IT203 Data Communication: Module II 2021

Module – II
Syllabus:
Data and signals, Analog Signals, Digital Signals - Transmission Impairments, Data Rate Limits: Channel
Capacity, Nyquist Bit Rate, Shannon Capacity, Performance parameters - Bandwidth, Throughput, Delay
& Jitter.
Digital-To-Digital Conversion: Line Coding Schemes: Unipolar, Polar, Bipolar - Block Coding,
Scrambling, Analog-To-Digital Conversion: Pulse Code Modulation, Delta Modulation - Digital-To-
Analog Conversion: ASK, FSK, PSK.

INTRODUCTION TO DATA COMMUNICATION

Communication model Simplex, half duplex and full duplex transmission.

DATA COMMUNICATION
• Data Communication is a process of exchanging data or information
• In case of computer networks this exchange is done between two devices over a
transmission medium.
• This process involves a communication system which is made up of hardware and
software. The hardware part involves the sender and receiver devices and the
intermediate devices through which the data passes. The software part involves certain
rules which specify what is to be communicated, how it is to be communicated and
when. It is also called as a Protocol.

Characteristics of Data Communication

The effectiveness of any data communications system depends upon the following four
fundamental characteristics:
1. Delivery: The data should be delivered to the correct destination and correct user.
2. Accuracy: The communication system should deliver the data accurately, without
introducing any errors. The data may get corrupted during transmission affecting the
accuracy of the delivered data.
3. Timeliness: Audio and Video data has to be delivered in a timely manner without any
delay; such a data delivery is called real time transmission of data.

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4. Jitter: It is the variation in the packet arrival time. Uneven Jitter may affect the
timeliness of data being transmitted

Components of Data Communication

1. Message: Message is the information to be communicated by the sender to

the receiver.
2. Sender: The sender is any device that is capable of sending the data (message).
3. Receiver: The receiver is a device that the sender wants to communicate the data (message).
4. Transmission Medium: It is the path by which the message travels from sender to receiver. It
can be wired or wireless and many subtypes in both.
5. Protocol: It is an agreed upon set or rules used by the sender and receiver to communicate data.
A protocol is a set of rules that governs data communication. A Protocol is a necessity in data
communications without which the communicating entities are like two persons trying to talk
to each other in a different language without know the other language.

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Communication model
Two devices communicate with each other by sending and receiving data. The data can
flow between the two devices in the following ways.
1. Simplex
2. Half Duplex

3. Full Duplex
(1) Simplex
In Simplex, communication is unidirectional
Only one of the devices sends the data and the other one only receives the data.
Example: in the above diagram: a CPU sends data while a monitor only receives data.

(2) Half Duplex

In half duplex both the stations can transmit as well as receive but not at the same time.
When one device is sending other can only receive and vice-versa (as shown in figure
above.)
Example: A walkie-talkie.

(3) Full Duplex

In Full duplex mode, both stations can transmit and receive at the same time.

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Example: mobile phones

Review Questions: -

1. Differentiate between data & information. What are the different forms in
which data can be represented?
2. What are the characteristics of data communication?
3. What are the components of a data communication system?

Analog & Digital data and signals

Time Domain and Frequency Domain concepts
• Computer networks are designed to transfer data from one point to another.
• During transit data is in the form of electromagnetic signals.
• Hence it is important to study data and signals before we move to further concepts in data
communication.

DATA & SIGNALS

To be transmitted, data must be transformed to electromagnetic signals.

1. Data can be Analog or Digital.
a) Analog data refers to information that is continuous; ex. sounds made
by a human voice
b) Digital data refers to information that has discrete states. Digital data take
on discrete values.
c) For example, data are stored in computer memory in the form of 0s and 1s

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2. Signals can be of two types:

a) Analog Signal: They have infinite values in a range.
b) Digital Signal: They have limited number of defined values

3. Periodic & Non Periodic Signals

➢ Signals which repeat itself after a fixed time period are called Periodic Signals.
➢ Signals which do not repeat itself after a fixed time period are called
Non-Periodic Signals.
➢ In data communications, we commonly use periodic analog signals and
non-periodic digital signals.

ANALOG SIGNAL

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 as it can be seen in the figure
below.
A simple analog signal is a sine wave that cannot be further decomposed
into simpler signals.

Figure: Sine wave

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A sine wave is characterized by three parameters:
1. Peak Amplitude
2. Frequency

3. Phase

Characteristics of an Analog Signal

(1) Peak Amplitude

• The amplitude of a signal is the absolute value of its intensity at time t
• The peak amplitude of a signal is the absolute value of the highest intensity.
• The amplitude of a signal is proportional to the energy carried by the signal

(1) Frequency
Frequency refers to the number of cycles completed by the wave in one second.
Period refers to the time taken by the wave to complete one second.

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(2) Phase
Phase describes the position of the waveform with respect to time (specifically
relative to time 0).
Phase indicates the forward or backward shift of the waveform from the axis
It is measured in degrees or radian
The figure above shows the sine waves with same amplitude and frequency but
different phases

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Relation between Frequency & Period

Frequency & Period are inverse of each other.

It is indicated by the following formula:

Example1. A wave has a frequency of 100hz. Its period(T) is given by T = 1/ F = 1/ 100 = 0.01
sec Example2. A wave completes its one cycle in 0.25 seconds. Its frequency is given by F = 1 /
T = 1 / 0.25 = 4 Hz.

WAVELENGTH
The wavelength of a signal refers to the relationship between frequency (or period) and
propagation speed of the wave through a medium. The wavelength is the distance a signal travels
in one period. It is given by
Wavelength = Propagation Speed X Period
OR
Wavelength =Propagation Speed X (1/ Frequency )

It is represented by the symbol : λ (pronounced as lamda)

It is measured in micrometers
It varies from one medium to another.

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Time Domain and Frequency domain representation of signals

A sine wave can be represented either in the time domain or frequency domain.
The time-domain plot shows changes in signal amplitude with respect
to time. It indicates time and amplitude relation of a signal.
The frequency-domain plot shows signal frequency and peak amplitude.
The figure below show time and frequency domain plots of three sine waves.
A complete sine wave in the time domain can be represented by one single

spike in the frequency domain

Composite Signal

A composite signal is a combination of two or more simple sine waves

with different frequency, phase and amplitude.

If the composite signal is periodic, the decomposition gives a series of signals

with discrete frequencies; if the composite signal is non-periodic, the

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decomposition gives a combination of sine waves with continuous frequencies.
For data communication a simple sine wave is not useful, what is used is a
composite signal which is a combination of many simple sine waves.
According to French Mathematician, Jean Baptist, any composite signal is a
combination of simple sine waves with different amplitudes and frequencies
and phases.
Composite signals can be periodic or non-periodic.
A periodic composite signal can be decomposed into a series of signals
with discrete frequencies.
A non-periodic signal when decomposed gives a combination of sine waves
with continuous frequencies.

The time and frequency domains of a non-periodic composite analog signal

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DIGITAL SIGNAL
Information can also be explained in the form of a digital signal.
A digital signal is a signal that has discrete values.
The signal will have value that is not continuous.

Voltage Level

Information in a digital signal can be represented in the form of voltage levels.

Ex. In the signal shown below, a ‘1’ is represented by a positive voltage
and a ‘0’ is represented by a Zero voltage.
A digital signal with two levels. ‘1’ represented by a positive voltage and ‘0’
represented by a negative voltage is shown below.

A Signal can have more than two levels

A digital signal with four levels is given below
In general, if a signal has L levels then, each level need Log2L bits

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Example: Consider a digital Signal with four levels, how many bits are required
per level?
Answer: Number of bits per level = Log2L = Log24 = 2 Hence, 2 bits are required
per level for a signal with four levels.

Bit Length or Bit Interval (Tb)

It is the time required to send one bit.
It is measured in seconds.

Bit Rate
It is the number of bits transmitted in one second.
It is expressed as bits per second (bps).

Relation between bit rate and bit interval can be as follows

Bit rate = 1 / Bit interval

Baud Rate
It is the rate of Signal Speed, i.e the rate at which the signal changes.
A digital signal with two levels ‘0‘ & ‘1‘ will have the same baud rate and bit rate & bit rate.
The diagram below shows three signal of period (T) 1 second
a) Signal with a bit rate of 8 bits/ sec and baud rate of 8 baud/sec
b) Signal with a bit rate of 16 bits/ sec and baud rate of 8 baud/sec
c) Signal with a bit rate of 16 bits/ sec and baud rate of 4 baud/sec

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TYPES OF CHANNELS
Each composite signal has a lowest possible (minimum) frequency and a highest
possible (maximum) frequency. From the point of view of transmission, there are two types
of channels:
a) Low pass Channel

b) Band pass channel

Low pass Channel
This channel has the lowest frequency as ‘0‘and highest frequency as some non-
zero frequency ’f1‘.
This channel can pass all the frequencies in the range 0 to f1.
Band pass channel
This channel has the lowest frequency as some non-zero frequency ‘f1‘ and
highest frequency as some non-zero frequency ‘f2‘.
This channel can pass all the frequencies in the range f1 to f2.

Transmission of Digital signal

Digital signal can be transmitted in the following two ways:
• Baseband Transmission
• Broad band Transmission

Baseband Transmission
The signal is transmitted without making any change to it (ie. Without modulation)
In baseband transmission, the bandwidth of the signal to be transmitted has to be less

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than the bandwidth of the channel.
Ex. Consider a Baseband channel with lower frequency 0Hz and higher frequency
100Hz, hence its bandwidth is 100 (Bandwidth is calculated by getting the difference
between the highest and lowest frequency).
We can easily transmit a signal with frequency below 100Hz, such a channel whose
bandwidth is more than the bandwidth of the signal is called Wideband channel
Logically a signal with frequency say 120Hz will be blocked resulting in loss of
information, such a channel whose bandwidth is less than the bandwidth of the signal is
called Narrowband channel

Broad band Transmission

➢ Given a bandpass channel, a digital signal cannot be transmitted directly through it
➢ In broadband transmission we use modulation, i.e we change the signal to analog signal
before transmitting it.
➢ The digital signal is first converted to an analog signal, since we have a bandpass channel
we cannot directly send this signal through the available channel. Ex. Consider the
bandpass channel with lower frequency 50Hz and higher frequency 80Hz, and the signal
to be transmitted has frequency 10Hz.
➢ To pass the analog signal through the bandpass channel, the signal is modulated using a
carrier frequency. Ex. The analog signal (10Hz) is modulated by a carrier frequency of
50Hz resulting in an signal of frequency 60Hz which can pass through our bandpass
channel.
➢ The signal is demodulated and again converted into an digital signal at the other end
as shown in the figure below.

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Fig: Broadband Transmission Involving Modulation & Demodulation

REVIEW QUESTIONS

1. Define analog and digital signals

2. Explain Composite analog signals.
3. Explain Time and Frequency Domain Representation of signals
4. Explain the characteristics of an Analog signal
5. Explain the characteristics of an Digital signal
6. Explain the difference between
Low pass and Band pass
channel
Narrowband and wideband channel
7. Explain why a digital signal requires to undergo a change before transmitting it
through a band pass channel.

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Transmission Impairments - Attenuation, Delay distortion, Noise -

Different types of noise
When signals travel through imperfect transmission media, 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 impairment of signals is mainly
caused due to attenuation, distortion, and noise in the medium.

➢ 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:

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.

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The decibel (dB) measures the relative strengths of two signals or one signal at two different points.
Note that the decibel is negative if a signal is attenuated and positive if a signal is amplified.
Variables PI and P2 are the powers of a signal at points 1 and 2, respectively

Example 1: Suppose a signal travels through a transmission medium and its power is reduced to
one-half. This means that P2 = ~ PI' In this case, the attenuation (loss of power) can be calculated
as

A loss of 3 dB (-3 dB) is equivalent to losing one-half the power.

Example 2
A signal travels through an amplifier, and its power is increased 10 times. This means that
P2=1OP1 In this case, the amplification (gain of power) can be calculated as

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. Figure below
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shows the effect of distortion on a composite signal.
Distortion changes the shape of the signal as shown below

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 ascending 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.

➢ 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.

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Different types of noise

Noise is another cause of impairment. Several types of noise, such as
➢ thermal noise
➢ induced noise
➢ crosstalk
➢ 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 act 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.

REVIEW QUESTIONS

1. Write a short note on transmission impairments

2. What are the different types of noises

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Channel capacity -Shannon's Theorem

The maximum rate at which data can be transmitted over a communication channel under
given conditions is referred as the channel capacity.
Four important parameters are related in the evaluation of channel capacity.
➢ Data rate: The rate at which data can be transmitted. Measured in bits per second
➢ Bandwidth: The bandwidth of the transmitted signal. Measured in cycles per second
(Hz).
➢ Noise: The average level of unwanted signals over communication path. Expressed as the
ratio between signal and noise.
➢ Error rate: The rate at which error can occur.

Data rate depends on three factors:

1. The bandwidth available
2. The level of the signals we use
3. The quality of the channel (the level of noise)

The quality of the channel indicates two types:

a) A Noiseless or Perfect Channel
➢ An ideal channel with no noise.
➢ The Nyquist Bit rate derived by Henry Nyquist gives the bit rate for a Noiseless
Channel.

b) A Noisy Channel
➢ A realistic channel that has some noise.
➢ The Shannon Capacity formulated by Claude Shannon gives the bit rate for a
Noisy Channel

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Nyquist Bit Rate

The Nyquist bit rate formula defines the theoretical maximum bit rate for a noiseless channel

Where,

Bitrate is the bitrate of the channel in bits per second

Bandwidth is the bandwidth of the channel
L is the number of signal levels.

Example What is the maximum bit rate of a noiseless channel with a bandwidth of 5000 Hz
transmitting a signal with two signal levels.
Solution: The bit rate for a noiseless channel according to Nyquist Bit rate can be calculated as
follows: BitRate = 2 x Bandwidth x Log2 L = 2 x 5000 x log2 2 =10000 bps
Example: Consider a noiseless channel with a bandwidth of 3000 Hz transmitting a signal with
two signal levels.
The maximum bit rate can be calculated as
Bit Rate =2 x 3000 x log2 2 =6000 bps

Shannon Capacity
The Channel capacity (in cycles per second) is given by Shannon’s theorem. The Shannon
Capacity defines the theoretical maximum bit rate for a noisy channel

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Where,

Capacity is the capacity of the channel in bits per second

Bandwidth is the bandwidth of the channel in Hz
SNR is the Signal to Noise Ratio (SNR = 10 log10 (Signal power/Noise power))

Here the SNR ratio assumes only white noise (thermal noise) where as other noise like impulse
noise, attenuation noise and delay noise are not taken into account. Hence the actual value of channel
capacity differs from the estimated value

Shannon Capacity for calculating the maximum bit rate for a noisy channel does not consider the
number of levels of the signals being transmitted as done in the Nyquist bit rate.
Example: Calculate the bit rate for a noisy channel with SNR 300 and bandwidth of 3000Hz

Solution: The bit rate for a noisy channel according to Shannon Capacity can be calculated as
follows: Capacity=bandwidth X log2 (1 +SNR) = 3000 x log2 (1 + 300) = 3000 x log2 ( 301) =
3000 x 8.23 = 24,690bps

REVIEW QUESTIONS

1. Explain the term bandwidth of a signal

2. Explain the term bandwidth of a channel.
3. Write short note on maximum data rate of a channel.

DIFFERENT CRITERIA FOR NETWORK PERFORMANCE

Performance of a network pertains to the measure of service quality of a network as perceived by the
user. There are different ways to measure the performance of a network, depending upon the nature
and design of the network. The characteristics that measure the performance of a network are:

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• Bandwidth
• Throughput
• Latency (Delay)
• Bandwidth – Delay Product
• Jitter

Bandwidth
Bandwidth is characterized as the measure of data or information that can be transmitted in a
fixed measure of time. The term can be used in two different contexts with two distinctive estimating
values. In the case of digital devices, the bandwidth is measured in bits per second(bps) or bytes per
second. In the case of analogue devices, the bandwidth is measured in cycles per second, or Hertz (Hz).
“Bandwidth” means “Capacity” and “Speed” means “Transfer rate”.

Bandwidth in Hertz: It is the range of frequencies contained in a composite signal or the range of
frequencies a channel can pass. For example, let us consider the bandwidth of a subscriber telephone
line as 4 kHz.

Bandwidth in Bits per Seconds: It refers to the number of bits per second that a channel, a link, or
rather a network can transmit. For example, we can say the bandwidth of a Fast Ethernet network is a
maximum of 100 Mbps, which means that the network can send 100 Mbps of data.

Relationship
There is an explicit relationship between the bandwidth in hertz and bandwidth in bits per seconds.
Basically, an increase in bandwidth in hertz means an increase in bandwidth in bits per second. The
relationship depends on whether we have baseband transmission or transmission with modulation.

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Throughput
Throughput is the number of messages successfully transmitted per unit time. It is controlled by
available bandwidth, the available signal-to-noise ratio and hardware limitations. The terms
‘throughput’ and ‘bandwidth’ are often thought of as the same, yet they are different. Bandwidth is the
potential measurement of a link, whereas throughput is an actual measurement of how fast we can send
data.
Throughput is a measure of how fast we can actually send data through a network. It is usually
measured in the unit of bits per second(bps), which has evolved to bytes per second(Bps), kilobytes
per second(KBps), megabytes per second(MBps) and gigabytes per second(GBps). For example, we
may have a link with a bandwidth of 1 Mbps, but the devices connected to the end of the link may
handle only 200 kbps. This means that we cannot send more than 200 kbps through this link.

Latency (Delay)
The latency or delay defines how long it takes for an entire message to completely arrive at the
destination from the time the first bit is sent out from the source. We can say that latency is made of
four components: propagation time, transmission time, queuing time and processing delay.
Latency =propagation time +transmission time +queuing time + processing delay

Propagation Time
Propagation time measures the time required for a bit to travel from the source to the destination. The
propagation time is calculated by dividing the distance by the propagation speed.
Propagation time = Distance /Propagation speed
The propagation speed of electromagnetic signals depends on the medium and on the frequency of
the signal. For example, in a vacuum, light is propagated with a speed of 3x108mfs. It is lower in air;
it is much lower in cable.

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Transmission time
In data communications we don't send just 1 bit, we send a message. The first bit may take a time equal
to the propagation time to reach its destination; the last bit also may take the same amount of time.
However, there is a time between the first bit leaving the sender and the last bit arriving at the receiver.
The first bit leaves earlier and arrives earlier; the last bit leaves later and arrives later. The time required
for transmission of a message depends on the size of the message and the bandwidth of the channel.
Transmission time =Message size / Bandwidth

Queuing Time
The third component in latency is the queuing time, the time needed for each intermediate or end device
to hold the message before it can be processed. The queuing time is not a fixed factor; it changes with
the load imposed on the network. When there is heavy traffic on the network, the queuing time
increases. An intermediate device, such as a router, queues the arrived messages and processes them
one by one. If there are many messages, each message will have to wait.
Bandwidth – Delay Product
Bandwidth and delay are two performance measurements of a link.
The bandwidth-delay product defines the number of buts that can fill the links

Case 1: Assume a link is of bandwidth 1bps and the delay of the link is 5s. Let us find the bandwidth-
delay product in this case. From the image, we can say that this product 1 x 5 is the maximum number
of bits that can fill the link. There can be close to 5 bits at any time on the link.

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Case 2: Assume a link is of bandwidth 3bps. From the image, we can say that there can be a maximum
of 3 x 5 = 15 bits on the line. The reason is that, at each second, there are 3 bits on the line and the
duration of each bit is 0.33s.

For both examples, the product of bandwidth and delay is the number of bits that can fill the link. This
estimation is significant in the event that we have to send data in bursts and wait for the
acknowledgement of each burst before sending the following one. To utilize the maximum ability of
the link, we have to make the size of our burst twice the product of bandwidth and delay. Also, we
need to fill up the full-duplex channel. The sender ought to send a burst of data of (2*bandwidth*delay)
bits. The sender at that point waits for the receiver’s acknowledgement for part of the burst before
sending another burst.
The amount: 2*bandwidth*delay is the number of bits that can be in transition at any time.

Jitter
Another performance issue that is related to delay is jitter. Jitter is a problem if different packets of
data encounter different delays and the application using the data at the receiver site is time- sensitive
(audio and video data, for example). If the delay for the first packet is 20 ms, for the second is 45 ms,
and for the third is 40 ms, then the real-time application that uses the packets endures jitter.

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Sampling theorem, Encoding digital data into digital signal

“A band limited signal can be reconstructed exactly if it is sampled at a rate at least twice
the maximum frequency component in it.”

(Spectrum of bandlimited signal g(t))

The Nyquist sampling theorem provides a prescription for the nominal sampling interval
required to avoid aliasing. It may be stated simply as follows:

“The sampling frequency should be at least twice the highest frequency contained in
the signal.”
Or in mathematical terms: fs ≥ 2 fc

Where fs is the sampling frequency (how often samples are taken per unit of time or space), and
fc is the highest frequency contained in the signal.

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INTRODUCTION TO SIGNAL ENCODING

Data can be analog or digital, so can be the signal that represents it.
Signal encoding is the conversion from analog/digital data to analog / digital signal.

In the Figure above,

A) Demonstrates Digital Signaling where data from an analog/digital source is encoded into
Digital Signal
B) Demonstrates Analog signaling in which the analog/digital source modulates a continuous
carrier signal to produce an analog signal.

The possible encodings are:

1. Digital data to Digital Signal
2. Digital data to Analog Signal
3. Analog data to Digital Signal
4. Analog data to Analog Signal

SYNCHRONIZATION
In order to receive the signals correctly, the receivers bit intervals must correspond exactly to
the senders bit intervals.
The clock frequency of the transmitter and receiver should be the same.
If the clock frequency at the receiver is slower or faster than the bit intervals are not matched
and the received signal is different than the transmitted one.

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In the above figure, the receiver clock frequency is twice that of the transmitter frequency.
Hence the received data is totally different than the transmitted one
To avoid this, receiver and transmitter clocks have to be synchronized.
To achieve this the transmitted digital signal should include timing information which forces
synchronization

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DIGITAL DATA TO DIGITAL SIGNAL
Data can be either digital or analog. We also said that signals that represent data can also
be digital or analog. In this section, we can represent digital data by using digital signals.

Figure: D2D Conversion

Coding methods are used to convert digital data into digital signals. There are three
types of coding methods:
1. Line Coding
2. Block Coding
3. Scrambling

Line Coding/Encoding

➢ It is the process of converting Digital data into digital signal. In other words, it is
converting of binary data (i.e. A sequence of bits) into digital signal (i.e. a sequence of
discrete, discontinuous voltage pulses)
➢ Line coding is the process of converting digital data to digital signals.
➢ Data are, in the form of text, numbers, graphical images, audio, or video, are stored in
computer memory as sequences of bits.
➢ Line coding converts a sequence of bits to a digital signal.
➢ At the sender, digital data are encoded into a digital signal, at the receiver, the digital
data are recreated by decoding the digital signal.

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Characteristics of Line Codes

SIGNAL ELEMENT VERSUS DATA ELEMENT

➢ A data element is the smallest entity that can represent a piece of information: this is the bit. In
digital data communications, a signal element carries data elements.
➢ A signal element is the shortest unit (time wise) of a digital signal.
➢ In other words, data elements are what we need to send, signal elements are what we can send.
Data elements are being carried signal elements are the carriers.
➢ We define here a ratio r which is the number of data elements carried by each signal element.
Figure shows several situations with different values of r.

In part a of the figure, one data element is carried by one signal element (r = 1). In part b of the figure, we
need two signal elements (two transitions) to carry each data element (r = 1/2 ), In part c of the figure, a
signal element carries two data elements (r = 2).
Finally, in part d, a group of 4 bits is being carried by a group of three signal elements (r = 4/3)

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DATA RATE VERSUS SIGNAL RATE

The data rate defines the number of data elements (bits) sent in 1s. The unit is bits per second (bps).
The signal rate is the number of signal elements sent in 1s. The unit is the baud. There are several
common terminologies used in the literature. The data rate is sometimes called the bit rate, the signal
rate is sometimes called the pulse rate, the modulation rate, or the baud rate. One goal in data
communications is to increase the data rate while decreasing the signal rate. Increasing the data rate
increases the speed of transmission; decreasing the signal rate decreases the bandwidth requirement. In
our vehicle-people analogy, we need to carry more people in fewer vehicles to prevent traffic jams. We
have a limited bandwidth in our transportation system.
We can formulate the relationship between data rate and signal rate as

Where, N is the data rate (in bps); c is the case factor, which varies for each case; S is the number of
signal elements; and r is the previously defined factor.

Bandwidth: the digital signal that carries information is non periodic. We also showed that the
bandwidth of a non-periodic signal is continuous with an infinite range. However, most digital signals
we encounter in real life have a bandwidth with finite values. In other words, the bandwidth is
theoretically infinite, but many of the components have such small amplitude that they can be ignored.
The effective bandwidth is finite.

Baseline Wandering: In decoding a digital signal, the receiver calculates a running average of the
received signal power. This average is called the baseline. The incoming signal power is evaluated
against this baseline to determine the value of the data element. A long string of 0s or 1s can cause a
drift in the baseline (baseline wandering) and make it difficult for the receiver to decode correctly. A
good line coding scheme needs to prevent baseline wandering.

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DC Components: When the voltage level in a digital signal is constant for a while, The spectrum creates
very low frequencies (results of Fourier analysis). These frequencies around zero, called DC (direct-
current) components, present of this component are problems for a system that cannot pass low
frequencies or a system that uses electrical coupling (via a transformer). For example, a telephone line
cannot pass frequencies below 200 Hz. For these systems, we need a scheme with no DC component.

Self-synchronization: To correctly interpret the signals received from the sender, the receiver's bit
intervals must correspond exactly to the sender's bit intervals. If the receiver clock is faster or slower,
the bit intervals are not matched and the receiver might misinterpret the signals. A self-synchronizing
digital signal includes timing information in the data being transmitted. This can be achieved if there are
transitions in the signal that alert the receiver to the beginning, middle, or end of the pulse. If the
receiver's clock is out of synchronization, these points can reset the clock. In a figure shows the situation
in which the receiver has shorter bit duration. The sender sends 10110001, while the receiver receives
110111000011.

Built-in Error Detection: It is desirable to have a built-in error-detecting capability in the generated
code to detect some of or all the errors that occurred during transmission. Some encoding schemes have
this capability to some extent.

Immunity to Noise and Interference: Another desirable code characteristic is a code I that is immune
to noise and other interferences. Some encoding schemes have this capability.

Complexity: A complex scheme is more costly to implement than a simple one. For example, a scheme
that uses four signal levels is more difficult to interpret than one that uses only two levels.

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Classification of Line Coding

The following figure shows the classification of Line coding schemes:

UNIPOLAR
All signal levels are either above or below the time axis.
NRZ - Non Return to Zero scheme is an example of this code. The signal level does not
return to zero during a symbol transmission.

NRZ (Non-Return-to-Zero) Traditionally, a unipolar scheme was designed as a non-return-to- zero

(NRZ) scheme in which the positive voltage defines bit I and the zero voltage defines bit O. It is
called NRZ because the signal does not return to zero at the middle of the bit. Figure 4.5 show a
unipolar NRZ scheme.

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POLAR
NRZ-voltages are on both sides of the time axis.
Polar NRZ scheme can be implemented with two voltages. E.g. +V for 1 and -V for 0.

There are two variations:

o NZR - Level (NRZ-L) - positive voltage for one symbol and negative for the
other

o NRZ - Inversion (NRZ-I) - the change or lack of change in polarity determines

the value of a symbol. E.g. a “1” symbol inverts the polarity a “0” does not.

Non-Return-to-Zero (NRZ) In polar NRZ encoding, we use two levels of voltage amplitude. We
can have two versions of polar NRZ: NRZ-L and NRZ-I.

The figure also shows the value of r, the average baud rate, and the bandwidth. In the first variation,
NRZ-L (NRZ-Level), the level of the voltage determines the value of the bit. In the second
variation, NRZ-I (NRZ-Invert), the change or lack of change in the level of the voltage determines
the value of the bit. If there is no change, the bit is 0; if there is a change, the bit is 1.

In NRZ-L the level of the voltage determines the value of the bit. In NRZ-I the inversion or the lack
of inversion determines the value of the bit.
NRZ-L and NRZ-J both have an average signal rate ofNI2 Bd.
NRZ-L and NRZ-J both have a DC component problem.

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Polar – RZ
The Return to Zero (RZ) scheme uses three voltage values. +, 0, -.
Each symbol has a transition in the middle. Either from high to zero or from low to zero
More complex as it uses three voltage level. It has no error detection capability

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Polar - Biphase: Manchester and Differential Manchester

o Manchester coding is a combination of NRZ-L and RZ schemes.
o Every symbol has a level transition in the middle: from high to low or low to
high.
o It uses only two voltage levels.

Differential Manchester coding consists of combining the NRZ-I and RZ schemes.

Every symbol has a level transition in the middle. But the level at the beginning of the
symbol is determined by the symbol value. One symbol causes a level change the other
does not.
The minimum bandwidth of Manchester and differential Manchester is 2 times that of
NRZ.
In bipolar encoding, we use three levels: positive, zero, and negative.

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Bipolar - AMI and Pseudoternary

This coding scheme uses 3 voltage levels: - +, 0, -, to represent the symbols

Voltage level for one symbol is at ―0ǁ and the other alternates between + & -.
Bipolar Alternate Mark Inversion (AMI) - the “0” symbol is represented by
zero voltage and the “1” symbol alternates between +V and -V.
Pseudoternary is the reverse of AMI

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Summary of Line Coding Schemes

REVIEW QUESTIONS

1. Explain the different ways of converting data to signal and vice versa.
2. Explain in detail what is signal encoding
3. Write short note on NRZ-L, NRZ-I, AMI and Manchester encoding
4. Differentiate between NRZ-L, NRZ-I, AMI, Manchester and Differential
Manchester encoding

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Block Coding

Block coding can give us this redundancy and improve the performance of line coding. In general, block
coding changes a block of m bits into a block of n bits, where n is larger than m. Block coding is referred
to as an mB/nB encoding technique. The slash in block encoding (for example, 4B/5B) distinguishes
block encoding from multilevel encoding (for example, 8B6T), which is written without a slash. Block
coding normally involves three steps: division, substitution, and combination. In the division step, a
sequence of bits is divided into groups of m bits. For example, in 4B/5B encoding, the original bit
sequence is divided into 4-bit groups. The heart of block coding is the substitution step. In this step, we
substitute an m-bit group for an n-bit group. For example, in 4B/5B encoding we substitute a 4-bit code
for a 5-bit group. Finally, the n-bit groups are combined together to form a stream. The new stream has
more bits than the original bits. The procedure for block coding is given below.

It normally involves three steps: division, substitution, and combination. In the division step,a sequence
of bits is divided into groups of m-bits. In the substitution step, we substitute an m-bit group for an n-bit
group. Finally, the n-bit groups are combined together to form a stream which has more bits than the
original bits.

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Examples of mB/nB coding:

4B/5B (four binary/five binary ) –

This coding scheme is used in combination with NRZ-I. The problem with NRZ-I was that it has
a synchronization problem for long sequences of zeros. So, to overcome it we substitute the bit stream
from 4-bit to 5-bit data group before encoding it with NRZ-I. So that it does not have a long stream of
zeros. The block-coded stream does not have more than three consecutive zeros (see encoding table).

At the receiver, the NRZ-I encoded digital signal is first decoded into a stream of bits and then decoded
again to remove the redundancy bits.

Drawback – Though 4B/5B encoding solves the problem of synchronization,it increases the signal rate of
[Link],it does not solve the DC component problem of NRZ-L.

8B/10B (eight binary/ten binary) –

This encoding is similar to 4B/5B encoding except that a group of 8 bits of data is now substituted by a
10-bit code and it provides greater error detection capability than 4B/5B. It is actually a combination of
5B/6B and 3B/4B [Link] most five significant bits of a 10-bit block is fed into the 5B/6B encoder;
the least 3 significant bits is fed into a 3B/4B encoder. The split is done to simplify the mapping table.

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A group of 8 bits can have 2^8 different combinations while a group of 10 bits can have 2^10 different
combinations. This means that there are 2^10-2^8=768 redundant groups that are not used for 8B/10B
encoding and can be used for error detection and disparity check.

Thus, this technique is better than 4B/5B because of better error-checking capability and better
synchronization.

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Scrambling

Scrambling is a technique that does not increase the number of bits and does provide synchronization.
Problem with technique like Bipolar AMI(Alternate Mark Inversion) is that continuous sequence of zero’s
create synchronization problems one solution to this is Scrambling.

There are two common scrambling techniques:

1. B8ZS(Bipolar with 8-zero substitution)
2. HDB3(High-density bipolar3-zero)

B8ZS(Bipolar with 8-zero substitution) –

This technique is similar to Bipolar AMI except when eight consecutive zero-level voltages are
encountered they are replaced by the sequence,”000VB0VB”.
Note –
➢ V(Violation), is a non-zero voltage which means signal have same polarity as the previous non-zero
voltage. Thus it is violation of general AMI technique.
➢ B(Bipolar), also non-zero voltage level which is in accordance with the AMI rule (i.e.,opposite
polarity from the previous non-zero voltage).

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Note – Both figures (left and right one) are correct, depending upon last non-zero voltage signal of previous
data sequence (i.e., sequence before current data sequence “100000000”).

HDB3(High-densitybipolar3-zero) –In this technique four consecutive zero-level voltages are replaced
with a sequence “000V” or “B00V”.
Rules for using these sequences:
• If the number of nonzero pulses after the last substitution is odd, the substitution pattern will be
“000V”, this helps maintaining total number of nonzero pulses even.
• If the number of nonzero pulses after the last substitution is even, the substitution pattern will be
“B00V”. Hence even number of nonzero pulses is maintained again.

Explanation – After representing first two 1’s of data we encounter four consecutive [Link] our last
substitutions were two 1’s(thus number of non-zero pulses is even).So,we substitute four zeros with “B00V”.

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Encoding analog data into digital signals - PCM, DM

Assume, we have an analog signal such as one created by a microphone or camera. A digital signal
is superior to an analog signal. The tendency today is to change an analog signal to digital data. In
this section we describe two techniques, pulse code modulation and delta modulation. After the
digital data are created (digitization), we can use one of the techniques to convert the digital data
to a digital signal.

Two methods of Analog to Digital Conversion:-

• Pulse Code Modulation (PCM)
• Delta Modulation (DM)

PCM (PULSE CODE MODULATION)

PCM is a general scheme for transmitting analog data in a digital and binary way,
independent of the complexity of the analog waveform. With PCM all forms of analog data
like video, voice, music and telemetry can be transferred.
To obtain PCM from an analog waveform at the source (transmitter), the analog signal
amplitude is sampled at regular time intervals. The sampling rate (number of samples per
second), is several times the maximum frequency of the analog waveform. The amplitude of
the analog signal at each sample is rounded off to the nearest binary level (quantization).
The number of levels is always a power of 2 (4, 8, 16, 32, 64, ...). These numbers can be
represented by two, three, four, five, six or more binary digits (bits) respectively.
At the destination (receiver), a pulse code demodulator converts the binary numbers back into
pulses having the same quantum levels as those in the modulator. These pulses are further
processed to restore the original analog waveform.

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The encoder has 3 processes:

1) Sampling
2) Quantization &
3) Encoding.

Sampling
• We convert the continuous time signal (analog) into the discrete time signal (digital).
• Pulses from the analog-signal are sampled every Ts sec where Ts is the sample-interval or period.
• The inverse of the sampling-interval is called the sampling-frequency (or sampling-rate).
• This method is sometimes referred to as PAM (pulse amplitude modulation).
• Sampling-frequency is given by
fs = 1/Ts
• Three sampling methods are:
1) Ideal Sampling: This method is difficult to implement.
2) 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.
3) Flat Top Sampling
The most common sampling method is sample and hold. Sample and hold method create flat-top
samples.

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Figure: Three Sampling methods for PCM

A PCM encoder has three processes: -
1. The analog signal is sampled.
2. The sampled signal is quantized.
3. The quantized values are encoded as streams of bits.

Sampling Rate
• According to Nyquist theorem,
“The sampling-rate must be at least 2 times the highest frequency, not the bandwidth“.
i) If the analog-signal is low-pass, the bandwidth and the highest frequency are the same value
ii) If the analog-signal is bandpass, the bandwidth value is lower than the value
of the maximum frequency

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Quantization
• The sampled-signal is quantized.
• Result of sampling is a set of pulses with amplitude-values b/w max & min amplitudes of the signal.
Four steps in quantization:
1) We assume that the original analog-signal has amplitudes between Vmin & Vmax.
2) We divide the range into L zones, each of height Δ(delta).

where L = number of levels.

3) We assign quantized values of 0 to (L-1) to the midpoint of each zone.
4) We approximate the value of the sample amplitude to the quantized values.
For example:
Let Vmin=-20 Vmax =+20 V L = 8 Therefore, Δ = [+20-(-20)]/8= 5 V

• In the chart ,
1) First row is normalized-PAM-value for each sample.
2) Second row is normalized-quantized-value for each sample.

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3) Third row is normalized error (which is diff. b/w normalized PAM value & quantized values).
4) Fourth row is quantization code for each sample.
5) Fifth row is the encoded words (which are the final products of the conversion).

Quantization Level
• Let L = number of levels.
• The choice of L depends on
→ range of the amplitudes of the analog-signal and
→ how accurately we need to recover the signal.
• If the signal has only 2 amplitude values, we need only 2 quantization-levels.
If the signal (like voice) has many amplitude values, we need more quantization-levels.
• In audio digitizing, L is normally chosen to be 256. In video digitizing, L is normally thousands.
• Choosing lower values of L increases the quantization-error.
Quantization Error
• Quantization-error is the difference b/w normalized PAM value & quantized values
• Quantization is an approximation process.
• The input values to the quantizer are the real values.
The output values from the quantizer 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, Then, there is no 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.
Uniform vs. Non Uniform Quantization
• Non-uniform quantization can be done by using a process called companding and expanding.
1) The signal is companded at the sender before conversion.
2) The signal is expanded at the receiver after conversion.

• Companding means reducing the instantaneous voltage amplitude for large values.
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Expanding means increasing the instantaneous voltage amplitude for small values.
• It has been proved that non-uniform quantization effectively reduces the SNRdB of quantization.
Encoding
• The quantized values are encoded as n-bit code word.
• In the previous example,
A quantized value 2 is encoded as 010.
A quantized value 5 is encoded as 101.
• Relationship between number of quantization-levels (L) & number of bits (n) is given by n=log2L or
2n=L
• The bit-rate is given by:

PCM decoder

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Three different sampling methods for PCM

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.

DELTA MODULATION (DM)

The simplest is delta modulation. PCM finds the value of the signal amplitude for each
sample; DM finds the change from the previous sample.
Note that there are no code words here; bits are sent one after another.

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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 δ. If the delta is
positive, the process records a I; 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
δ up; it the next bit is 0, it moves it δ down. Note that we need a delay unit to hold the staircase
function for a period between two comparisons.

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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 0 is not fixed. In adaptive delta modulation,
the value of 0 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.

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Encoding digital data into analog signals - ASK, FSK, PSK

Digital-to-analog conversion is the process of changing one of the characteristics of an analog

signal based on the information in digital data.

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Digital modulation is used to convert digital data to analog signal.
It can be accomplished in the following ways:
1. ASK
2. FSK
3. PSK
4. QAM

ASPECTS OF DIGITAL-TO-ANALOG CONVERSION

Data Element Versus Signal Element
Data Rate Versus Signal Rate
Bandwidth

Carrier Signal

AMPLITUDE SHIFT KEYING (ASK)

In amplitude shift keying, the amplitude of the carrier signal is varied to create signal elements.
Both frequency and phase remain constant while the amplitude changes.

Binary ASK (BASK)

ASK is normally implemented using only two levels and is hence called binary amplitude shift
keying. Bit 1 is transmitted by a carrier of one particular amplitude. To transmit Bit 0 we change
the amplitude keeping the frequency is kept constant

The value of d is between 0 and 1. This means that the bandwidth can be expressed as shown,
where 5 is the signal rate and the B is the bandwidth.

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Example
We have an available bandwidth of 100 kHz which spans from 200 to 300 kHz. What are the carrier frequency

and the bit rate if we modulated our data by using ASK with d =I?

Solution

The middle of the bandwidth is located at 250 kHz. This means that our carrier frequency can be atfe =250

kHz. We can use the formula for bandwidth to find the bit rate (with d =1 and r =1).

FREQUENCY SHIFT KEYING (FSK)

In Frequency shift keying, we change the frequency of the carrier wave.

Bit 0 is represented by a specific frequency, and bit 1 is represented by a different frequency.
In the figure below frequency used for bit 1 is higher than frequency used for bit 0

Example
We have an available bandwidth of 100 kHz which spans from 200 to 300 kHz. What should be
the carrier frequency and the bit rate if we modulated our data by using FSK with d =1?

Solution
This problem is similar to Example 5.3, but we are modulating by using FSK. The midpoint of
the band is at 250 kHz. We choose 2Δf to be 50 kHz; this means

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PHASE SHIFT KEYING (PSK)

Phase shift keying (PSK) is a method of transmitting and receiving digital signals in which
the phase of a transmitted signal is varied to convey information.
Both amplitude and frequency remain constant as the phase changes.
The simplest from of PSK has only two phases, 0 and 1.
If the phase of the wave does not change, then the signal state stays the same (low or high).
If the phase of the wave changes by 180 degrees, that is, if the phase reverses, then the signal
state changes (from low to high or from high to low)

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Quadrature PSK (QPSK)

The simplicity of BPSK enticed designers to use 2 bits at a time in each signal element, thereby
decreasing the baud rate and eventually the required bandwidth. The scheme is called quadrature PSK or
QPSK because it uses two separate BPSK modulations; one is in-phase, the other quadrature (out-of-

phase).
Example
Find the bandwidth for a signal transmitting at 12 Mbps for QPSK. The value of d =O.

Solution
For QPSK, 2 bits is carried by one signal element. This means that r =2. So the signal rate (baud rate) is
S =N x (lIr) =6 Mbaud. With a value of d =0, we have B =S =6 MHz.

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QAM
The concept of Quadrature Amplitude Modulation (QAM) involves use of two carriers, one for
phase and the other for quadrature, with different amplitude levels for each carrier.
It is a combination of ASK & PSK.

The above Figure (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,
Figure (d) shows a 16-QAM constellation of a signal with eight levels, four positive and four negative.

Bandwidth for QAM

The minimum bandwidth required for QAM transmission is the same as that required for ASK and PSK
transmission. QAM has the same advantages as PSK over ASK.

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REVIEW QUESTIONS

1. Define analog transmission.

2. Define carrier signal and its role in analog transmission.
3. Define digital-to-analog conversion.
4. Which characteristics of an analog signal are changed to represent the digital signal in each of the
following digital-to-analog conversion?
a. ASK b. FSK c. PSK [Link]
5. Which of the four digital-to-analog conversion techniques (ASK, FSK, PSK or QAM) is the most
susceptible to noise? Defend your answer.

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Transmission media- twisted pair, Coaxial cable, optical fiber,

terrestrial microwave, satellite microwave.

In Data Communication networking, it is worth understanding the medium through

which data passes and what are the available mediums and their types.
TRANSMISSION MEDIA

Transmission media is a means by which a communication signal is carried from one

system to another
A transmission medium can be defined as anything that can carry information from a
source to a destination.
The transmission medium is usually free space, metallic cable or fiber – optic cable.

Categories of transmission media

In telecommunications, transmission media can be divided into two broad categories:
Guided
o twisted-pair cable
o coaxial cable
o fiber-optic cable
Unguided.
o free space

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GUIDED MEDIA

➢ Guided Transmission media uses a cabling system that guides the data signals
along a specific path
➢ Guided media also known as Bounded media, which are those that provide a
conduit from one device to another, include twisted-pair cable, coaxial cable,
and fiber-optic cable.
➢ Out of these twisted-pair cable, coaxial cable transport signals in the form of
electric signals and fiber-optic cable transport signals in the form of light.
➢ Types:
1. Twisted-Pair Cable
i. Unshielded Twisted Pair Cable (UTP)
ii. Shielded Twisted Pair Cable (STP)
2. Coaxial Cable
3. Fiber-Optic Cable

(1) Twisted-pair cable

➢ The wire is twisted together in pairs.

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➢ Each pair would consist of a wire used for the +ve data signal and a wire
used for the —ve data signal. Any noise that appears on +ve/—ve wire of the
pair would occur on the other wire.
➢ Because the wires are opposite polarities, they are 180 degrees out of phase
(180 degree phases or definition of opposite polarity) when the noise appears
on both wires, it cancels or nulls itself out at the receiving used.
➢ Twisted pair cables are most effectively used in a system that uses a balanced
line method of transmission.

Unshielded Twisted Pair Cable (UTP)& Shielded Twisted Pair Cable (STP)

(UTP)

(STP)

Cables with the shield are called shielded twisted pair and commonly abbreviated STP.
Cables without a shield are called unshielded twisted pair or UTP.
Twisting the wires together results in characteristics impedance for the cable.
UTP or unshielded twisted pair cable is used on Ethernet
UTP cables are used for Ethernet cabling where 4 twisted pair cables (a total of 8 wires are
used)

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(2) Co-Axial Cable

Coaxial cable consists of 2 conductors.

The inner conductor is contained inside the insulator with the other conductor
weaves around it providing a shield.
An insulating protective coating called a jacket covers the outer conductor.
The outer shield protects the inner conductor from outside electrical signals.
The distance between the outer conductor (Shield) and inner conductor plus the
type of material used for insulating the inner conductor determine the cable
properties or impedance.
The excellent control of the impedance characteristics of the cable allow higher
data rates to be transferred than twisted pair cable.

(3) Fibre Optic Cable

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Optical fiber consists of thin glass fiber that can carry information at frequencies
in the visible light spectrum.
The typical optical fiber consists of a very narrow strand of glass called the cladding.
A typical core diameter is 62.5 microns.
Typically cladding has a diameter of 125 minors. Coating the cladding is a
protective coating consisting of plastic, it is called the jacket.

The device generating the message has it in electromagnetic form (electrical signal);
this has to be converted into light (i.e. optical signal) to send it on optic fiber cable.
The process of converting light to electric signal is done on the receiving side.

Advantages:
1. Small size and light weight: The size of the optical fibers is very small. Therefore a
large number of optical fibers can fit into a cable of small diameter.
2. Easy availability and low cost: The material used for the manufacturing of optical
fibers is “Silica glass”. This material is easily available. So the optical fibers cost lower
than the cables with metallic conductors.
3. No electrical or electromagnetic interference: Since the transmission takes place in
the form of light rays the signal is not affected due to any electrical or electromagnetic
Interference.
4. Large Bandwidth: As the light rays have a very high frequency in GHz range, the
bandwidth of the optical fiber is extremely large.
5. Other advantages: - No cross talk inside the optical fiber cable. Signal can be sent up
to 100 times faster.

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STEP- INDEX AND GRADED-INDEX FIBER

There are two methods to classify fiber optic cables.

1. Index of refraction variation across the cable cross section.

2. Using mode i.e. various paths light rays usually take to pass through the cable. Figure
depicts fiber optic cable cross section. As shown all the fiber optic cable consists of two
main layers. They are core(inner most) and cladding.

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SINGLE MODE STEP INDEX FIBER

In Single mode light takes single path through the fiber core. Step index means sharp step
in the index of refraction between core and cladding interface. This indicates that in step
index, core and cladding have their own constant index of refractions N1 and N2
respectively. The step index cable cross section is depicted in the figure

MULTIMODE GRADED INDEX FIBER

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In multimode light follows multiple paths through the fiber core. In graded index, index
of refraction is not constant but vary smoothly across the diameter of the core. Index of
refraction is increasing as one goes near the center while decreasing near outer core edges.
Index of refraction is maximum at the center of the core. Index of refraction is constant
for cladding part of the fiber.
Based on above, there are three types of fiber optic cable.
• multimode step index
• single mode step index
• multimode graded index
Multimode step index fiber cable is widely used due to benefits of large size. It is easier
to make and hence less expensive. It is used for shorter distances at very low pulse
frequencies. The figure-3 above depicts graded index cable cross section.

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UNGUIDED (WIRELESS) TRANSMISSION MEDIUM

➢ Unguided media transport data without using a physical conductor.

This type of communication is often referred to as wireless
communication. It uses wireless electromagnetic signals to send
data.
➢ There are three types of Unguided Media
(i) Radio waves
(ii) Micro waves
(iii) Infrared.

➢ Before understanding the different types of wireless transmission medium, let us

first understand the ways in which wireless signals travel. These signals can be
sent or propagated in the following three ways:
1. Ground-wave propagation
2. Sky-wave propagation
3. Line-of-sight propagation

(1) Ground-wave propagation

Characteristics of Ground-wave propagation are as follows:

i. Follows contour of the earth
ii. Can Propagate considerable distances
iii. Frequencies up to 2 MHz
iv. Example - AM radio

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(2) Sky-wave propagation

Characteristics of Sky Propagation are as follows:

i. Signal reflected from ionized layer of atmosphere back down to earth
ii. Signal can travel a number of hops, back and forth between ionosphere and earth‘s surface
iii. Reflection effect caused by refraction
iv. Examples
i. Amateur radio
ii. CB radio

(3) Line-of-sight propagation

Characteristics of Line-of-Sight Propagation are as follows:

➢ Transmitting and receiving antennas must be within line of sight
a. Satellite communication – signal above 30 MHz not reflected by ionosphere
b. Ground communication – antennas within effective line of site due to refraction

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RADIO WAVES:

➢ Electromagnetic wave ranging in frequencies between 3 KHz and 1GHz are

normally called radio waves.
➢ Radio waves are omni-directional when an antenna transmits radio waves they
are propagated in all directions. This means that sending and receiving antenna do
not have to he aligned. A sending antenna can send waves that can be received by
any receiving antenna.
➢ Radio waves particularly those waves that propagate in sky mode, can travel long
distances. This makes radio waves a good candidate for long-distance
broadcasting such as AM radio.
➢ Radio waves particularly those of low and medium frequencies can penetrate
walls. It is an advantage because; an AM radio can receive signals inside a
building. It is the disadvantage because we cannot isolate a communication to first
inside or outside a building.

MICROWAVES

➢ Electromagnetic waves having frequencies between 1 and 300 GHz are called microwaves.
➢ Microwaves are unidirectional; when an antenna transmits microwaves they can
be narrowly focused. This means that the sending and receiving antennas need to
be aligned. The unidirectional property has an obvious advantage. A pair of
antennas can be aligned without interfering with another pair of aligned antennas.
➢ Microwaves propagation is line-of-sight. Since the towers with the mounted
antennas needs to be in direct sight of each other, towers that are far apart need to
be very tall, the curvature of the earth as well as other blocking obstacles do not

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allow two short towers to communicate using microwaves, Repeaters are often
needed for long distance communication very high frequency microwaves cannot
penetrate walls.
➢ Parabolic dish antenna and horn antenna are used for this means of transmission

INFRARED

➢ Infrared signals with frequencies ranges from 300 GHz to 400 GHz can be
used for short range communication.
➢ Infrared signals, having high frequencies, cannot penetrate walls. This helps to
prevent interference between one system and another. Infrared Transmission in
one room cannot be affected by the infrared transmission in another room.
➢ Infrared band, has an excellent potential for data transmission. Transfer digital
data is possible with a high speed with a very high frequency. There are number
of computer devices which are used to send the data through infrared medium e.g.
keyboard mice, PCs and printers. There are some manufacturers provide a special
part called the IrDA port that allows a wireless keyboard to communicate with a
PC

COMPARISON BETWEEN WIRED AND WIRELESS MEDIA

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COMPARISON BETWEEN TWISTED PAIR CABLE, CO-AXIAL CABLE AND
OPTICAL FIBER

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