .. . .. . . . . . . . . . ..10

Or C(P) = 1 – h(p), where

h(p) = - p logp – (1-p) log (1-p) is the binary entropy function for the probability pair

(p,1-p). (Compare with equation 7).

Finally, the capacity of additive white-noise channel is given by:


C = W log2(1 + S/N) bit/sec. . . . . . . . . . . . .. . . . . .. . . . . . . . . . . . . . . . . 11
Where,
W- Is the channel bandwidth available,
S- Signal power and
N- Noise power.
Equation (11) is the famous SHANNON’S theorem, which gives an upper bound to the information
capacity of a channel, based on it bandwidth and signal-to-noise ratio.

BANDWIDTH
Let us briefly provide definitions to the following: bandwidth, bandwidth of an analog signal,
bandwidth of digital signal and bandwidth of a channel.
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. Bandwidth of analog
and bandwidth of digital signals are calculated in different ways.
Bandwidth of Analog Signal: Analog signal bandwidth is measured in terms of its frequency (Hz). It
is defined as the range of frequencies that the composite analog signal carries. It is calculated by the
difference between the maximum and the minimum frequency. For example if frequency f1 =30 Hz
and f2 =90 Hz, then Bandwidth (W) =f2 – f1 =90 - 30 = 60 Hz..
Bandwidth of Digital Signal: Digital Signal bandwidth is measured in terms of bit rate second (bps).
It is defined as the maximum bit rate of the signal to be transmitted.
Bandwidth of a Channel (Wc): Bandwidth of signal is different from bandwidth of the medium or
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. 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. Generally, Wc >Ws.

CHANNEL CAPACITY
The Channel capacity- Is the maximum rate at which data can be transmitted over a given
communication path, or channel and under given conditions.
Data rate - is the rate at which data can be communicated in bits per second (bps)
Bandwidth – is the permissible rate of transmission expressed in cycles per second or Hertz (Hz).

Nyquist Bandwidth
In a noise free environment, the data rate equals the bandwidth of the signal.
27
Given a bandwidth of W, the highest (i.e., maximum) signal rate that can be carried is 2W. Thus by
Nyquist Bandwidth:
C = 2W.
For example: if the bandwidth is 3100Hz, then the capacity C = 2W = 2 x 3100bps. (Assume binary
signals of two levels).

If M possible voltage levels are used as signals, then each signal element can represent two bits.
C = 2W log2M – By Nyquist formula
where M is number of discrete signal or voltage levels.

SHANNON CAPACITY FORMULA


For a given level of noise, we would expect that a greater signal strength would improve the ability to
receive data correctly in the presence of noise.
Signal-to-noise ratio (SNR or S/N)
SNRdB = (10log10 signal power/noise power) dB
or (SNRdB = 10log10(SNR)
Shannon deduced that, the theoretical maximum channel capacity is:
[C = W log2 (1 + SNR)]
Where C - is the capacity of the channel in bps and;
W - is the Bandwidth.
The Shannon’s formula (c) sets the upper-bound on the achievable data rate.
Note that the wider the bandwidth, the more noise is admitted to the system, thus, as W increases,
SNR decreases.

Solved Example1:
Suppose that the spectrum of a channel is between 3MHz and 4 MHz and SNRdB is 24 dB. Find (i) the
channel capacity C, (ii) the number of discrete signal levels (M).
Solution:
(i) W = 4 MHz – 3 MHz = 1MHz = 106Hz
SNRdB = 24 dB = 10 Log10 (SNR) 24 = log10 (SNR)
SNR = 251 24 = log10 SNR
Using Shannon’s Formula SNR = 102.4
C = W log2 (1 + SNR) = 106 x log2 (1+251)  106 x 8 = 8Mbps
(ii) Using Nyquist Formula:
C = 2 *W* log2 M
8 x 106 = 2 x (106) x log2M
4 = log2 M  M = 24 = 16
M = 16

Example 2: Telephones with a bandwidth of roughly 3 KHz and a signal-to-noise ratio of


approximately 30 dB. The maximum data rate that such a signal can support is therefore
C = 3000 log2 (1+1000) = 29.9Kb/S

Since STN ratio = 10log10 Wanted signal power w ) = 3  dB


= 1  log10( unw
Unwanted noise power
wanted signal
Here, unwanted noise power

Since log10 = 3
28
Example 3: If the signal power equals the noise power in a channel of bandwidth 1Hz, what is the
theoretical information rate in bits/s which can be carried through the channel?
Solution:
By Shannon’ theorem:
S
C = W log2 (1+ ).
N
If W =1, and S = N, then
S
C = log2 (1+ ) = log2 2 = 1 bit/s
N

TRANSMISSION MEDIA (or LINES)


A transmission medium can be defined as anything that can carry information from a source to a
destination. A transmission line consists of a pair of copper conductors that are separated from one
another by a dielectric. We can distinguish two broad types transmission media in common use
nowadays. These are wired and wireless media. The wired media may be subdivided into three types,
namely:
(i) The two-wire or twin line, or twisted-pair cable
(ii) The coaxial line cable and
(iii) The optical fibre cables.
Figure ( ) below depicts the two-wire and the coaxial line
Jacket
a) (b) Inner
conductor
Outer
conductor
Fig (a) Twin line: two types (i) Shielded (STP) and (ii) Unshielded (UTP)
(b) Coaxial line: two types (i) Thinnet (flexible, light and is about 0.25 inches
thickness) and (ii) Thicknet (does not bend, about 0.5 inches13mm in diameter).

Unshielded TP

The quality of UTP may vary from telephone-grade wire to extremely high-speed cable. The cable has
four pairs of wires inside the jacket. Each pair is twisted with a different number of twists per inch to
help eliminate interference from adjacent pairs and other electrical devices. The tighter the twisting,
the higher the supported transmission rate and the greater the cost per foot. The EIA/TIA (Electronic
Industry Association/Telecommunication Industry Association) has established standards of UTP and
rated six categories of wire (additional categories are emerging).

Categories of Unshielded Twisted Pair

Category Speed Use


29

1 1 Mbps Voice Only (Telephone Wire)

2 4 Mbps LocalTalk & Telephone (Rarely used)

3 16 Mbps 10BaseT Ethernet

4 20 Mbps Token Ring (Rarely used)

100 Mbps (2 pair) 100BaseT Ethernet


5
1000 Mbps (4 pair) Gigabit Ethernet

5e 1,000 Mbps Gigabit Ethernet

6 10,000 Mbps Gigabit Ethernet

Unshielded Twisted Pair Connector

The standard connector for unshielded twisted pair cabling is an RJ-45 connector. This is a plastic
connector that looks like a large telephone-style connector (See fig. below). A slot allows the RJ-45 to
be inserted only one way. RJ stands for Registered Jack, implying that the connector follows a
standard borrowed from the telephone industry. This standard designates which wire goes with each
pin inside the connector.

Fig.( ). RJ-45 connector

Shielded Twisted Pair (STP) Cable


Although UTP cable is the least expensive cable, it may be susceptible to radio and electrical
frequency interference (it should not be too close to electric motors, fluorescent lights, etc.). If you
must place cable in environments with lots of potential interference, or if you must place cable in
extremely sensitive environments that may be susceptible to the electrical current in the UTP, shielded
twisted pair may be the solution. Shielded cables can also help to extend the maximum distance of the
cables.
30
Shielded twisted pair cable is available in three different configurations:

1. Each pair of wires is individually shielded with foil.


2. There is a foil or braid shield inside the jacket covering all wires (as a group).
3. There is a shield around each individual pair, as well as around the entire group of
wires (referred to as double shield twisted pair).

Coaxial Cable
Coaxial cabling has a single copper conductor at its center. A plastic layer provides insulation between
the center conductor and a braided metal shield (See fig. below). The metal shield helps to block any
outside interference.

Fig. Coaxial cable

Although coaxial cabling is difficult to install, it is highly resistant to signal interference. In addition, it
can support greater cable lengths between network devices than twisted pair cable. The two types of
coaxial are:

(i) Thin coaxial cable is also referred to as thinnet. 10Base2 refers to the specifications for thin coaxial
cable carrying Ethernet signals. The 2 refers to the approximate maximum segment length being 200
meters. In actual fact the maximum segment length is 185 meters. Thin coaxial cable has been
popular.

(ii) Thick coaxial cable is also referred to as thicknet. 10Base5 refers to the specifications for thick
coaxial cable carrying Ethernet signals. The 5 refers to the maximum segment length being 500
meters. Thick coaxial cable has an extra protective plastic cover that helps keep moisture away from
the center conductor. This makes thick coaxial a great choice when running longer lengths in a linear
bus network. One disadvantage of thick coaxial is that it does

Coaxial Cable Connectors

The most common type of connector used with coaxial cables is the Bayone-Neill-Concelman (BNC)
connector (See fig. below). Different types of adapters are available for BNC connectors, including a
T-connector, barrel connector, and terminator. Connectors on the cable are the weakest points in any
network. To help avoid problems with your network, always use the BNC connectors that crimp,
rather

Fig. . BNC connector


31
Fiber Optic Cable
Fiber optic cabling consists of a center glass core surrounded by several layers of protective materials
(See fig. below). It transmits light rather than electronic signals eliminating the problem of electrical
interference. This makes it ideal for certain environments that contain a large amount of electrical
interference. It has also made it the standard for connecting networks.

Fiber optic cable has the ability to transmit signals over much longer distances than coaxial and
twisted pair. It also has the capability to carry information at vastly greater speeds. This capacity
broadens communication possibilities to include services such as video conferencing and interactive
services. The cost of fiber optic cabling is comparable to copper cabling.

The center core of fiber cables is made from glass or plastic fibers (see fig below). A plastic coating
then cushions the fiber center, and kevlar fibers help to strengthen the cables and prevent breakage.
The outer insulating jacket made of teflon or PVC.

Fig.. Fiber optic cable

There are two common types of fiber cables -- single mode and multimode.

Multimode cable has a larger diameter; however, both cables provide high bandwidth at high speeds.
Single mode can provide more distance, but it is more expensive.

Ethernet Cable Summary

Specification Cable Type

10BaseT Unshielded Twisted Pair

10Base2 Thin Coaxial

10Base5 Thick Coaxial

100BaseT Unshielded Twisted Pair

100BaseFX Fiber Optic

100BaseBX Single mode Fiber

100BaseSX Multimode Fiber

1000BaseT Unshielded Twisted Pair

1000BaseFX Fiber Optic


32

1000BaseBX Single mode Fiber

1000BaseSX Multimode Fiber

An optical-fibre cable consists of a cylindrical glass core that is surrounded by a glass cladding and it
is able to transmit a light wave with very little loss of energy.
Advantages of optical-fibre cable over copper transmission lines are:
i) Light-weight, small-dimensioned cables;
ii) Very wide bandwidth;
iii) Freedom from electromagnetic interference;
iv) Low attenuation, i.e. low decay in signal;
v) High reliability and long life;
vi) Cheap raw materials and
vii) Scalable (negligible) crosstalk between fibres in the same line.
For these reasons, optical fibre is particularly suited to the transmission of digital signals and it
is often used for the cabling of a local area network (LANS).
We can generally classify two types of parameters of a medium. These are
a) Primary parameters and
b) Secondary parameters.
Let us briefly examine them in turn.
a) Primary parameters:
The conductors that form a pair in a telephone line usually comprise of four parameters which
are as follows:
i) Resistance (R)
ii) Inductance (L)
iii) Capacitance (C) and
iv) Leakance (g) or conductance of a line.
All these four parameters are uniformly distributed over the length of the line.
The resistance, R, is the loop resistance in ohms of a one-kilometre length of the line (i.e. the
sum of the resistances of each conductor).
The inductance, L, is the total series inductance of both conductors, or loop inductance. It is
measured in henry’s per kilometer.
The capacitance, C, is the total capacitance between a one-kilometre length of the two
conductors measured in microfarads per kilometer.
The leakance, G, represents the leakage of current between the two conductors. This leakage
occurs partly because the insulation resistance between the conductors is not infinite, and partly
because current must be supplied to supply the power losses in the dielectric as the line capacitance is
charged and discharged. Figure below show a typical transmission line:

d L2d 2
R1 d1 2 L1 2
1 1

d
R2 2
C G 1

dl
33

Generally, the line is considered to consist of a very large number of very short lengths dL of line
connected in cascade. Each short section has a total shunt capacitance CdL and total shunt leakage
GdL. The total series resistance and inductance are RdL and LdL, respectively.

b) Secondary Parameters of a line


The secondary parameters of a transmission line are its:
i) Characteristic impedance,
ii) Attenuation coefficient
iii) Phase-change coefficient
iv) Velocity of propagation
i) Characteristic impedance, Z0 of a transmission line is the ratio between the voltage across
input terminals and the current flowing into the terminals.
V
That is, Z 0  s (in ohms)
Is I s

Or Vs  Z 0 I s
Z0
Vs

ii) Attenuation- is the term given to the decay in the amplitude of a current, voltage or wave
along a transmission line, which happens in an exponential manner. Attenuation therefore refers to the
progressive reduction in propagated signal.
The percentage reduction in amplitude is exactly the same in each kilometer of the line. If for
instance, the input voltage is 12V and 10% is lost in every kilometer of the line. Then the voltage that
will enter the second kilometer is 10.8V, and in the third kilometer is 9.72V etc. It is measured in
Decibels per kilometer.
iii) Phase-change coefficient
The phase different between the voltages 1 Km apart is known as the phase-change coefficient
 of the line. The phase-change coefficient is measured in radians per kilometer. In each kilometer
length of the line there will be the same phase shift and hence for a line that is L kilometers long the
total phase difference is equal to  L rad.
iv) Velocity of propagation
The phase velocity vp of a line is the velocity with which a sinusoidal wave travels along that
line. The phase velocity is equal to the angular velocity (w) of the signal divided by the phase-change
w
coefficient; ( m / s )

w
For a digital data waveform the ratio must be constant at all frequencies.

Transmission Media Problems


Round-trip delay (a) – is the time delay between the first bit of a block being transmitted by the sender
and the last bit of its associated acknowledgment being received.
Round trip delay (a) = TP/Tx
TP – propagation delay and Tx – transmission delay
Where Tp = Physical separation (S) in meters/Velocity of propagation (V) in
meters per second
and Tx = number of bits to be transmitted (N)/Link bit rate (R) in bits per second
34
Solved Example:
A 1000 – bit block of data is to be transmitted between two computers. Determine the ratio of the
propagation delay to the transmission delay (a), for the following types of data link:
(i) 100m of twisted-pair wire and a transmission rate of 10kbps,
(ii) 10km of coaxial cable and a transmission rate of 1Mbps,
(iii) 50,000km of free space (satellite link) and a transmission rate of 10 Mbps. Assume that the
velocity of propagation of an electrical signal within each type of cable is 2 x 108ms-1, and that
of free space 3 x 108ms-1
Solution:
(i) Tp = S/V = 100/2x108 = 5 x 10-7 S a = Tp/Tx = 5x10-7/0.1 = 5.10-6
Tx = N/R = 1000/10x103 = 0.1s

(ii) Tp = S/V = 10x103/2x108 = 5 x 10-5 S a = Tp/Tx = 5x10-5/1x10-3 = 5x10-2


Tx = N/R = 1000/1x106 = 1x10-3 S

(iii) Tp = S/V = 5x107/3x108 = 1.67 x 10-1S a=Tp/Tx= 1.67x10-1/1x10-4 =


1.67x103
Tx = N/R = 1000/10x106 = 1x10-4S

Conclusion
If a < 1, then the round-trip delay is determined by the transmission delay Tp.
If a = 1, then both delays have equal effect
If a > 1, then the propagation delay dominates.
Other important characteristics include distortion, bit error, noise, etc.

Different Methods for Digital Signal Transmission


A digital signal periodic or non-periodic, is a composite analog signal with frequencies between zero
and infinity. We can transmit a digital signal by using one of two different approaches: baseband
transmission or broadband transmission (using modulation).

1. Baseband Transmission
Baseband transmission means sending a digital signal over a channel without changing the digital
signal to an analog signal. The following figure shows baseband transmission.

Baseband transmission requires a low-pass channel, a channel with a bandwidth that starts from zero.
This is the case if we have a dedicated medium with a bandwidth constituting only one channel. For
example, the entire bandwidth of a cable connecting two computers is one single channel. As another
example, we may connect several computers to a bus, but not allow more than two stations to
communicate at a time.
35
2. Broadband Transmission (Using Modulation)
Broadband transmission or modulation means changing the digital signal to an analog signal for
transmission. Modulation allows us to use a band pass channel-a channel with a bandwidth that does
not start from zero. This type of channel is more available than a low-pass channel. The following
figure shows a band pass channel.

Differences between Baseband and Broadband


The Table below shows the differences between the above.
S/N Baseband (CDMA 1.25 MHZ) Broadband (WCDMA 5 MHZ)
1 Digital signals are used. Analog signals are used over fibre cables, etc
2 Frequency Division Multiplexing FDM is possible.
is NOT possible.
3 It is bi-directional transmission Transmission of data is unidirectional.
4 Signal traveling distance is short. Signal traveling distance is long.
5 Entire bandwidth of the cable is The bandwidth is divided into a number of
used by a single signal. channels with multiple frequencies to be
transmitted simultaneously.
6 Signals are of very low Signals are of high frequencies thus used for
frequencies since Not used for long distances..
long distances.
7 Used in LAN implementation. Used in WAN implementation and may travel
over cables that are buried underground..

Transmission Impairment
Signals travel through transmission media, which are not perfect. The imperfection causes signal
impairment. This means that the signal at the beginning of the medium is not the same as the signal at
the end of the medium. What is sent is not what is received.
The three different causes of impairment are attenuation, distortion, and noise.
36
Attenuation:
Attenuation means a loss of energy. When a signal, simple or composite, travels through a medium, it
loses some of its energy in overcoming the resistance of the medium. That is why a wire carrying
electric signals gets warm, if not hot, after a while. Some of the electrical energy in the signal is
converted to heat. To compensate for this loss, amplifiers are used to amplify the signal. The following
figure shows the effect of attenuation and amplification.

Distortion:
Distortion means that the signal changes its form or shape. Distortion can occur in a composite signal
made of different frequencies. Each signal component has its own propagation speed (see the next
section) through a medium and, therefore, its own delay in arriving at the final destination. Differences
in delay may create a difference in phase if the delay is not exactly the same as the period duration. In
other words, signal components at the receiver have phases different from what they had at the sender.
The shape of the composite signal is therefore not the same. The following figure shows the effect of
distortion on a composite signal.

When a digital signal is transmitted over a telephone circuit the characteristics of that circuit will cause
the received signal to be both reduced in amplitude and distorted. This is observed if the regular time
interval between successive 1’s and 0’s of the transmitted signal is either lengthened or shortened at
the receiver end. When the various frequency components making up the signal arrive at the receiver
with varying delays, this is called delay distortion.
This can result in some of the bits incorrectly received or lost.
37

1 1 2 3 4 5 6

Transmitted signal
0

1 2 3 4 5 6
1
5
Received signal
0

The term positive bias refers to the binary 1 pulses being lengthened and negative bias to the 0 pulses
becoming longer.
T1T0 100
The percentage bias distortion = x %
2(T1  T0 ) 1

Where T1 and T0 are the time durations of the binary 1 and the binary 0 pulses, respectively.

Noise:
Noise is another cause of impairment. Several types of noise, such as thermal noise, induced noise,
crosstalk, and impulse noise, may corrupt the signal. Thermal noise is the random motion of electrons
in a wire which creates an extra signal not originally sent by the transmitter. Induced noise comes from
sources such as motors and appliances.

These devices act as a sending antenna, and the transmission medium acts as the receiving antenna.
Crosstalk is the effect of one wire on the other. One wire acts as a sending antenna and the other as the
receiving antenna. Impulse noise is a spike (a signal with high energy in a very short time) that comes
from power lines, lightning, and so on. The following figure shows the effect of noise on a signal.
Noise is also a random signal obtained as the result of measuring some physical quantity. One
characteristic of physical measurements is that in addition to physical quantity of interest, other effects
can influence the outcome. Noise is not the physical process itself, but rather the incomplete
representation of a complex process by a signal having few degrees of freedom. Noise comes about
because we operate measuring equipment in an environment that is subject to unavoidable interactions
with a large number of particles in random motion.

Sources of noise
The various sources of noise that can affect a data communication circuit are:
a) Thermal agitation noise in conductors, resistors and semiconductors. Thermal noise are on the line
even when no signal is being transmitted.
b) Short noise and flicker noise in semiconductors,
c) Faulty electrical connections which may cause short breaks in the transmission path,
38
d) Electrical and magnetic couplings to other circuits, causing cross-talk in equipment wiring and in
cables.
A signal –to –noise ratio (STN) may be defined as the ratio of the wanted signal to the unwanted noise
power;
STN = (wanted signal power/ unwanted noise power);

STN ratio = 10 log10 [(wanted signal power)/unwanted noise power)] (in decibel).

BIT ERROR

The data transitions in the received data waveform tend to move around from their ideal positions in
time. This results in an effect that is known as bit jitter. See figure ( ) below

Transmitted
pulse train

t Received
pulse train

If  is the duration of a pulse and t is the movement of a pulse from it s idea position, then,

Bit jitter = t max  t min or

t max  t min
Bit jitter = = x100%

BIT ERROR RATE (BER)

Any data circuit is always subjected to noise and interference voltages that originate from a wide
variety of sources. These unwanted voltages are superimposed upon the received data voltage and
usually corrupt the waveform.
At each sampling instant the receiver must determine whether the bit received at that moment
is a 1 or a 0 and any waveform corruption increases the probability of this determination being
incorrect and hence of an error occurring. The bit error rate (BER) is given by
Number of bits wrongly received
BER = Total number of bits transmitted

Example:
A message is transmitted at 2400 bits/s and it occupies a time period of 1 minute and 20 seconds. If
two of the received bits are in error calculate the BER.

Solution:
At 2400 bits/s there will be no start and stop bits and so the total number of bits transmitted is 80 x
2400 =192000. Hence,

BER= 2/19200 = 10.42x10-6


39
MODULATION AND DEMODULATION

Signal Encoding Techniques


Time Domain Concepts
Viewed as a function of time, an electromagnetic signal can be either ANALOG or DIGITAL.
An analog signal is one in which the signal intensity varies in a smooth fashion over time. In
other words, there are no breaks or discontinuities in the signal.
A digital signal is one in which the signal intensity maintains a constant level for some period
of time and then changes to another constant level.
Both analog and digital information can be encoded as either analog or digital signals. The
particular encoding that is chosen depends on the specific requirements to be met and the media and
communications facilities available. Four combinations are available. These are:
1. Digital data, digital signal: This is the simplest form of digital encoding of digital data. Here,
one voltage level is assigned to binary one, and another to binary zero.
2. Digital data, Analog Signal: In this technique, a modem is used to convert digital data to an
analog signal so that it can be transmitted over an analog line. The basic techniques used are
amplitude shift keying (ASK), frequency shift keying (FSK), and phase shift keying (PSK).
All involve altering one or more characteristics of a carrier frequency to represent binary data.
3. Analog data, digital signal: Analog data, such as voice and video, are often digitized to be
able to use digital transmission facilities. The simplest technique is pulse code modulation
(PCM), which involves sampling the analog data periodically and quantizing the samples.
4. Analog data, analog signal: Analog data are modulated by a carrier frequency to produce an
analog signal in a different frequency band, which can be utilized on an analog transmission
system. The basic techniques are amplitude modulation (AM), frequency modulation (FM)
and phase modulation (PM).

Analog to Digital Conversion Techniques


If we have an analog signal such as one created by a microphone or camera. To change an analog
signal to digital data we use two techniques, pulse code modulation and delta modulation. After the
digital data are created (digitization) then we convert the digital data to a digital signal.

1. Pulse Code Modulation (PCM):

Pulse Code Modulation (PCM) is the most common technique used to change an analog signal to
digital data (digitization). A PCM encoder has three processes as shown in the following Figure.
(i)The analog signal is sampled.
(ii)The sampled signal is quantized.
(iii)The quantized values are encoded as streams of bits.
40

Sampling

The first step in PCM is sampling. The analog signal is sampled every Ts s, where Ts is the sample
interval or period. The inverse of the sampling interval is called the sampling rate or sampling
frequency and denoted by ƒs, Where ƒs = 1/ Ts.

There are three sampling methods-ideal, natural, and flat-top. In ideal sampling, pulses from the
analog signal are sampled. This is an ideal sampling method and cannot be easily implemented. In
natural sampling, a high-speed switch is turned on for only the small period of time when the sampling
occurs.
The result is a sequence of samples that retains the shape of the analog signal. The most common
sampling method, called sample and hold, however, creates flat-top samples by using a circuit. The
sampling process is sometimes referred to as pulse amplitude modulation (PAM). The different
sampling methods are as shown in the following figure.

Sampling Rate
One important consideration is the sampling rate or frequency. What are the restrictions on Ts?
According to the Nyquist theorem, to reproduce the original analog signal, one necessary condition is
that the sampling rate be at least twice the highest frequency in the original signal.
As for this Theorem, First, we can sample a signal only if the signal is band-limited i.e a signal with an
infinite bandwidth cannot be sampled. Second, the sampling rate must be at least 2 times the highest
frequency, not the bandwidth. If the analog signal is low-pass, the bandwidth and the highest
frequency are the same value. If the analog signal is bandpass, the bandwidth value is lower than the
value of the maximum frequency.

Quantization
The result of sampling is a series of pulses with amplitude values between the maximum and
minimum amplitudes of the signal. The set of amplitudes can be infinite with non-integral values
between the two limits. These values cannot be used in the encoding process. The following are the
steps in quantization:
41
1. We assume that the original analog signal has instantaneous amplitudes between Vmin and Vmax
2. We divide the range into L zones, each of height ∆ (delta).
∆=(Vmax-Vmin)/L
3. We assign quantized values of 0 to L - I to the midpoint of each zone.
4. We approximate the value of the sample amplitude to the quantized values.
As a simple example

assume that we have a sampled signal and the sample amplitudes are between -20 and +20 V.
We decide to have eight levels (L = 8). This means that ∆ =5 V.

We have shown only nine samples using ideal sampling (for simplicity). The value at the top of each
sample in the graph shows the actual amplitude. In the chart, the first row is the normalized value for
each sample (actual amplitude/∆).
The quantization process selects the quantization value from the middle of each zone. This means that
the normalized quantized values (second row) are different from the normalized amplitudes. The
difference is called the normalized error (third row). The fourth row is the quantization code for each
sample based on the quantization levels at the left of the graph. The encoded words (fifth row) are the
final products of the conversion.

Quantization Levels:
In the above example, we showed eight quantization levels. The choice of L, the number of levels,
depends on the range of the amplitudes of the analog signal and how accurately we need to recover the
signal. If the amplitude of a signal fluctuates between two values only, we need only two levels; if the
signal, like voice, has many amplitude values, we need more quantization levels. In audio digitizing, L
42
is normally chosen to be 256; in video it is normally thousands. Choosing lower values of L increases
the quantization error if there is a lot of fluctuation in the signal.

Quantization Error:
One important issue is the error created in the quantization process. Quantization is an approximation
process. The input values to the quantizer are the real values; the output values are the approximated
values. The output values are chosen to be the middle value in the zone. If the input value is also at the
middle of the zone, there is no quantization error; otherwise, there is an error. In the previous example,
the normalized amplitude of the third sample is 3.24, but the normalized quantized value is 3.50. This
means that there is an error of +0.26. The value of the error for any sample is less than ∆/2. In other
words, we have ∆/2<=error<= ∆/2.

Uniform Versus Non uniform Quantization:


For many applications, the distribution of the instantaneous amplitudes in the analog signal is not
uniform. Changes in amplitude often occur more frequently in the lower amplitudes than in the higher
ones. For these types of applications it is better to use nonuniform zones. In other words, the height of
∆ is not fixed; it is greater near the lower amplitudes and less near the higher amplitudes.

Nonuniform quantization can also be achieved by using a process called companding and expanding.
The signal is companded at the sender before conversion; it is expanded at the receiver after
conversion. Companding means reducing the instantaneous voltage amplitude for large values;
expanding is the opposite process. Companding gives greater weight to strong signals and less weight
to weak ones. It has been proved that nonuniform quantization effectively reduces the SNRdB of
quantization.

Encoding
The last step in PCM is encoding. After each sample is quantized and the number of bits per sample is
decided, each sample can be changed to an nb-bit code word. In the above figure the encoded words
are shown in the last row. A quantization code of 2 is encoded as 010; 5 is encoded as 101; and so on.
Note that the number of bits for each sample is determined from the number of quantization levels. If
the number of quantization levels is L, the number of bits is nb=log2 L. In our example L is 8 and nb
is therefore 3. The bit rate can be found from the formula.
Bit-rate = Sampling rate X Number of bites per sample= ƒs X nb

II. Delta Modulation (DM)


PCM is a very complex technique. Number of other techniques has been developed to reduce the
complexity of PCM. The simplest is delta modulation. PCM finds the value of the signal amplitude for
each sample; DM finds the change from the previous sample. The following figure shows the process.
Note that there are no code words here; bits are sent one after another.
43

Modulator
The modulator is used at the sender site to create a stream of bits from an analog signal. The process
records the small positive or negative changes, called delta . If the delta is positive, the process
records a 1; if it is negative, the process records a 0. However, the process needs a base against which
the analog signal is compared. The modulator builds a second signal that resembles a staircase.
Finding the change is then reduced to comparing the input signal with the gradually made staircase
signal.

The modulator, at each sampling interval, compares the value of the analog signal with the last value
of the staircase signal. If the amplitude of the analog signal is larger, the next bit in the digital data is
1; otherwise, it is O. The output of the comparator, however, also makes the staircase itself. If the next
bit is I, the staircase maker moves the last point of the staircase signal up; it the next bit is 0, it moves
it down. Note that we need a delay unit to hold the staircase function for a period between two
comparisons.

Demodulator
The demodulator takes the digital data and, using the staircase maker and the delay unit, creates the
analog signal. The created analog signal, however, needs to pass through a low-pass filter for
smoothing.

Adaptive DM
A better performance can be achieved if the value of is not fixed. In adaptive delta modulation, the
value of changes according to the amplitude of the analog signal.

Quantization Error
It is obvious that DM is not perfect. Quantization error is always introduced in the process. The
quantization error of DM, however, is much less than that for PCM.
Digital to Analog Conversion Techniques:
Digital-to-analog conversion is the process of changing one of the characteristics of an analog signal
based on the information in digital data.
A sine wave is defined by three characteristics: amplitude, frequency, and phase. When we change
anyone of these characteristics, we create a different version of that wave. So, by changing one
characteristic of a simple electric signal, we can use it to represent digital data.
44
There are three mechanisms for modulating digital data into an analog signal: amplitude shift keying
(ASK), frequency shift keying (FSK), and phase shift keying (PSK). In addition, there is a fourth (and
better) mechanism that combines changing both the amplitude and phase, called quadrature amplitude
modulation (QAM).

Bandwidth
The required bandwidth for analog transmission of digital data is proportional to the signal rate except
for FSK, in which the difference between the carrier signals needs to be added.
Carrier Signal
In analog transmission, the sending device produces a high-frequency signal that acts as a base for the
information signal. This base signal is called the carrier signal or carrier frequency. The receiving
device is tuned to the frequency of the carrier signal that it expects from the sender. Digital
information then changes the carrier signal by modifying one or more of its characteristics
(amplitude, frequency, or phase). This kind of modification is called modulation (shift keying).

1.Amplitude Shift-Keying

In amplitude shift keying, the amplitude of the carrier signal is varied to create signal elements. In
ASK, the two binary values are represented by two different amplitudes of the carrier frequency.
Usually, one of the amplitudes is zero and the other by the absence of the carrier. The resulting
transmitted signal becomes as follows. Both frequency and phase remain constant while the amplitude
changes.
 ACos(2f0t ) for binary1
ASK: S(t)= 
0 for binary0
ASK is susceptible to sudden gain changes and is a rather inefficient modulation technique.

Binary ASK (BASK)


ASK is normally implemented using only two levels. This is referred to as binary amplitude shift
keying or on-off keying (OOK). The peak amplitude of one signal level is 0; the other is the same as
the amplitude of the carrier frequency. The following figure gives a conceptual view of binary ASKS.
45

Implementation:
If digital data are presented as a unipolar NRZ digital signal with a high voltage of 1V and a low
voltage of 0V, the implementation can achieved by multiplying the NRZ digital signal by the carrier
signal coming from an oscillator which is represented in the following figure. When the amplitude of
the NRZ signal is 1, the amplitude of the carrier frequency is held; when the amplitude of the NRZ
signal is 0, the amplitude of the carrier frequency is zero.

Bandwidth for ASK:


The carrier signal is only one simple sine wave, but the process of modulation produces a non-periodic
composite signal. This signal has a continuous set of frequencies. As we expect, the bandwidth is
proportional to the signal rate (baud rate).

However, there is normally another factor involved, called d, which depends on the modulation and
filtering process. The value of d is between 0 and 1. This means that the bandwidth can be expressed
as shown, where S is the signal rate and the B is the bandwidth.

B = (1 +d) x S

The formula shows that the required bandwidth has a minimum value of S and a maximum value of
2S. The most important point here is the location of the bandwidth. The middle of the bandwidth is
where fc the carrier frequency, is located. This means if we have a bandpass channel available, we can
choose our fc so that the modulated signal occupies that bandwidth. This is in fact the most important
46
advantage of digital-to- analog conversion.

2. Frequency Shift Keying


In frequency shift keying, the frequency of the carrier signal is varied to represent data. In FSK the
higher frequency is used to represent binary 0, while the lower represents binary 1. The frequency of
the modulated signal is constant for the duration of one signal element, but changes for the next signal
element if the data element changes. Both peak amplitude and phase remain constant for all signal
elements.

Binary FSK (BFSK)


One way to think about binary FSK (or BFSK) is to consider two carrier frequencies. In the following
Figure, we have selected two carrier frequencies f1 and f2. We use the first carrier if the data element
is 0; we use the second if the data element is 1. The resulting transmitted signal for one bit time is

 A Cos(2f1t ) binary1
BFSK: S(t) = 
 A Cos(2f 2t ) Binary0
where f1 and f2 are typically offset from the carrier frequency fc by equal but opposite amounts.

The above figure shows, the middle of one bandwidth is f1 and the middle of the other is f2. Both f1
and f2 are ∆f apart from the midpoint between the two bands. The difference between the two
frequencies is 2∆f.

Implementation:
There are two implementations of BFSK: non-coherent and coherent. In non-coherent BFSK, there
may be discontinuity in the phase when one signal element ends and the next begins. In coherent
BFSK, the phase continues through the boundary of two signal elements. Non-coherent BFSK can be
implemented by treating BFSK as two ASK modulations and using two carrier frequencies. Coherent
BFSK can be implemented by using one voltage-controlled oscillator (VCO) that changes its
frequency according to the input voltage.

The following figure shows the simplified idea behind the second implementation. The input to the
oscillator is the unipolar NRZ signal. When the amplitude of NRZ is zero, the oscillator keeps its
regular frequency; when the amplitude is positive, the frequency is increased.
47

Bandwidth for BFSK:


The above figure shows the bandwidth of FSK. Again the carrier signals are only simple sine waves,
but the modulation creates a non-periodic composite signal with continuous frequencies. We can think
of FSK as two ASK signals, each with its own carrier frequency f1 and f2. If the difference between
the two frequencies is 2∆f, then the required bandwidth is

B=(l+d)XS+2∆f

3. Phase Shift Keying:

In phase shift keying, the phase of the carrier is varied to represent two or more different signal
elements. Both peak amplitude and frequency remain constant as the phase changes. There are many
variations of PSK. These are Two-level PSK or Binary PSK, Four-level PSK or Quadrature PSK and
Multi-level PSK.

Binary PSK
The simplest PSK is binary PSK, in which we have only two signal elements, one with a phase of 0°,
and the other with a phase of 180°. The following figure gives a conceptual view of PSK. Binary PSK
is as simple as binary ASK with one big advantage-it is less susceptible to noise. In ASK, the criterion
for bit detection is the amplitude of the signal. But in PSK, it is the phase. Noise can change the
amplitude easier than it can change the phase. In other words, PSK is less susceptible to noise than
ASK. PSK is superior to FSK because we do not need two carrier signals. The resulting transmitted
signal for one bit time is:
 ACos(2f ct )  A Cos(2f ct ) binary1
BPSK: S(t) =  
 A Cos(2f ct   )   ACos(2f ct ) binary0
48

Bandwidth:
The bandwidth is the same as that for binary ASK, but less than that for BFSK. No bandwidth is
wasted for separating two carrier signals.

Implementation:
The implementation of BPSK is as simple as that for ASK. The reason is that the signal element with
phase 180° can be seen as the complement of the signal element with phase 0°. This gives us a clue on
how to implement BPSK. We use a polar NRZ signal instead of a unipolar NRZ signal, as shown in
the following figure . The polar NRZ signal is multiplied by the carrier frequency. The 1 bit (positive
voltage) is represented by a phase starting at 0° the 0 bit (negative voltage) is represented by a phase
starting at 180°.

Quadrature Amplitude Modulation (QAM)

PSK is limited by the ability of the equipment to distinguish small differences in phase. This factor
limits its potential bit rate. So far, we have been altering only one of the three characteristics of a sine
wave at a time; but what if we alter two? Why not combine ASK and PSK? The idea of using two
carriers, one in-phase and the other quadrature, with different amplitude levels for each carrier is the
concept behind quadrature amplitude modulation (QAM).

The possible variations of QAM are numerous. The following figure shows some of these schemes. In
the following figure Part a shows the simplest 4-QAM scheme (four different signal element types)
using a unipolar NRZ signal to modulate each carrier. This is the same mechanism we used for ASK
(OOK). Part b shows another 4-QAM using polar NRZ, but this is exactly the same as QPSK. Part c
shows another QAM-4 in which we used a signal with two positive levels to modulate each of the two
carriers. Finally, Part – d shows a 16-QAM constellation of a signal with eight levels, four positive and
four negative.
49

Analog To Analog Conversion Techniques


Analog-to-analog conversion, or analog modulation, is the representation of analog information by an
analog signal. Modulation is needed if the medium is bandpass in nature or if only a bandpass channel
is available to us.
An example is radio. The government assigns a narrow bandwidth to each radio station. The analog
signal produced by each station is a low-pass signal, all in the same range. To be able to listen to
different stations, the low-pass signals need to be shifted, each to a different range.
Analog- to digital conversion can be accomplished in three ways, namely:
(i) Amplitude Modulation (AM)
(ii) Frequency Modulation (FM)
(iii) Phase Modulation (PM)

1. Amplitude Modulation:
In AM transmission, the carrier signal is modulated so that its amplitude varies with the changing
amplitudes of the modulating signal. The frequency and phase of the carrier remain the same. Only the
amplitude changes to follow variations in the information. The following figure shows how this
concept works. The modulating signal is the envelope of the carrier.

AM is normally implemented by using a simple multiplier because the amplitude of the carrier signal
needs to be changed according to the amplitude of the modulating signal.
50
AM Bandwidth:
The modulation creates a bandwidth that is twice the bandwidth of the modulating signal and covers a
range centered on the carrier frequency. However, the signal components above and below the carrier
frequency carry exactly the same information. For this reason, some implementations discard one-half
of the signals and cut the bandwidth in half.

The total bandwidth required for AM can be determined from the bandwidth of the audio signal:

BAM =2B

Standard Bandwidth allocation for AM Radio:


The bandwidth of an audio signal (speech and music) is usually 5 kHz. Therefore, an AM radio station
needs a bandwidth of 10kHz. In fact, the Federal Communications Commission (FCC) allows 10 kHz
for each AM station.
AM stations are allowed carrier frequencies anywhere between 530 and 1700 kHz (1.7 MHz).
However, each station's carrier frequency must be separated from those on either side of it by at least
10 kHz (one AM bandwidth) to avoid interference. If one station uses a carrier frequency of 1100 kHz,
the next station's carrier frequency cannot be lower than 1110 kHz.

2. Frequency Modulation

In FM transmission, the frequency of the carrier signal is modulated to follow the changing voltage
level (amplitude) of the modulating signal. The peak amplitude and phase of the carrier signal remain
constant, but as the amplitude of the information signal changes, the frequency of the carrier changes
correspondingly.

The following figure shows the relationships of the modulating signal, the carrier signal, and the
resultant FM signal. FM is normally implemented by using a voltage-controlled oscillator as with
FSK. The frequency of the oscillator changes according to the input voltage which is the amplitude of
the modulating signal.

FM Bandwidth

The actual bandwidth is difficult to determine exactly, but it can be shown empirically that it is several
times that of the analog signal or 2(1 + β)B where β is a factor depends on modulation technique with
a common value of 4.
51

Standard Bandwidth allocation for FM Radio:


The bandwidth of an audio signal (speech and music) broadcast in stereo is almost 15 kHz. The FCC
allows 200 kHz (0.2 MHz) for each station. This mean β = 4 with some extra guard band. FM stations
are allowed carrier frequencies anywhere between 88 and 108 MHz. Stations must be separated by at
least 200 kHz to keep their bandwidths from overlapping.
To create even more privacy, the FCC requires that in a given area, only alternate bandwidth
allocations may be used. The others remain unused to prevent any possibility of two stations
interfering with each other. Given 88 to 108 MHz as a range, there are 100 potential PM bandwidths in
an area, of which 50 can operate at any one time.

3. Phase Modulation:
In PM transmission, the phase of the carrier signal is modulated to follow the changing voltage level
(amplitude) of the modulating signal. The peak amplitude and frequency of the carrier signal remain
constant, but as the amplitude of the information signal changes, the phase of the carrier changes
correspondingly. It is proved mathematically that PM is the same as FM with one difference.

In FM, the instantaneous change in the carrier frequency is proportional to the amplitude of the
modulating signal; in PM the instantaneous change in the carrier frequency is proportional to the
derivative of the amplitude of the modulating signal. The following figure shows the relationships of
the modulating signal, the carrier signal, and the resultant PM signal.
52

PM is normally implemented by using a voltage-controlled oscillator along with a derivative. The


frequency of the oscillator changes according to the derivative of the input voltage which is the
amplitude of the modulating signal.

PM Bandwidth

The actual bandwidth is difficult to determine exactly, but it can be shown empirically that it is several
times that of the analog signal. Although, the formula shows the same bandwidth for FM and PM, the
value of β is lower in the case of PM (around 1 for narrowband and 3 for wideband).

MULTIPLEXING (MUX)

Multiplexing and Types of Multiplexing


Multiplexing is the set of techniques that allows the simultaneous transmission of multiple signals
across a single data link. Whenever the bandwidth of a medium linking two devices is greater than the
bandwidth needs of the devices, the link can be shared. In a multiplexed system, n lines share the
bandwidth of one link.
The following figure shows the basic format of a multiplexed system. The lines on the left direct their
transmission streams to a multiplexer (MUX), which combines them into a single stream (many-to-
one). At the receiving end, that stream is fed into a demultiplexer (DEMUX), which separates the
stream back into its component transmissions (one-to-many) and directs them to their corresponding
lines.
53
The three basic multiplexing techniques are: (i) Frequency Division Multiplexing, (ii) Wavelength
Division Multiplexing and Time Division Multiplexing.
The first two are techniques designed for analog signals, while the third is for digital signal.
1. Frequency-Division Multiplexing
Frequency-division multiplexing (FDM) is an analog technique that can be applied when the
bandwidth of a link (in hertz) is greater than the combined bandwidths of the signals to be transmitted.
In FDM, signals generated by each sending device modulate different carrier frequencies. These
modulated signals are then combined into a single composite signal that can be transported by the link.
Carrier frequencies are separated by sufficient bandwidth to accommodate the modulated signal.
These bandwidth ranges are the channels through which the various signals travel.

M1(t)
f1
M2(t) Mn(t) st
Time f2  fe
FDM
Mn(t) signal
fn

f1 f2 f3 f4 f5 f6 Transmitter

Channels can be separated by strips of unused bandwidth-guard bands-to prevent signals from
overlapping. The following figure gives a conceptual view of FDM. In this illustration, the
transmission path is divided into three parts, each representing a channel that carries one transmission.

Multiplexing Process:
The following figure is a conceptual illustration of the multiplexing process. Each source generates a
signal of a similar frequency range. Inside the multiplexer, these similar signals modulates different
carrier frequencies (f1, f2 and f3). The resulting modulated signals are then combined into a single
composite signal that is sent out over a media link that has enough bandwidth to accommodate it.
Demultiplexing Process:
The demultiplexer uses a series of filters to decompose the multiplexed signal into its constituent
component signals. The individual signals are then passed to a demodulator that separates them from
their carriers and passes them to the output lines.

Applications of FDM:
To maximize the efficiency of their infrastructure, telephone companies have traditionally multiplexed
signals from lower-bandwidth lines onto higher-bandwidth lines.
54
A very common application of FDM is AM and FM radio broadcasting.

The first generation of cellular telephones (still in operation) also uses FDM.

Implementation:
FDM can be implemented very easily. In many cases, such as radio and television broadcasting, there
is no need for a physical multiplexer or demultiplexer. As long as the stations agree to send their
broadcasts to the air using different carrier frequencies, multiplexing is achieved. In other cases, such
as the cellular telephone system, a base station needs to assign a carrier frequency to the telephone
user. There is not enough bandwidth in a cell to permanently assign a bandwidth range to every
telephone user. When a user hangs up, her or his bandwidth is assigned to another caller.
Time-Division Multiplexing
Time-division multiplexing (TDM) is a digital process that allows several connections to share the
high bandwidth of a linle Instead of sharing a portion of the bandwidth as in FDM, time is shared.
Each connection occupies a portion of time in the link.

We can divide TDM into two different schemes: synchronous and statistical.

Synchronous Time-Division Multiplexing:


In synchronous TDM, each input connection has an allotment in the output even if it is not sending
data. In synchronous TDM, the data flow of each input connection is divided into units, where each
input occupies one input time slot.

A unit can be 1 bit, one character, or one block of data. Each input unit becomes one output unit and
occupies one output time slot. However, the duration of an output time slot is n times shorter than the
duration of an input time slot. If an input time slot is T s, the output time slot is T/n s, where n is the
number of connections. In other words, a unit in the output connection has a shorter duration; it travels
faster. The following figure shows an example of synchronous TDM where n is 3.
55

In synchronous TDM, a round of data units from each input connection is collected into a frame. If we
have n connections, a frame is divided into n time slots and one slot is allocated for each unit, one for
each input line. If the duration of the input unit is T, the duration of each slot is Tin and the duration of
each frame is T.
Time slots are grouped into frames. A frame consists of one complete cycle of time slots, with one slot
dedicated to each sending device. In a system with n input lines, each frame has n slots, with each slot
allocated to carrying data from a specific input line.

Interleaving
TDM can be visualized as two fast-rotating switches, one on the multiplexing side and the other on the
demultiplexing side. The switches are synchronized and rotate at the same speed, but in opposite
directions. On the multiplexing side, as the switch opens in front of a connection, that connection has
the opportunity to send a unit onto the path. This process is called interleaving. On the demultiplexing
side, as the switch opens in front of a connection, that connection has the opportunity to receive a unit
from the path.

Empty Slots
Synchronous TDM is not as efficient as it could be. If a source does not have data to send, the
corresponding slot in the output frame is empty. The following figure shows a case in which one of the
input lines has no data to send and one slot in another input line has discontinuous data.

The first output frame has three slots filled, the second frame has two slots filled, and the third frame
has three slots filled. No frame is full. We learn in the next section that statistical TDM can improve
the efficiency by removing the empty slots from the frame.
56
Data Rate Management
One problem with TDM is how to handle a disparity in the input data rates. If data rates are not the
same, three strategies, or a combination of them, can be used. The three different strategies are
multilevel multiplexing, multiple-slot allocation, and pulse stuffing.

Multilevel Multiplexing:
Multilevel multiplexing is a technique used when the data rate of an input line is a multiple of others.
For example, if we have two inputs of 20 kbps and three inputs of 40 kbps. The first two input lines
can be multiplexed together to provide a data rate equal to the last three. A second level of
multiplexing can create an output of 160 kbps.

Multiple-Slot Allocation:
Sometimes it is more efficient to allot more than one slot in a frame to a single input line. For
example, we might have an input line that has a data rate that is a multiple of another input. The input
line with a 50-kbps data rate can be given two slots in the output. We insert a serial-to-parallel
converter in the line to make two inputs out of one.

Pulse Stuffing:
Sometimes the bit rates of sources are not multiple integers of each other. Therefore, neither of the
above two techniques can be applied. One solution is to make the highest input data rate the dominant
data rate and then add dummy bits to the input lines with lower rates. This will increase their rates.
This technique is called pulse stuffing, bit padding, or bit stuffing. The input with a data rate of 46 is
pulse-stuffed to increase the rate to 50 kbps. Now multiplexing can take place.

Frame Synchronizing
The implementation of TDM is not as simple as that of FDM. Synchronization between the
multiplexer and demultiplexer is a major issue. If the, multiplexer and the demultiplexer are not
synchronized, a bit belonging to one channel may be received by the wrong channel.

For this reason, one or more synchronization bits are usually added to the beginning of each frame.
These bits, called framing bits, follow a pattern, frame to frame, that allows the demultiplexer to
synchronize with the incoming stream so that it can separate the time slots accurately. In most cases,
this synchronization information consists of 1 bit per frame, alternating between 0 and 1.

Statistical Time-Division Multiplexing:

In synchronous TDM, each input has a reserved slot in the output frame. This can be inefficient if
some input lines have no data to send. In statistical time-division multiplexing, slots are dynamically
allocated to improve bandwidth efficiency. Only when an input line has a slot's worth of data to send
is it given a slot in the output frame.
In statistical multiplexing, the number of slots in each frame is less than the number of input lines. The
multiplexer checks each input line in round robin fashion. It allocates a slot for an input line if the line
has data to send otherwise it skips the line and checks the next line.

The following figure shows a synchronous and a statistical TDM example. In the former, some slots
are empty because the corresponding line does not have data to send. In the latter, however, no slot is
left empty as long as there are data to be sent by any input line.
57

Addressing:
The above figure also shows a major difference between slots in synchronous TDM and statistical
TDM. An output slot in synchronous TDM is totally occupied by data, in statistical TDM, a slot needs
to carry data as well as the address of the destination.

In synchronous TDM, there is no need for addressing. Synchronization and preassigned relationships
between the inputs and outputs serve as an address. We know, for example, that input 1 always goes to
input 1. If the multiplexer and the demultiplexer are synchronized, this is guaranteed. In statistical
multiplexing, there is no fixed relationship between the inputs and outputs because there are no
preassigned or reserved slots. We need to include the address of the receiver inside each slot to show
where it is to be delivered.

The addressing in its simplest form can be n bits to define N different output lines with n =log 2 n. For
example, for eight different output lines, we need a 3-bit address.

Slot Size
Since a slot carries both data and an address in statistical TDM, the ratio of the data size to address
size must be reasonable to make transmission efficient. For example, it would be inefficient to send 1
bit per slot as data when the address is 3 bits. This would mean an overhead of 300 percent. In
statistical TDM, a block of data is usually many bytes while the address is just a few bytes.
No Synchronization Bit
There is another difference between synchronous and statistical TDM, but this time it is at the frame
level. The frames in statistical TDM need not be synchronized, so we do not need synchronization
bits.

Bandwidth
In statistical TDM, the capacity of the link is normally less than the sum of the capacities of each
channel. The designers of statistical TDM define the capacity of the link based on the statistics of the
load for each channel. If on average only x percent of the input slots are filled, the capacity of the link
reflects this. Of course, during peak times, some slots need to wait.
Wavelength-Division Multiplexing
Wavelength-division multiplexing (WDM) is designed to use the high-data-rate capability of fiber-
optic cable. The optical fiber data rate is higher than the data rate of metallic transmission cable. Using
58
a fiber-optic cable for one single line wastes the available bandwidth. Multiplexing allows us to
combine several lines into one.
WDM is conceptually the same as FDM, except that the multiplexing and demultiplexing involve
optical signals transmitted through fiber-optic channels.

The following figure gives a conceptual view of a WDM multiplexer and demultiplexer. Very narrow
bands of light from different sources are combined to make a wider band of light. At the receiver, the
signals are separated by the demultiplexer.

In this method, we combine multiple light sources into one single light at the multiplexer and do the
reverse at the demultiplexer. The combining and splitting of light sources are easily handled by a
prism. Recall from basic physics that a prism bends a beam of light based on the angle of incidence
and the frequency. Using this technique, a multiplexer can be made to combine several input beams of
light, each containing a narrow band of frequencies, into one output beam of a wider band of
frequencies. A demultiplexer can also be made to reverse the process.

TRANSMISSION MODES
Data is transmitted between two digital devices on the network in the form of bits. Transmission mode
refers to the mode used for transmitting the data. The transmission medium may be capable of
sending only a single bit in unit time or multiple bits in unit time.

When a single bit is transmitted in unit time the transmission mode used is Serial Transmission and
when multiple bits are sent in unit time the transmission mode used is called Parallel transmission.

Types of Transmission Modes:


There are two basic types of transmission modes (Serial and Parallel) as shown in the figure below.
Serial transmission is further categorized into Synchronous and Asynchronous Serial transmission.

Synchronous
Serial
Transmission
Transmission Asynchronous
Mode
Parallel
Transmission
59

Fig.1.1 Types of Transmission Modes

Parallel Transmission
It involves simultaneous transmission of N bits of N different channels. With the parallel transmission
of data all the bits making up a character are transmitted simultaneously over separate conductors. The
number of conductors required for a parallel interface is known as the bus width. Since several
conductors are necessary the parallel transmission system is only economic over fairly short distances.

Parallel Transmission increases transmission speed by a factor of N over serial transmission.


Disadvantage of parallel transmission is the cost involved, N channels have to be used, hence, it can be
used for short distance communication only.

N = 8 channels used, 8 bits


can be sent in unit time

R
S E
E C
E
N
I
D V
E E
R R

Fig.1.2 Parallel Transmission of Data over N = 8 channels

Example of Parallel Transmission is the communication between CPU and the Projector.

Serial Transmission
In Serial Transmission, as the name suggest data is transmitted serially, i.e. bit by bit, one bit at a time.
Since only one bit has to be sent in unit time only a single channel is required.

N = 8 channels used, 1 bits


can be sent in unit time

R
S E
E C
N E
D I
E V
R E
R

Fig.1.3 Serial Transmission of Data over N = 8 channels

Types of Serial Transmission:


Depending upon the timing of transmission of data, there are two types of serial transmission as
described below:
60
Asynchronous Transmission
In asynchronous serial transmission, the sender and receiver are not synchronized.
The data is sent in group of 8 bits i.e. in bytes.
The sender can start data transmission at any time instant without informing the receiver.
To avoid confusing the receiver while receiving the data, “start” and “stop” bits are inserted before
and after every group of 8 bits as shown below.

Character bits

Data bits

0 X X X X X X X X 1 0 X X X X X X X X 1 0
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0

START PARITY STOP X = 1 or 0


BIT BIT BIT
Fig.1.4 The structure of an asynchronous frame consists of four-key-bit components.
This frame has four components:
1) A starts bit: This component signals that a frame is starting and enables the receiving device
to synchronize itself with the message.
2) Data bits: This component consists of a group of seven or eight bits when character data is
being transmitted.
3) A parity bit: This component is optionally used as a crude method of detecting transmission
errors. This bit can be set to either binary 1 or 0 to ensure either even for even-parity or odd for
odd-parity.
4) A stop bit or bits: This component signals the end of the data frame.
Thus each character has a length of ten bits.
The start bit is indicated by “0” and stop bit is indicated by “1”.
The sender and receiver may not be synchronized as seen above but at the bit level they have to be
synchronized i.e. the duration of one bit needs to be same for both sender and receiver for accurate
data transmission. There may be gaps in between the data transmission indication that there is no data
being transmitted from sender. Ex. Assume a user typing at uneven speeds, at times there is no data
being transmitted from keyboard to the CPU.
Asynchronous transmission is a simple, inexpensive technology ideally suited for transmitting small
frames at irregular intervals. Because start, stop and parity bits must be added to each character being
transmitted, however, overhead for asynchronous transmission is high-often in the neighbourhood of
nearly 20 to 30 percent. This high overhead wastes bandwidth and makes asynchronous transmission
undesirable for transmitting large amounts of data.
Asynchronous transmission is frequently used for PC – to - PC and terminal-to-host communication.
Data in these environments is often of the busty, character – oriented nature that is ideal for
asynchronous communication. Asynchronous transmission generally requires less expensive hardwire
than synchronous transmission. Following fig 1.5 is the Diagram for Asynchronous Serial
Transmission.
61

Start bit Stop bit R


S E
E C
N E
0 1 BYTE 0 0 1 BYTE 0 I
D
E V
R E
Gap R

Fig:1.5 Asynchronous Serial Transmission


Advantages
1. Cheap and Effective implementation
2. Can be used for low speed communication
Disadvantages
Insertion of start bits, stop bits and gaps make asynchronous transmission slow.
Application
Keyboard

Synchronous Transmission
In Synchronous Serial Transmission, the sender and receiver are highly synchronized.
No start, stop bits are used. Instead a common master clock is used for reference.
Synchronous transmission refers to data transmission in which the time of occurrence of each signal
representing a bit is related to a fixed time frame. Synchronous transmission eliminates the need for
start and stop bits by synchronizing the clocks on the transmitting and receiving devices. This
synchronization is accomplished in two ways:
1. By transmitting synchronization signals with data. Some data encoding techniques, by
guaranteeing a signal transition with each bit transmitted, are inherently self-clocking.
2. By using a separate communication channel to carry clock signals. This technique can
function with any signal encoding technique.

Figure 1.6 below, illustrates the two possible structures of messages associated with synchronous
transmission.
a)
SYNCH SYNCH CHARACTER …. CHARACTER CRC END
b)
SYNC SYNC BINARY CR EN FIL SYNC SYNC DAT CR EN
H H DATA C D L H H A C D
MESSAG BIT
E S

Fig.1.6 Structures of synchronous transmission


Both synchronous transmission methods begin with a series of SYNCH signals, which notify the
receiver of the beginning of a frame.
SYNCH signals generally utilize a bit pattern that cannot appear elsewhere in messages, ensuring that
the signals always are distinct and easily recognizable by the receiver.
A wide variety of data types can be transmitted. Fig. above, illustrates both character-oriented and bit-
oriented data. Notice that under synchronous transmission, multiple characters or long series of bits
62
can be transmitter and receiver remain in synchronization for the duration of the transmission, frames
maybe very long.
When frames are long, parity is no longer a suitable method for detecting errors. If errors occur,
multiple bits are more likely to be affected, and parity 0 techniques are less likely to report an error. A
more appropriate error-control technique for synchronous transmission is the Cyclic Redundancy
Check (CRC).
In this technique, the transmitter uses an algorithm to calculate a CRC value that summarizes the
entire value of the data bits. This value is then appended to the data frame,(just like a check sum
value).
The receiver uses the same algorithm, recalculates the CRC, and compares the CRC, and compares the
CRC in the frame almost definitely was transmitted without error.
When synchronous transmission links are idle, communicating devices generally send FILL BITS to
the devices synchronized.

The sender simply send stream of data bits in group of 8 bits to the receiver without any start or stop
bit. It is the responsibility of the receiver to regroup the bits into units of 8 bits once they are received.
When no data is being transmitted, a sequence of 0’s and 1’s indicating IDLE is put on the
transmission medium by the sender.

R
S E
E C
N 10010010 11001010 11101101 10101101 E
D I
E V
E
R
R

Fig:1.7 Asynchronous Serial Transmission

Advantage
1. There are no start bits, stop bits or gaps between data units. The overhead bits (SYNCH, CRC,
and END) comprise a smaller portion (15%) of the overall data frame, which provides for more
efficient use of available bandwidth.
2. Synchronization improves error detection and enables the devices to operate at higher speeds.
.3. Due to synchronization, there are no timing errors.

Disadvantages
Synchronous transmission requires more complex circuitry for communication, which is more
expensive.
The choice between the two methods: asynchronous versus synchronous, must be based upon the
required speed of response, and telephone circuit costs.
Table 1.1Comparison of serial and parallel transmission
S/N Parameter Parallel Serial transmission
transmission
1 Number of wire required to N wire 1 wire
transmit N bits
2 Number of bits transmitted N bits 1 bit
simultaneously
63
3 Speed of data transfer False Slow
4 Application Higher due to more Low, since only one
number of wire is used
conductor
5 Short distance Long distance
communication such computer to
as computer to computer
printer communication
communication

Transmission Impairments & Types


Data is transmitted through transmission medium which are not perfect.
The imperfection causes signal impairment.
Due to the imperfection error is introduced in the transmitted data i.e. the original signal at the
beginning of the transmission is not the same as the signal at the Receiver.
There are three causes of impairment attenuation, distortion, and noise as shown below:

Transmission
Impairment

Attenuation Distortion Noise

Fig:1.8 Transmission Impairment Types


Attenuation
 Attenuation results in loss of energy. When a signal travels through a medium, it loses some of
its energy in overcoming the resistance of the medium.
 The electrical energy in the signal may be converted to heat.
 To compensate for this loss, amplifiers are used to amplify the signal. Figure below shows the
effect of attenuation and amplification.

Amplifi
er

At Attenuated At Receiver
Sender during after
transmission amplification

Fig.1.9 Attenuation
64
Distortion
Distortion changes the shape of the signal as shown below:

Original Signal Distorted Signal


Received after
transmission

Fig.1.10 Distortion
Noise
Noise is any unwanted signal that is mixed or combined with the original signal during transmission.
Due to noise the original signal is altered and signal received is not same as the one sent.

ERRORS, DETECTION & CORRECTION

INTRODUCTION
Errors in the data are basically caused due to the various impairments that occur during the process of
transmission. When there is an imperfect medium or environment exists in the transmission it prone to
errors in the original data.
Errors can be classified as follows: Attenuation, Noise and Distortion

TYPES OF ERRORS
If the signal comprises of binary data, there can be two types of errors which are possible during the
transmission.
1. Single bit errors
2. Burst Errors

Transmission Errors

Single-bit
Burst Errors
Errors

Figure2.1 Transmission errors

1. Single-bit errors:
In single-bit error, a bit value of 0 changes to bit value 1 or vice versa. Single bit errors are
more likely to occur in parallel transmission. Figure below(a)
2. Burst errors:
In Burst error, multiple bits of the binary value changes. Burst error can change any two or
more bits in a transmission. These bits need not be adjacent bits. Burst errors are more likely to occur
in serial transmission. Figure below (b)
65

Original data Received data

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

Figure (a) Single bit error

Original data Received data

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

Figure(b) Burst error


REDUNDANCY
In order to detect and correct the errors in the data communication we add some extra bits to the
original data. These extra bits are nothing but the redundant bits which will be removed by the
receiver after receiving the data.
Their presence allows the receiver to detect or correct corrupted bits. Instead of repeating the entire
data stream, a short group of bits may be attached to the entire data stream. This technique is called
redundancy because the extra bits are redundant to the information, they are discarded as soon as the
accuracy of the transmission has been determined.
Receiver Sender

0 1 1 0 1 1 0 0 1 1 0 1 1 0
Data Data

Rejecte Yes
Data
d
Ok?
No

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

Data and Redundancy Data and Redundancy

Medium

Figure 2.3

There are different techniques used for transmission error detection and correction.
66
Detection
methods

Parity Cyclic Redundancy Checksum


Check Check

Figure 2.4

2. Parity Check
In this technique, a redundant bit called a parity bit is added to every data unit so that the total
number of 1’s in the unit (including the parity bit) becomes even (or odd).

Figure below shows this concept when transmitting the binary data: 100101.
Receiver Sender

Drop parity bit and accept the


data
1 0 0 1 0 1
Data
Yes
Rejecte
Data
d Even
?
No Calculate
Parity bit

Calculate
Parity bit

1 0 01 0 1 1
Bits

Medium

Simple parity check can detect all single-bit errors. It can also detect burst errors as long as the total
number of bits changed is odd. This method cannot detect errors where the total number of bits
changed is even.
Two-Dimensional Parity Check
A better approach is the two dimensional parity checks. In this method, a block of bits is
organized in a table (rows and columns). First we calculate the parity bit for each data unit. Then we
organize them into a table. We then calculate the parity bit for each column and create a new row of 8
bits.
Consider the following example, we have four data units to send. They are organized in the tabular
form as shown below:
67
Original Data

0110110 1101001 1110011 0001110

0 1 1 0 1 1 0 0 Row
0 parities
0 1 1 0 1 1 0
0 `0
0 1 1 0 1 1 0 1
0
0 1 1 0 1 1 0 1
0
Data and Parity bits
Column
0 1 1 0 1 1 0 parities
0

01101100 11010010 11100111 00011101 01000100

We then calculate the parity bit for each column and create a new row of 8 bits; they are the parity bits
for the whole block. Note that the first parity bit in the fifth row is calculated based on all first bits;
the second parity bit is calculated based on all second bits; and so on. We then attach the 8 parity bits
to the original data and send them to the receiver.

Two-dimensional parity check increases the likelihood of detecting burst errors. A burst error
of more than `n’ bits is also detected by this method with a very high probability.

3. Cyclic Redundancy Check (CRC)


Most powerful of the redundancy checking techniques is the cyclic redundancy check (CRC).
This method is based on the binary division. In CRC, the desired sequence of redundant bits are
generated and is appended to the end of data unit. It is also called as CRC reminder. So that the
resulting data unit becomes exactly divisible by a predetermined binary number. At its destination, the
incoming data unit is divided by the same number. If at this step there is no remainder then the data
unit is assumed to be correct and is therefore accepted. A remainder indicates that the data unit has
been damaged in transit and therefore must be rejected. The redundancy bits used by CRC are derived
by dividing the data unit by a predetermined divisor; the remainder is the CRC. To be valid, a CRC
must have two qualities: it must have exactly one less bit than the divisor, and appending it to the end
of the data string must make the resulting bit sequence exactly divisible by the divisor.

The following figure shows the process:


68
Receiver Sender

Data CRC Data 00…0


n bits

Divisor Divisor n + 1 bits

Data CRC
remainder

Remainder CRC
Zero, accept
Non-zero, Reject n
bits

Step 1: A string of 0’s is appended to the data unit. It is n bits long. The number n is 1 less if-number
of bits in the predetermined divisor which is n + 1 bits.
Step 2: The newly generated data unit is divided by the divisor, using a process known as binary
division. The remainder resulting from this division is the CRC.
Step 3: The CRC of n bits derived in step 2 replaces the appended 0’s at the data unit. Note that the
CRC may consist of all 0’s.
The data unit arrives at the receiver data first, following by the CRC. The receiver treats the whole
string as a unit and divides it by the same divisor that was used the CRC remainder. If the string
arrives without error, the CRC checker yields a remainder of zero, the data unit passes. If the string
has been changed in transit, the division yields zero remainder and the data unit does not pass.

Cyclic Redundancy Check (CRC) Technique


CRC refers to an error detection technique in which the code is the remainder resulting from dividing
the bits to be checked by a predetermined binary number. In transmission the data block or message is
thought of as a stream of serial data bits. The bits in this n-bit block are considered the coefficients of
a characteristic polynomial M (x).
M(x) = bn X0+ bn-1 X + bn-2 X2 + … + b1 Xn-1 + b0 Xn. Where, b0 is the LSB and bn is the MSB.

Example 1: Calculate the data polynomial M (x) for the 16-bit data stream 26F0H.
Solution: First representing 26F0H in binary becomes 0010 0110 1111 00002

Now write this as M(x):


M(x) = 0 + 0X1 + 1X2 + 0X3 + 0X4 +1X5 + 1X6 + 0X7 + 1X8 + 1X9 + 1X10 + 1X11 + 0X12 + 0X13 +
0X14 + 0X15 and eliminating the zero terms given:
M(x) = X2 + X5 + X6 + X8 + X9 + X10 + X11………………………(1)

Equation 1 is a unique polynomial representing the data in the 16-bit block. If one or more of the data
bits were to be changed the polynomial would also change.
The CRC is therefore given by the formulae; CRC = M(x) *Xn / G(x) = Q(x) + R(x).
Or CRC = M(x) * 2n / G(x)………………………………………(2)
Where, M(x)- is a k-bit number;
R(x) – an n-bit number such that k is greater than n; G(x) – is an (n+1) -bit number.
69
In equation 2, G(x) is called the generator polynomial. For the Bisync protocols G(x) = X + X + X2
16 15

+ 1. Similarly, for SDLC (Synchronous Data link Control) protocol: G(x) = X16 + X13 + X5 + 1. When
the division is performed, the result will be a quotient Q(x) and a remainder R(x). The CRC technique
consists of calculating R(x) for the data stream and appending the result to the data block. With
modulo-2 division of one binary number by another the rules for performing the division is as follows:
(a) If the divisor has the same number of bits as the dividend, the quotient is 1; if the divisor has
fewer bits than the dividend the quotient is 0.
(b) There are no carries and 1 – 1 =0, 0 - 0 =0, 1 – 0 = 1 and 0 – 1 = 1.
Since the division is binary any remainder will always be one bit shorter than the divisor.
Following figure shows the process of generating CRC reminder:
Figure:
Quotient
1 1 1 1 0 1
Divisor 1101 1 0 0 1 0 0 0 0 0 Extra bits
1 1 0 1

1 0 0 0
1 1 0 1

1 0 1 0
1 1 0 1

1 1 1 0
1 1 0 1

0 1 1 0
0 0 0 0

1 1 0 0
1 1 0 1

0 0 1 Remainder
A CRC checker functions does exactly as the generator does. After receiving the data appended with
the CRC, it does the same modulo-2 division. If the remainder is all 0’s, the CRC is dropped and the
data is accepted, otherwise, the received stream of bits is discarded and data is resent.
Following Figure shows the same process of division in the receiver.

Figure:
70
Quotient
1 1 1 1 0 1
Divisor 1101 1 0 0 1 0 0 0 0 1 CRC
1 1 0 1

1 0 0 0
1 1 0 1

1 0 1 0
1 1 0 1

1 1 1 0
1 1 0 1

0 1 1 0
0 0 0 0

1 1 0 1
1 1 0 1

0 0 0 0 Result

Solved Example: Given a divisor as X4 + X3 + 1 or G(x) =11001 and a data stream of M(x)
=1010110101. Find the CRC?

Solution:

Performance:
CRC is a very effective error detection method. If the divisor is chosen according to the previously
mentioned rules:
1. CRC can detect all burst errors that affect an odd number of bits.
2. CRC can detect all burst errors of length less than or equal to the degree of the polynomial.
3. CRC can detect, with a very probability, burst error of length greater than the degree of the
polynomial.

3. Checksum
A checksum is fixed length data that is the result of performing certain operations on the data
to be sent sender to the receiver. The sender runs the appropriate checksum algorithm to compute the
checksum of the data, appends it as a field in the packet that contains the data to be sent, as well as
various headers.
Example: Calculate the checksum byte for the following four hex data bytes: 10, 23, 45, and 04.
71
Solution;
The sum is calculated first as 10 + 23+45+ 04 =7C, which is 0111 1100, then invert and add 1 to the
LSB (that is , forming 2”s complement code of 7C) giving 84H.

When the receiver receives the data, the receiver runs the same checksum algorithm to compute a
fresh checksum. The receiver compares this freshly computed checksum with the checksum that was
computed by the sender. If the two checksum matches, the receiver of the data is assured that the data
has not changed during the transit.

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