EEE 4201 (Monirul - Sir)

Telecommunication-I; Course No.
: EEE-4201 1
1. Telecommunication
Telecommunication is the transmission of signs, signals, messages, writings, images and
sounds or intelligence of any nature by wire, radio, optical or other electromagnetic systems.
Telecommunication occurs when the exchange of information between communication
participants includes the use of technology. It is transmitted either electrically over physical
media, such as cables, or via electromagnetic radiation. Such transmission paths are often
divided into communication channels which afford the advantages of multiplexing.
Fig. 1.1: Communication system.
A typical communication system is modeled in Fig. 1.1. The components of a communication

system are as follows:
Input transducer: The source originates a message, such as, a human voice, a television picture,
a teletype message or data. If the data is nonelectrical then it must be converted by an input
transducer into an electrical waveform referred to as the baseband signal or message signal.
Transmitter: The transmitter modifies the baseband signal for efficient transmission.
Channel: The channel is a medium, such as, wire, coaxial cable, a waveguide, an optical fiber
or, a radio link through which the transmitter output is sent.
Receiver: The receiver reprocesses the signal received from the channel by undoing the signal
modifications made at the transmitter and the channel.
Output transducer: The receiver output is fed to the output transducer, which converts the
electrical signal to its original form, i.e., the message.
Destination: The destination is the unit to which the message is communicated.
2. Signal
Telecommunication-I; Course No.: EEE-4201 2
A signal is a set of information or data. For example, a telephone or, a television signal,
monthly sales of a corporation, or, the daily closing prices of a stock market. In these cases, the
signals are the functions of the independent variable time. On the other hand, when an electrical
charge is distributed over a body, the signal is the charge density, which is a function of space
rather than time. In this course, we deal almost exclusively with signals that are the functions of
time.
3. Systems
A system is an entity that processes a set of signals (i.e., input) to yield another signals (i.e.,
output). A system may be made up of physical components, such as, electrical, mechanical, or,
hydraulic systems (i.e., hardware realization), or, it may be an algorithm that computes an
output from an input signal (i.e., software realization).
In other words, signals may be processed further by systems, which may modify them or,
extract additional information from them. For example, anti aircraft operator may want to
know the future location of a hostile moving target that is being tracked by his radar. Knowing
the radar signal he knows the past location and velocity of the target. By properly processing
the radar signal (i.e., the input), he can approximately estimate the future location of the target.
4. Size of a Signal
Generally, the size of any entity is a number that indicates the largeness or strength of that
entity. In case of signal, the amplitude varies with time. In this regard, the measurement of size
of a signal is very much difficult in terms of largeness and strength. In order to measure the size
of a signal, in this case, considering the signal amplitude and its duration are important.
For instance, if we measure the size of a human being (i.e., volume V), then width and height
must be considered. It is assumed that, the shape of a person is a cylinder of variable radius r
which varies with the height h. Then, a reasonable measure of the size of a person of height H
is the person’s volume V, given by,
4.1 Signal Energy
Considering the signal amplitude and its duration to measure the size of a signal, f(t) is a
defective concept. The reason is that, the positive and negative areas of a signal f(t) could
cancel each other. This difficulty can be corrected by defining the signal size as the area under
f2(t), which is always positive. This measure is called the signal energy, Ef, defined as,
………………………………………………………..…(3.1)
This definition can be generalized to a complex valued signal f(t) as
…………………………………………………………..(3.2)
4.2 Signal Power

Signal energy is whether finite or infinite depends on the signal behavior. For example, a signal
shown in Fig. 3.1(a) having the amplitude whereas the amplitude does not
in Fig. 3.1(b).
Fig. 3.1: Examples of signals: (a) a signal with finite energy, (b) a signal with finite
power.
It is noted that, the signal energy must be finite for it to be a meaningful measure of the signal
size. The integral in Eq. (3.1) will converge for the signal shown in Fig. 3.1(a). Thus, the signal
energy is finite in this case. On the other hand, the integral in Eq. (3.1) will not converge for the
next signal shown in Fig. 3.1(b). For this, the signal energy is infinite.
A more meaningful measure of the signal size in such a case (i.e., signal in Fig. 3.1(b)) would
be the time average of the energy. This measure is called the power of the signal. For a signal
f(t), the power Pf is defined as,
……………………………………………………….
(3.3)
This definition can be generalized to a complex valued signal f(t) as,
……………………………………………………..
(3.4)
It is observed that, the signal power Pf is the time average of the signal amplitude squared, that
is, the mean-squared value of f(t). Indeed, the square root of Pf is the familiar rms (i.e., root
mean square) value of f(t).
Note: The mean of an entity averaged over a large time interval approaching infinity exists if
the entity is either periodic or has a statistical regularity. If such a condition is not satisfied,
average may not exist. For example, a ramp signal f(t) = t increases infinitely as then
neither the energy nor the power exists for this signal.
Note:
1. The signal energy computed by Eq. (3.1) & (3.2) does not indicate the actual energy of the
signal. Because, the signal energy depends not only on the signal, but also on the load.
Q1. Determine the suitable measures if the signals shown in Fig. 3.2
Fig. 3.2: Signal curves.
Solution:
Q1(a). Here, the signal amplitude therefore, the suitable measure for this signal
is its energy, Ef given by,
Q1(b). Here, the signal amplitude does not However, it is periodic and
therefore its power exists. For this, Eq. (3.3) can be used to determine its power as,
Here, a periodic signal repeats regularly each period that is T = 2 seconds. To measure the
power of that signal, we can average over an infinitely large interval that is identical to
averaging this quantity over one period (i.e., T = 2 sec.). Therefore, the measured power is,
It is noted that, signal power is the square of its rms value. Therefore, the rms value of this
signal is,
5. Classification of Signals
There are several classes of signals, which are:
a) Continuous-time and discrete-time signals
b) Analog and digital signals

c) Periodic and aperiodic signals
d) Energy and power signals
e) Deterministic and probabilistic signals
5.1 Continuous-time and discrete-time signals

A signal that is specified for every value of time t is a continuous-time signal (shown in Figs.
4.1(a, b)), and a signal that is specified only at discrete values of t is a discrete-time signal
(shown in Figs. 4.1(c, d)). For examples, telephone and video camera outputs are continuous-
time signals, whereas, monthly sales of a corporation and stock market daily averages are
discrete-time signals.
Fig. 4.1: Examples of signals: (a) analog, continuous-time, (b) digital, continuous-
time, (c) analog, discrete-time, (d) digital, discrete-time.
5.2 Analog and digital signals

5.3 Periodic and Aperiodic signals
A signal f (t) is said to be periodic if for some positive constant T0 , that is,
………………………………….. (4.1)
Here, the smallest value of T0 that satisfies the periodicity condition of Eq. (4.1) is the period of
f (t). The Fig. 4.2 and Fig. 3.2 (b) show the examples of periodic signals. On the other hand, a
signal is aperiodic if it is not periodic shown in Figs. 3.2(a), 4.3(a,b,c,d).
Fig. 4.2: A periodic signal of period T0.
Fig. 4.3: Aperiodic signals.
5.4 Energy and Power signals

A signal with finite energy is an energy signal, and a signal with finite and nonzero power is a
power signal. Here, signals in Fig. 3.2(a) and 3.2(b) are examples of energy and power signals,
respectively. It is observed that, power is the time average of energy. Since the averaging is
made over an infinitely large interval, a signal with finite energy has zero power, and a signal
with finite power has infinite energy. Therefore, a signal cannot be both an energy and a power
signal. On the other hand, there are signals that are neither energy nor power signals, which are
ramp signals.
5.5 Deterministic and Random signals

The physical description of a signal is known completely, either in a mathematical form or a
graphical form, is a deterministic signal. On the other hand, a signal whose values cannot be
predicted precisely but are known only in terms of probabilistic description, such as, mean
value, mean squared value, and so on is a random signal.
6. Signal Operations
There are three useful signal operations: shifting, scaling, and inversion. Since the independent
variable in our signal description is time, these operations are discussed as time shifting , time
scaling, and time inversion.
6.1 Time Shifting

Considering a signal f (t) shown in Fig. 5.1(a) and the same signal delayed by T seconds shown
in Fig. 5.1(b), which is denoted by  (t). Whatever happens in f (t) (Fig. 5.1(a)) at some instant
t, whereas, same thing is happened in  (t) (Fig. 5.1(b)) T seconds later at the instant tT.
Therefore,
……………………………………………………………….(5.1)
and
……………………………………………………………….(5.2)
Therefore, to time-shift a signal by T, t is replaced by t –T. Thus, f (t –T) represents f (t) time-
shifted by T seconds. If T is positive, the shift is to the right (i.e., delay). If T is negative, the
shift is to the left (i.e., advance). Thus, f (t –2) is f (t) delayed (i.e., right-shifted) by 2 seconds,
and f (t + 2) is f (t) advanced (i.e., left-shifted) by 2 seconds.
Fig. 5.1: A signal of time shifting.
6.2 Time Scaling

The compression or expression of a signal in time is known as time scaling. Consider a signal
f (t) in Fig. 5.2(a). The signal  (t) in Fig. 5.2(b) is f (t) compressed in time by a factor of 2.
Therefore, whatever happens in f (t) at some instant t also happens to  (t) at the instant t/2, so
that,
Fig. 5.2: Time scaling a signal.
…………………………………………………….(5.3)
and
……………………………………………….……(5.4)
It is observed that, f (t) = 0 at t = T1 and T2 , whereas,  (t)=0 at t = T1/2 and T2/2 as shown in
Fig. 5.2(b). Fig. 5.2(c) shows which is f (t) expanded in time by a factor of 2.
6.3 Time Inversion

Consider the signal f (t) in Fig. 5.3(a), it is observed that, f (t) is a rigid wire frame hinged at the
vertical axis. To time-invert f (t), we rotate this frame 180 about the vertical axis. This time
inversion or folding gives us signal  (t) shown in Fig. 5.3(b). Observe that, whatever happens
in Fig. 5.3(a) at some instant t also happens in Fig. 5.3(b) at instant –t. Therefore,
…………………………………………………………..…..(5.5)
and
………………………………………………………………(5.6)
Fig. 5.3: Time inversion of a signal.
7. Some Useful Signal Models

Some useful signal models are: the step, the impulse and the exponential functions which are
very useful to simplify many aspects of signals and systems.
7.1 Unit Step Function

A unit step function u (t) is shown in Fig. 6.1 that can be defined by,
Fig. 6.1: Unit step function u(t).
If we want a signal to start at t = 0, it is needed to multiply the signal with u(t). Unit function is
useful in specifying a function with different mathematical descriptions over different intervals.
Fig. 6.2: Representation of rectangular pulse by step functions.
In the Fig. 6.2(a), the rectangular pulse is presented. Such a pulse can be expressed in terms of
step functions by observing that the pulse f(t) can be expressed as the sum of the two delayed
unit step functions as shown in Fig. 6.2(b). The unit step function u(t) delayed by T seconds is
u(t-T). Thus, the pulse f(t) is expressed by,
7.2 Unit Impulse Function

The unit impulse function is shown in Fig. 6.3 that can be defined by,
……………………………….……………………………..(6.1)
Fig. 6.3: A unit impulse and its approximation.
An impulse is a tall, narrow rectangular pulse of unit area shown in Fig. 6.3(b). The width of
this rectangular pulse is a very small value Consequently, its height is a very large value
1/. The unit impulse therefore can be regarded as a rectangular pulse with a width that has
become infinitesimally small, a height that has become infinitely large, and an overall area that
has been maintained at unity.
Thus,  (t) = 0 everywhere except at t = 0, where it is undefined. For this reason, an unit
impulse is represented by the spear like symbol shown in Fig. 6.3 (a).
7.3 Multiplication of a Function by an Impulse

If a unit impulse (t) is multiplied by a function (t) then the impulse is known to be continuous
at t = 0. Since, the impulse exists only at t = 0, where the value of (t) at t = 0 is (0), we get,
……………………………………………………..(6.2)
Similarly, if (t) is multiplied by an impulse (t-T) (impulse located at t = T), then
……………………………………………...(6.3)
Here, (t) is continuous at t = T.
7.4 Sampling Property of the Unit Impulse Function

By the integration of Eq. (6.2), we get,
Here, (t) is continuous at t = 0. This result means that the area under the product of a function
with an impulse (t) is equal to the value of that function at the instant where the unit impulse is
located. This property is very important and useful that is known as the sampling or shifting
property of the unit impulse.
By the integration of Eq. (6.3), we get,

……………………………………………………..(6.4)
This equation is another form of sampling or shifting property. Here, the impulse (t-T) is
located at t = T. Therefore, the area under (t)(t-T) is (T), the value of (t) at the instant
where the impulse is located at t =T. In these derivations, we have assumed that the function is
continuous at the instant where the impulse is located.
Chapter-3
3.1 Signal Transmission through a Linear System

For a linear, time-invariant, continuous-time system, the input-output relationship is given by,
…………………………………………………………….(3.1)
Where, g(t) is the input, y(t) is the output, and h(t) is the unit impulse response of the linear
time-invariant system. If
where, is the system transfer function, then application of the time convolution property
of Eq. (3.1) yields,
………………………………………………………………(3.2)
3.1.1 Signal Distortion during Transmission

The transmission of an input signal g(t) through a system changes it into the output signal y(t).
The nature of this change or modification is shown in Eq. (3.2). Here, G(w) and Y(w) are the
spectra of the input and output, respectively. Therefore, H(w) is the spectral response of the
system. Thus, the Eq. (3.2) refers to the output spectrum that is given by the input spectrum
multiplied by the spectral response of the system. Furthermore, the spectral modification of the
signal by the system is cleared in the above equation.
Eq. (3.2) can be expressed in polar form as,

………………………………………….(3.3)
Therefore,
…………………………………………………………….(3.4)
During transmission, the input signal amplitude spectrum is changed to

Similarly, the input signal phase spectrum is changed to
An input signal spectral component of frequency w is modified in amplitude by
a factor and is shifted in phase by an angle Clearly, is the amplitude
response and the is the phase response of the system. Thus, the system modifies the
amplitudes and phases of various sinusoidal inputs. This is why, is called the
frequency response of the system.
3.1.2 Distortionless Transmission

During the transmission of signal amplification or message signal over a communication

channel, it is required the output waveform to be a replica of the input waveform. In such cases,
distortion needs to be minimized caused by the amplifier or, the communication channel.
Transmission is said to ne distortionless if the input and the output have identical wave shapes
within a multiplicative constant. A delayed output that retains the input waveform is also
considered distortionless. Thus, in distortionless transmission, the input g(t) and the output y(t)
satisfy the condition,
………………………………………………………………………..(3.5)
The Fourier transform of Eq. (3.5) yields,
……………………………………………………………………..(3.6)
By the comparison of Eqs. (3.4) & (3.6), we get,
…………………………………………………………………………..(3.7)
This is the transfer function required for distortionless transmission. From this equation, it
follows that,
and …………………………………………………………(3.8)
This shows that, in case of distortionless transmission, the amplitude response must be
a constant and the phase response must be a linear function of  shown in Fig. 3.1. The
slope of with respect to  is -td, where td is the delay of the output with respect to the
input.
Fig. 3.1: Linear time-invariant system frequency response for distortionless transmission.
Example: Fourier wave forms from the original signals.
3.1.3 Intuitive Explanation of the Distortionless Transmission Conditions

Suppose that, a signal g(t) is passing through a communication channel. For the distortionless
case, the output signal is the input signal multiplied by k and delayed by td. To synthesize such
a signal, we need exactly the same components as those of g(t), with each component
multiplied by k and delayed by td. This means that, the system transfer function H() should be
such that each sinusoidal component suffers the same attenuation k and each component
undergoes the same time delay of td seconds. Thus, from Eq. (3.8), we get,
and
The time delay resulting from the signal transmission through a system is the negative of the
slope of the system phase response, that is,
……………………………………………………………………(3.9)
If the slope of is constant, all the components are delayed by the same time interval td .But,
if the slope is not constant, the time delay td varies with frequency. This means that, different
frequency components undergo different amounts of time delay and consequently the output
waveform will not be a replica of the input waveform.
3.1.4 Nature of Distortion in Audio and Video Signals

Generally speaking, a human ear can readily recognize amplitude distortion, although it is
relatively insensitive to phase distortion. In case of audio signals, each spoken syllable can be
considered as an individual signal. The average duration of spoken syllable is of a magnitude
on the order of 0.01 to 0.1 second. The audio systems may have nonlinear phases, yet no
noticeable signal distortion is detected by human ear. The reason is that the maximum
variation in the slope of is only a small fraction of a millisecond. That’s why, the human
ear is relatively insensitive to phase distortion. As a result, the manufacturers of audio
equipments make available only , the amplitude response characteristic of their
systems.
In case of video signal, the scenario is exactly opposite. The human eye is sensitive to phase
distortion but is relatively insensitive to amplitude distortion. The amplitude distortion in
television signals reveals itself as a partial destruction of the resulting picture, which is not
apparent to the human eye. On the other hand, the phase distortion (i.e., nonlinear phase) causes
different time delays in different picture elements. This results in a smeared picture, which is
readily apparent to the human eye. Phase distortion is also very important in digital
communication systems because the nonlinear phase characteristic of a channel cause pulse
dispersion (i.e., spreading out), which in turn causes pulses to interfere with neighboring pulses.
This interference can cause an error in the pulse amplitude at the receiver: a binary 1 may read
as 0, and vice versa.
Example: 3.16
If g(t) and y(t) are the input and the output, respectively, of a sample RC low-pass filter shown
in Fig. 3.2, determine the transfer function H() and sketch For
distortionless transmission through this filter, what is the requirement on the bandwidth of g(t)
if amplitude response variation within 2% and time delay variation within 5% are tolerable?
What is the transmission delay? Find the output y(t).
Fig. 3.2: Sample RC filter.

Solution:
According to the voltage division rule, we know that,
As per the notion, the transfer function,
Where,
Hence,
We know, time delay,
Now, the amplitude and phase response characteristics are presented in Fig. 3.3(a), whereas, the
time delay td as a function of  is shown in Fig. 3.3(b).
(a)
(b)
Fig. 3.3: Frequency response (a) and time delay (b) of Fig. 3.2.
Here, the amplitude response is practically constant and the phase shift is nearly linear. The
phase linearity results in a constant time delay characteristic. The filter therefore can transmit
low-frequency signals with negligible distortion.
In our case, the amplitude response variation 2% and time delay variation within 5% are
tolerable. Let, be the highest bandwidth of signal that can be transmitted within these
specifications. In order to compute , it is observed that, the filter is a low-pass filter with
gain and time delay both at maximum when and
Therefore, so that,
Here, the smaller of the two values, rad/s or, 32.31 kHz, is the highest bandwidth
that satisfies both constraints on
The time delay, over this band (see Fig. 3.2(b)). Also, the amplitude response is
almost unity (see Fig. 3.2(a)).

Therefore, the output,
Example: 3.17
A low-pass filter shown in Fig. 3.4(a) having the transfer function H() given by,
A pulse g(t) band-limited to B Hz shown in Fig. 3.4(b) is applied at the input of this filter. Find
the output y(t).
Solution:
The filter has ideal phase and nonideal magnitude characteristics. Because,
and
…………………………………(3.10)
Fig. 3.4: A pulse passes through a system that is not distortionless.
According to the Time-Shifting property, we know that,

If then
……………………………………………………..(3.11)
According to the Fourier transform, we get,
………………………………………..(3.12)
Using the comparison of Eq. (3.10) with Eqs. (3.11) and (3.12), then, we get,
Here, the output is actually delayed by
3.1.5 Distortion Caused by Multipath Effects

A multipath transmission takes place when a transmitted signal arrives at the receiver by two or
more paths of different delays. For example, if a signal is transmitted over a cable that has
impedance irregularities (i.e., mismatching) along the path, the signal will arrive at the receiver
in the form of a direct wave plus various reflections with various delays. In radio links, the
signal can be received by direct path between the transmitting and receiving antennas and also
by reflections from other objects, such as, hills, buildings, and so on. In long-distance radio
links using the ionosphere, similar effects occur because of one-hop and multi-hop paths. In
each of these cases, the transmission channel can be represented as several channels in parallel,
each with a different relative attenuation and a different time delay.
Example: Transmission distances for: (a) single-hop, (b) double-hop, (c) triple-hop, (d) quad-hop.
For the above-mentioned cases, the transmission channel can be represented as several channels
in parallel, each with a different relative attenuation and a different time delay.
Let us consider the case of only two paths: one with a unity gain and a delay td , and the other
with a gain  and a delay as shown in Fig. 3.5(a). The transfer functions of the two
paths are given by, and respectively.
The overall transfer function of such a channel is H(), given by
As we know, the polar form of

is
…………………………………………(3.13)
Here, both the magnitude and phase characteristics of H() are periodic in  with a period of
shown in Fig. 3.5(b).
Fig. 3.5: Multipath transmission.
The multipath therefore causes nonidealities in the magnitude and phase characteristics of the
channel that causes linear distortion. If the gains of the two paths are very closed, i.e.,
then the signals received by the two paths can very nearly cancel each other at certain
frequencies, where their phases are π rad apart.
Eq. (3.13) shows that at frequencies where then
These frequencies are multipath null frequencies. On the other hand, at
frequencies the two signals interfere constructively to enhance the gain.
Such channels are called frequency-selective fading of transmitted signals. Such distortion can
partly be corrected by using tapped delay-line equalizer. These equalizers are useful in several
applications in communications.
For Example:
For Example:
3.1.6 Signal Energy
………………………………………(3.14)
This is the statement of the well-known Parseval’s theorem. This result allows us to determine
the signal energy from either the time-domain specification g(t) or the frequency-domain
specification G(ω) of the same signal.
Example: 3.17
3.1.7 Energy Spectral Density(ESD)

Eq. (3.14) can be interpreted to mean that the energy of a signal g(t) is the result of energies
contributed by all the spectral components of the signal g(t). The contribution of a spectral
component of frequency  is proportional to
Fig. 3.6: Interpretation of the energy spectral density of a signal.
In detail, considering a signal g(t) is applied at the input of an ideal band pass filter, whose
transfer function H() is shown in Fig. 3.6(a). This filter suppresses all frequencies except a
narrow band centered at a frequency 0 shown in Fig. 3.6(b).
If the filter output is y(t), then its Fourier transform is Thus, the energy
of the output y(t) is,
Here, H()=1 over the passband and zero everywhere else, the integral on the right-hand
side is the sum of the two shaded areas (see Fig. 3.6(b)), and we have
[Note that,
Thus, is the energy contributed by the spectral components within two narrow
bands, each of width Hz, centered at Therefore, can be interpreted as the
energy per unit band width in hertz of the spectral components of g(t) centered at frequency .
In other words, is the energy spectral density (per unit band width in hertz) of g(t).
The energy spectral density (ESD) is thus defined as,
Example: 3.20
Example: 3.21
Estimate the essential bandwidth of a rectangular pulse shown in Fig. 3.7,

where the essential bandwidth is to contain at least 90% of the pulse energy.
This ESD is shown in Fig. 3.7 as a function of ωT as well as fT, where f is the frequency in
hertz. The energy Ew within the band from 0 to W rad/s is given by,
Fig. 3.7: Gate function and its energy spectral density.
Now, the energy Ew within the band from 0 to W rad/s is given by,
Setting T = x in the integral so that we obtain
As Eg = T, we have
……………………………………………………………….(3.15)
The plot of Eq. (3.15), that is to say, EW/Eg vs WT is shown in Fig. 3.7(c). It is noted that,
90.28% of the total energy of the pulse g(t) is contained within the band rad/s or,
B=1/T Hz. Therefore, using 90% criterion, the bandwidth of a rectangular pulse of width T
seconds is 1/T Hz.
3.1.8 Signal Power

As discussed earlier, the power Pg of a real signal g(t) is given by,
……………………………………………………………(3.16)
The signal power and the related concepts can readily be understood by defining a truncated
signal (see Fig. 3.8), as,
Now, Eq. (3.16) can be written as,
Here, EgT refers to the energy of truncated signal.
Fig. 3.8: Limiting process in derivation of PSD.
3.1.9 Power Spectral Density (PSD)

Self Study
Chapter-2
(Ref: Principles of Communication Engineering
Authors: Anokh Singh, A.K. Chhabra)
2.1 Telephony
Telephony is the system by which speech communication between two points can be carried
out. This communication is carried out by converting speech signals into electrical signals
which are then sent through telephone lines to the distant receiver which reproduces sound
waves form these currents. The connection with the desired distant subscriber is established
through telephone exchanges.
2.1.1 Telephone Transmitter

Carbon granule microphones are most widely used in modern telephone handsets. The essential
components of such a microphone are shown in Fig. 2.1. The transmitter comprises a chamber
in which the carbon granules are put. Carbon electrodes provide electrical contacts with the
granules. One of the electrodes is fixed, while the other is movable and fixed with the
diaphragm. The pressure of sound waves causes diaphragm and hence the movable carbon
electrode to move to and fro.
The electrical resistance of the granules varies with the pressure on the movable electrode and
as a result, the current in the external circuit also varies with sound pressure. If the movement
of the diaphragm takes place sinusoidally at a frequency f, then the resistance of granules at any
instant t is given by,
Where, R0 is the resistance of granules when there is no pressure, Rt is the total resistance under
signal conditions and r is the maximum change in the resistance due to sound pressure.
The instantaneous current in the circuit is given by,
Therefore,
Since, K ≤ 1, the amplitude of these harmonic contents is quite small.

Therefore, which is similar to the equation for amplitude modulation.
2.1.2 Telephone Receiver

A telephone receiver performs the function of converting electrical energy into sound energy.
Fig. 2.2 shows the essential parts of a telephone receiver. The current is passed through a pair
of coils. As a result, the magnetic flux is produced in proportion to the current. This magnetic
flux is in the magnetic path formed by the permanent magnet, the pole-pieces, the diaphragm
and the air-gap between the pole-pieces and the diaphragm.
The flux produced in the air-gap equals the sum of the magnetic flux produced by the
permanent magnet (ΦP) and the flux produced by the audio current (ΦA).
The instantaneous flux Φt is given by,
…………………………………………………………………………..(2.1)
Assuming the audio current to be the flux ΦA produced by it is given by,
Thus, Eq. (2.1) can be written as,
We know that, force acting on the diaphragm is,

…………………………………………………………(2.2)
Here,
Since, Eq. (2.2) can be written as

where K2 is a proportionality constant and K3=
2.1.3 Telephone Sets
A telephone set is a device with which a subscriber can send or receive a telephone call. Fig.
2.3 gives the block diagram of a telephone set.
Fig.: Block diagram of a telephone set
a) Ringer
b) Cradle switch
c) Dialer
d) Speech circuit
e) Balancing network
2.1.4 Types of Telephone Sets

There are basically two types of telephone sets, such as, Local battery (LB) system and central
battery (CB) system. In LB system, the polarizing battery required for the telephone is provided
in the telephone set itself. On the other hand, this potential is provided by a battery located at
the exchange in CB system. The use of former system is restricted to isolated areas only, while
the latter system is very popular and employed in most of the applications.
Fig.: Subscribers telephone set of a LB exchange
Side-tone: The voice frequency generated by the transmitter also passes through the receiver.
Hence, the speaker will also hear his own voice. This sound which a speaker hears in his
receiver originating from his own transmitter is known as the side-tone. Some interesting
points about the side-tone are as follows:
i) A complete absence of side-tone is undesirable as then the telephone appears to be dead
to the speaker, which is a very uncomfortable feeling.
ii) Too much side-tone, on the other hand, makes the speaker voice lower unwillingly. This
reduces the transmission efficiency and may cause inconvenience to the distant listener.
iii) In a noisy place, the noise appears as side-tone and interferes with the listening to the
distant speaker.
iv) The desirable amount of side-tone is that much which speaker would have heard in a
free air conversation with the other party across the table.
A circuit arrangement by which side-tone can be considerably reduced is known as an anti
side-tone circuit.
Fig.: Anti side-tone circuit in a LB telephone
A transformer is most commonly employed for reduction of side-tone. The connections of this
transformer commonly termed as Anti-Side-Tone Induction Coil (ASTIC) for LB telephone.
The AF current produced by the microphone completes its path through L2. As a result of this
current, emf is induced in the windings L1 and L3. This entire voltage is fed into the line as AF
signal. AF voltage developed across L1 produces side-tone in the receiver minimum. Thus,
reduced side-tone is obtained in LB telephones using this simplified circuit.
Fig.: A modern CB telephone set
In modern CB system, battery may utilize manual or automatic switching at the exchange. The
telephone set has a pair of contacts GS1 and GS2 operated by cradle switch. In addition, there
are 3 contacts D1 to D3 operated by the dial. When the set is at rest, the contact GS1 and GS2
are opened and there is no DC path across the line but bell is connected across the line in order
that incoming calls may be received. On lifting of handset, contacts D1 to D3 and GS1 and GS2
being closed permitting energizing current to flow through microphone. Current from
microphone divides in coils L1 and L2. If the line impedance equals the impedance of the
balancing network comprising R1, C1 and R2, C2 then equal and opposite current flows in L1
and L2 and a very small voltage is induced in the coil L3. This ensures to keep the side-tone
level minimum.
Fig.: Anti side-tine circuit in a CB telephone
2.1.5 Tone Dialing

While the pulse dialing is the most common method of dialing, some telephone sets employ
dual tone multi-frequency (DTMF) for dialing. This method can be used only if the telephone
exchange is equipped to process such calls. Instead of rotary dials, these telephones employ
push button key pads as shown in the below Fig.
Fig.: Frequencies used in tone dialing

Usually, a keypad with 12 keys representing numbers 0 through 9 and the symbols * and # are
employed bust some special purpose telephones have fourth column of keys as shown in the
above Fig.
When a key is pressed, the electronic circuit generates two tones corresponding to that key. For
example, if we press key 8, tones of 852 and 1336 Hz will be generated.
Advantages of DTMF:
i) Dialing is very fast as compared to pulse dialing
ii) It uses solid state circuit for tone generation/detection.
iii) After the call that has been connected, it can be used for low speed data
transmission.
iv) It is more compatible with electronically controlled exchanges.
2.1.6 Rotary Switches

The main parts of a step-by-step exchange are the uniselectors and two-motion selectors. Both
these types of selectors belong to the same class of switches which can be conveniently called
rotary switches. The ordinary switches have one, two, or more poles which can be thrown at
best in two different positions. A pole in a rotary switch, on the other hand, can be thrown in
more than two positions. The moving contact which is known as a pole in a toggle switch is
called a wiper in a rotary switch. The different conductors with which the wiper makes contact
are called bank contacts. More often there are more than one wiper moving on more than one
set of bank contacts. In such cases, the drum on which the weepers are mounted along with the
wipers is known as wiper assembly and the set of contacts served by one wiper is known as one
bank.
Uniselector: A uniselector is a rotary switch where the wiper assembly moves only in one
direction.
Two-motion selector: A two-motion selector is a rotary switch whose wipers are capable of
horizontal as well as vertical motion. During the vertical motion the wipers do not make any
contact, but simply move upward. It is during the horizontal movement that the wipers actually
touch the bank contact. In the normal or low position, the wipers rest just below the 1 st level.
Thus their horizontal movement on the bank contact is always preceded by a vertical
movement.
Fig. 2.48 of Ref: Anokh Singh
Necessity of Rotary switch: As we know that, the rotary switch is composed by two selectors,
such as, uniselector and two-motion selector. Every subscriber has a uniselector switch which
connects a subscriber to the desired line when he dials the number. If uniselector switches with
25 contact points are employed, twenty-five subscribers could be served and the system would
be termed as a 25 line telephone exchange.
The system could be extended to form a 100 line telephone exchange using two-motion
selectors each having 10 levels of 10 bank contacts. Numbering in such a system involves two
decimal digits; the first digit dialed moves the selector wiper assembly vertically upwards to the
required level while the second digit dialed connects the wipers to the called subscriber.
The rotary switches as used in a step-by-step exchange satisfy the following conditions:
The usual driving mechanism of a rotary switch is the pawl and ratchet mechanism, which
shown in the following figure. The pawl which is rigidly fixed with the armature can engage
with the ratchet as shown in Fig. (A). In this method, when the driving magnet is energized due
to current flowing through its winding, the armature is attracted and the pawl slips over one
tooth of the ratchet wheel. The detent prevents any movement of the ratchet in anti-clockwise
direction. When current ceases to flow through the driving magnet winding, it is de-energized.
The armature comes back to the normal position due to restoring action of the spring. During
this reverse motion of the armature and thereby of the pawl, the ratchet wheel moves in
clockwise direction by one tooth and the wipers moves to the next contact. The play of the
armature is so adjusted that during the forward motion of the armature the pawl slips just over
one tooth. It should also be seen that during half a revolution of the ratchet wheel the wiper
moves over all the contact.
Fig. (B) shows the arrangement of the pawl when the drive is of the forward acting type. It is
evident that the ratchet wheel moves during the forward movement of the armature. This type
of drive is utilized for both vertical and rotary stepping of two-motion selectors.
Fig. 3.2 of Ref: N. N. Biswas with descriptions
2.1.7 Trunk Principle

1000-line exchange:
In a 1000-line exchange all the numbers are of three digits, and they are divided into 10
groups as follows:
Groups Numbers
1. 100-199
2. 200-299
3. 300-399
4. 400-499
5. 500-599
6. 600-699
7. 700-799
8. 800-899
9. 900-999
10. 000-099
In this scheme shown in the below Fig. each subscriber is provided with a uniselector. At the
outlet of the uniselector are connected two-motion selectors known as group selectors. As soon
as a subscriber lifts his handset, his uniselector moves from the home position and stops on an
outlet to which a free group selector is connected. This group selector then responds to the
first digit dialed by making that number of vertical steps. Immediately, after the vertical
stepping is completed, the group selector automatically makes horizontal stepping and stops on
a contact of the level where another free two-motion selector is connected. These two-motion
selectors which are connected at the group selector outlets are known as final selectors.
Suppose, this final selector is connected to the 8 th level of the group selector. Then, the 100
subscriber terminated at the 100 outlets of the final selectors are those of the 8 th group. So, the
numbers of the subscribers available at this final selector are from 800 to 899. Hence, the
subscriber must have dialed 8 as the first digit to get access to this final selector through the 8 th
level of the preceding group selector stage. When he dials the second digit, the final selector
makes that number of vertical steps and stops. When the third digit is dialed, the final selector
makes that number of rotary steps and the call is established.
Fig. 3.8 of Ref: NN Biswas with descriptions
10,000-line exchange:
In 10,000-line exchange all the numbers are of 4 digits. The trunking principle shown in the
below Fig. is an extension of the principle employed for 1000-line exchange. In this case, two
stages of group selectors are utilized to select the first two digits. The last two digits are
selected by the final selector.
Fig. 3.9 of Ref: NN Biswas with descriptions

2.1.8 Traffic and Trunking

Calling rate: The average number of calls a subscriber makes per hour is termed as calling
rate.
Busy hour: The period during which maximum calls are originated during the day in an
exchange is called busy hour.
Busy hour calling rate: The average number of calls originated per subscriber during the busy
hour is termed as the busy hour calling rate. This is an important factor in the design of an
automatic exchange and its usual value lies between 0.8 to 1.1.
Problem-1:
In a teleprinter exchange serving 5000 subscribers, the total number of calls in a busy hour is
6000. If the holding time is 2.5 minutes, find (i) calling rate, (ii) rate of flow of traffic.
Solution:
The total number of calls in busy hour=6000
Number of subscribers=5000
Therefore, busy hour calling rate=
The rate of traffic flow, A=C×T

Here, C is the number of calls originating per hour and T is the average duration of calls in
hours.
A= traffic units (Ans)
Problem-2:
In an exchange with 10,000 lines, the total number of calls in the busy hour is found to be
16000 with an average holding time of 2 min of 36 sec. Calculate the calling rate and the traffic
flow rate.
Solution:
The total number of subscribers=1000
Total number of calls in busy hour=16000
Busy hour calling rate= (Ans)
Holding time, T=2 min 36 sec= 2.6 min

There, the rate of traffic flow, traffic units (Ans)
Problem-3:
If 1000 subscribers originate 60 Erlangs of traffic in the busy hours of a telephone exchange
with an average holding time of 2.4 min, what is the busy hour calling rate per subscriber? If
these subscribers lose 30 calls in the busy hour, what is the grade of service?
Solution:
Number of subscribers=1000
Holding time, T=2.4 min=0.04 hrs
Traffic flow rate, A=60 Erlangs
We know that, A=C×T
Therefore, Number of calls per hour, C=A/T=60/0.04=1500.
We know, busy hour calling rate=
It is noted that, the ratio of calls lost to the total number of calls initiated in a busy hour
determined the grade of service for that switching stage.
(Ans)
Chapter-4
The key issue in evaluating the performance of a digital communication system concerns the
efficiency with which information from a given source can be represented. Another key issue
relates to the rate at which information can be transmitted reliably over a noisy channel. The
basis of these aspects depends on the information theory. Such theory is originally known as the
mathematical theory of communication. It deals only with mathematical modeling and analysis
of a communication rather than with physical sources and physical channels.
4.1 Uncertainty, Information, and Entropy

The entropy of a source is a function of the probabilities of the source symbols that constitute
the alphabet of the source. Since entropy is a measure of uncertainty, the entropy is maximum
when the associated probability distribution generates maximum uncertainty.
Suppose that, a probabilistic experiment involves the observation of the output emitted by a
discrete source every unit of time. The source output is modeled as a discrete random variable,
S, which takes on symbols from a fixed finite alphabet:
with probabilities:
Of course, this set of probabilities must satisfy the condition, .
By getting a derivation it is obtained that,
Here, is called the entropy of a discrete memoryless source with source alphabet . It
is a measure of the average information content per source symbol. Note that, the entropy
depends only on the probabilities of the symbols in the of the source.
It is assumed that, the symbols emitted by the source during successive signaling intervals are
statistically independent. A source having the properties just described is called a discrete
memoryless source. Here, memoryless in the sense that the symbol emitted at any time is
independent of previous choices.
4.1.1 Properties of Entropy

Considering a discrete memoryless source whose mathematical model is defined by Eqs. 4.1-
4.2. The entropy of such a source is bounded as follows:
Where, K is the number of symbols of the alphabet of the source. Thus, we may state that,
i. if and only if the probability for some k, and the remaining
probabilities in the set are all zero. This lower bound on entropy corresponds to no
uncertainty.
ii. if and only if for all k (i.e., all the symbols in the alphabet
are equiprobable). This upper bound on entropy corresponds to maximum
uncertainty.
4.1.2 Extension of a Discrete Memoryless Source

In discussing information theoretic concepts, it is useful to consider blocks rather than

individual symbols, with each block consisting of n successive source symbols. We may view
each such block as being produced by an extended source with a source alphabet that has
distinct blocks, where K is the number of distinct symbols in the source alphabet of the
original source.
Thus, the entropy of the extended source, is equal to n times the entropy of the original
source. That is we may write,
Problem-1:
A discrete memoryless source with alphabet with respective probabilities
Verify the extension property of entropy. Here, the order of extension source
is 2.
Solution: We know that, the entropy of the source is,
According to this problem,
bits
Here, alphabet particulars of 2-order extension of a discrete memoryless source are

mentioned below:
Symbols of
Corresponding
sequences of
symbols of
Probability
i=0,1,
….,8
Now,
Thus, (verified)
4.1.3 Source Coding Theorem

An important problem in communications is the efficient representation of data generated by a
discrete source. The process by which this representation is accomplished is called source
encoding. The device that performs the representation is called source encoder. A source
encoder to be efficient, the knowledge of statistics of the source is required. In particular, if
some source symbols are known to be more probable than others, then we may utilize this
feature in the generation of a source code by asserting short cord-words to frequent source
symbols, and long code-words to rare source symbols. Such a source code is referred as a
variable-length code. The Morse code is an example of a variable-length code. In the Morse
code, the letters of the alphabet and numerals are encoded into streams of marks and spaces,
denoted as dots “.” and dashes “-”, respectively.
For example, in the English language, the letter E occurs more frequently than the letter Q, the
Morse code encodes E into a single dot “.”, the shortest code-word in the code, and it encodes
Q into “--.-”, the longest code-word in the code. An efficient source encoder needs to be
satisfied the following requirements:
i) The code-words produced by the encoder are in binary form.
ii) The source code is uniquely decodable, so that the original source sequence can be
reconstructed perfectly from the encoded binary sequence.
4.1.4 Channel Coding Theorem

The expected presence of noise in a channel causes errors between the output and input data
sequences of a digital communication system. In order to achieve a high level of performance,
channel coding is necessary. Particularly, channel coding increases the resistance of a digital
communication system to channel noise. Channel coding consists of mapping the incoming
data sequence into a channel input sequence and inverse mapping the channel output sequence
into an outgoing data sequence in such a way that the overall effect of channel noise on the
system is minimized.
The first mapping operation is performed in the transmitter by means of an encoder, whereas
the inverse mapping operation is performed in the receiver by means of a decoder as shown in
the below Fig.
The channel coding theorem for a discrete memoryless channel is stated in two parts as follows:
i) Let a discrete memoryless source with an alphabet have entropy and produce
symbols once every Ts seconds. Let a discrete memoryless channel have capacity C and
be used once every Tc seconds. Then if there exists a coding scheme for
which the source output can be transmitted over the channel and be reconstructed with
an arbitrarily small probability of error. The parameter C/Tc is called the critical rate.
When the above condition is satisfied with the equality sign, the system is said to be
signaling at the critical rate.
ii) Conversely, if it is not possible to transmit information over the channel
and reconstruct it with an arbitrarily small probability of error.
Problem-4:
Calculate the capacity of a standard 4-kHz telephone channel with a 32 dB signal-to-noise ratio.
Solution: The actual signal-to-noise ratio= antilog (32/10) = antilog (3.2) =1585.
We know,
Problem-5:
A system has a bandwidth of 4-kHz and a signal-to-noise ratio of 28 dB at the input to the
receiver. Calculate:
(a) Its information carrying capacity
(b) The capacity of the channel if its bandwidth is doubled, while the transmitted signal power
remains constant.
Solution:
(a) S/N=antilog(28/10)=antilog(2.8)=631
(Ans)
(b) If S/N in the 4-kHz channel is 631:1, this can be interpreted as a noise power of 1 mW at
some point in the channel where the signal power is 631 mW. It is noted that, when the
bandwidth is doubled in a system, the signal power is unchanged here, but the noise power
is doubled.
Therefore,
(Ans)

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Telecommunication-I; Course No.

Fig. 1.1: Communication system.

A typical communication system is modeled in Fig. 1.1. The components of a communication

4.1 Signal Energy

This definition can be generalized to a complex valued signal f(t) as

4.2 Signal Power

This definition can be generalized to a complex valued signal f(t) as,

Fig. 3.2: Signal curves.

b) Analog and digital signals

5.1 Continuous-time and discrete-time signals

5.2 Analog and digital signals

Fig. 4.2: A periodic signal of period T0.

Fig. 4.3: Aperiodic signals.

5.4 Energy and Power signals

5.5 Deterministic and Random signals

6.1 Time Shifting

Fig. 5.1: A signal of time shifting.

6.2 Time Scaling

Fig. 5.2: Time scaling a signal.

6.3 Time Inversion

Fig. 5.3: Time inversion of a signal.

7. Some Useful Signal Models

7.1 Unit Step Function

Fig. 6.1: Unit step function u(t).

Fig. 6.2: Representation of rectangular pulse by step functions.

7.2 Unit Impulse Function

Fig. 6.3: A unit impulse and its approximation.

7.3 Multiplication of a Function by an Impulse

Similarly, if (t) is multiplied by an impulse (t-T) (impulse located at t = T), then

Here, (t) is continuous at t = T.

7.4 Sampling Property of the Unit Impulse Function

By the integration of Eq. (6.3), we get,

3.1 Signal Transmission through a Linear System

3.1.1 Signal Distortion during Transmission

Eq. (3.2) can be expressed in polar form as,

During transmission, the input signal amplitude spectrum is changed to

3.1.2 Distortionless Transmission

During the transmission of signal amplification or message signal over a communication

Example: Fourier wave forms from the original signals.

3.1.3 Intuitive Explanation of the Distortionless Transmission Conditions

3.1.4 Nature of Distortion in Audio and Video Signals

Fig. 3.2: Sample RC filter.

As per the notion, the transfer function,

We know, time delay,

gain and time delay both at maximum when and

almost unity (see Fig. 3.2(a)).

Fig. 3.4: A pulse passes through a system that is not distortionless.

According to the Time-Shifting property, we know that,

Here, the output is actually delayed by

3.1.5 Distortion Caused by Multipath Effects

The overall transfer function of such a channel is H(), given by

As we know, the polar form of

Fig. 3.5: Multipath transmission.

Eq. (3.13) shows that at frequencies where then

These frequencies are multipath null frequencies. On the other hand, at

frequencies the two signals interfere constructively to enhance the gain.

3.1.6 Signal Energy

3.1.7 Energy Spectral Density(ESD)

Fig. 3.6: Interpretation of the energy spectral density of a signal.

of the output y(t) is,

Estimate the essential bandwidth of a rectangular pulse shown in Fig. 3.7,

Fig. 3.7: Gate function and its energy spectral density.