M01 - Communication Basics

TC31009
Certified Wireless Maintenance Professional
MODULE - 01
COMMUNICATION BASICS
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Communication Basics Module - 01
Module - 01
Communication Basics
INDEX
Chapter Topic Pages
Transmission
01 Transmission Units and Impairments 4 - 18

02 Modulation and Multiplexing 19 – 47
03 Transmission Media 48 – 60
04 Optical Fiber Communication 61 – 79
05 Optical Fiber Cable Splicing 80 – 97
06 OFC Test and Measurements 98 – 104
07 Optical Connectors and Couplers 105 – 115
08 Coding Theory 116 – 125
09 Basics of Transmission Systems 126 - 147
Switching
10 Introduction to Electronic Exchanges 148 – 174

11 Switching Concepts 175 – 180
12 Digital Switching 181 – 192
13 Signaling in Telecom Systems 193 – 219
14 ISDN 220 – 225
15 Telephone Traffic Engineering 226 – 247
16 Long Distance Switching Plans 248 – 291
17 Network Management 292 - 303
TC31009D10 2
Teleman Certified Wireless

Course Name Maintenance Professional
(TCWMP)
Course Code TC31009
Document Code TC31009D10
Module No. Topic No. of Chapters
01 / 05 Communication Basics 17
Issue No. Revision No. Issue Date Effective Date
1 0 15-12-2008 01-01-2009
Teleman (A Division of Plug-n-Work Directions)

January 2009
Copyright © P-n-W 2009, All rights reserved
No part of this publication may be reproduced, stored in retrieval system or transmitted in any form or by any
means, electronic, mechanical, photocopying, recording or otherwise, without the prior permission of the
publisher.
Trademark acknowledgements:
All products are registered trademarks of their respective organisation.
TC31009D10 3
Module - 01
Chapter – 01
Transmission Units and Impairments
Session Objectives:
On completion of this session, you will be able to understand the

concepts and able to explain:
• Transmission Units
• Transmission Impairments
• Decibel and Neper
• Basic derived decibel units
• Signal-to-Noise Ratio
• Digital Transmission - Performance Criteria
• Cross-Talk in Transmission Media
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1. Historical overview
Telecommunication systems started with the transmission of

digital signals. In fact, non–electric signaling systems date back over
2000 years. The Greek General Polybius is known to have used a scheme
based on an array of 10 torches in 300 B.C. and Roman armies made
extensive use of a form of semaphore signaling. Claude Chappe,
Sommering, Whetstone and Cook were all experimenting with different
kinds of Telegraphy till it was perfected by Mores. In all this, only written
message was transmitted and message was converted to a coded signal
to match the characteristics of a transmission line. Gary, Baudot and
others developed other codes, which were mainly used in Telegraph
network. Thus, we can say, by 1972 most of the basic techniques of
digital transmission had been discovered.
In 1876, Alexander Graham Bell invented the Telephone and as

means of communication, the telephone was fast, personal and
convenient. It needed no training in the use of codes and so made
electrical communications directly accessible to the general public. Thus,
telephone began to dominate the development of communications.
Telephony involves the transmission of analog signals and when a
practical amplifying service appeared in the form of the thermionic valve,
this also proved suitable for dealing with analog signals. Hence, after
1880, the developing Telecom networks were basically designed to handle
analog transmissions and to an increasing extent, the digital
transmission in the form of telegraphy had to be adopted to fit in with
the characteristics of these networks. By 1950s, the world's
communications systems were based entirely on analog transmission.
However, interest in the digital transmission received an impetus

after the publications of classic papers of Nyquist and Shannon. With the
invention of pulse code modulation by Reeves in 1938, the basic
principles for digitising analog speech signals were established. However,
the technical means for transmitting digitized speech signals were not
available at that time. It was not until the transistor came into use that
indications of the economic advantages of digital techniques as compared
to analog methods became apparent. LSI and VLSI techniques that are
now available have made digital communications far more economical as
compared to analog methods became apparent. LSI and VLSI techniques
that are now available have made digital communications far more
economical as compared to analog systems.
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Digital transmission systems are gaining more acceptances in view

of:
• Introduction of digital switching systems
• The need to transmit non-voice signals which are increasingly
becoming important instead of the plain old Telephone service.
• The introduction of new media like optical fibers, wave-guide
that are more suitable for digital transmission systems, will be
introduced in the network and by the turn of the century, most
of the countries would have gone completely digital.
2. Transmission Units
The study of transmission units has a unique importance for

communication engineer who has to maintain and install
telecommunication equipments achieving the standards set up by
international consultation committees.
In order to control the quality of wanted signal in the presence of
many undesired signals, we should be able to specify the amount of
wanted and unwanted signals at a point in the telecommunications
network.
The components used in the telecommunication circuit either give
loss or gain to the signals they handle. There are certain specific
operating conditions to be satisfied for various components without
which the optimum performance cannot be obtained from these
components. For this, it is essential to define conditions that control
those operating conditions. This can be done only if the conditions are
specified in terms of certain units of the quantity the components are to
handle.
3. Transmission Impairments
With analog transmission systems using copper cable there are

three major categories of impairments. They are attenuation, noise, and
distortion.
• Attenuation: There are two commonly used processes to

compensate (overcome) for attenuation or loss:
(a) Repeaters are the most commonly used devices to compensate

for "Loss." However, repeaters amplify the noise along with the
signal resulting in a poor signal to noise ratio.
(b) Signal to Noise Ratio: The ratio of the average signal power
(strength) to the average noise power (strength) at any point in a
transmission path.
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• Noise: Any random disturbance or unwanted signal on a

transmission facility that obscures the original signal. The
environment in which the system is operating generally causes
noise.
• Distortion: Inaccurate reproduction of a signal caused by changes
in the signal's waveform, either amplitude or frequency, to
compensate for distortion equalizers may be used. One type of
equalizer used in the analog environment is the load coil. Load
coils are used to flatten the frequency response.
The higher voice frequencies attenuate at a higher rate than the

lower voice frequencies.
4. Decibel and Neper

Historically speaking ‘attenuation’ was first of all defined in terms
of the attenuation produced by a standard reference cable known as
“mile of standard cable”. It consists of 88 ohms series impedance and
0.54 µF as shunt impedance.
The fundamental objection to this unit was the fact that the
attenuation of the standard cable varied with frequency. With the
introduction of systems operating over different frequency ranges, it
became necessary to define a unit which was independent of frequency
.The unit which represents the useful and convenient concepts in
connection with the transmission of signals over telephone lines has
been named and defined as “Bel” (which comes from the name Alexander
Graham Bell -the inventor of Telephone). In practice, however, a smaller
and more convenient unit called decibel (abbreviated as dB) is used.
DECIBEL (dB)
One tenth of the common logarithm of the ratio of relative powers,

equal to 0.1 B (bel). The decibel is the conventional relative power ratio,
rather than the bel, for expressing relative powers because the decibel is
smaller and therefore more convenient than the bel. The ratio in dB is
given by
X = log P2/P1 B i.e. = 10 log P2/P1 dB
Where P 1 and P 2 are the actual powers. Power ratios may be expressed
in terms of voltage and impedance, E and Z, or current and impedance, I
and Z. Thus dB is also given by;
X = 20 log V2/ V1 dB. (When Z 1 =Z 2 )
TC31009D10 7
Note: The dB is used rather than arithmetic ratios or percentages

because when circuits are connected in tandem, expressions of power
level, in dB, may be arithmetically added and subtracted. For example, in
an optical link if a known amount of optical power, in dBm, is launched
into a fiber, and the losses, in dB, of each component (e.g., connectors,
splices, and lengths of fiber) are known, the overall link loss may be
quickly calculated with simple addition and subtraction.
Example 1
Let us look at the following network:
Net Work
1W 2W
The input is 1W and its output 2W, therefore,

Gain = 10 log (output)/ (input) dB.
= 10 log 2/1 dB= 10 (0.3010) dB=3.010 dB
= 3dB approximately
Example 2
Let us look at another network:
Net Work
1000 W 1W
Loss = 10 log Input/Output =10 log 1000/1 dB =10 log 103

dB
=30 log 10 dB
= 30 dB
Thus a network with an input of 5 W and output of 10 W is said to have
Gain = 10 log 10/5 dB
= 10 log 2 dB
=3.103 dB
= 3 dB.
Let us remember that doubling the power means a 3 dB gain; likewise
halving the power means a 3dB loss.
Example 3
Consider a network with a 13 dB gain:
0.1W Network 13 db gain ?
Gain = 10 log P2/P1 dB = 10 log P2/0.1 dB =13db

i.e., log P2/0.1 = 1.3 or P2/0.1 = antilog 1.3 or
P2 = 0.1 antilog 1.3
P2 = 2W
TC31009D10 8
Example 4
Consider the following network
1W Network 27 dB loss ?W
What is the power output of this network?

To do this without pencil and paper, we would proceed as follows:
• Suppose the network attenuated the signal by 30 dB. Then the
output would be 1/1000 of the input or 1mW.
• Now 27 dB loss is 3dB less than 30 dB.
• Thus the output would be twice 1m W i.e., 2mW.
• (Because the loss is less by 3 dB, the corresponding output will
be more i.e. double but not half)
• It is quite simple. Thus, if we have multiples of 10 or 3 up or 3
down from these multiples, we can work it out in our mind
without pencil and paper.
NEPER
The natural logarithm of the ratio of two voltages (or currents)
expresses the loss or gain in Nepers, N
i.e. X= loge V1/V2 (N)
When the loss (gain) is X Neper, V1 and V2 are voltages, then
ex = V1/V2
Example
The loss of a transmission system is 1N when 2.72 V input voltage
produces 1 V output voltage.
Comparing powers, X= 1/2 loge P1/P2 (N) or

e2x = P1/P2
Other transmission units 1 deciNeper (dN) = 0.1 N

1 Centi Neper (cN) = 0.01N
1 MilliNeper (mN) = 0.001 N
5. Basic derived decibel units
DBm
Till now decibel has referred to ratios or relative units. We cannot
say that the output of an amplifier is 33 dB. We can say that an amplifier
has a gain of 33 dB or that a certain attenuator has a 6 dB loss. These
figures or units don't give any idea whatsoever of absolute level.
Whereas, several derived decibels units do. Perhaps the dBm is the most
TC31009D10 9
common of these. By definition dBm is a power level related to 1 mw. The

most important relationship to remember is:
0 dBm = 1mW.
The dBm formula may then be written as:

Power (in dBm) = 10 log Power (mW)/(1mW)
Example
An amplifier has an output of 20 W; what is its output in dBm?
Power (dBm) = 10 log 20 W/1 mW = 10 log 20x103 mW/1mW = +43 dBm.

(The plus sign indicates that the quantity is above the level of reference,
0 dBm.)
dBmO
Decibel referred to 1 mw at zero (0) Transmission level point.
dBmO is a measure of power with reference to Zero dBm at the Reference
Transmission Level Point (RTLP).
The RTLP is also known as Zero Transmission Level Point (0TLP).
Powers measured at any transmission level point can be expressed in
dBmO, by correcting the power measured for the difference in level
between the point of measurement and the RTLP.
For example, a level of +25 dBm measured at a +17 dB
transmission level point is equivalent to 8 dBmO. Conversely a level of +8
dBmO is also equivalent to +3 dBm measured at a -5 dB transmission
level point. A level expressed in dBmO is, therefore, only a relative level.
Conversion from Neper to decibel and Vice Versa
We know that decibel is fundamentally a unit of power ratio but it

can be used to express current ratios when the resistive components of
the impedance, through which the current flows, are equal.
The Neper, on the other hand, is fundamentally a unit of current
ratio but it can also be used to express power ratios when the resistive
components, of the impedance, through which the current flows, are
equal.
Because of its derivation from the exponential e, the Neper is the
most convenient unit for expressing attenuation in theoretical works. The
decibel, on the other hand, being defined in terms of logarithms to base
10, is a more convenient unit in practical calculations using the decimal
system.
TC31009D10 10
The conditions under which the two units may be used can be
summarised in the following equations, the notation of which is indicated
in Fig below.
Where Z1 and Z2 are characteristic impedances

R1 and R2 are pure resistances
G1 and G2 are leakances
β1 and β 2 are phase angles
X1 and X2 are reactance.
Atténuation in decibels = 10 log10 |PS/PR |.................. (i)

Attenuation in Nepers = log e| IS/IR|.......................................... (ii)
Equation (i) can be expressed as current ratio and also as voltage ratio as
follows
Attenuation in dB = 20 log10 IS/IR if R1= R2 (iii)

= 20 log10 ES/ER if G1 = G2 (iv)
Attenuation in Nepers = log e ES/ER if Z1 =Z2
(v)
=1/2 log e PS/PR (if R1=R2 (vi)
If the resistive components of the impedance at the input and

output of the network are equal, then the attenuation may be readily
converted from one notation to another, for:
Attenuation in dB = 20 log10 IS/IR
= 20 loge IS/IR log10 e (log10 e= .4343)
= 8.686 loge IS/IR
= 8.686 x (attenuation in Nepers)
Thus,
Attenuation in dB = 8.686 x attenuation in Nepers
(vii)
(provided that R1 = R2)
Attenuation in Nepers = 0.1151 x attenuation in dB (viii)
(provided that R1 =R2)
TC31009D10 11
Other Units
In Analogue Transmission system, the quality of communication is

mainly assessed by the value of Signal to noise ratio.
6. Signal-to-Noise Ratio
It is popularly known as SNR. SNR is the ratio of signal power to

the noise power at any point in a circuit. This ratio is usually expressed
in Decibels (dB). For satisfactory operation of a channel the value of SNR
should be sufficiently high i.e., the signal power should be sufficiently
higher than the noise power.
SNR at any point in a circuit is given as:
SNR = S/N = Signal Power / Noise Power
Both powers are expressed in watts.
Expressing dBs: SNR = 10 log10 (S/N) dB.
Example: Signal voltage Vs = 0.923 µV; Noise voltage Vn = 0.267 µV, then
calculate the signal-to-noise ratio.
S/N = Vs2 / Vn2 = 0.923/0.267)2 = 11.95
In decibels, S/N = 10 log10 (11.95) = 10.77 dB.
In Digital Transmission system, the quality of communication is

mainly assessed by two factors.
1. BER (Bit Error Ratio)

2. Jitter
These two factors can be taken as Quality Factors as they are used
for judging the quality of Digital Transmission.
Bit Errors
In the digital transmission, the bits transmitted at the transmitting
end (1 or 0 ) are not always detected as 1 or 0 at the receiving end. When
the transmitted bit 1 or 0 is not identified as 1 or 0 at the receiver, the
bit is counted as an error bit.
For assessing the real error performance, the bit error ratio (BER)
is to be calculated instead of actual error bits.
Bit Error Rate (BER)

The BER is the measure or error bits with respect to the total
number of bits transmitted in a given time. The total number of bits
transmitted can be known from the bit rate of the digital signal. The bit
rate is the number of bits transmitted in one second and is specified for
each transmission system. Hence, the total number of bits transmitted in
a given time can be counted. In the measurement of BER, generally the
TC31009D10 12
measuring instrument measures the number of bits transmitted in a

given time.
The time setting can be from a few seconds to a few hours,
depending on the feasibility. The standards are set by ITU (International
Telecommunication Union). The time set for the measurement of BER, is
called gating time. Larger the gating time better is the assessment of
BER. But for the measurement of BER, the Digital Equipment has to be
taken off-line.
Digital communication can just run with one error bit in one
thousand bits received. For more than one error bit, in one thousand bits
received, communication gets affected. For good quality communication,
the requirement is not more than one error bit in one million bits.
Jitter
Abrupt and unwanted variations of one or more signal
characteristics, such as the interval between successive pulses, the
amplitude of successive cycles, or the frequency or phase of successive
cycles. Jitter must be specified in qualitative terms (e.g., amplitude,
phase, and pulse width or pulse position) and in quantitative terms (e.g.,
average, RMS, or peak-to-peak). The low-frequency cut-off for jitter is
usually specified at 1 Hz. Contrast with drift, wander.
Short-term variations of the significant instances of a digital signal
from their reference position in time. (Short term frequency equal to or
greater than 10 Hz.). Long term variations of significant instances of a
digital signal from their ideal positions in time, are called wander. (Long-
term variations – frequency less than 10 Hz).
Drift
A comparatively long-term change in an attribute or value of a
system or equipment operational parameter. The drift should be
characterized, such as "diurnal frequency drift" and "output level drift."
Drift is usually undesirable and unidirectional, but may be bi-directional,
cyclic, or of such long-term duration and low excursion rate as to be
negligible.
Wander
Relative to Jitter and swim, long-term random variations of the
significant instants of a digital signal from their ideal positions. Wander
variations are those that occur over a period greater than 1 s (second).
Jitter, swim, wander, and drift have increasing periods of variation in
that order.
Digital Transmission Analyser (DTA) is used for the measurement
of both BER and Jitter.
TC31009D10 13
7. Digital Transmission - Performance Criteria (General)
1 in 106 (1.OE – 6) : Better

1 in 105 (1.OE – 5) : Good
1 in 104 (1.OE – 4) : Reasonably good
1 in 103 (1.OE – 3) : Just Acceptable
More than 1 in 103 : Unacceptable
Bit errors greatly affect data service.
For data channels 1 in 109 (1.OE – 9) is normally realizable.
Quality Parameters
To pin point the exact number of seconds or minutes, in which the
bit errors take place and up to what extent, the quality parameters are
defined.
The quality parameters are:
1. Error Seconds (ES)

2. Severely Error Seconds (SES)
3. Non Severely Error Seconds (NSES)
4. Degraded Minutes (DM).
Error Seconds (ES):

Number of one-second intervals with one or more errors.
Severely Error Seconds (SES):

Number of one-second intervals with an error rate, worse than
1.OE-3
Non-Severely Error Seconds (NSES):

Number of one-second intervals with an error rate, better than or
equal to 1.OE-3.
Degraded Minutes (DM):

Number of one-second intervals with a bit error rates worse than
1.OE-6.
Available and non-available time

A period of available time begins with a period of ten consecutive
seconds each of which has a BER better than 1.0E-3. These 10 seconds
are considered to be available time.
TC31009D10 14
A period of unavailable time begins when the bit error rate in each
second is worse than 1.0E-3 for a period of 10 consecutive seconds.
These 10 consecutive seconds are considered to be unavailable time.
8. Cross-Talk in Transmission Media
Any disturbing signal produced by transfer of unwanted power

from one transmission path (called disturbing circuit) to another
transmission path (called disturbed circuit) is known as cross talk.
Cross talk may be produced by:
• Galvanic, capacitive or inductive couplings between

transmission media (Linear cross-talk) e.g. between pairs of a
VF (voice frequency) cable system.
• Poor control of frequency response i.e. defective filters or poor
filter design is the cause.
• Non-linear performance in analogue (FDM) multiplex systems. A
signal transmitted on one circuit or channel of a transmission
system (multi-channel) creates an undesired effect in another
circuit or channel (non-linear cross talk)
Types of cross talk
Broadly speaking, cross talk is of six types.
1. Near-end cross- talk (NEXT).

2. Far-end cross talk (FEXT).
3. Intelligible cross-talk
4. Unintelligible cross-talk
5. Interaction cross-talk
6. Reflected cross-talk
Near-end cross talk (NEXT)
Near-end cross talk occurs if the cross talk power in the disturbed
channel propagates in the direction opposite to the propagation of useful
power in the disturbing channel. Refer to Fig 1.1 for illustration of near-
end cross talk.
The terminals of the disturbed channel, at which the near-end
cross talk is present, and the energized terminal of the disturbing
channel, are usually near each other. The near-end cross talk is much
stronger than far-end cross talk because the magnetic (or galvanic) and
electrostatic inductions are additive in the case of near-end cross talk
and the inducing current in the disturbing circuit is much stronger.
TC31009D10 15
Fig. 1.1 Near End Cross Talk
Far-end cross-talk (FEXT)

It occurs if the cross talk power in the disturbed channel
propagates in the direction of the propagation of the useful power in the
disturbing channel. Refer to Fig.1.2 for illustration of far-end cross talk.
The terminals of the disturbed channel, at which the far-end cross talk is
present, and the energized terminals of the disturbing channel, are
usually remote from each other. Far-end cross talk is less effective in
impairment of the original signal in the disturbed circuit because the
magnetic and electrostatic inductions are subtractive. Also the inducing
current in the disturbing circuit gets very much attenuated after it has
traveled to the far end.
Fig. 1.2 Far End Cross Talk
Intelligible cross talk

The cross talk is intelligible when the whole or an important part,
of the speech on the disturbing circuit is intelligible on the disturbed
circuit. Between circuits transmitting the same frequency band or
working without frequency translation (audio-frequency) only intelligible
cross talk can arise. As the secrecy of the conversation is affected by
TC31009D10 16
intelligible cross talk, steps should be taken to see that intelligibility of

sentence articulation of the cross talk should be less than 10%.
Unintelligible cross talk (also called noise)

The cross talk is unintelligible when the disturbing circuit gives
rise only to noise in the disturbed circuit. It decreases the intelligibility
but does not endanger the secrecy of conversation. Unintelligible cross
talk occurs
• Between carrier channels having different frequency allocations.
• Between carrier channels having virtual carrier frequencies
essentially differing from each other and
• In consequence of non-linear distortion.
Interaction cross talk (Indirect cross talk)

Interaction cross talk conveyed by a third circuit from the
disturbing circuit to the disturbed circuit, where it causes far end cross
talk (fig.3). This type of cross talk is also called double near-end cross
talk. It occurs mainly in two-wire carrier systems fitted with intermediate
repeaters.
Reflected cross-talk
Indirect cross talk caused by reflection due to mismatch of the
circuit is called reflected cross talk.
Causes of cross-talk
Cross talk is mainly caused by two types of induction viz.,
Magnetic and Electrostatic.
Magnetic induction
It is well known that a change in magnetic lines of forces is
associated with the flow of electric currents. The magnetic lines of forces
due to currents flowing through circuit A will also embrace the wires of
circuit B. As the current in circuit A alternates, the magnetic field also
alternates, and according to Faraday' law it induces e.m.fs in the wires of
circuit B
Electrostatic induction
Electrostatic induction occurs due to the capacitance between four
wires of the two circuits that are built side by side.
Practically it is noted that the current due to magnetic induction
flows in one direction in the entire circuit, whereas that due to the
electric induction flows through the two sections in opposite directions
TC31009D10 17
Methods for reducing cross talk

There are a number of methods for eliminating or at least
substantially reducing cross talk in open-wire lines. Since the cross talk
reduction in open wire lines depends upon three factors viz. Wire
configuration, transposition and resistance unbalance. One possible
method is to arrange the wires in such a configuration that the effect of
the electric and magnetic fields of one pair will be the same on both wires
of the disturbed pair, thus leaving no residual difference to cause
currents in the disturbed circuit.
Another method to reduce the cross-talk is to reduce the
separation between the wires of either or both disturbing and the
disturbed pairs and, if possible, to increase the separation between the
pairs themselves.
The most commonly used method is the use of “transpositions".
Transposition means interchanging the position of the two wires forming
the pair at regular intervals on the pole route, right through the length of
the pair. The transposition is by far the most effective and practical
method of reducing cross talk.
TC31009D10 18
Module - 01
Chapter – 02
Modulation and Multiplexing
Session Objectives:

• Need for Modulation

• Analog-to-Analog Modulation
• Analog-To-Digital Conversion
• Pulse Code Modulation
• Digital-To-Analog Conversion
• Multiplexing Techniques
• Digital Hierarchies
• Multiple Access Methods
TC31009D10 19
1. Introduction
The basics of modulation is to take a message bearing signal like
an audio signal and superimpose it upon a carrier signal for
transmission. For ease of transmission such carrier signals use generally
high frequencies:
For easy propagation as electromagnetic waves with low loss

and low dispersion
Simultaneous transmission without interference from other
signals
Enables the construction of small antennas (a fraction, usually
a quarter of the wavelength)
Enables the multiplexing (combining) multiple signals for

transmission at the same time over the same carrier
Different modulation schemes are possible. Well known examples
of high frequency carrier signals are:
• AM radio is 550-1600 KHz
• FM radio is 88 MHz-108 MHz
• TV is 52-88 MHz (channels 1-6), 174-216 MHz (channels 7-12)
and 470-900 MHz (UHF)
• Microwave and satellite signals are of the order of several GHz
• Infra red fiber optic signals are of the order of 200-300 THz.
2. Analog-to-Analog Modulation
Analog-to-analog modulation is the representation of analog
information by an analog signal. Radio, that familiar utility, is an
example of an analog-to-analog communication. The following Fig 2.1
shows the relationship between the analog information, the analog-to-
analog modulation hardware, and the resultant analog signal.
Fig 2. 1 Analog-to-analog modulation
Analog-to-analog modulation can be accomplished in three ways

• Amplitude modulation (AM)
• Frequency modulation (FM)
• Phase modulation (PM)
TC31009D10 20
2.1 Amplitude modulation (AM)

In AM transmission, the carrier signal is modulated so that its
amplitude varies with the changing amplitudes of the modulating signal.
The frequency and phase of the carrier remain the same; only the
amplitude changes to follow variations in the information. The following
fig show this concept works. The modulating signal becomes an envelope
to the carrier.
Fig 2.2 Amplitude modulation
2.2 Frequency Modulation (FM)
In FM transmission, the frequency of the carrier signal is

modulated to follow the changing voltage level (amplitude) of the
modulating signal. The peak amplitude and phase of the carrier signal
remain constant, but as the amplitude of the information signal changes,
the frequency of the carrier changes correspondingly. The following Fig
2.3 shows the relationships of the modulating signal, the carrier signal,
and the resultant FM signal.
TC31009D10 21
Fig 2.3 Frequency modulation
2.3 Phase Modulation

Due to simpler hardware requirements, phase modulation (PM) is
used in some systems as an alternative to frequency modulation. In PM
transmission, the phase of the carrier signal is modulated to follow the
changing voltage level (amplitude) of the modulating signal. The peak
amplitude and frequency of the carrier signal remain constant, but as
the amplitude of the information signal changes, the phase of the carrier
changes correspondingly. The analysis and the final result (modulated
signal) are similar to those of frequency modulation.
3. Analog-To-Digital Conversion
We sometimes need to digitize an analog signal. For example, to
send human voice over a long distance, we need to digitize it since digital
signals are less prone to noise. This is called an analog-to-digital
conversion or digitizing an analog signal. The following Fig 2.4 shows the
analog-to-digital converter, called a codec (coder-decoder).
Fig 2.4 Analog-to-digital conversion
TC31009D10 22
In analog-to-digital conversion, we are representing the

information contained in a continuous wave form as a series of digital
pulses (1s or 0s).
3.1 Pulse Amplitude Modulation (PAM)

The first step in analog-to-digital conversion is called pulse
amplitude modulation (PAM). This technique takes an analog signal,
samples it, and generates a series pulses based on the results of the
sampling. The term sampling means measuring the amplitude of the
signal at equal intervals.
The method of sampling used in PAM is more useful to other areas
of engineering than it is to data communication. However, PAM is the
foundation of an important analog-to-digital conversion method called
pulse code modulation (PCM).
Fig 2.5 PAM
In PAM, the original signals sampled at equal intervals as shown

above. PAM uses a technique called sample and hold. At a given
moment, the signal level is read, and then held briefly. The sampled
value occurs only instantaneously in the actual waveform, but is
generalized over a still short but measurable period in the PAM result.
The reason PAM is not useful to data communications is that,
although translates the original wave form to a series of pulses, these
pulses are still of any amplitude (still an analog signal, not digital). To
make them digital, we must modify them using pulse code modulation
(PCM).
4. Pulse Code Modulation (PCM)
PCM modifies the pulses created by PAM to create a completely

digital signal. To do so, PCM first quantizes the PAM pulses.
Quantization is a method of assigning integral values in a specific range
to sampled instances. The result of quantization is presented in the Fig.
2.6.
TC31009D10 23
Fig 2.6 Quantized PAM signal
The binary signals are then transformed into a digital signal using
one of the digital-to-digital encoding techniques. The following Fig 2.7
shows the result of the pulse code modulation of the original signal
encoded finally into a unipolar signal. Only the first three sampled
values are shown.
Fig 2.7
4.1 From analog signal to PCM digital code
PCM is actually made up of four separate processes

• PAM
• Quantization
• Binary encoding
• Digital-to-Digital encoding
TC31009D10 24
The following Fig shows the entire process in graphic form.
Fig 2.8 (a) From Analog to PCM
Fig 2.8 (b) PCM Frame structure
TC31009D10 25
4.2 Sampling Rate

The accuracy of any digital reproduction of an analog signal
depends on the number of samples taken. Using PAM and PCM, we can
reproduce the waveform exactly by taking infinite samples, or we can
reproduce the barest generalization of its direction of change by taking
three samples. According to the Nyquist theorem, to ensure the accurate
reproduction of an original analog signal using PAM, the sampling rate
must be at least twice the highest frequency of the original signal. So if
we want to sample telephone voice with maximum frequency 4000 Hz,
we need a sampling rate of 8000 samples per second.
5. Digital-To-Analog Conversion
Digital-to-analog conversion or digital-to-analog modulation is the
process of changing one of the characteristics of an analog signal based
on the information in a digital signal (0s and 1s). The following Fig 2.9
shows the relationship between the digital information, the digital-to-
analog modulating hardware, and the resultant analog signal.
Fig 2.9 Digital-to-analog modulation
The following are the techniques adapted for digital-to-analog modulation
Fig 2.10 Types of digital-to-analog encoding
TC31009D10 26
5.1 Aspects of Digital-to-Analog conversion
Bit Rate and Baud Rate
Two terms used frequently in data communication are bit rate and
baud rate. Bit rate is the number of bits transmitted during one second.
Baud rate refers to the number of signal unit per second that are
required to represent those bits. The baud rate determines the
bandwidth required to send the signal. Bit rate equals the baud rate
times the number of bits represented by each signal unit. The baud rate
equals the bit rate divided by the number of bit represented by each
signal shift. Bit rate is always greater than or equal to the baud rate.
Carrier signal
In analog transmission, the sending device produces a high-

frequency signal that acts as a basis for the information signal. This
base signal is called the carrier signal or carrier frequency. The receiving
device is tuned to the frequency of the carrier signal that it expects from
the sender. Digital information is then modulated on the carrier signal
by modifying one or more of its characteristics (amplitude, frequency,
and phase). This kind of modification is called modulation (or shift
keying) and the information signal is called a modulating signal.
5.2 Amplitude Shift Keying (ASK)
In amplitude shift keying (ASK), the strength of the carrier signal is

varied to represent binary 1 or 0. Both frequency and phase remain
constant while the amplitude changes. Which voltage represents 1 and
which 0 is left to the system designers. A bit duration is the period of
time that defines one bit. The speeds of transmission using ASK is
limited by the physical characteristics of the transmission medium. The
following fig gives a conceptual view of ASK.
TC31009D10 27
Fig 2.11 ASK

ASK transmission is highly susceptible to noise interference. The
term noise refers to unintentional voltages introduced onto a line by
various phenomena such as heat or electromagnetic induction created by
other sources. Noise usually affects the amplitude; therefore, ASK is the
modulating method most affected by noise.
5.3 Frequency Shift Keying (FSK)

In frequency Shift Keying (FSK), the frequency of the carrier signal
is varied to represent binary 1 or 0. The frequency of the signal during
each bit duration is constant and its value depends on the bit (0 or1);
both peak amplitude and phase remain constant.
The following Fig gives the conceptual view of FSK.
Fig 2.12 FSK
TC31009D10 28
5.4 Phase Shift Keying (PSK)

In phase shift keying (PSK), the phase of the carrier is varied to
represent binary 1 or 0. Both peak amplitude and frequency remain
constant as the phase changes. For example, if we start with a phase of
0 degrees to represent binary 0, then we can change the phase to 180
degrees to send binary 1. The phase of the signal during each bit
duration is constant and its value depends on the bit (0 or1). The
following Fig gives a conceptual view of PSK.
Fig 2.13 PSK
The above method is often called 2-PSK, or binary PSK, because

two different phases (0 and 180 degrees) are used. The following figure
makes this point clearer by showing the relationship of phase to bit
value. A second diagram, called a constellation or phase-state diagram,
shows the same relationship by illustrating only the phases.
Fig 2.14 PSK constellation
PSK is not susceptible to the noise degradation that affects ASK, or

to the bandwidth limitations of FSK. This means that smaller variations
in the signal can be detected reliably by the receiver. Therefore, instead
TC31009D10 29
of utilizing only two variations of a signal, each representing one bit, we

can use four variations and let each phase shift represent two bits.
Fig 2.15 4-PSK
The constellation diagram for the signal shown above, a phase of 0

degrees now represents 00; 90 degrees represents 01; 180 degrees
represents 10; and 270 degrees represents 11. This technique is called
4-PSK or Q-PSK. The pair of bits represented by each phase is called a
dibit. We can transmit data two times as fast using 4-PSK as we can use
2-PSK.
Fig 2.16 4-PSK characteristics
We can extend this idea to 8-PSK. Instead of 90 degrees, we now

vary the signal by shifts of 45 degrees. With eight different phases, each
shift can represent three bits (one tribit) at a time. The following figure
shows the relationships between the phase shifts and tribits each one
represents: 8-PSK is three times faster than 2-PSK.
TC31009D10 30
Fig 2.17 8-PSK characteristics
5.5 Quadrature Amplitude Modulation (QAM)
PSK is limited by the ability of the equipment to distinguish small

differences in phase. This factor limits its potential bit rate.
So far, we have been altering only one of the three characteristics

of a sine wave at a time, but what if we alter two? Bandwidth
limitations make combinations of FSK with other changes practically
useless. But why not combine ASK and PSK? Then we could have x
variations in phase and y variations in amplitude, giving us x times y
possible variations and the corresponding number of bits per variation.
Quadrature amplitude modulation (QAM) does just that.
Possible variations of QAM are numerous. Theoretically, any

measurable number of changes in amplitude can be combined with any
measurable number of changes in phase. The following figure shows two
possible configurations, 4-QAM and 8-QAM. In both cases, the number
of amplitude shifts is fewer than the number of phase shifts. Because
amplitude changes are susceptible to noise and require greater shift
differences than do phase changes, the number of phase shifts used by a
QAM system is always larger than the number of amplitude shifts.
TC31009D10 31
Fig 2.18 4-QAM and 8-QAM constellations
The time-domain plot corresponding to the 8-QAM signal is shown below
Fig 2.19 Time domain for an 8-QAM signal
5.6 Bit/Baud Comparison

Assuming that an FSK signal over voice-grade phone lines can
send 1200 bits per second, the bit rate is 1200 bps. Each frequency shift
represents a single bit; so it requires 1200 signal elements to send 1200
bits. Its baud rate, therefore, is also 1200 bps. Each signal variation in
an 8-QAM system, however, represents three bits. So a bit rate of 1200
bps, using 8-QAM, has a baud rate of only 400. The following table gives
a comparative bit and baud rates for the various methods of digital-to-
analog modulation.
TC31009D10 32
Modulation Units Bits/Baud Baud Rate Bit

Rate
ASK, FSK, 2-PSK Bit 1 N N
4-PSK, 4-QAM Dbit 2 N 2N
8-PSK, 8-QAM Tribit 3 N 3N
16-QAM Quadbit 4 N 4N
32-QAM Pentabit 5 N 5N
64-QAM Hexabit 6 N 6N
128-QAM Septabit 7 N 7N
256-QAM Octabit 8 N 8N
5.7 PDM (Pulse Duration Modulation)

PDM is a method of pulse modulation in which the duration of the
pulse train is used to transfer the binary signal information.
5.8 PPM (Pulse Position Modulation)

The amplitude and width of the pulse is kept constant in the
system. The position of each pulse, in relation to the position of a
recurrent reference pulse, is varied by each instantaneous sampled value
of the modulating wave. PPM has the advantage of requiring constant
transmitter power since the pulses are of constant amplitude and
duration. It is widely used but has the big disadvantage that it needs a
synchronization between transmitter and receiver.
5.9 PWM (Pulse Width Modulation)
Pulse Width Modulation refers to a method of carrying information

on a train of pulses, the information being encoded in the width of the
pulses. In applications to motion control, it is not exactly information we
are encoding, but a method of controlling power in motors without
(significant) loss.
In battery systems PWM is the most effective way to achieve a
constant voltage for battery charging by switching the system controller's
power devices on and off.
6. Multiplexing Techniques
Suppose a company with a link between two cities wished to
maximize the traffic between them. First, the data must be sent faster.
Then, more links must be acquired. This is the basis of multiplexing,
transmission technology deals with.
The transport network has been defined as a set of links between
telecommunication sites. Before multiplexing was discovered, each
TC31009D10 33
telephone call needed its own link to be transmitted. Many telephone

calls needed many links, which was expensive.
A way to put more than one telephone call on each link must be
found to save money. The best way to put more than one telephone call
on each link is to multiplex the calls. This makes best use of the links.
The easiest way to understand multiplexing is to remember the
transmission game one played as a child: two tin cans connected by a
piece of string (see Figure 2.20). In essence, that was a private link.
Multiplexing, however, enables several telephone calls to be sent on the
same line. The end users have the illusion of being on their own private
link. In effect, multiplexing creates a virtual telephone link for all of the
users, which is an early telephone version of virtual reality. Transmission
systems use a different type of multiplexing.
Figure 2.20 Multiplexing
Transmission systems that are designed according to European

rules work with groups of 30 telephone calls. A group of 30 telephone
calls is multiplexed into a 2–Mbps digital signal, and throughout most
transmission documents and presentations, constant references to 2–
Mbps channels may be found. These 2–Mbps streams are the basic
building blocks for multiplexing.
There are basically three types of multiplexing techniques.

• Frequency Division Multiplexing (FDM)
• Time division Multiplexing (TDM)
• Code Division Multiplexing (CDM)
TC31009D10 34
6.1 Frequency Division Multiplexing (FDM)

Frequency Division Multiplexing (FDM) technique is the process of
translating the frequencies of individual channels, having the bandwidth
300-3400 Hz, into pre-assigned frequency slots within the bandwidth of
the transmission medium. The frequency translation is done by
Amplitude Modulation (AM) of an appropriate carrier frequency by the
Audio Frequency. At the output of the modulator, a filter network is
connected to select either the lower or upper of the side band. Since the
intelligence is carried in either of the two side bands, Single Side Band
Suppressed Carrier (SSBSC) mode of AM is used. This has advantages of
substantial saving of bandwidth and also permits the use of low power
amplifiers. In this way number of channels can be combined by using
different carrier frequencies for different channels, and transmitting all of
them on a single medium.
Application of FDM Techniques

FDM techniques usually find application in the following analogue
transmission systems i.e., in a system used for transmitting
continuously varying signals.
• Carrier Systems
• Coaxial Systems
• Microwave systems
• Satellite systems
Limitations of FDM Analogue Systems

• Interference (Near-end and Far-end cross-talk)
• Noise (increases with the length of the system)
• Distortion (increases with the length of the system)
Because of the various limitations and transmission impairments
in analogue transmission system, need for digital systems using Time
Division Multiplexing arises.
6.2 Time Division Multiplexing (TDM)

Basically, TDM involves sharing a transmission medium by a
number of circuits, in time domain. This can be achieved by establishing
a sequence of time periods during which individual channel (circuit) is
transmitted. Thus the entire bandwidth is periodically available to each
channel.
Each channel is sampled at a specified rate and transmitted for a
fixed duration. All channels are sampled, one by one and transmitted.
Normally, all the time durations are equal. Each channel is assigned time
duration with a specific common repetition period. The channels are
connected to individual gates, which are opened one by one, in a fixed
sequence. At the receiving end also, similar gates are opened in unison
with the gates at the transmitting end.
TC31009D10 35
The signal received at the receiving end will be in the form of

discrete samples, which are combined to reproduce the original signal.
Thus, at a given instant of time, only one channel is transmitted through
the medium, and by sequential sampling, a number of channels can be
staggered in time, instead of transmitting all the channels at the same
time as in FDM systems. TDM Technique is used in Digital Transmission
Systems and Digital Switching Systems, where the discrete values of the
codified signal are employed.
6.3 Code Division Multiplexing (CDM)

Code Division Multiplexing is a technique in which each channel
transmits its bits as a coded channel-specific sequence of pulses. This
coded transmission typically is accomplished by transmitting a unique
time-dependent series of short pulses, which are placed within chip
times within the larger bit time. All channels, each with a different code,
can be transmitted on the same media and asynchronously
demultiplexed.
Code Division Multiple Access (CDMA), a method for transmitting
simultaneous signals over a shared portion of the spectrum. The
foremost application of CDMA is the digital cellular phone technology
from QUALCOMM.
Unlike the other digital systems that divide the spectrum into
different time slots, CDMA's spread spectrum technique overlaps every
transmission on the same carrier frequency by assigning a unique code
to each conversation. The often-used analogy for this is your ability to
detect your own language in a room full of people speaking other
languages.
7. Digital Hierarchies
The term “digital hierarchy” has been created when developing
digital transmission systems. It was laid down when by multiplexing a
certain number of PCM primary multiplexers were combined to form
digital multiplexers of higher order (e.g. second-order multiplex
equipments). Consequently, a digital hierarchy comprises a number of
levels. Each level is assigned a specific bit rate which is formed by
multiplexing digital signals, each having the bit rate of the next lower
level.
7.1 Definition
In CCITT Rec. G.702, the term “Digital Multiplex Hierarchy” is

defined as follows:
TC31009D10 36
“A series of digital multiplexes graded according to capability so

that multiplexing at one level combines a defined number of digital
signals, each having the digit rate prescribed for the next lower order,
into a digital signal having a prescribed digit rate which is then available
for further combination with other digital signals of the same rate in a
digital multiplex of the next higher order”.
7.2 Why Hierarchies?

FDM System
Before considering in detail the digital hierarchies under

discussion we are going to recapitulate in brief, why there are several
digital hierarchies instead of one only. It has always been pointed out
that as far as the analogue FDM technique is concerned, the C.C.I.T.T.
recommends the world wide use of the 12-channel group (secondary
group). Relevant C.C.I.T.T. Recommendation exists also for channel
assemblies with more than 60 channels so that with certain exceptions –
there is only one world-wide hierarchy for the FDM system (although the
term “hierarchy” is not used in the FDM technique).
Digital Scenario
In the digital transmission technique it was unfortunately not
possible to draw up a world-wide digital hierarchy. In practice,
equipment as specified in C.C.I.T.T. Recommendation G.732 and 733,
they not only differ completely in their bit rates, but also in the frame
structures, in signaling, frame alignment, etc. Needless to say that, as a
consequence, the higher order digital multiplexers derived from the two
different PCM primary multiplexers and thus the digital hierarchies differ
as well.
Digital Hierarchy based on the 2048 Kbps PCM

For this digital hierarchy, two specifications have at present been
laid down only for the first level at 2048 Kbps and for the second level at
8448 Kbps.
As for the higher levels, the situation is just contrary to that
existing in the case of digital hierarchies derived from 1544 Kbps primary
multiplex. General agreement has more or less been reached on the
fourth level having a bit rate of 139264 Kbps. 5th order system where bit
rate of 565 Mb/s have also been planned now.
The critical point in this hierarchy is whether or not the third level
at 34368 Kbps should exist.
The C.C.I.T.T. has agreed after long discussions on the following
(Recommendation G.751) “that there should be a 4th order bit rate of
TC31009D10 37
139264 Kbps in the digital hierarchy which is based on the 2nd order bit
rate of 8448 Kbps”.
There should be two methods of achieving the 4th order bit rate:
Method 1 by using a 3rd order bit rate of 34368 Kbps in the digital
hierarchy.
Method 2 by directly multiplexing sixteen digital signals at 8448 Kbps.
The digital signals at the bit rate of 139264 Kbps obtained by
these two methods should be identical.
The existence of the above two methods implies that the use of the
bit rate of 34368 Kbps should not be imposed on an Administration that
does not wish to realize the corresponding equipment.
Encoded TDM (European)

In accordance with the above two methods the following
realizations of digital multiplex equipments using positive justification
are recommended:
Method 1: Realization by separate digital multiplex equipments:
one type which operates at 34368 Kbps and multiplexes four digital
signals at 8448 Kbps; the other type which operates at 139264 Kbps and
multiplexes four digital signals at 34368 Kbps.
Method 2: Realization by single digital multiplex equipment which
operates at 139264 Kbps and multiplexes sixteen digital signals at 8448
Kbps.
Method 1 has been put into practice.
Where the fifth level is concerned, some preliminary proposals
(e.g. 565148 Kbps) have been submitted which were not discussed in
detail.
Therefore, the present structure of this digital hierarchy is as given
in Figure below.
Fig. 2.21 Encoded TDM (European)
TC31009D10 38
Most of the administrations favor the specification of a third level

at 34368 Kbps. The same is mainly as a suitable flexibility point for the
operation of the network and as an adequate bit rate for digital line
systems which are to be set up either on new cables (screened
symmetrical or micro-coaxial cables) or a radio-relay links. Other
administrations do not consider the specification of a third level to be
advantageous for their networks. On the contrary they regard it to be
more economical to go directly from the second level at 8448 Kbps so the
fourth level at 139264 Kbps, is also achieved by multiplexing four digital
signals at 34368 Kbps, each of which is obtained by multiplexing first
four digital signals at 8448 Kbps. However, this is a matter of internal
multiplexing only, i.e. digital multiplex equipment of this type has no
external input or output at 34368 Kbps.
All administrations interested in the third level at 34368 Kbps
would thus be offered the possibility of using this level. Their digital
multiplex equipment which multiplexes in the same way each of the four
digital signals at 8448 Kbps has to provide external outputs for the
resulting signal at 34368 Kbps. The digital multiplex equipment which
multiplexes each of the four digital signals at 34368 Kbps has to provide
four inputs for these bit rates and one output for the resulting bit rate of
139264 Kbps.
7.3 Higher order Multiplexing
Second order Multiplexing

The CCITT Recommendation G.742 deals with Second order digital
multiplex equipment operating at 8448 Kbps and using positive
justification.
This 2nd order digital multiplex equipment using positive
justification is intended for use on digital paths using 2048 Kbps primary
multiplex equipments.
Bit rates
The nominal bit rate should be 8448 Kbps. The tolerance on
this rate should be +30 PPM.
Third order digital multiplexing
Third order digital multiplexing equipment operates at 34368
Kbps. The nominal bit rate should be 34368 Kbps. The tolerance on the
rate should be + 20 PPM
Fourth order digital multiplexing
The nominal bit rate should be 139264 kb/s. The tolerance on the
rate should be +15 PPM
TC31009D10 39
8. Multiple Access Methods

The radio frequency spectrum, a finite natural resource, has
greater demands placed on it every day. In an effort to make the most
efficient use of this resource, various technologies have been developed
so that multiple, simultaneous users can be supported in a finite amount
of spectrum. This concept is called "multiple access." The three most
commonly used access methods are frequency division multiple access
(FDMA), time division multiple access (TDMA), and code division multiple
access (CDMA).
FDMA and TDMA are currently being used to support conventional
and trunked radio systems, as well as commercial cellular systems.
CDMA is being used primarily in cellular systems at this time.
8.1 FDMA, TDMA, and CDMA - a non technical example

The best way to describe the differences between FDMA, TDMA,
and CDMA technologies is with an example of how they work. The
following example is one of the best. Picture a large room with a group of
people divided up into pairs. Each pair would like to hold their own
conversation with no interest in what is being said by the other pairs. For
these conversations to take place without interruption from other
conversations, it is necessary to define an isolated environment for each
conversation. In this example, the room should be considered as a slice
of the radio spectrum specifically allocated to be used by this group of
people. Imagine each pair communicating through cellular telephones or
radios. Applying an FDMA system to this analogy, the single large room
(slice of spectrum) would be partitioned with many dividing walls and
creating a large number of smaller rooms. A single pair of people would
enter each small room and hold their conversation. Each room is like a
single frequency/channel. No one else could use the room (or frequency)
until the conversation was complete, whether or not the parties were
actually talking. When the conversation is completed, the first pair of
people would leave and another pair would then be able to enter that
small room.
In a TDMA environment, each of the small rooms would be able to
accommodate multiple conversations “simultaneously.” For example,
with a three-slot TDMA system, each “room” would contain up to three
pairs of people, with the different pairs taking turns talking. According to
this system, each pair can speak for 20 seconds during each minute.
Pair A would use 0:01 second through 0:20 second, pair B would use
0:21 second through 0:40 second, and pair C would use 0:41 second
through 0:60 second. However, even if there were fewer than three pairs
in the small room, each pair would still be limited to 20 seconds per
minute.
Using the CDMA technology, all the little rooms would be
eliminated. All pairs of people would enter the single large room (our
spectrum space). Each pair would be holding their conversations in a
TC31009D10 40
different language and therefore they could use the air in the whole room
to carry their voices while experiencing little interference from the other
pairs. The air in the room is analogous to a wideband “carrier” and the
languages represent the “codes” assigned by the CDMA system. In
addition, language “filters” would be incorporated so that, for example,
people speaking French would hear virtually nothing from those
speaking another language. Additional pairs could be added, each
speaking a unique language (as defined by the unique code) until the
overall “background noise” (interference from other users) made it too
difficult to hold a clear conversation. By controlling the voice volume
(signal strength) of all users to a minimum, the number of conversations
that could take place in the room could be maximized (i.e., maximize the
number of users per carrier). Additional pairs can be easily added to the
room without much interference to the other pairs.
8.2 FDMA - Frequency Division Multiple Access

Frequency division is the original multiple access technique.
Currently, most legacy public safety wireless networks use FDMA to
improve spectrum efficiency. FDMA is used throughout the commercial
wireless industry. Legacy commercial telecommunication networks
(analog networks based on Advanced Mobile Phone Service [AMPS] and
Total Access Communications System [TACS] standards) are built on a
backbone of cellular base stations, using the FDMA technology. However,
due to increased spectrum efficiency of CDMA and TDMA systems, very
few, if any, new cellular systems are using FDMA.
How it Works: FDMA systems separate a client's large frequency

band into several smaller individual bands/channels. Each channel has
the ability to support a user. Guard bands are used to separate channels
to prevent interference. They are used to isolate channels from adjacent-
channel interference.
Figure 2.22 FDMA

FDMA permits only one user per channel because it allows
the user to use the channel 100 percent of the time. Therefore,
only the frequency “dimension” is used to define channels. Each
block represents a different user
TC31009D10 41
When the FDMA technique is employed, each user is assigned a

discrete slice of the radio frequency (RF) spectrum, a “channel” of
spectrum space that will vary in size depending on the type of signal
being transmitted. In a given amount of spectrum space, the user is
granted access to a small sliver of the overall allocation. As long as the
user is engaged in “conversation,” no other user can access the same
spectrum space. An example of this type of access is use of the spectrum
by commercial radio broadcasters. In the commercial radio broadcast
bands, 535–1705 kHz for amplitude modulation (AM) and 88–108
megahertz (MHz) for frequency modulation (FM), each local broadcast
station (user) is assigned a specific slice of spectrum within the
frequency band allocated for that purpose. As long as the station
broadcasts, no other radio station in the same area can use that radio
frequency bandwidth to send a signal. Another broadcast station can use
that same bandwidth only when the distance between the stations is
sufficient to reduce the risk of interference.
In a conventional two-frequency public safety radio system, one
frequency is used to transmit and the other is used to receive. Each
channel has its own center frequency and each channel has a bandwidth
that is a fraction of the original allotted bandwidth. In this type of
system, if an FDMA channel is in use, other users cannot use it until the
“conversation” is complete. This is one of the inefficiencies of FDMA
systems. Figure 2 graphically displays a two-frequency conventional
system. The mobile and portable radio users transmit on frequency F1 to
the repeater; the repeater then retransmits back to the users on
frequency F2. In Figure 2, the F1 lightning symbol is an uplink to the
repeater while the F2 lightning symbol is a downlink.
Project 25's (P25) Phase I standard requires upgrades from standard
analog technology with a 25 kHz bandwidth to digital technology with a
narrower bandwidth of 12.5 kHz. Implementation of an FDMA system
would give each user access to two separate frequency allotments, each
with a 12.5 kHz bandwidth. Under P25, this newer equipment is also
required to be “backward compatible” to the legacy 25 kHz analog
equipment to allow a smooth transition.
Because adjacent channel interference is an important factor in
channel quality, frequency planning is a key consideration when
selecting fixed or base station locations. Frequency planning is
complicated and difficult. Available frequency bands must be researched
and analyzed. Transceiver transmission strength affects fixed station
range while antenna design affects its coverage patterns. These are also
important factors in frequency planning. Figure 3 is a sample base
station coverage scheme for a cellular system.
TC31009D10 42
Advantages
• Simple to implement, from a hardware standpoint
• Fairly efficient with a small base population and when traffic is
constant
• Backward compatible to analog radio equipment
Disadvantages
• Network and spectrum planning are intensive
• Poor spectrum efficiency, because channels are allocated for one
user.
• Frequency planning is time-consuming
8.3 TDMA - Time Division Multiple Access

As the frequency spectrum experiences more traffic, spectrum
efficiency becomes increasingly important. TDMA systems were
developed as FDMA system spectrum efficiency became insufficient. Not
only do TDMA systems split users into an available pair of channels, but
they also assign each user an available time-slot/cell within that
channel. TDMA systems have the capability to split users into time slots
because they transfer digital data, instead of analog data commonly used
in legacy FDMA systems. Each of the users takes turns transmitting and
receiving in a round-robin fashion. Frequency division is still employed,
but these frequencies are now further subdivided into a defined number
of time slots per frequency. In reality, only one user is (actually) using
the channel at any given moment. Each user is transmitting and
receiving in short “bursts.” Because TDMA systems do not transmit all of
the time, their mobile phones have an extended battery life and talk time.
How it works: Similar to an FDMA trunked system, when a user
depresses the Push-To-Talk (PTT) switch in a TDMA system, a control
channel registers the radio to the closest base station. During
registration, the base station assigns the user an available pair of
channels, one to transmit and the other to receive. But, unlike an FDMA
system registration, a TDMA system registration also assigns an
available time-slot within the channel. The user can only send or receive
information at that time, regardless of the availability of other time-slots.
Information flow is not continuous for any user, but rather is sent and
received in bursts. The bursts are re-assembled at the receiving end and
appear to provide continuous sound because the process is very fast.
In Figure 2.23, each row of blocks represents a single channel
divided into three time-slots. Calls in a TDMA system start in analog
format and are sampled, transforming the call into a digital format. After
the call is converted into digital format, the TDMA system places the call
into an assigned time slot.
TC31009D10 43
Figure 2.23 TDMA

TDMA increases the number of users who have access to
particular channel by dividing that channel into time-slots.
Figure 6 is also a graphical display of the efficiency of a TDMA

system. The improved efficiency of TDMA over FDMA can be realized
through a quick glance at Figures 2.22 and 2.23. In Figure 2.22, the
FDMA system supports 4 users while in Figure 2.23, the TDMA system
supports 12 users within the same bandwidth as the FDMA system.
There are systems in place today that allow an increase of up to six times
the capacity of FDMA alone.
Because TDMA systems also split an allotted portion of the
frequency spectrum into smaller slots (channels), they require the same
level of frequency planning as FDMA systems. The same careful steps in
frequency planning must be taken in both FDMA and TDMA systems.
Advantages
• Extended battery life and talk time
• More efficient use of spectrum, compared to FDMA
• Will accommodate more users in the same spectrum space than an
FDMA system which improves capacity in high traffic areas, such
as large metropolitan areas
• Efficient utilization of hierarchical cell structures – pico, micro,
and macro cells
• Can handle video and audio data efficiently
Disadvantages
• Network and spectrum planning are intensive
• Multipath interference affects call quality
• Dropped calls are possible when users switch in and out of
different cells
• Frequency planning is time consuming
• Frequency guard bands add to spectrum inefficiency
• Too few users result in idle channels (rural versus urban
environment)
• Higher costs due to greater equipment sophistication
TC31009D10 44
8.4 CDMA - Code Division Multiple Access

CDMA is a spread spectrum technique used to increase spectrum
efficiency over current FDMA and TDMA systems. Although spread
spectrum’s application to cellular telephony is relatively new, it is not a
new technology. Spread spectrum has been used in many military
applications, such as anti-jamming (because of the spread signal, it is
difficult to interfere with or jam), ranging (measuring the distance of the
transmission to determine when it will be received), and secure
communications (the spread spectrum signal is very hard to detect).
How it works: With CDMA, unique digital codes (Walsh Codes),
rather than separate radio frequencies/ channels, are used to
differentiate users. The Walsh codes are shared by the mobile phone and
the base station, and are called “pseudo-Random Code Sequences.” All
users access the entire spectrum allocation all of the time. That is, every
user uses the entire block of allocated spectrum space to carry his/her
message. A user's unique Walsh Code separates the call from all other
calls. Figure 2.24 graphically shows each user simultaneously accessing
the fully allotted frequency spectrum.
Figure 2.24 CDMA

CDMA allows all users access to their entire allocated spectrum.
CDMA, being a “spread-spectrum” technology, spreads the

information contained in a signal over the entire available bandwidth and
not simply through one frequency. Due to the wide bandwidth of a
spread-spectrum signal, it is very difficult to cause jamming, difficult to
interfere with, and difficult to identify. It appears as nothing more than a
slight rise in the “noise floor” or interference level, unlike other
technologies where the power of the signal is concentrated in a narrower
band making it easier to detect. Therefore CDMA systems provide more
privacy than FDMA or TDMA systems. These are great advantages over
technologies using a narrower bandwidth.
CDMA channels can handle an unspecified number of users. There
is not a fixed number. The capacity of the system depends on the quality
of current calls. As more users are added, noise is added to the wideband
frequency and therefore decreases the quality of current calls. Each
TC31009D10 45
user's transmission power increases the level of the frequency spectrum's

"noise floor" and therefore decreases the overall call quality for all users.
To help eliminate the "noise floor," CDMA mobile phones and base
stations use the minimum amount of power required to communicate
with each other. They use precise power control to decrease users'
transmission power. By decreasing a user's transmission power, the
mobile phone has added battery life, increased talk time, and smaller
batteries.
Because CDMA is a spread spectrum technology, it requires less
frequency planning. The full original spectrum is not divided into
separate blocks/channels, like it is in FDMA and TDMA systems.
Therefore, there is no need to plan for multiple frequency guard bands.
Because all users have access to the entire spectrum at all times,
frequency planning only needs to consider one frequency/channel.
Advantages
• Greatest spectrum efficiency: capacity increases of 8 to 10 times

that of an analog system and 4 to 5 times that of other digital
systems which makes it most useful in high traffic areas with a
large number of users and limited spectrum
• CDMA improves call quality by filtering out background noise,
cross-talk, and interference
• ”Soft handoffs”— Because of the multiple diversities in use,
handoffs between cells are undetected by the user
• Simplified frequency planning - all users on a CDMA system use
the same radio frequency spectrum
• Detailed frequency plans are not necessary.
• Frequency re-tunes for expansion are eliminated.
• Fewer cells are required for quality coverage
• Random Walsh codes enhance user privacy; a spread-spectrum
advantage
• Precise power control increases talk time and battery size for
mobile phones
Disadvantages
• Backwards compatibility techniques are costly

• Currently, base station equipment is expensive
• Difficult to optimize to maximize performance
• Low traffic areas lead to inefficient use of spectrum and equipment
resources
TC31009D10 46
The following table shows the differences between each of the

technologies. Each technology is rated based on their performance with
respect to the ideal performance level.
Performance FDMA TDMA CDMA

Capacity (Spectrum Poor Medium Good
Efficiency)
Security Poor Poor Good
Ease of Network Planning Poor Poor Medium
Ease of Implementation Very Good Very Good Medium
Cost of Implementation Good Good Medium
Backwards Compatibility Very Good Good Poor
TC31009D10 47
Module - 01
Chapter – 03
Transmission Media
Session Objectives:

• Various types of Transmission Media

• Overhead Lines
• Underground Copper Cables
• Coaxial Cables
• Optical Fiber Cables
• Fiber Geometry
• Construction of the OFC
• Optical Fiber Cable Splicing
TC31009D10 48
1. Transmission Media
Basically there are two ways in which information of any type can
be transmitted over telecommunication media – analog or digital. Analog
means that the amplitude of the transmitted amplitude signal varies over
a continuous range. Digital transmission means that streams of on/off
pulses are sent on the transmission media. The pulses are referred to as
bits. Examples of analog signals are human voice, hi–fi music,
temperature reading, etc. while those of digital are data, telegraphy
signals.
There are four types of media that can be used in transmitting
information in the telecommunications world:
• Overhead Lines
• Underground Copper Cables
• Coaxial cable (actually an adaptation of copper wire)
• Fiber
• Wireless
In days of old, copper wire was the only means of transmitting
information. Technically known as unshielded twisted pair (UTP), this
consisted of a large number of pairs of copper wire of varying size in a
cable. The cable did not have a shield and therefore the signal—primarily
the high-frequency part of the signal—was able to leak out. Also, the
twisting on the copper pair was very casual, designed as much to identify
which wires belonged to a pair as to handle transmission problems.
However, this is the way it was done, and for voice communications it
was quite satisfactory. Consequently, there are millions of miles of
copper in the PSTN—miles that must be used.
2. Overhead Lines
All are familiar with kilometers of pole lines on city streets, rural
highways and along railway tracks without knowing what the conductors
of these lines are meant for.
These conductors are the transmission lines consists of a pair of
conductors across which voltage is applied for transmission to some
distant point. Since transmission of voltage takes place from one end
to another end over these conductors, it is called a transmission line.
In telecommunications, the open wire transmission lines may
transmit telegraph or telephone signals at milliwatt levels from direct
current to alternating currents of up to some kilo Hertz.
The open wire lines may be used as:

• Telephone subscriber lines
• Telegraph lines
• Long distance, trunks or carrier lines.
TC31009D10 49
3. Underground Copper Cables

Cables are bunch of high conductivity annealed copper conductors
of small diameter, well insulated from each other and compactly packed
inside a sheath. The sheath is mechanically protected by providing steel
tape, etc. The cables nomenclature identifies the size in number of pairs,
gauge of the conductor, type of insulation, core filling material, armored
or unarmoured, etc.
The Advantages of Underground cables over overhead lines are:
1. Compactness in construction.
2. More number of subscribers can be provided using small space.
3. Better and efficient performance.
4. Less constructional difficulties.
5. Less maintenance cost
6. Less fault liability
Junction Network
The junction network comprises of the linkages between local
exchanges and between tandem exchange and local exchange. Provision
of junctions, routing of traffic and transmission media for junction
networks has to be carefully planned. It is very important to select
proper transmission media and junction network.
In the majority of networks the most commonly used medium for
junctions is underground cables. These suffer from several drawbacks
like poor transmission quality due to high attenuation and poor
reliability. With the advent of digital transmission systems PCM systems
(2 Mb) are also in use in our networks. The transmission rate cannot be
increased beyond 2 Mb due to poor X-Talk performance of cables. The
data over voice technology enables combined voice and data operation on
the existing unloaded subscriber cable loops. The data and speech are
integrated at customer premises equipment and separated and
segregated at exchange. The limitation is that the data calls through
PSTN are limited to voice band modem speeds in two wire mode and
require external modems.
Subscriber Network
The subscriber network consists of the circuits between the local
exchange and subscribers up to customer premises include cables, cross
connection points, and wires. The Local Exchange network diagram is
given in Figure 3.1. As the over head line is not reliable cable is extended
to the nearest locations in the area for serving the customer needs. It
consists of Primary cable, Secondary cable and Distribution cables.
These cables are terminated at cross connection points like cabinets,
pillars and DPs of various capacities to suit different cable network
systems.
TC31009D10 50
Fig. 3.1 External Plant Structure in a Local Telephone Exchange System
The subscriber network consists of the following characteristics.

• One pair exclusively for each subscriber
• Very low traffic occupancy and subs network difficult to replace.
• Transmission quality is poor and accounts for large part of overall
transmission loss which directly shows the efficiency of the system.
• Substantial part of overall investment goes in to the subscriber
network.
The factors driving towards replacement of present predominantly

analog network by digital subs network are mainly :
1. Technological evolution, with Optical fibers, VLSI
2. Reduced costs due to OFC
3. Need for new services with a long term objective of ISDN lines.
The services which can be provided with existing subscriber
network based on copper cable pairs are Telephony, Telefax, and
Teletext. With phased digitalisation the services that can be made
available on copper wire are Telephony (7KHz), Audio conferencing
(64Kb/s), Videotext, Image transmission, Computer communications and
ISDN
Classification of Underground Cables with regard to design
features are:
1. Place where it is used - Underground / overhead / submarine
2. Insulation material used - Paper / polythene cables
3. The filling compound - Dry core / jelly filled cables
4. Mechanical protection - Armored / un-armored cables
5. Place of utilization - Primary / Distribution / Junction cable.
6. System for which used - Co-axial / PCM
7. Type of conductor - Copper cable / Optical fiber cable
8. Gauge of the conductor - 0.40 mm / 0.50 mm / 0.63 mm/0.90mm
9. Pressurization of core - Pressurized / un-pressurised cables
TC31009D10 51
Limitations copper cable in local loop

1. The local loop is the suspected link for most of the local faults. The
old and worn-out cables have poor insulation which results in cross talk.
2. Water seepage during rainy season is the major cause of faults in
copper cables. Repeated digging and opening of joints further increase
the possibility of occurrence of faults.
3. The local loop is also vulnerable to electromagnetic interception
and hence misuse.
4. The limitations on loop length and unpredictable growth of
demand restrict the quick expansion capability and ease in network
flexibility.
5. Large size copper cable calls for more space for leading in cables in
MDF and congestion in duct.
6. Copper pairs are band width limited and are not able to support
new broad band services.
4. Coaxial Cables
Coaxial cable consists of a single strand of copper running down
the axis of the cable. This strand is separated from the outer shielding by
an insulator made of foam or other dielectrics. A conductive shield covers
the cable. Usually an outer insulating cover is applied to the overall
cable—this has nothing to do with the carrying capacity of the cable.
Because of the construction of the cable, obviously coaxial in nature,
very high frequencies can be carried without leaking out. In fact, dozens
of TV channels, each 6 MHz wide, can be carried on a single cable.
The coaxial cable consists of an inner solid cylindrical conductor
placed along the axis of an outer hollow cylindrical conductor. A coaxial
cable may consist of two or more cores layered up with suitable lay with
proper insulation along with quads laid in the interstices between them
all enclosed in a lead sheath. The cable is recognized with:
• Number of cores i.e. either 2 core or 4 core
• Size of the inner diameter of tube - r i.e. 0.375 types (large
tube – 0.375”) or 0.174 types (small tube – 0.174”).
The interstice Quads or pairs having diameter of 0.9mm
Fig: 3.2 Two Core

375 coaxial cable
TC31009D10 52
5. Optical Fiber Cables
Fiber-optic communications is based on the principle that light in

a glass medium can carry more information over longer distances than
electrical signals can carry in a copper or coaxial medium. The purity of
today's glass fiber, combined with improved system electronics, enables
fiber to transmit digitized light signals well beyond 100 km without
amplification. With few transmission losses, low interference, and high
bandwidth potential, optical fiber is an almost ideal transmission
medium.
Fiber is the third transmission media, and it is unquestionably the

transmission medium of choice. Whereas transmission over copper
utilizes frequencies in the megahertz range, transmission over fiber
utilizes frequencies a million times higher. This is another way of saying
that the predominant difference between electromagnetic waves and light
waves is the frequency. This difference, in turn, permits transmission
speeds of immense magnitudes. Transmission speeds of as high as 9.9
Gbps have become commonplace in the industry today. At this speed,
the entire fifteen-volume set of Encyclopedia Britannica can be
transmitted in well under one second.
In order to over come the limitations of copper cable and to be able

to support value added broad band service like data cable video HDTV
and increased use of computer which requires band width on demand,
optical fiber is introduced in the local loop . In the light of infinite
bandwidth and the reliability, optical fiber is the automatic choice in the
local loop
Laying fiber, on a per-mile basis, still costs somewhat more than

laying copper. However, on a per-circuit basis there is no contest; fiber
wins hands down. However, if a local loop is being laid to a residence,
there is little justification to installing fiber—there will never be a need
for more than one or two or three circuits. This realization has led to a
transition in our thinking. Shortly after the commercialization of fiber, we
talked about fiber-to-the-home (FTTH). It was then realized that there
was little need to install fiber for a final several hundred yards, so the
industry shied away from fiber-to-the-curb (FTTC). In such a system,
fiber would carry a plurality of channels to the "curb," whereupon they
would be broken down and applied to the copper drop leading to the
home. In many cases even this was overkill, and fiber-to-the-
neighborhood (FTTN) is now being used. The message is clear: apply fiber
when it is economical to do so and otherwise rely on copper.
TC31009D10 53
6. Fiber Geometry
An Optical fiber consists of a core of optically transparent material
usually silica or borosilicate glass surrounded by a cladding of the same
material but a slightly lower refractive index. Fiber themselves have
exceedingly small diameters. Figure shows cross section of the core and
cladding diameters of commonly used fibers. The diameters of the core
and cladding are as follows.
µm)
Core (µ µm)
Cladding (µ
8 125
50 125
62.5 125
100 140
Fig. 3.3 Typical Core and Cladding Diameter
Fiber sizes are usually expressed by first giving the core size
followed by the cladding size. Thus, 50/125 means a core diameter of 50
µm and a cladding diameter of 125 µm.
Fiber Types
The refractive index profile describes the relation between the
indices of the core and cladding. Two main relationships exist:
Step Index
Graded Index
The step index fiber has a core with uniform index throughout. The
profile shows a sharp step at the junction of the core and cladding. In
contrast, the graded index has a non–uniform core. The index is highest
at the center and gradually decreases until it matches with that of the
cladding. There is no sharp break in indices between the core and the
cladding.
By this classification, there are three types of fibers:

• Multimode Step Index Fiber (Step Index Fiber).
• Multimode Graded Index Fiber (Graded Index Fiber).
• Single–mode Step Index Fiber (Single Mode Fiber).
TC31009D10 54
Figure 3.4(a) OF Cable construction
Figure 3.4 (b) Single-Mode and Multimode Fibers
Figure 3.4 (c) Single-Mode and Multimode Fibers – Cross section
TC31009D10 55
Step Index Multimode Fiber

This fiber is called "Step Index" because the refractive index
changes abruptly from cladding to core. The cladding has a refractive
index somewhat lower than the refractive index of the core glass. As a
result, all rays within a certain angle will be totally reflected at the core–
cladding boundary. Rays striking the boundary at angles greater than
the critical angle will be partially reflected and partially transmitted out
through the boundary. After many such bounces the energy in these rays
will be lost from the fiber.
The paths along which the rays (modes) of this step index fiber
travel differ, depending on their angles relative to the axis. As a result,
the different modes in a pulse will arrive at the far end of the fiber at
different times, resulting in pulse spreading which limits the bit–rate of a
digital signal which can be transmitted.
This type of fiber results in considerable model dispersion, which
affects the fiber's bandwidth.
Fig. 3.5 Step Index fiber
Graded Index Multi–mode Fiber

This fiber is called graded index because there are many changes
in the refractive index with larger values towards the center. As light
travels faster in a lower index of refraction, so, the farther the light is
from the center axis, the greater is its speed. Each layer of the core
refracts the light. Instead of being sharply reflected as it is in a step
index fiber, the light is now bent or continuously refracted in an almost
sinusoidal pattern. Those rays that follow the longest path by traveling
near the outside of the core have a faster average velocity. The light
traveling near the center of the core has the slowest average velocity. As
a result all rays tend to reach the end of the fiber at the same time. That
causes the end travel time of different rays to be nearly equal, even
though they travel different paths.
TC31009D10 56
The graded index reduces model dispersing to 1 ns/km or less.

Graded Index fibers have core diameter of 50, 62.5 or 85 µm and a
cladding diameter of 125 µm. The fiber is used in applications requiring a
wide bandwidth and low model dispersion. The number of modes in the
fiber is about half that of step index fiber having the same diameter and
numerical aperture.
Fig. 3.6 Graded Index fiber
Singe Mode Fiber
Another way to reduce model dispersion is to reduce the core's

diameter, until the fiber only propagates one mode efficiently. The single
mode fiber has an exceedingly small core diameter of only 5 to 10 µm.
Standard cladding diameter is 125 µm. Since this fiber carries only one
mode, model dispersion does not exists. Single mode fibers easily have a
potential bandwidth of 50 to 100 GHz–km. The core diameter is so small
that the splicing technique and measuring techniques are more difficult.
Light source must have a very narrow spectral width and they must be
very small and bright in order to permit efficient coupling into the very
small core diameter of these fibers.
One advantage of single mode fiber is that once they are installed,
the system's capacity can be increased as newer, higher capacity
transmission system becomes available. This capability saves the high
cost of installing a new transmission medium to obtain increased
performance and allows cost effective increases from low capacity system
to higher capacity system.
TC31009D10 57
Fig . 3.7 Optical Fibers – Principle and Types
As the wavelength is increased, the fiber carries fewer and fewer

modes until only one remains. Single mode operation begins when the
wavelength approaches the core diameter. At 1300 nm, the fiber permits
only one mode; it becomes a single mode fiber.
As optical energy in a single mode fiber travels in the cladding as
well as in the core, therefore, the cladding must be a more efficient
carrier of energy. In a multimode fiber cladding modes are not desirable;
a cladding with inefficient transmission characteristic can be tolerated.
The diameter of the light appearing at the end of the single mode fiber is
larger than the core diameter, because some of the optical energy of the
mode travels in the cladding. Mode field diameter is the term used to
define this diameter of optical energy.
Optical Fiber Parameters

Optical fiber systems have the following parameters:
• Wavelength
• Frequency
• Window
• Attenuation
• Dispersion
• Bandwidth
Fiber Optic – Cable being used in India
The cable being used for long distance route has been supplied by
M/s SIECOR, U.S.A., a joint venture of Siemens and Corning glass
works. The cable is non metallic with 12 fibers single mode, to be
operated at 1300 nm. In fact, DOT has standardised single mode fibers
at 1300 nm for all long distance routes and for junction working.
TC31009D10 58
7. Construction of the OFC

An optical fiber consists of a core of optically transparent material
usually doped silica or borosilicate glass surrounded by a cladding of the
same material but of a slightly lower refractive index.
Nominal diameter of core is 9 µm and of cladding is 125 µm in
Siecor optical fiber cable single mode. Material used for primary coating
is Dzg resin and for secondary coating is UV curable resin.
OF Cable Construction
Cabling is an outer protective structure surrounding one or more
fibers. Cabling protect fibers environmentally and mechanically from
being damaged or degraded in performance. SIECOR Fiber Optic Cables
have the following parts:
• Optical Fiber
• Buffer tube
• Strength member
• Jacket
The cable buffer tube is one of two types, namely Loose buffer or
Tight buffer. The loose buffer uses a hard plastic tube having an inside
diameter several times that of the fiber. One or more fibers lie within the
buffer tube. As the cable expands and shrinks with temperature
changes, it does not affect the fiber as much; the fiber in the tubes is
slightly longer than the tube itself. SIECOR cable is loose buffer tube
cable.
Fig. 3.8 Cross Section of Optical Fiber (Single Mode)
TC31009D10 59
Fiber Optic Construction Specifications
Primary coating – 250 + 15 micron of UV cured acryl

ate.
Secondary coating – Nominal 2.4 mm loose dual layered
buffer tube filled with paraffin
based gel.
Central member – Solid GRP non–metallic in the
center of the cable core : Outer dia.
= 2.1 + 0.15 mm.
Core – Loose buffer tubes and fillers
stranded around the central
member.
Buffer tube filling – Paraffin gel inside tube
compound
Interstitial filling – Petroleum gel filled areas between
compound the loose buffer tubes.
Filler – Nominal 2.4 mm natural coloured
solid P.E. filler
Core Wrap – Continuous layer of non
hygroscopic dielectric material
applied longitudinally
Inner P.E. Sheath – Not less than 2.0 mm thick P.E.
inner sheath (Black colour).
Outer Nylon sheath – Not less than 0.7 mm thick Nylon
sheath (Orange colour).
TC31009D10 60
Module - 01
Chapter – 04
Optical Fiber Communication
Session Objectives:

• Introduction
• Fiber Optics Characteristics
• Fiber types
• Splicing
• Bend Radius & Tensile Loading
• Fiber optic Communication System
TC31009D10 61
1. Introduction
An Optical Fiber Transmission System uses light waves as carrier
of the information signals. These light waves are transmitted from one
place to another place through an optical fiber using the principle of
Total Internal Reflection. Over an Optical Fiber System, large bulk of
information can be sent with very low distortion.
2. Fiber-Optic Characteristics
Optical-fiber systems have many advantages over metallic-based
communication systems. These advantages include interference,
attenuation, and bandwidth characteristics. Furthermore, the relatively
smaller cross section of fiber-optic cables allows room for substantial
growth of the capacity in existing conduits. Fiber-optic characteristics
can be classified as linear and nonlinear. Nonlinear characteristics are
influenced by parameters, such as bit rates, channel spacing, and power
levels.
Wavelength
It is a characteristic of light that is emitted from the light source
and is measured in nanometers (nm). In the visible spectrum, wavelength
can be described as the colour of the light.
For example, Red light has a longer wavelength than Blue light.
Typical wavelengths for fiber optic are 850nm, 1300nm and 1550nm, all
of which are invisible.
Frequency
It is the number of pulse per second emitted from a light source.
Frequency is measured in units of hertz (Hz). It terms of optical pulse 1
Hz = 1 pulse/sec.
Window
A narrow window is defined as the range of wavelengths at which a
fiber best operates. Typical windows are given below:
Window Operational Wavelength

800 nm–900 nm 850 nm
1250 nm – 1305 nm 1300 nm
1500 nm – 1600 nm 1550 nm
Interference
Light signals traveling via a fiber-optic cable are immune from
electromagnetic interference (EMI) and radio-frequency interference (RFI).
Lightning and high-voltage interference is also eliminated. A fiber
network is best for conditions in which EMI or RFI interference is heavy
TC31009D10 62
or safe operation free from sparks and static is a must. This desirable
property of fiber-optic cable makes it the medium of choice in industrial
and biomedical networks. It is also possible to place fiber cable into
natural-gas pipelines and use the pipelines as the conduit.
Linear Characteristics
Linear characteristics include attenuation, chromatic dispersion
(CD), polarization mode dispersion (PMD), and optical signal-to-noise
ratio (OSNR).
Attenuation
Several factors can cause attenuation, but it is generally
categorized as either intrinsic or extrinsic. Intrinsic attenuation is caused
by substances inherently present in the fiber, whereas extrinsic
attenuation is caused by external forces such as bending. The
attenuation coefficient α is expressed in decibels per kilometer and
represents the loss in decibels per kilometer of fiber.
Intrinsic Attenuation
Intrinsic attenuation results from materials inherent to the fiber. It
is caused by impurities in the glass during the manufacturing process.
As precise as manufacturing is, there is no way to eliminate all
impurities. When a light signal hits an impurity in the fiber, one of two
things occurs: It scatters or it is absorbed. Intrinsic loss can be further
characterized by two components:
• Material absorption
• Rayleigh scattering
Material Absorption Material absorption occurs as a result of the

imperfection and impurities in the fiber. The most common impurity is
the hydroxyl (OH-) molecule, which remains as a residue despite
stringent manufacturing techniques.
Figure shows the variation of attenuation with wavelength
measured over a group of fiber-optic cable material types. The three
principal windows of operation include the 850-nm, 1310-nm, and 1550-
nm wavelength bands. These correspond to wavelength regions in which
attenuation is low and matched to the capability of a transmitter to
generate light efficiently and a receiver to carry out detection.
TC31009D10 63
Fig 4.1 : Attenuation versus Wavelength
The OH- symbols indicate that at the 950-nm, 1380-nm, and

2730-nm wavelengths, the presence of hydroxyl radicals in the cable
material causes an increase in attenuation. These radicals result from
the presence of water remnants that enter the fiber-optic cable material
through either a chemical reaction in the manufacturing process or as
humidity in the environment. The variation of attenuation with
wavelength due to the water peak for standard, single-mode fiber optic
cable occurs mainly around 1380 nm. Recent advances in
manufacturing have overcome the 1380-nm water peak and have
resulted in zero-water-peak fiber (ZWPF). Examples of these fibers
include SMF-28e from Corning and the Furukawa-Lucent OFS AllWave.
Absorption accounts for three percent to five percent of fiber attenuation.
This phenomenon causes a light signal to be absorbed by natural
impurities in the glass and converted to vibration energy or some other
form of energy such as heat. Unlike scattering, absorption can be limited
by controlling the amount of impurities during the manufacturing
process. Because most fiber is extremely pure, the fiber does not heat up
because of absorption.
Rayleigh scattering As light travels in the core; it interacts with the

silica molecules in the core. Rayleigh scattering is the result of these
elastic collisions between the light wave and the silica molecules in the
fiber. Rayleigh scattering accounts for about 96 percent of attenuation in
TC31009D10 64
optical fiber. If the scattered light maintains an angle that supports

forward travel within the core, no attenuation occurs. If the light is
scattered at an angle that does not support continued forward travel,
however, the light is diverted out of the core and attenuation occurs.
Depending on the incident angle, some portion of the light propagates
forward and the other part deviates out of the propagation path and
escapes from the fiber core. Some scattered light is reflected back toward
the light source. This is a property that is used in an optical time domain
reflectometer (OTDR) to test fibers. The same principle applies to
analyzing loss associated with localized events in the fiber, such as
splices.
Short wavelengths are scattered more than longer wavelengths.
Any wavelength that is below 800 nm is unusable for optical
communication because attenuation due to Rayleigh scattering is high.
At the same time, propagation above 1700 nm is not possible due to high
losses resulting from infrared absorption.
Extrinsic Attenuation
Extrinsic attenuation can be caused by two external mechanisms:
macro bending or micro bending. Both cause a reduction of optical
power. If a bend is imposed on an optical fiber, strain is placed on the
fiber along the region that is bent. The bending strain affects the
refractive index and the critical angle of the light ray in that specific area.
As a result, light traveling in the core can refract out, and loss occurs.
A macro bend is a large-scale bend that is visible, and the loss is
generally reversible after bends are corrected. To prevent macro bends,
all optical fiber has a minimum bend radius specification that should not
be exceeded. This is a restriction on how much bend a fiber can
withstand before experiencing problems in optical performance or
mechanical reliability.
The second extrinsic cause of attenuation is a micro bend. Micro
bending is caused by imperfections in the cylindrical geometry of fiber
during the manufacturing process. Micro bending might be related to
temperature, tensile stress, or crushing force. Like macro bending, micro
bending causes a reduction of optical power in the glass. Micro bending
is much localized, and the bend might not be clearly visible on
inspection. With bare fiber, micro bending can be reversible.
Chromatic Dispersion
Chromatic dispersion is the spreading of a light pulse as it travels
down a fiber. Light has a dual nature and can be considered from an
electromagnetic wave as well as quantum perspective. This enables us to
quantify it as waves as well as quantum particles. During the
propagation of light, all of its spectral components propagate accordingly.
These spectral components travel at different group velocities that lead to
dispersion called group velocity dispersion (GVD). Dispersion resulting
TC31009D10 65
from GVD is termed chromatic dispersion due to its wavelength

dependence. The effect of chromatic dispersion is pulse spread.
Polarization Mode Dispersion

Polarization mode dispersion (PMD) is caused by asymmetric
distortions to the fiber from a perfect cylindrical geometry. The fiber is
not truly a cylindrical waveguide, but it can be best described as an
imperfect cylinder with physical dimensions that are not perfectly
constant. The mechanical stress exerted upon the fiber due to
extrinsically induced bends and stresses caused during cabling,
deployment, and splicing as well as the imperfections resulting from the
manufacturing process are the reasons for the variations in the
cylindrical geometry. PMD is not an issue at low bit rates but becomes an
issue at bit rates in excess of 5 Gbps.
Fig 4.2: Polarization Mode Dispersion
Polarization Dependent Loss

Polarization dependent loss (PDL) refers to the difference in the
maximum and minimum variation in transmission or insertion loss of an
optical device over all states of polarization (SOP) and is expressed in
decibels. A typical PDL for a simple optical connector is less than .05 dB
and varies from component to component.
Optical Signal-to-Noise Ratio

The optical signal-to-noise ratio (OSNR) specifies the ratio of the
net signal power to the net noise power and thus identifies the quality of
the signal. Attenuation can be compensated for by amplifying the optical
signal. However, optical amplifiers amplify the signal as well as the noise.
Over time and distance, the receivers cannot distinguish the signal from
the noise, and the signal is completely lost. Regeneration helps mitigate
these undesirable effects before they can render the system unusable
and ensures that the signal can be detected at the receiver.
Optical amplifiers add a certain amount of noise to the channel.
Active devices, such as lasers, also add noise. Passive devices, such as
taps and the fiber, can also add noise components. In the calculation of
TC31009D10 66
system design, however, optical amplifier noise is considered the

predominant source for OSNR penalty and degradation. OSNR is
measured in decibels. The higher the bit rate, the higher the OSNR ratio
required.
3. Fiber Types
The MMF and SMF types currently used for premise, metro, aerial,
submarine, and long-haul applications. The International
Telecommunication Union (ITU-T), which is a global standardization body
for telecommunication systems and vendors, has standardized various
fiber types. These include the 50/125-µm graded index fiber (G.651),
Non dispersion-shifted fiber (G.652), dispersion-shifted fiber (G.653),
1550-nm loss-minimized fiber (G.654), and NZDSF (G.655).
Multimode Fiber with a 50-Micron Core (ITU-T G.651)

The ITU-T G.651 is an MMF with a 50-µm nominal core diameter
and a 125-µm nominal cladding diameter with a graded refractive index.
The attenuation parameter for G.651 fiber is typically 0.8 dB/km at 1310
nm. The main application for ITU-T G.651 fiber is for short-reach optical
transmission systems. This fiber is optimized for use in the 1300-nm
band. It can also operate in the 850-nm band.
Non dispersion-Shifted Fiber (ITU-T G.652)

The ITU-T G.652 fiber is also known as standard SMF and is the
most commonly deployed fiber. This fiber has a simple step-index
structure and is optimized for operation in the 1310-nm band. It has a
zero-dispersion wavelength at 1310 nm and can also operate in the
1550-nm band, but it is not optimized for this region. The typical
chromatic dispersion at 1550 nm is high at 17 ps/nm-km. Dispersion
compensation must be employed for high-bit-rate applications. The
attenuation parameter for G.652 fiber is typically 0.2 dB/km at 1550
nm, and the PMD parameter is less than 0.1 ps/√ km. An example of this
type of fiber is Corning SMF-28.
Low Water Peak Non dispersion-Shifted Fiber (ITU-T G.652.C)

The legacy ITU-T G.652 standard SMFs are not optimized for WDM
applications due to the high attenuation around the water peak region.
ITU G.652.C-compliant fibers offer extremely low attenuation around the
OH peaks. The G.652.C fiber is optimized for networks where
transmission occurs across a broad range of wavelengths from 1285 nm
to 1625 nm. Although G.652.C-compliant fibers offer excellent
capabilities for shorter, unamplified metro and access networks, they do
not fully address the needs for 1550-nm transmission. The attenuation
parameter for G.652 fiber is typically 0.2 dB/km at 1550 nm, and the
PMD parameter is less than 0.1 ps/√ km. An example of this type of fiber
is Corning SMF-28e.
TC31009D10 67
Dispersion-Shifter Fiber (ITU-T G.653)

Conventional SMF has a zero-dispersion wavelength that falls near
the 1310-nm window band. SMF shows high dispersion values over the
range between 1500 nm and 1600 nm (third window band). The trend of
shifting the operating transmission wavelength from 1310 nm to 1550
nm initiated the development of a fiber type called dispersion-shifted
fiber (DSF). DSF exhibits a zero-dispersion value around the 1550-nm
wavelength where the attenuation is minimum. The DSFs are optimized
for operating in the region between 1500 to 1600 nm. With the
introduction of WDM systems, however, channels allocated near 1550
nm in DSF are seriously affected by noise induced as a result of
nonlinear effects caused by FWM. This initiated the development of
NZDSF. G.53 fiber is rarely deployed any more and has been superseded
by G.655.
4. Splicing
Fiber-optic cables might have to be spliced together for a number
of reasons—for example, to realize a link of a particular length. Another
reason might involve backhoe fade, in which case a fiber-optic cable
might have been ripped apart due to trenching work. The network
installer might have in his inventory several fiber-optic cables, but none
long enough to satisfy the required link length. Situations such as this
often arise because cable manufacturers offer cables in limited lengths—
usually 1 to 6 km. A link of 10 km can be installed by splicing several
fiber optic cables together. The installer can then satisfy the distance
requirement and avoid buying a new fiber-optic cable. Splices might be
required at building entrances, wiring closets, couplers, and literally any
intermediate point between a transmitter and receiver.
Connecting two fiber-optic cables requires precise alignment of the

mated fiber cores or spots in a single-mode fiber-optic cable. This is
required so that nearly all the light is coupled from one fiber-optic cable
across a junction to the other fiber-optic cable. Actual contact between
the fiber-optic cables is not even mandatory.
The splices offer sophisticated, computer-controlled alignment of
fiber-optic cables to achieve losses as low as 0.02 dB. Typical fusion-
splice losses can be estimated at 0.02 dB for loss-budget calculation
purposes. Mechanical splices are easily implemented in the field, require
little or no tooling, and offer losses of about 0.5 to 0.75 dB.
5. Bend Radius and Tensile Loading

An important consideration in fiber-optic cable installation is the
cable’s minimum bend radius. Bending the cable farther than its
minimum bend radius might result in increased attenuation or even
broken fibers. Cable manufacturers specify the minimum bend radius for
TC31009D10 68
cables under tension and long-term installation. The ANSI TIA/EIA-

568B.3 standard specifies a bend radius of 1.0 inch under no pull load
and 2.0 inches when subject to tensile loading up to the rated limit.
Cable tensile load ratings, also called cable pulling tensions or
pulling forces, are specified under short-term and long-term conditions.
The short-term condition represents a cable during installation and it is
not recommended that this tension be exceeded. The long-term condition
represents an installed cable subjected to a permanent load for the life of
the cable. Typical loose-tube cable designs have a short-term (during
installation) tensile rating of 600 pounds (2700 N) and a long-term (post
installation) tensile rating of 200 pounds (890 N).
6. Fiber-Optic Communications System

As shown Figure below, information (voice, data, and video) from
the source is encoded into electrical signals that can drive the
transmitter. The fiber acts as an optical waveguide for the photons as
they travel down the optical path toward the receiver. At the detector, the
signals undergo an optical-to-electrical (OE) conversion, are decoded,
and are sent to their destination.
Fig 4.3 : Fiber Optic Communication System
TC31009D10 69
Transmitter
The transmitter component of Figure 4.3 serves two functions.

First, it must be a source of the light launched into the fiber-optic cable.
Second, it must modulate this light to represent the binary data that it
receives from the source. A transmitter’s physical dimensions must be
compatible with the size of the fiber-optic cable being used. This means
that the transmitter must emit light in a cone with a cross-sectional
diameter of 8 to 100 microns; otherwise, it cannot be coupled into the
fiber-optic cable. The optical source must be able to generate enough
optical power so that the desired BER can be met over the optical path.
There should be high efficiency in coupling the light generated by the
optical source into the fiber-optic cable, and the optical source should
have sufficient linearity to prevent the generation of harmonics and inter-
modulation distortion. If such interference is generated, it is extremely
difficult to remove.
This would cancel the interference resistance benefits of the fiber-
optic cable. The optical source must be easily modulated with an
electrical signal and must be capable of high-speed modulation;
otherwise, the bandwidth benefits of the fiber-optic cable are lost.
Finally, there are the usual requirements of small size, low weight, low
cost, and high reliability. The transmitter is typically pulsed at the
incoming frequency and performs a transducer electrical-to-optical (EO)
conversion. Light-emitting diodes (LEDs) or vertical cavity surface
emitting lasers (VCSELs) are used to drive MMF systems, whereas laser
diodes are used to drive SMF systems. Two types of light-emitting
junction diodes can be used as the optical source of the transmitter.
These are the LED and the laser diode (LD). LEDs are simpler and
generate incoherent, lower-power light. LEDs are used to drive MMF. LDs
generate coherent, higher-power light and are used to drive SMF.
Figure 4.4 below shows the optical power output, P, from each of
these devices as a function of the electrical current input, I, from the
modulation circuitry. As the figure indicates, the LED has a relatively
linear P-I characteristic, whereas the LD has a strong nonlinearity or
threshold effect. The LD can also be prone to kinks when the power
actually decreases with increasing input current. LDs have advantages
over LEDs in the sense that they can be modulated at very high speeds,
produce greater optical power, and produce an output beam with much
less spatial width than an LED. This gives LDs higher coupling efficiency
to the fiber-optic cable. LED advantages include a higher reliability,
better linearity, and lower cost.
TC31009D10 70
Fig 4.4 : LED and LD- PI Characteristics
A key difference between the optical output of an LED and a LD is

the wavelength spread over which the optical power is distributed. The
spectral width, σ, is the 3-dB optical power width (measured in
nanometers or microns). The spectral width impacts the effective
transmitted signal bandwidth. A larger spectral width takes up a larger
portion of the fiber-optic cable link bandwidth. Figure 4.5 shows the
spectral width of the two devices.
The optical power generated by each device is the area under the
curve. The spectral width is the half-power spread. An LD always has a
smaller spectral width than an LED. The specific value of the spectral
width depends on the details of the diode structure and the
semiconductor material. However, typical values for an LED are around
40 nm for operation at 850 nm and 80 nm at 1310 nm.
Typical values for an LD are 1 nm for operation at 850 nm and 3

nm at 1310 nm.
TC31009D10 71
Fig 4.5: LED and LD Spectral Width
Other transmitter parameters include packaging, environmental

sensitivity of device characteristics, heat sinking, and reliability. With
either an LED or LD, the transmitter package must have a transparent
window to transmit light into the fiber-optic cable. It can be packaged
with either a fiber-optic cable pigtail or with a transparent plastic or
glass window. Some vendors supply the transmitter with a package
having a small hemispherical lens to help focus the light into the fiber-
optic cable. Packaging must also address the thermal coupling for the
LED or LD. A complete transmitter module can consume more than 1
watt, which could result in significant heat generation. Plastic packages
can be used for lower-speed and lower reliability applications. However,
high-speed and high-reliability transmitters need metal packaging with
built-in fins for heat sinking.
There are several different schemes for carrying out the modulation
function. These include intensity modulation (IM), frequency shift keying
(FSK), phase shift keying (PSK), and polarization modulation (PM). Within
the context of a premise fiber-optic data link, the only one really used is
IM. IM is used universally for premise fiber-optic data links because it is
well matched to the operation of both LEDs and LDs. The carrier that
each of these sources produces is easy to modulate with this technique.
Passing current through them operates both of these devices. The
amount of power that they radiate (sometimes referred to as the
radiance) is proportional to this current. In this way, the optical power
takes the shape of the input current. If the input current is the waveform
m (t) representing the binary information stream, the resulting optical
signal looks like bursts of optical signal when m (t) represents a 1 and
TC31009D10 72
the absence of optical signal when m (t) represents a 0. This is also

known as direct modulation of the LED or LD.
Receiver
Figure 4.6 below shows a schematic of an optical receiver. The
receiver serves two functions: It must sense or detect the light coupled
out of the fiber-optic cable and convert the light into an electrical signal,
and it must demodulate this light to determine the identity of the binary
data that it represents. The receiver performs the OE transducer
function.
Fig 4.6 : Schematic of an Optical Receiver
A receiver is generally designed with a transmitter. Both are

modules within the same package. The light detection is carried out by a
photodiode, which senses light and converts it into an electrical current.
However, the optical signal from the fiber-optic cable and the resulting
electrical current will have small amplitude. Consequently, the
photodiode circuitry must be followed by one or more amplification
stages. There might even be filters and equalizers to shape and improve
the information-bearing electrical signal.
The receiver schematic in Figure shows a photodiode, bias resistor
circuit, and a low-noise pre-amp. The output of the pre-amp is an
electrical waveform version of the original information from the source.
To the right of this pre-amp are an additional amplification, filters, and
equalizers. All of these components can be on a single integrated circuit,
a hybrid, or discretely mounted on a printed circuit board.
The receiver can incorporate a number of other functions, such as
clock recovery for synchronous signaling, decoding circuitry, and error
detection and recovery. The receiver must have high sensitivity so that it
can detect low-level optical signals coming out of the fiber-optic cable.
The higher the sensitivity, the more attenuated signals it can detect. It
TC31009D10 73
must have high bandwidth or a fast rise time so that it can respond fast
enough and demodulate high-speed digital data. It must have low noise
so that it does not significantly impact the BER of the link and counter
the interference resistance of the fiber-optic cable transmission medium.
There are two types of photodiode structures: positive intrinsic negative
(PIN) and the avalanche photodiode (APD). In most premise applications,
the PIN is the preferred element in the receiver. This is mainly due to fact
that it can be operated from a standard power supply, typically between
5 and 15V. APD devices have much better sensitivity. In fact, APD
devices have 5 to 10 dB more sensitivity. They also have twice the
bandwidth. However, they cannot be used on a 5V printed circuit board.
They also require a stable power supply, which increases their cost. APD
devices are usually found in long-haul communication links and can
increasingly be found in metro-regional networks (because APDs have
decreased in cost). The demodulation performance of the receiver is
characterized by the BER that it delivers to the user. The sensitivity
curve indicates the minimum optical power that the receiver can detect
compared to the data rate, to achieve a particular BER. The sensitivity
curve varies from receiver to receiver. The sensitivity curve considers
within it the SNR parameter that generally drives all communications-
link performance. The sensitivity depends on the type of photodiode used
and the wavelength of operation.
1. Optical Sources
The Figure 4.7 shown below is a basic block diagram of an optical
transmitter. It contains an electronic processing circuit and a light
source. The input to the processing circuit is the signal from the carrier
multiplexing equipment (for example 140 CMI signal). The output from
the electronic processing circuit is the current required to operate the
source.
Fig. 4.7 Basic Transmitter Block Diagram
The circuit of transmitter must accept standard analog or digital

signal levels. Digital systems use either Transistor–transistor logic (TTL)
or emitter–coupled logic (ECL) or complementary metal oxide
semiconductor (CMOS) logic levels. The circuitry has to perform the
following functions in a digital transmitter.
TC31009D10 74
• Separation of clock and data.

• Retiming and reshaping of pulses.
• Line coding–conversion to optical line code.
• Scrambling or randomizing base band input signal.
• Combining order wire, supervisory, protection switching signals
with the main signal.
The two types of sources employed in fiber optics communication
systems as carriers of information are the light emitting diode (LED) and
laser diode (LD). Analogous to electronic oscillators in radio systems,
these are the optic oscillators in fiber system. The source provides
suitable power at the required wavelengths for long distance
applications. The sources found suitable in communication applications
are the semiconductor sources (LED and LD).
2. Optical Sources Requirements

To meet the communication needs in optical fiber technology, the
requirements have been summarized in Table 1 below:
Property Requirement Target
It must operate at a wavelength
Primary Characteristic 850, 1300 and
which gives low loss and low
Wavelength 1550 nm.
dispersion in fibers
Long life, good stability of
operation and good
Reliability Life = 106 hrs.
reproducibility of output
characteristics are necessary.
Output power System demands must be met > 1 mW
It must operate with a low
Power efficiency power and low voltage and heat > 10%
generated must be small.
It must have a spectral width Target must be
Desirable Properties
which enables the maximum determined by the
Spectral width
bandwidth of the optical fiber to scale of the
(temporal coherence)
be realized system.
It should be possible to focus
Focusing effect
the output onto the fiber and to
(spatial coherence)
obtain high coupling efficiency.
Direct modulation must be
Modulation possible or coupling to an
external modulation made easy.
It must be small and light in
Size and weight
weight.
Cost Mass production and low cost.
TC31009D10 75
Like in many other cases, the practical sources fall short of the
above requirements particularly in respect of stability and emission of
single frequency. The efforts required to achieve stability are not as
difficult as the generation of a single frequency output. Continuous
development work is in progress to achieve the latter requirement.
Light emitting diode optical sources find application in short haul
and medium haul communication systems where the power requirement
is small and bit rates are low. For long distance, high bit rate
applications, Laser diode is the answer. The two types of sources are
discussed below. The optic beams generated by these two light sources
carry the information through the process of intensity modulation. The
operating principles, transfer characteristics, modulation and
stabilization of output pertaining to the two types (LD and LED) of
semiconductor optical sources are considered.
3. Light Emitting Diodes
Operating Principle: An LED is a semiconductor p–n junction diode and

is forward biased. It emits light under this condition. When the p and n
types of semiconductor materials are brought together, the resulting
energy barrier under zero external applied voltage prevents the
movement of the n and p charge carriers. When the barrier energy level
is lowered by the application of an external voltage (in forward direction),
the movement of electrons and holes have taken place and some of the
charge carriers recombine in the transition region. The energy lost in the
transition is converted to optical energy giving rise to photon.
The diode is modulated by a current source that turns the LED on
or off as shown below. Unlike analog modulators, digital drivers need not
provide d–c bias current. In the off state, the emission of the LED should
be low.
Fig. 4.8 Series–switched and parallel–switched digital modulators
TC31009D10 76
Long Term Stability

Lifetime of the LED is about 106 hours and large temperature
variations are tolerated but at the cost of output power. The decline in
output power is approximately 1% per degree centigrade. Different types
of packages are available. The pigtail construction and micro lens type
are a few.
4. Laser Diode
A laser has an optical cavity formed by two parallel mirrors at the
end facets of the semiconductor crystal as in Fig. 4.9. For most
applications, it is sufficient to use the natural cleavage facets of the
crystal as mirrors. To increase the reflectivity, the mirrors can be coated
with a metallic film.
Fig. 4.9 Resonant Cavity
Operating Principles
A few characteristics of lasers are:
1) Pumping threshold: The input power must be above a threshold

before emission of light takes place. This is not required in LED.
2) Radiation pattern: The angle of radiation depends on the size of the
emitting area and on the modes of oscillation within the laser.
3) Output spectrum: The output power is spread over a range of
frequencies and power does not vary smoothly over this range but has
peaks and valleys. The number of such peaks depends on spectral width,
refractive index of material and dimension of cavity.
Operating Characteristics
The output power–forward input current characteristics are shown

in Fig. 4.10. The threshold current is approximately 75 mA and the
TC31009D10 77
voltage at threshold is 1 to 2 volts. Operating current is generally about

20 to 40 mA above the threshold current. A short kink is observed near
the threshold. Digital modulation of laser diode differs from the
modulation used in LED, in that a d–c bias current is added.
Fig. 4.10 Operating Characteristics
Temperature stability
Laser diode is more temperature sensitive than LEDs. At high

temperatures, threshold current increases. The threshold increases 1.5%
per degree increase in temperature. If the current remains constant, the
output of the laser diode decreases.
Output Spectrum
When the drive current is near threshold, lasers produce

multimode spectra. As the current increases, total line width decreases
and number of longitudinal modes decreases. At sufficiently high
currents, the spectrum contains just one mode. The light from laser
beam is confined to a narrow angular region.
TC31009D10 78
5. Comparison of LED and LASER Diode (Table 2)
Single
Laser
Property LED Mode
Diode
Laser Diode
Spectral Width (nm) 20–100 1–5 <0.2
Rise Time (ns) 2–250 0.1–1 0.05–1

Modulation Bandwidth
< 300 < 2000 ~ 2000
(MHz)
Coupling Frequency Very low Moderate High
MM
Compatible Fiber MMGI, SM Single Mode
(SI&GI)
Temperature Sensitivity Low High High
Circuit Complexity Simple Complex Complex
Lifetime (Hours) 105 104–105 104–105
Costs Low High Highest
Path Length Moderate Long Very Long
Data Rate Moderate High Very High
TC31009D10 79
Module - 01
Chapter – 05
Optical Fiber Cable Splicing
Session Objectives:

• Definition
• Splicing Methods
• Example of Mechanical Splicing
• Preparations of Splicing Closure
• Fusion Splicer (X 76)
• Fusion Splicing methods
TC31009D10 80
1. Definition
Splices are permanent connection between two fibers. The splicing
involves cutting of the edges of the two fibers to be spliced.
There are several reasons for splicing a fiber cable, these include:
• To join two fibers due to a breakage.

• To connect some of the cores straight through a patch cabinet.
• To extend a cable run.
• To reduce losses, a fusion splice has much lower losses than
two connectorized cables joined through a coupler.
• To attach a pre-terminated pigtail.
A Pigtail is a short length of fiber with a factory fitted and polished

connector. In the past these were used in preference to field terminations
because of the complexities at the time of manually terminating optical
fibers. These days pigtails are mainly used where the environment isn't
suitable for manual terminations or where speed is a factor.
2. Splicing Methods
Single–Fibre Mechanical Splicing

• Single Fibre Capillary
• Aligns two fiber ends to a common centerline, thereby aligning
cores.
• Clean, cleaved fibers are butted together and index matched.
• Permanently secured with epoxy or adhesive.
Examples: Siecor, See Splice GTE Elastomeric Splice.
Splice
Uncos Cos
ted ted
Fig. 5.1 Mechanical Splice
The following three types are widely used:

• Adhesive bonding or Glue splicing.
• Mechanical splicing.
• Fusion splicing.
TC31009D10 81
Adhesive Bonding or Glue Splicing

This is the oldest splicing technique used in fiber splicing. After
fiber end preparation, it is axially aligned in a precision V–groove.
Cylindrical rods or another kind of reference surfaces are used for
alignment. During the alignment of fiber end, a small amount of adhesive
or glue of same refractive index as the core material is set between and
around the fiber ends. A two component epoxy or an UV curable
adhesive is used as the bonding agent. The splice loss of this type of joint
is same or less than fusion splices. But fusion splicing technique is more
reliable, so at present this technique is very rarely used.
Mechanical Splicing
This technique is mainly used for temporary splicing in case of
emergency repairing. This method is also convenient to connect
measuring instruments to bare fibers for taking various measurements.
The mechanical splices consist of 4 basic components:
(i) An alignment surface for mating fiber ends.
(ii) A retainer
(iii) An index matching material.
(iv) A protective housing
A very good mechanical splice for M.M. fibers can have an optical
performance as good as fusion spliced fiber or glue spliced. But in case of
single mode fiber, this type of splice cannot have stability of loss.
Fusion Splicing
The fusion splicing technique is the most popular technique used
for achieving very low splice losses. The fusion can be achieved either
through electrical arc or through gas flame.
The process involves cutting of the fibers and fixing them in micro–
positioners on the fusion splicing machine. The fibers are then aligned
either manually or automatically core aligning (in case of S.M. fiber)
process. Afterwards the operation that takes place involve withdrawal of
the fibers to a specified distance, preheating of the fiber ends through
electric arc and bringing together of the fiber ends in a position and
splicing through high temperature fusion.
If proper care taken and splicing is done strictly as per schedule,
then the splicing loss can be minimized as low as 0.01 dB/joint. After
fusion splicing, the splicing joint should be provided with a proper
protector to have following protections:
• Mechanical protection
• Protection from moisture.
Sometimes the two types of protection are combined. Coating with
Epoxy resins protects against moisture and also provides mechanical
strength at the joint.
TC31009D10 82
Now–a–days, the heat shrinkable tubes are most widely used,

which are fixed on the joints by the fusion tools.
The fusion splicing technique is the most popular technique used
for achieving very low splice losses. The introduction of single mode
optical fiber for use in long haul network brought with it fiber
construction and cable design different from those of multimode fibers.
The splicing machine begins to the core profile alignment system,
the main functions of which are:
• Auto active alignment of the core.
• Auto arc fusion.
• Video display of the entire process.
• Indication of the estimated splice loss.
The two fiber ends to be spliced are cleaved and then clamped in
accurately machined vee–grooves. When the optimum alignment is
achieved, the fibers are fused under the microprocessor control; the
machine then measures the radial and angular off–sets of the fibers and
uses these figures to calculate a splice loss. The operation of the machine
observes the alignment and fusion processes on a video screen showing
horizontal and vertical projection of the fibers and then decides the
quality of the splice.
The splice loss indicated by the splicing machine should not be
taken as a final value as it is only an estimated loss and so after every
splicing is over, the splice loss measurement is to be taken by an OTDR
(Optical Time Domain Reflectometer).
The manual part of the splicing is cleaning and cleaving the fibers.
For cleaning the fibers, Dichlorine Methyl or Acetone or Alcohol is used
to remove primary coating.
With the special fiber cleaver or cutter, the cleaned fiber is cut. The
cut has to be so precise that it produces an end angle of less than 0.5
degree on a prepared fiber. If the cut is bad, the splicing loss will
increase or machine will not accept for splicing. The shape of the cut can
be monitored on the video screen, some of the defect noted while cleaving
are listed below:
(i) Broken ends.
(ii) Ripped ends.
(iii) Slanting cuts.
(iv) Unclean ends.
It is desirable to limit the average splice loss to be less than 0.1 dB.
3. Example of Mechanical splicing

The 3M brand Optical Fiber Splicing System provides permanent
mechanical splices for single or multi mode fiber having 125 µm
diameter cladding. Three color coded versions of the Fibrlok Splice are
available for splicing 250 µm and 900 µm diameter plastic coated fiber.
TC31009D10 83
Fig. 5.2 Mechanical Splice (3M brand)
Fig. 5.3 Fiber assembly tool (3M brand)
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Splicing Set-Up
• The splicing area should be clean, dry and well-lighted. A

clean, well organized splicing area will improve splicing
efficiency and minimize the risk of contamination of fibers or
splices.
• Open the buffer tubes, expose and clean the fibers as specified
in practice.
• Load the splice into the assembly tool by pressing firmly at the
ends of the splice.
• Remove the minimum length of fiber required to prepare and
splice the fibers.
• Strip approximately 1 to 2 inches (2.5 to 5 cm) of plastic
coating from the fiber using a mechanical stripper.
• Clean the bare glass by pulling the fiber through an alcohol
soaked lint-free wipe. This will remove any fragments or dirt
remaining on the fiber.
• Push the fiber down into the fiber retention pad on the proper
side of the splice. Fiber should be inserted into the splice
immediately following cleaning and placing in retention pad to
minimize exposure to the atmosphere and reduce the risk of
contamination.
• Prepare second fiber (strip, clean and cleave) as described for
fiber 1.
• Lay fiber into foam retention pad and begin to insert the fiber
end into the splice.
TC31009D10 85
• Pivot the handle of the Assembly Tool down until it contacts

the cap of the Splice. Squeeze the handle of the assembly tool
as shown in order to close
• cap and actuate the splice. When possible, secure the tool to a
work surface for added support. A snap sound will be heard
when the splice is actuated.
• Remove the Splice from the Assembly Tool by first removing the
fibers from the foam retention pads and then lifting the splice
from the splice holding cradle.
4. Preparation of Splicing Closure

There are two types of splice closures – one to be installed in a pit
(SC–4) and the other to be installed on wall (Wall Splicing Closure -
WSC).
The general steps related to splice closure are given below:
• Locate the splice pit.
• Mark the splice number and route name on the joint indicator.
• Open the splice pit.
• Record the meter marks of the cable.
• Coil the spare cable.
• Prepare end caps including fixing of cable in end cap.
• Assemble the closure.
The details of WSC and SC4 are given below:
Wall Splice Closure (WSC)

The front view of the WSC with its components is shown in Fig.
Pigtail cables–tight–buffered fibers (with connectors on one end) that are
spliced to the outside cable fibers.
TC31009D10 86
• Anchoring screw – used to secure the strength member to the

strain relief bracket.
• Strain relief brackets – used as strain relief for multifibre cable.
• Fastener locations – used with fasteners to secure the front
panel to the unit.
• Cable ties – used to secure cables.
Fig. 5.4 Components of WSC
• Cable contains fibers from the outside environment which are

spliced to pigtail cables or other multifibre cables.
• Knockouts – when removed provide location for cable routing.
• Cable retaining guide – used to hold the cable in position.
• Wire stacker – used to store splice trays.
• Screw holes – locations used to mount strain relief brackets.
• Splice tray – used to store splices.
• Identification label – used to make an installation plan as well
as a record of splicing information; located on the front panel.
All vital information about the installation should be recorded
on this label.
SC–4 Type Closure

The application of SC–4 closure is shown in Fig 5.5. The diagram
shows the placement of SC4 closures and the cable racking arrangement.
There is a long list of parts in SC4 due to its underground applications.
Underground application needs the water proofing of the closure.
TC31009D10 87
Fig. 5.5 Application of SC-4 Type closure
Loading of Splice Trays in SC4:

The following steps are followed for field splicing:
• Remove 150 cms of buffer tubes.
• Clean the fibers.
• Load the buffer tubes in the tray
• Place the fibers in the tray
• Close the trays.
5. Fusion Splicer (X 76)

As with all fiber termination methods, safety is very important so
first some safety tips.
• Always work in a clean and tidy area.

• Fiber off cuts are hard to see and can easily penetrate the skin
especially if they get into your clothes, so care must be taken to
ensure the safe disposal of all off cuts. Dispose of fiber scraps
immediately using a suitable container and do not throw into a
waste paper bin. * Because of the dangers of ingesting a fiber,
do not eat or drink in the termination area.
• Fusion splicers use an electric arc to fuse the fibers together so
they should never be used in an environment where flammable
gases or liquids are present.* Never look into the end of a live
fiber connector. Holding some multimode fibers up to a piece of
paper may prove the presence of light and therefore prove that
it is live, but it doesn't prove that it isn't live! Some laser
powered equipment use light which is outside of the visible
spectrum, so err on the side of caution.
A fusion splice is a way of joining two fiber cores by melting the

ends together using an electric arc. A splicing machine is used because
TC31009D10 88
an extremely high degree of accuracy is needed; the machine first has to

align the cores and then apply the exact amount of heat to melt the ends
before pressing them together.
There are four basic steps to fusion splicing
1 - Strip back all coatings down to the bare fibers and clean using
isopropyl alcohol.
2 - Cleave the fibers using a precision cleaving tool and put the heat
shrink tube on to one of the ends.
3 - Fuse the fibers together in the fusion splicer.
4 - Put the heat shrink protector on the fiber joint.
6. Fusion Splicing Method
Stripping
Strip back the external sheathing of the cable using a rotary

stripping tool. Cut back the aramid strength member using ceramic or
kevlar scissors. Strip the primary buffer from the fiber using fiber
strippers not ordinary wire strippers. Do this a small section at a time to
prevent the fiber breaking, about 10mm (3/8 in) on each cut is fine until
you get used to it. Strip back about 35mm (1.5 in).
Clean the bare fiber with a lint free wipe and isopropyl alcohol, it
will "squeak" when it is clean.
Cleaving
The cleaver first scores the fiber and then pulls the fiber apart to
make a clean break. It is important that the ends are smooth and
perpendicular to get a good joint; this is why a hand held cleaver will not
do.
Cleavers vary from manufacturer to manufacturer and you should
read the instructions for the one you are using. Basically the operation
consists of putting the fiber into the groove and clamping, then close the
lid and press the lever.
Fusion Process
Once the fiber ends are prepared they are placed in the fusion
splicer. Press the button and the machine takes care of the rest of the
fusion process automatically.
First the two fibers are aligned; you can see this on the photo
where a much magnified image shows the two fiber ends. The display
also shows how well the cleaver does its job of producing a perfect 90
degree cut.
TC31009D10 89
If you watch very carefully in the video you can see the X and Y
alignment that takes place. The splicer aligns the fibers on one axis and
then from another camera angle set at 90 degrees, it aligns the other
axis. This high precision alignment is critical for a low loss joint; any
mismatch of the fiber cores will significantly reduce the propagation of
light through the joint.
Bearing in mind that we are dealing with two very small glass rods
of only 125 microns in diameter, it brings it home as to how extremely
accurate these machines are.
Once the fibers are aligned the splicer fires an electric arc between
the two ends which melts them immediately and pushes them together,
or fuses them into one piece of fiber.
The fusion splicer then tests for dB loss and tensile strength before
giving the "OK" beeps for you to remove the splice from the machine
Protection
The splicer in the video has a built in heat shrink oven, so when
the fiber is taken out of the machine the protective tube is slid into place
and the whole assembly is put into the oven to shrink the tube on to the
splice.
The protective tube gives physical protection to the splice and

further protection is provided by placing the splice into a splice tray.
Once all of the fibers have been joined the whole tray is then fixed
into a splice box which protects the cable joint as a whole and the cable
clamps are then tightened to prevent any external forces from pulling on
the splices.
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Fusion Splicing Procedure
TC31009D10 91
Preparing the Fiber Ends
Note: Minimum splice loss can only be achieved if the fiber ends have
been prepared carefully.
Preparation of the Fiber includes the following steps

• Stripping the coating
• Cleaning the fibers
• Cutting the fibers with cleaver
• Assessing quality of the fibers and faces (monitor)
Follow the instructions for the fiber cleaver

Remove the coating
Fig. 5.6 Stripping the coating and cleaning the Fiber
Cleaning the fibers
Clean the ends of the fibers over the length of about 10cm (coated and
uncoated fibers)
With a lint free paper cloth dipped in alcohol or with alcohol in ultrasonic
bath
Danger : there is a risk of fire if the solvent is spilled
TC31009D10 92
Cutting the fiber
Using the cleaver S46999 M9-A8
• Select the correct fiber guides for the coating diameter (250 µm,
500 µm, or 900 µm
• Open the flap of the cleaver and place the fiber in the cleaver so
that the end of the coating is 10mm mark
• Close the flap of the cleaver and then press down carefully
• Open the flap before you cut fiber in order to avoid the damage
to the end face of the fiber.
Fig. 5.7 Cutting the Fiber with RXS cleaver
Note: If you press down the flap too quickly the fiber and the face quality
will be poor and you will damage the diamond.
Inserting the fibers
Insertion in the fiber holder
• Open the electrode flap and the clamps flap for the tensile test
equipment and the fiber holder flaps
• Place the fibers in the V Groove of the slide so that the ends of
the fibers are between the electrodes.
TC31009D10 93
Fig. 5.8 Inserting the Fiber in the Fiber Holder
• Close the fiber holder flap prepare the second fiber and insert
in the same way on the other side.
• Close the Electrode flap. If you wish to perform a tensile test on
the splice after fusing, close the flaps of the tensile test device
as well.
Note: Ensure that the fiber ends are visible on the monitor after
insertion. Otherwise the resulting splice may be poor or the travel of the
positioner may not be adequate.
Checking the fiber end faces
• After inserting the fibers in the in the splicers select menu

items (“searching fibers”?) or Automatic or fully automatic.
• The fiber ends are displayed in two views from the direction of
the X axis ( upper half of the monitor) and from the direction of
the Y axis ( lower half of the monitor)
Note: the fibers must be

• Clean
• Free of protrusions and indentation
• Flat and square to the fiber axis
• The quality of the splice is always determined by the quality of
the fiber end faces.
TC31009D10 94
Fig. 5.9 Assessing the quality of the Fiber end faces
• If the quality of the fiber end is poor the end face must be
prepared again.
• If you are in “fully automatic” or “automatic mode” and notice
that a fiber end is of poor quality you can terminate the
proceeding process by pressing key.
Splicing fibers
Splicing the fibers includes the following steps
• Cleaning the fibers

• Adjusting the fibers (Z axis gap X/Y offset)
• Fusing the fibers
• Assessing the splice quality (with fully automatic and automatic)
Cleaning
Clean the fibers, repeat the process if the fiber ends are still visibly
contaminated. If the contamination is still not removed completely after
this the fiber must be prepared again.
Adjusting and fusing
Adjustment of the fibers in X-Y-Z –axis and initiation of the fusion

process can be implemented fully automatically, automatically or
manually depending on which splicing process you have selected.
Manual position is only possible for the Z-axis. Procedure for each
splicing process is already described.
TC31009D10 95
Assessment of the Splice and Refusing
Visual assessment
If a good splice has been made, the cladding has a smooth surface.
There should be no visible faults or shadows inside the fiber image as
shown in the next page. The bright lines in the center of the fiber are
result of the lens effect.
Fig. 5.10 Visual Splice assessment
Splice loss indication
In fully automatic and automatic mode the splice loss is indicated

in dB to an accuracy of +/- 0.1 dB.
Causes of Poor Splice quality
• Quality of the fiber end faces unsatisfactory

• Fusion parameters are not set correctly and not optimized.
Fit the splice protector as follows:
Connect the heater to one of the terminals on the splicer. Make

sure that the splicer is connected to a battery or AC power supply. Close
the cover of the heater and check that the red LED lights. Open the cover
again. Push the heat shrink splice protector over one of the fiber ends
before splicing. Splice the fibers as described and then push the splice
protector centrally over the splice.
TC31009D10 96
Note: The splice protector must be pushed over one of the fiber
ends before fusion since this is no longer possible once the fiber ends
have been fused.
Place the fiber with the splice protector in the heater. Only apply
light tension to the splice in order to avoid damage. Make sure that the
fiber remains in position in the splice protector.
Cleaning: Remove any dirt and pieces of fiber.

Flaps closed?
Close all flaps on the splicer.
Case lid: If you are working with a splicer integrated in the case,
clip in the lid of the case and close the case.
Warning: Do not use force to close the case. Make sure that all
parts of the splicer are in their transport position.
TC31009D10 97
Module - 01
Chapter – 06
OFC Test and Measurements
Session Objectives:

• Introduction
• Calibrated Light Source
• The Light Power Meter
• The Optical Attenuator
• Optical Time Domain Reflectometer (OTDR)
• Fiber Optic Power Meter (FOT-12A-50)
• Optical Talk Set (TSH-01 Ver.2 1550nm)
• Continuity Test
• Cable Loss
TC31009D10 98
1. Introduction
Low loss, wide bandwidth, non–inductive transmission without
crosstalk and highly insulated, thin and lightweight optical fiber cables
have revolutionised the field of Telecommunication.
Testing a Fiber–Optic based system requires special
instrumentation if the link itself must be checked. The basic structure of
communication system using Optical Fiber is shown in Fig. 6.1 below:
Fig. 6.1 System Composition
Measurement Requirements
The most commonly used tests in field are Optical Power and Cable
loss. The instruments needed for these tests include a calibrated light
source, a light power meter, an optical attenuator, and optical Time
Domain Reflectometer.
Optical Devices
2. Calibrated Light Source

The calibrated light source is the equivalent of a signal generator.
It must generate light energy signals of known power levels. These light
signals come out through an LED or LASER. The other requirement may
be the need to vary the wave–length (frequency) of the light wave, just as
a signal generator allows the user to vary the frequency. The reason for
this is that optical fibers perform best at specific light wavelengths only
(depending on the exact type of glass used) and the light supplied to the
optical fiber must have the matching wave-length for that fiber type.
3. The Light Power Meter

The light power meter measures the power of any light signal,
which typically ranges from 1 nanowatt (nW) to 2 milliwatts (mW). This is
similar to measuring the voltage or power of an electrical signal. The
power meter must also be calibrated carefully, since its readings form the
TC31009D10 99
basis for determining the amount of light energy that is present. The
power meter uses a precise light–to–electrical–energy transducer and
then measures its electrical output.
4. The Optical Attenuator

The optical attenuator is similar to a simple potentiometer or
circuit used to reduce a signal level. The attenuator is used whenever
performance tests must be run. For example, to see how the bit error
rate is affected by varying the signal level in the link. The optical
equivalent is much more complex than the one used in electronic
circuitry. It can be implemented in two ways. One way is to have a
precise mechanical setup in which the optical signal passes through a
glass plate with differing amounts of darkness and then back to the
optical fiber, as shown in Fig. 6.2. The glass plate has gray density
ranging from 0 percent at one end to 100 percent at the other end. As the
plate is moved across the gap, more or less light energy is allowed to
pass. This type of attenuator is very precise, and can handle any light
wavelength (since the plate attenuates any light energy by the same
amount, regardless of wavelength), but it is mechanically expensive.
Fig. 6.2 An attenuator for fiber–optic
5. Optical Time Domain Reflectometer (OTDR)

The OTDR is an important tool to characterize a fiber’s
attenuation, uniformity, splice loss, breaks and length. Its main
advantage is one–port operation at the fiber input with no need to access
the fiber output. These are measured by launching an optical pulse into
one end of the optical fiber and detecting reflected pulses (back–scattered
light) returning to the instrument. The measuring conditions, items and
measured results are displayed on CRT.
TC31009D10 100
An optical pulse is launched into one end of fiber. This optical

pulse, on encountering microscopic non–uniformities of route, will cause
light scattering in all directions. If the size of non–uniformity is much
less then, the scattering phenomenon is governed by Rayleigh scattering
principle (Scattering coefficient = 1/λ4, i.e. light scattering decreases as
the wavelength increases) out of this scattered light a portion of it, i.e.
back scattered light, may reach back to the fiber input. Infect, due to
limited numerical aperture of the fiber, only a portion of back-scattered
light reaches the fiber input.
When the optical pulse launched into the fiber encounters a
transition in media, for example, at a fiber 'break (Fig. 6.3), connectors,
splicing points and free fiber ends, reflection occurs in addition to
refraction. Depending upon the reflection coefficient at that point, the
reflected light is received back at the sending end of the fiber. This
reflected light is typically 4% of incident light. Thus, the reflected light
received at the sending has a much higher power level as compared to
back–scattered light, which is typically 50 dB lower than the incident
optical power. A typical OTDR arrangement is shown in the Fig. 6.4.
Fig. 6.3 Break in Fiber
Fig. 6.4 OTDR Instrument Principle
TC31009D10 101
A pulse generator drives a laser diode which the launches high

power optical pulses (100 nW up to several mW) with pulse widths of 100
ns to 4 µs and repetition rate of few kHz into fiber using a polarising
beam splitter. Refractive index of fiber core, under test, up to four
decimal points can be entered in the instrument for exact display of fiber
length on x–axis. An APD is used as a detector. Its signal is fed to
amplifier. The diagram in Fig. 6.5 shows a typical signal display on OTDR
screen.
The reflection from the fiber front is the largest signal in an OTDR.
The attenuation of cable section can be obtained from equation:
α = 1/L x 0.5 x 10 log (p1/p2) dB/km
Back scattered light from two non–uniformities reach the input at the
same time if they are spaced apart by half of pulse length in fiber as
shown below.
Fig. 6.5 OTDR Signal Display
Fig. 6.6 Explanation of the ∆z/2 Uncertainty of the OTDR Signal
TC31009D10 102
It may be noted that at the receiving end, it cannot be

distinguished whether the received back–scattered light pertained to
discontinuities of z or z' points. It may be concluded that z–z', i.e. dz/2 is
the unresolvable length of OTDR signal.
For example: A pulse width W of 10 ns and a refractive index n = 1.5
results in Group velocity = (3 x 108)/1.5.
Pulse length in fiber = Pulse width W x Group velocity
Pulse length in fiber = (3 x 108)/(1.5 x 10–8) = 2 m.
Uncertainty(minimum resolvable length in fiber)=2m/2 =1m or +0.5 m
Improvement in received power is possible by increasing pulse
width at the expense of length resolution. Back-scattered signals are
normally buried in noise because of their tiny amplitudes. Special
techniques are used to improve S/N of received signal in OTDR.
Concept of Dead zone

The dead zone is the distance corresponding to the initial length of
fiber under test whose trace is missed in OTDR (as shown in Fig. 6.6)
due to the saturation of the receiver circuit of OTDR by strong signal like
Fresnel reflection and due to imperative property of this circuit that a
certain time is lost in transforming the time into distance. The back-
scattered signals received from the initial distance (dead zone) take too
less a time to be transformed into corresponding trace by the receiver
circuit. The dead zone is the sum of pulse–width and the distance
corresponding to recovery time of the receiver circuit. The smaller the
pulse–width, the smaller the dead zone.
7. Optical Talk Set

This is a dedicated Talk set for fiber optic engineers involved in
telecommunications plant installations and maintenance. This set allows
Full-Duplex conversation between up to four operators with headset or
voice gated speakerphone over a single optical fiber. Designed for long
distance single mode applications.
The features of the set include volume control (both in headset and
speakerphone modes) for the best listening comfort and a Call function
for alerting the companion user.
8. Continuity Test
The simplest is the continuity test, to find out whether there is any
break in the fiber. It requires only a light source and optical power meter.
Some light energy is coupled into the cable, and some should come out
the other end. The test is a good/bad test, which doesn't indicate
anything about the cable loss. The light source and power meter do not
have to be calibrated or accurate for this simple test as in figure below.
TC31009D10 103
Fig. 6.7 Continuity Test
Sometimes, a simple flashlight is used as the light source, and the

light coming out can be seen by eye. This is not a recommended practice
because in a multifibre bundle, the fiber the technician looks into, may
not be the one with the flashlight at the other end. Instead, it may be an
active fiber that has a laser LED as its source, and eye damage can
result. The flashlight and eye should only be used for a single strand of
fiber, when the entire length of fiber is visible, and there can be no
mistakes.
Fig. 6.8 Visual Inspection
9. Cable Loss
To check cables loss, a measured amount of light energy at a
specific wavelength is coupled into the cable, and the amount of output
light power is measured. The cable loss is specified in decibels. Since the
amount of energy coupled into the cable must be known, the light source
must be calibrated, or the power meter must first be used to measure the
output of the source directly and then the output through the cable
using the same source.
TC31009D10 104
Module - 01
Chapter – 07
Optical Connectors and Couplers
Session Objectives:

• Connectors
• Connector Requirement
• Connector Composition
• SC Connectors
• ST Connectors
• LC Connectors
• FC/PC Connector
• MT-RJ Connectors
• MTP/MPO Connectors
• Fibre Connecting Techniques
• Optical Connector types and insertion loss
• Couplers
TC31009D10 105
Fiber-Optic Cable Termination
1. Connectors
The connectors are rematable interconnect devices which provide
flexibility required in a Fiber Optical Transmission system. The basic
function required of connectors is to allow transfer of optical power from
one fiber component to another with minimum loss and possibility of
disconnection and remitting number of times with minimum insertion
loss.
2. Connector Requirement
• The attenuation in optical fiber connectors should be less than
1 dB.
• The connector must provide consistent performance on each
remitting.
• The connector must provide protection to the fiber so that it
does not break while being handled.
• The connectorisation technique should be simple.
• The connector size should not be very much bigger than the
fiber size and it should not be too small.
• Connector must be cost effective.
3. Connector Composition
Connector fundamentally consists of two parts, a plug and an
adapter. For fiber to fiber connections, the fibers are terminated in
individual plugs and mated in the adapter.
For fiber to device connection, the devices may be housed in the
adapter part and the fiber in the plug part. The fixing of the fiber in the
plug may be achieved directly or by using sleeves commonly known as
ferrules. The proper centering in these ferrules could be achieved by
using precision drilled holes, jewels or rods depending on the
arrangement. The adapter provides the alignment mechanism.
The performance of the connectors depends on the accuracy of the
alignment of the optical elements to be connectorised. The basic
elements in the connectors are fiber fixing mechanism and the alignment
mechanism. The alignment accuracy required is of very high to avoid
losses and are consequently quite costly.
There are many types of optical connectors. The one you use
depends on the equipment you are using it with and the application you
are using it on. The connector is a mechanical device mounted on the
end of a fiber-optic cable, light source, receiver, or housing. The
connector allows the fiber-optic cable, light source, receiver, or housing
to be mated to a similar device. The connector must direct light and
collect light and must be easily attached and detached from equipment.
TC31009D10 106
A connector marks a place in the premises fiber-optic data link where

signal power can be lost and the BER can be affected by a mechanical
connection. Of the many different connector types, those for glass fiber-
optic cable and plastic fiber-optic cable are discussed in this chapter.
Other considerations for terminations are repeatability of connection and
vibration resistance. Physical termination density is another
consideration. Commonly used fiber-optic connectors are shown in
Figure. 7.1
Fig 7.1 : Fibre Optic Connectors

4. SC Connectors
SC connectors are used with single-mode and multimode fiber-

optic cables. They offer low cost, simplicity, and durability. SC
connectors provide for accurate alignment via their ceramic ferrules. An
SC connector is a push-on, pull-off connector with a locking tab. Typical
matched SC connectors are rated for 1000 mating cycles and have an
insertion loss of 0.25 dB. From a design perspective, it is recommended
to use a loss margin of 0.5 dB or the vendor recommendation for SC
connectors.
TC31009D10 107
5. ST Connectors
The ST connector is a keyed bayonet connector and is used for

both multimode and single mode fiber-optic cables. It can be inserted
into and removed from a fiber-optic cable both quickly and easily.
Method of location is also easy. ST connectors come in two versions: ST
and ST-II. These are keyed and spring-loaded. They are push-in and
twist types. ST connectors are constructed with a metal housing and are
nickel-plated. They have ceramic ferrules and are rated for 500 mating
cycles. The typical insertion loss for matched ST connectors is 0.25 dB.
From a design perspective, it is recommended to use a loss margin of 0.5
dB or the vendor recommendation for ST connectors.
6. LC Connectors
LC connectors are used with single-mode and multimode fiber-

optic cables. The LC connectors are constructed with a plastic housing
and provide for accurate alignment via their ceramic ferrules. LC
connectors have a locking tab. LC connectors are rated for 500 mating
cycles. The typical insertion loss for matched LC connectors is 0.25 dB.
From a design perspective, it is recommended to use a loss margin of 0.5
dB or the vendor recommendation for LC connectors.
7. FC/PC Connector
FC/PC has been one of the most popular single-mode connectors for
many years. It screws on firmly, but make sure you have the key aligned
in the slot properly before tightening. SCs and LCs are replacing it.
8. MT-RJ Connectors
MT-RJ connectors are used with single-mode and multimode fiber-

optic cables. The MT-RJ connectors are constructed with a plastic
housing and provide for accurate alignment via their metal guide pins
and plastic ferrules. MT-RJ connectors are rated for 1000 mating cycles.
TC31009D10 108
The typical insertion loss for matched MT-RJ connectors is 0.25 dB for
SMF and 0.35 dB for MMF. From a design perspective, it is
recommended to use a loss margin of 0.5 dB or the vendor
recommendation for MT-RJ connectors.
9. MTP/MPO Connectors
MTP/MPO connectors are used with single-mode and multimode

fiber-optic cables. The MTP/MPO is a connector manufactured
specifically for a multifiber ribbon cable. The MTP/MPO single-mode
connectors have an angled ferrule allowing for minimal back reflection,
whereas the multimode connector ferrule is commonly flat. The ribbon
cable is flat and appropriately named due to its flat ribbon-like structure,
which houses fibers side by side in a jacket. The typical insertion loss for
matched MTP/MPO connectors is 0.25 dB. From a design perspective, it
is recommended to use a loss margin of 0.5 dB or the vendor
recommendation for MTP/MPO connectors.
TC31009D10 109
• Connectors are rematable devices

• Provide flexibility for interconnection of different section of fiber in
Fiber Optical Transmission System
Requirements
• Attenuation should be low <1db

• Should be good in performance
• Should protect the fiber while handling
• Should be made with simple method - No special tools should be
required for making the connection
• Small in size
• Cost effective
Connectors & Couplers composition
• Connectors are plug in type

• Couplers are socket or adapter type
• Fibers terminated in plug
• Adapters have two sides. Each side plug is screwed
10. Fibre Connecting Techniques
• Core Centered Connectors – Core is centered in the plug. Pig tails are
this type. Made in factory only
• Core Aligned Connectors – Core is aligned in the plug by a special
machine. Made in factory only
• Precision Mould Connectors – Made from plastic materials and
molded in machine
• Optical Tech. – Optically collimating, Focussing connection, Fibre
taper connection
TC31009D10 110
• Optically collimating and focusing tech. Use lenses to collimate or

focus the light
• Fibre taper provide large area of contact and reduce loss
Types of Connectors
• FC, D3, D4, Dorran type connectors are used
• These connectors have pin and notches arrangement to provide good
performance and less loss using repeatedly
• FC type connectors are used widely now
Fiber-to-fiber interconnection can consist of a splice, a permanent
connection or a connector, which differs from the splice in its ability to
be disconnected and reconnected.
Ferrule
• The fiber is mounted in a long, thin cylinder, the ferrule, which acts
as a fiber alignment mechanism.
• The ferrule is bored through the center at a diameter that is slightly
larger than the diameter of the fiber cladding.
• The end of the fiber is located at the end of the ferrule.
• Ferrules are typically made of metal or plastic.
Connector Body
• The connector body holds the ferrule.
• It is usually constructed of metal or plastic and includes one or more
assembled pieces which hold the fiber in place.
• The Connector body assemblies vary among connectors.
• The ferrule extends past the connector body to slip into the coupling
device.
TC31009D10 111
Connector and Coupling Device

The cable is attached to the connector body. It acts as the point of
entry for the fiber. Typically, a strain-relief boot is added over the
junction between the cable and the connector body, providing extra
strength to the junction.
The Coupling Device
• Do not use the male-female configuration.

• Instead, a coupling device such as an alignment sleeve is used to
mate the connectors.
• These devices are also known as feed-through bulkhead adapters.
11. Optical Connector types and insertion loss
C o n n e c to r In s e r tio n L o s s R e p e a ta b ility F ib e r T y p e
0 .5 0 -1 .0 0 d B 0 .2 0 d B SM, MM
FC
0 .2 0 -0 .7 0 d B 0 .2 0 d B SM, MM
FDDI
0 .1 5 d b (S M )
0 .2 d B SM, MM
LC 0 .1 0 d B (M M )
0 .3 0 -1 .0 0 d B 0 .2 5 d B SM, MM
M T A r ra y
0 .2 0 -0 .4 5 d B 0 .1 0 d B SM, MM
SC
0 .2 0 -0 .4 5 d B 0 .1 0 d B SM, MM
S C D u p le x
T y p . 0 .4 0 d B (S M ) T y p . 0 .4 0 d B (S M )
SM, MM
ST T y p . 0 .5 0 d B (M M ) T y p . 0 .2 0 d B (M M )
TC31009D10 112
Various Optical Connectors
E2000 Connector: E2000/LX-5 is like a LC but with a shutter over the

end of the fiber
FDDI Connector
Developed by ANSI for use in FDDI network, this is a duplex connector
using two 2.5mm ferrules.
TC31009D10 113
12. Couplers
FC Coupling SC Coupling
ST Coupling LC Coupling
Damaged Connector
• A number of events can damage fiber optic connectors.
• Unprotected connector ends can experience damage by impact,
airborne dust particles, or excess humidity or moisture.
Effects on Fibre Optics Connectors

• Never clean an optical connector attached to a fiber that is carrying
light.
• The micro-explosions at the tip of the connector can leave pits in the
end of the connector and crack the connector’s surface, destroying its
ability to carry light with low loss.
Cleaning
• The fiber end face and ferrule must be absolutely clean before it is
inserted into a transmitter or receiver.
TC31009D10 114
• Dust, lint, oil (from touching the fiber end face), or other foreign
particles obscure the end face, compromising the integrity of the
optical signal being sent over the fiber.
• Larger dust particles (9 µm or larger) can completely obscure the core
of a single-mode fiber.
• Fiber optic connectors need to be cleaned every time they are mated
and unmated.
• Cover a fiber optic connector when it is not in use.
• Unprotected connector ends are most often damaged by impact, such
as hitting the floor.
• Most connector manufacturers provide some sort of protection boot.
• The protectors cover the entire connector end, but they are generally
simple closed-end plastic tubes that fit snugly over the ferrule only.
• These boots will protect the connector's polished ferrule end from
impact damage that might crack or chip the polished surface.
Cleaning Technique
• Use only industrial grade 99% pure isopropyl alcohol and lint-free
tissue required.
• Fold the tissue twice so it is four layers thick.
• Saturate the tissue with alcohol. First clean the sides of the connector
ferrule.
• Place the connector ferrule in the tissue, and apply pressure to
the sides of the ferrule.
• Rotate the ferrule several times to remove all contamination from the
ferrule sides.
• Now take a clean tissue and saturated with alcohol and that it is still
four layers thick.
• Put the tissue against the end of the connector ferrule.
• Put your fingernail against the tissue so that it is directly over the
ferrule.
• Now scrape the end of the connector until it squeaks. It will sound
like a crystal glass that has been rubbed when it is wet.
• Mate the connector immediately! Don’t let the connector lie around
and collect dust before mating.
• Air can be used to remove lint or loose dust from the port of a
transmitter or receiver to be mated with the connector. Never insert
any liquid into the ports.
Handling
• Never touch the fiber end face of the connector.
• Connectors that are not in use should be covered over the ferrule by a
plastic dust cap.
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Module - 01
Chapter – 08
Coding Theory
Session Objectives:

concepts and explain the features of transmission systems such as:
• Digital-To-Digital encoding
• Unipolar
• Polar
• Bipolar (Biphase)
• Line Codes
• AMI Code
• HDB3 Code
• CMI Code (Coded Mark Inversion Code)
• 5B6B Code
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1. Digital-To-Digital encoding
Digital-to-digital encoding is the representation of digital
information by a digital signal. For example, when we transmit data
from the computer to a printer, both the original data and the
transmitted data are digital. In this type of encoding, the binary 1s and
0s generated by a computer are translated into a sequence of voltage
pulses that can be propagated over a wire.
Fig 8.1 Digital to digital encoding
There are three types of digital encoding :

• Unipolar
• Polar
• Bipolar
2. Unipolar
Unipolar encoding is very simple and very primitive. Although it is
almost obsolete today, its simplicity provides an easy introduction to the
concepts developed with the more complex encoding systems and allows
us to examine the kinds of problems that any digital transmission
system must overcome. Unipolar encoding is so named because it uses
only one polarity. Therefore, only one of the two binary states is
encoded, usually the 1. The other state, usually the 0, is represented by
zero voltage, or an idle line.
Fig 8.2 Unipolar encoding
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3. Polar
Polar encoding uses two voltage levels; one positive and one
negative. By using both levels, in most polar encoding methods, the
average voltage level on the line is reduced.
Types of polar encoding
1. NRZ
• NRZ-L
• NRZ-I
2. RZ
3. BIPHASE
• Manchester
• Differential Manchester
Non-Return to Zero (NRZ)
In NRZ encoding, the level of the signal is always either positive or

negative. Unlike in unipolar encoding, where a 0 bit is represented by an
idle line, in NRZ systems if the line is idle, it means no transmission is
occurring at all. The two most popular methods of NRZ transmission are
discussed below.
NRZ-L
In NRZ-L encoding, the level of the signal depends on the type of
bit is represents. A positive voltage means the bit is a 1, and negative
voltage means the bit is a 0; thus, the level of the signal is dependent
upon the state of the bit.
Fig 8.3 NRZ-I and NRZ-L Encoding
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NRZ-I
In NRZ-I, an inversion of the voltage level represents a 1 bit. It is
the transition between a positive and negative voltage, not the voltages
themselves that represents a 1 bit. A 0 bit is represented by no change.
An advantage of NRZ-I over NRZ-L is that because the signal changes
every time a 1 bit is encountered, it provides some synchronization. A
series of seven 1s will cause seven inversions. Each of those inversions
allows the receiver to resynchronize its timer to the actual arrival of the
transmission. Statistically, strings of 1s occur more frequently in
transmissions than do strings of 0s.
Return to Zero (RZ)

As we can see, anytime the original data contain strings of
consecutive 1s or 0s, the receiver can lose its place. As we mentioned in
our discussion of unipolar encoding, one way to assure synchronization
is to send a separate timing signal on a separate channel. However, this
solution is both expensive and prone to errors of its own. A better
solution is to somehow include synchronization in the encoded signal,
something like the solution provide by NRZ-I, but one capable of
handling strings of 0s as well as 1s.
To assure synchronization, there must be a signal change for each
bit. The receiver can use these changes to build up, update, and
synchronize its clock. As seen above, NRZ-I accomplishes this for
sequences of 1s. But to change with every bit, we need more than just
two values. One solution is return to zero (RZ) encoding, which uses
three values: positive, negative, and zero. In RZ, the signal changes not
between bits but during each bit. Like NRZ-L, a positive voltage means 1
and a negative voltage means 0. But, unlike NRZ-L, halfway through
each bit interval, the signal returns to zero. A 1 bit is actually
represented by positive-to-zero, and a 0 bit by negative-to-zero, rather
than by positive and negative alone. The following figure illustrates the
concept.
Fig 8.4 RZ Encoding
TC31009D10 119
The main disadvantage of RZ encoding is that it requires two signal

changes to encode one bit and therefore occupies more bandwidth. But
of the three alternatives we have examined so far, it is the most effective.
4. Bipolar (Biphase)
Bipolar encoding, like RZ, uses three voltage levels: Positive,

negative and zero. Unlike RZ, however, the zero level in bipolar encoding
is used to represent binary 0. Positive and negative voltages represent
alternating 1s. If the first 1 bit is represented by the positive amplitude,
the second will be represented by the negative amplitude, the third by
the positive amplitude, and so on. This alternation occurs even when the
1 bits are not consecutive.
Probably the best existing solution to the problem of
synchronization is bipolar encoding. In this method, the signal changes
at the middle of the bit interval but does not return to zero. Instead, it
continues to the opposite pole. As in RZ, these mid interval transitions
allow for synchronization. As mentioned earlier, there are two types of
bipolar encoding in use on networks today:
• Manchester
• Differential Manchester.
Manchester
Manchester encoding uses the inversion at the middle of each bit
interval for both synchronization and bit representation. A negative-to-
positive transition represents binary 1 and positive-to-negative transition
represents binary 0. By using a single transition for a dual purpose,
Manchester encoding achieves the same level of synchronization as RZ
but with only two levels of amplitude.
Differential Manchester
In Differential Manchester, the inversion at the middle of the bit
interval is used for synchronization, but the presence or absence of an
additional transition at the beginning of the interval is used to identify
the bit. A transition means binary – and no transition means binary 1.
Differential Manchester requires two signal changes to represent binary 0
but only one to represent binary1.
TC31009D10 120
Fig 8.5 Manchester and Differential Manchester Encoding
Three types of bipolar encoding are in popular use by the data

communications industry, such as AMI, B8ZS and HDB3.
5. Line Codes
If encoder output of PCM equipment is transmitted over the

transmission medium, viz. VF cable pair, the signal is likely to under go
high frequency attenuation distortion and cross–talk. Moreover, the
signal has strong DC content and thus prevents the use of AC coupled
circuits.
For distortion free transmission, the encoder output should be
converted into a suitable code which will match the characteristics of the
medium. This code is called the "Line Code" and the signal converted to
the line code is called the line signal.
Characteristics
The line code needs to have the following characteristics:
• Restricted bandwidth
• Low energy in the upper part of the signal spectrum to
reduce attenuation distortion.
• Low energy in the lower part of the spectrum to reduce
cross–talk.
• No DC component so that transformers can be used for
coupling purposes.
• Contain adequate timing information.
• Have an in–built error monitoring capability.
TC31009D10 121
Following codes are available as Line codes.

• NRZ Binary
• RZ Binary
• Bipolar (AMI)
• HDB3
• CMI
• 5B6B
Because of the strong DC component and low frequency content,

NRZ and RZ codes are not suitable for transmission.
6. AMI Code
AMI stands for "Alternate Mark Inversion". This code solves the DC
content problem. Here, a logic '0' is represented by 0 volt and logic '1' is
alternately encoded with positive and negative voltages. Therefore, the
average voltage is maintained very close to zero and hence there is no DC
component. Under steady state conditions, a low DC of the order of 0.4
to 0.8 volts only remains.
By inverting on each occurrence of a 1, bipolar AMI accomplishes
two things:
• The DC component is zero,
• A long sequence of 1s stays synchronized.
Three is no mechanism to ensure the synchronization of a long
string of 0s.
Fig. 8.6 AMI Code Signal Waveform
Two variations of bipolar AMI have been developed to solve the

problem of synchronizing sequential 0s. The first, used in North
America, is called bipolar 8-zero substitution (B8ZS). The second, used
in Europe and Japan, is called high-density bipolar3 (HDB3). Both are
TC31009D10 122
adaptations of bipolar AMI that modify the original pattern only n the
case of multiple consecutive 0s.
7. HDB3 Code
To overcome the timing difficulties in the AMI code, another code
called the HDB3 code has been devised. The abbreviation HDB stands for
HIGH DENSITY BIPOLAR code.
The HDB3 code is actually a code from a family of codes derived
from what is called Binary N Zero Substitution or BNZS method.
In this method, the PCM signal is usually transmitted according to
the AMI code, but when a string of N zeros is encountered, the N zeros
are replaced by a special code which will deliberately introduce a bipolar
deviation or violation. Normally in the AMI code, if there are N zeros, they
will be transmitted as such. But in the BNZS method, a '1' pulse is
introduced deliberately. The polarity of this '1' depends upon the polarity
of the previous mark encountered. This additional '1' pulse introduced in
place of a '0' is called a 'violation'.
When the substitution of a zero by a violation pulse is done for 4
zeros, i.e. N = 4, the BNZS code is called the B4ZS code. Since this code
precludes strings of zeros greater than three, it is also referred to as a
HDB3 code. Here, when the number of zeros is more than 3, the fourth
bit position is filled with a violation pulse.
Consecutive violations are made to be of opposite polarity so that
these violations themselves do not produce any DC component.
The violation pulse is always placed in the last bit position.
Suppose there are 4 zeros coming in a row. Then the HDB3 code for this
would be ‘ B00V ’ in general where V is the violation pulse. The polarity
of this depends on the polarity of the last '1' and the number of '1's
encountered prior to the four zeros.
a) NRZ Code b) RZ Code c) HDB3 Code

Fig. 8.7 Example of HDB3 Code Conversion
TC31009D10 123
Fig 8.8 HDB3 Encoding
The first bit of the code was shown as B in above. B is set to '0' if
the number of '1's encountered prior to the violation is ODD. If it is
EVEN or ZERO, then the "B" bit is filled with a '1' whose polarity is in
accordance with the AMI.
Examples of HDB3 Code Conversion
Notice that up to pulse Z, the HDB3 code follows the AMI code.
After pulse Z, we have four consecutive zeros. This calls for a violation.
Prior to the arrival of these zeros, three '1's were encountered, i.e.
number of '1's preceding the violation is ODD.
This means that the HDB3 substitution for the zeros will be of the
form 000V. Also, the polarity of the last '1' before the arrival of the zeros
is positive. Therefore, code, i.e. if the previous '1' was positive, then B is
'1' with negative polarity and vice versa.
8. CMI Code (Coded Mark Inversion Code)
This is a 2 level NRZ code in which a binary '0' is coded as '1' and
binary '1's are coded alternatively as a logic '0' or '1'. In case of a binary
'0', the two CMI bits '0' and '1' are for half clock duration whereas for
binary '1's the '0' and '1's are for full clock duration. This is illustrated in
Fig. 8.9.
This is basically a binary code and the bit rate of the code is twice
the Bipolar AMI code. For this reason CMI code is grouped with IB2B
family of line codes. The CMI code has high clock content and for this
TC31009D10 124
reason the CMI code is recommended by CCITT for 140 Mb/s multiplex
equipment.
Fig. 8.9 CMI Code – An Example
9. 5B6B Code
The objective of this coding is to modify the input bit pattern such
that the continuity of ones or zeros is reduced and a mark ratio of nearly
½ is maintained in the resultant output bit pattern.
TC31009D10 125
Module - 01
Chapter – 09
Basics of Transmission Systems
Session Objectives:

concepts and explain the features of:
• Plesiochronous Digital Hierarchy

• Synchronous Digital Hierarchy
• Microwave Systems
• MW Applications
• MW System Capacity
• Communication through Satellite
• Features of Satellite Communications
• Advantages of Satellite Communications
• Frequency Bands
• VSAT System Components and Specifications
• HVNET – DOT VSAT Network
TC31009D10 126
1. Plesiochronous Digital Hierarchy (PDH)

The PDH is a technology used in telecommunications networks to
transport large quantities of data over digital transport equipment such
as fiber optics and microwave systems. The term Plesiochronous is
derived from Greek plesio, meaning near, and chronos, time, and refers
to the fact that PDH networks run in a state where different parts of the
network are almost, but not quite perfectly, synchronised.
PDH is now being replaced by Synchronous Digital Hierarchy
(SDH) equipment in most telecommunications networks. PDH allows
transmission of data streams that are nominally running at the same
rate, but allowing some variation on the speed around a nominal rate. By
analogy, any two watches are nominally running at the same rate,
clocking up 60 seconds every minute. However, there is no link between
watches to guarantee they run at exactly the same rate, and it is highly
likely that one is running slightly faster than the other.
The European and American versions of the PDH system differ
slightly in the detail of their working, but the principles are the same. In
European system, basic data transfer rate is a data stream of 2.048
Mbps (megabits/second). For speech transmission, this is broken down
into 30 x 64 kbit/s (kilobits/second) channels plus 2 x 64 kbit/s
channels used for signaling and synchronisation. Alternatively, the whole
2 Mbit/s may be used for non speech purposes, for example, data
transmission.
The exact data rate of the 2 Mbit/s data stream is controlled by a
clock in the equipment generating the data. The exact rate is allowed to
vary some percentage either side of an exact 2.048 Mbit/s. This means
that different 2 Mbit/s data streams can be (probably are) running at
slightly different rates to one another.
In order to move multiple 2 Mbit/s data streams from one place to
another, they are combined together, or "multiplexed" in groups of four.
This is done by taking 1 bit from stream #1, followed by 1 bit from
stream #2, then #3, then #4. The transmitting multiplexer also adds
additional bits in order to allow the far end receiving multiplexer to
decode which bits belong to which 2 meg data stream and so correctly
reconstitute the original data streams. These additional bits are called
"justification" or "stuffing" bits.
Because each of the four 2 Mbit/s data streams is not necessarily
running at the same rate, some compensation has to be made. The
transmitting multiplexer combines the four data streams assuming that
they are running at their maximum allowed rate. This means that
occasionally, (unless the 2 Mbit/s really is running at the maximum rate)
the multiplexer will look for the next bit but it will not have arrived. In
this case, the multiplexer signals to the receiving multiplexer that a bit is
"missing". This allows the receiving multiplexer to correctly reconstruct
TC31009D10 127
the original data for each of the four 2 Mbit/s data streams, and at the
correct, different Plesiochronous, rates.
The resulting data stream from the above process runs at 8.448
Mbit/s (about 8 Mbit/s). Similar techniques are used to combine four x 8
Mbit/s together, giving 34 Mbit/s. Four x 34 Mbit/s, gives 140. Four x
140 gives 565.
565 Mbit/s is the rate typically used to transmit data over a fiber
optic system for long distance transport. Recently, telecommunications
companies have been replacing their PDH equipment with SDH
equipment capable of much higher transmission rates.
2. Synchronous Digital Hierarchy (SDH)

Synchronous Digital Hierarchy (SDH) signals the beginning of a
new phase in evolution of the world’s telecommunication network. SDH
will bring a revolution in telecommunication services, which will have
far-reaching effects for end users, service providers and equipment
manufacturers alike. With introduction of SDH, the transmission
network will enter a new era, which can be compared in scale to that
occurred following the introduction of PCM and Optical fiber.
As end users (particularly business users) become more dependent
on effective communications, pressure builds up for a reliable and
flexible network with unlimited bandwidth. The complexity of the current
network, based on Plesiochronous transmission systems, means that
network operators are unable to meet this demand.
The current Plesiochronous Digital Hierarchy (PDH) evolved in
response to the demand for plain voice telephony. (Sometimes called
POTS- Plain old Telephone Service) is not ideally suited to the efficient
delivery of and management of high bandwidth connections.
Synchronous transmission systems address the shortcomings of the
PDH. Using essentially the same fiber, the synchronous network is able
to significantly increase the available bandwidth while reducing the
amount of equipment in the network. In addition the provision within the
SDH for sophisticated network management includes significantly more
flexibility in the network.
Deployment of synchronous transmission systems will be
straightforward due to their ability to inter work with existing
Plesiochronous systems. The SDH defines a structure which enables
Plesiochronous signals to be combined together and encapsulated within
a standard SDH signal. This protects network operators’ investment in
Plesiochronous Equipments and enables them to deploy synchronous
equipment in a manner suited to the particular needs of their network.
As Synchronous Equipments become established within the
network, the full benefits it brings will become apparent. The network
operator will experience significant cost savings associated with reduced
amount of hardware in the network and increased efficiency and
TC31009D10 128
reliability of the network will lead to savings resulting from a reduction in

operation and maintenance costs.
The sophisticated network management capabilities of a
synchronous network will give a vast improvement in the control of
transmission networks. Improved network restoration and
reconfiguration capabilities will result in better availability and faster
provisioning of services.
The SDH offers network operators a future proof network solution.
It has been designed to support future services such as Metropolitan
Area Network (MANs), Broadband ISDN etc.
2.1 SDH Evolution

PDH has reached a point where it is no longer sufficiently flexible
or efficient to meet the demands being placed on it. As a result,
synchronous transmission was thought to overcome the problems
associated with Plesiochronous transmission. In particular, the inability
of PDH to extract individual circuits from high capacity systems without
having to demultiplex the whole system.
Attempts to formulate a set of standards covering optical
transmission of synchronous signals began in the U.S. at beginning of
1984. The aim was to have a synchronous standard to allow the
interconnection of equipment from more than one vendor. In order to
move away from proprietary interfaces and achieve true interconnectivity
between vendors, subcommittee T1X1of the American National standard
institute (ANSI) began work in 1985 on developing a Standard Optical
NETwork (SONET) based on a proposal by Bell core.
In 1986, the CCITT became interested in the work being carried
out on SONET and after much debate on how to incorporate both US and
European transmission hierarchies, final agreement was reached in
February 1988 and CCITT working group XVIII brought out the
recommendations on SDH, published in CCITT Blue Book 1989. Since
then, an ongoing standards effort has continued to develop and refine
the SDH standards.
S.D.H. evolution is possible because of the following factors:

• Fiber Optic Bandwidth: The bandwidth in Optical Fibre can be
increased and there is no limit for it. This gives a great advantage
for using SDH.
• Technical Sophistication: Although, SDH circuitry is highly
complicated, it is possible to have such circuitry because of VLSI
technique which is also very cost effective.
• Intelligence: The availability of cheaper memory opens new
possibilities.
TC31009D10 129
• Customer Service Needs: The requirement of the customer with

respect to different bandwidth requirements could be easily met
without much additional equipment.
The different services it supports are:
• Low/High speed data.
• Voice
• Interconnection of LAN
• Computer links
• Feature services like H.D.T.V.
• Broadband ISDN transport (ATM transport)
2.2 Advantages of SDH
The advantages of the SDH system are listed below:

• First world standard in digital format.
• First optical Interfaces.
• Transversal compatibility reduces networking cost. Multi-
vendor environment drives price down
• Flexible synchronous multiplexing structure.
• Easy and cost-efficient traffic add-and-drop and cross connect
capability.
• Reduced number of back-to-back interfaces improves network
reliability and serviceability.
• Powerful management capability.
• New network architecture. Highly flexible and survivable self
healing rings available.
• Backward and forward compatibility: Backward compatibility
to existing PDH
• Forward compatibility to future B-ISDN, etc.
2.3 When do we use SDH?
• When networks need to increase capacity, SDH simply acts as a

means of increasing transmission capacity.
• When networks need to improve flexibility, to provide services
quickly or to respond to new change more rapidly.
• When networks need to improve survivability for important user
services.
• When networks need to reduce operation costs , which are
becoming a heavy burden
TC31009D10 130
3. Microwave Systems
With the advent of mass scale industrialisation in our country, the
demand for more communication facilities came up. Several new
telephone exchanges have been installed throughout the country for local
communication and more and more carrier channels have been provided
for carrying the trunk traffic. With the planned introduction of
Subscriber Trunk Dialing throughout the country, the number of carrier
chls required to interconnect different cities became too high to be
accomplished by overhead lines. Thus, U/G Cables Carrier Systems were
introduced, the first of them being the symmetrical pair Cable Carrier
System between Calcutta and Asansol with an ultimate capacity of 480
channels. Then came the Co–axial Cable Carrier System linking all major
cities in the country.
With the development of Microwave technique, which can provide
large block of circuits at comparative cost, the problem of long distance
communication circuits appear virtually solved. A brief description of the
Microwave technique is attempted in the following paragraphs.
Electromagnetic waves can be broadly classified in terms of frequencies
as follows
Range Name Wavelength Uses

0–30 KHz V.L.F. Up to 10 km. Used for long communication.
Has limited information.
Bandwidth requires very high
power.
30–300 KHz L.F. 10 km to 1 km
0.3–3 MHz M.F. 1 km to 100 m Radio Broadcast, Marine Power
in KW, ground wave
propagation, i.e. follows the
curvature of the Earth.
3–30 MHz H.F. 100 m to 10 m Long haul point to point
communication. Propagation is
by one or more reflections from
ionosphere layers and so
subject to variations.
30–300 MHz V.H.F. 10 m to 1 m Line of sight, Troposcatter
communication.
0.3–3 GHz U.H.F. 1 m to 10 cm. –––––– do ––––––
3–30 GHz S.H.F. 10 cm to 1 cm. Line of sight, terrestrial M/W
and Satellite communication.
30–300 GHz E.H.F. 1 cm to 1 mm. Experimental.
TC31009D10 131
The term SHF corresponds to "MICROWAVE" Centrimetric waves.

As a convention frequencies, above 1 GHz and up to 40 GHz are termed
as Microwave. However, most of the m/w systems available are in the
range of 1 to 18 GHz.
3.1 MW Applications
M/W frequency bands are used for the following services:

• Fixed Radio Communication Services.
• Fixed Satellite Services.
• Mobile Services.
• Broadcasting Services.
• Radio Navigation Services.
• Meteorological Services.
• Radio Astronomy Services.
To meet the requirements of all above mentioned services, co–

ordination among the users of M/W spectrum is necessary. In this
regard (in the national context) the wireless planning and co–ordination
wing (WPC) of the ministry of communication has allotted m/w
frequencies spectrum, on the basis of various wireless users classified as
general users and major users.
Wireless users who are permitted to plan their services and take
action for the development of the required equipments are major users.
DOT has been nominated as a major wireless user by the WPC in
1981 in the following sub base band of the m/w spectrum for fixed radio
communication.
Microwave Spectrum Available for DOT (Table 1)
Band Bandwidth Available Spectrum Space

2 GHz 300 MHz 2000–2300 MHz
4 GHz 900 MHz 3300–4200 MHz
6 GHz 1185 MHz 5925–7110 MHz
7 GHz 300 MHz 7425–7725 MHz
11 GHz 1000 MHz 10,700–11,700 HHz
13 GHz 500 MHz 12,750–13,250 MHz
In India the first M/w System was completed in December, 1965

between Calcutta and Asansol with a system capacity of 1200 channels.
At present thousands of kilometers of M/W systems are scattered
throughout the country and further expansion is taking place at a very
large rate.
TC31009D10 132
3.2 Frequency Characteristics

Microwaves are very short frequency radio waves that have many
of the characteristics of light wave in that they travel in line–of–sight
paths and can be reflected, boomed and focused. By focusing these ultra
high radio waves into a narrow beam, their energies are concentrated
and relatively low transmitting power is required for reliable transmission
over long distance.
3.3 MW System Capacity

Microwave communication systems are used to carry telephony,
television and data signals. Majority of the systems, however, carry
multi–channel telephone signals. The spectrum of the multichannel
telephone signal is shown in Table 1. This signal is also called base band.
Individual telephone channels, 4 KHz wide (300 to 3400 Hz for speech
and the remaining for signaling and guard band) are multiplexed
together in a multiplex equipment to get the base band. The base band
frequency occupied by some typical channel capacity system is given in
Table below:
Channel Capacity Base band frequency(in KHz)

60 channels 12–252
1800 channels 312–8120/316–8204
2700 channels 312–12336/316–12388
The system capacity of line of sight systems ranges from 60

telephone channels to 2700 channels over a Radio bearer with a few
systems of lower capacities varying from 60 to 600 channels. On the
same m/w route one can use more than one radio channels, thus getting
still larger capacity. As an example one can accommodate 8 go and 8
return RF channels each with a capacity of 1800 telephone channels in a
500 MHz bandwidth. Of course, in such cases usually one or two RF
channels are kept as a standby which is switched over automatically on
fading or equipment failure. Usually the system with capacities up to 300
channels is called narrow band system and the systems providing more
than 300 channels are called wide band system. Microwave systems
used to provide communication on major trunk routes with high traffic
density and serving long distances are classified as long haul m/w
systems. 2, 4, and 6 GHz systems are long haul systems. Systems used
to provide communication over short distances for trunk routes with
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light traffic density are classified as short haul system. 7 and 11 GHz
systems are short haul systems.
Fig. 9.1 Typical Microwave Radio System
Fig. 9.2 Schematic of a Microwave Terminal
TC31009D10 134
4. Communication through Satellite

Long distance communication using conventional techniques like
coaxial cable or microwave radio relay links involves a large number of
repeaters. For radio relay links of repeater spacing is limited by line of
sight and is of the order of tens of kms. As the number of repeaters
increase system performance and reliability are degraded. Tropo scatter
propagation can cover several hundred kms. but the channel capacity is
limited and costs are high due to necessity of large antennas and high
transmit power. HF communication is subject to fading due to
ionosphere disturbances and channel capacity is severely restricted due
to limited bandwidth available. Large areas could be covered if the height
of microwave repeater could be increased by putting it on board an
artificial earth satellite. Science Fiction writer Arthur C. Clarke in an
article in Wireless World in 1945 proposed that worldwide coverage could
be obtained by using three microwave repeaters placed in a geostationary
orbit at the height of about 36000 kms. with a period of 24 hours (Fig.
9.3).
Fig. 9.3 Modes of Communication
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Fig. 9.4 Global Coverage with Geo stationary Satellite
Satellite communication provides a practical and economical

means of long haul communication traffic in a country with a large
geographical area. It also enables communication service to those areas
which are virtually INACCESSIBLE by other conventional forms of
communication system due to natural physical barriers.
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4.1 Principles and Features of Satellite Communications
Principles
In satellite communications, a satellite with microwave radio
repeater equipment receives and amplifies radio waves sent from earth
stations and returns them to the earth.
A geo-stationary satellite is launched above the equator 36,000 km
high above the earth. Its period round the earth coincides with that of
the earth rotation. Therefore, the satellite looks as if it is stationary from
the earth. If three (3) communication satellites are launched
equidistantly above the equator (See Fig. 9.4), it can serve almost all
communication networks round the world. Therefore, to facilitate public
international telecommunications, INTELSATS IV and V have been
launched above the Atlantic, Pacific, and Indian Oceans. These networks
cover almost all countries around the world.
Features
For international communication, a submarine cable along the
Atlantic Ocean was installed in 1857. Also, short–wave radio
communication (invented by Marconi in 1886) has been in use. However,
short wave radio communication has disadvantages of:
• Small transmission capacity; only small telephone channels
can be used to transmit.
• Fading in wave propagation; interferes with stability of
transmission. Although over–the–horizon propagation is
used for short distance international communications, it is
impossible to apply it to transoceanic long distance
communications.
Unlike other system, geo-stationary satellite communication
systems have the following features
• Stable and large capacity communication.
• Costs of establishment and maintenance do not depend on
communication distance. The costs of submarine and over–
the–horizon systems are proportional to the length, but those
of the satellite system do not affect the communication
distance. Therefore, the satellite system is ideal for long
distance communications.
• Multiple accesses are possible. Signals sent from an earth
station can be received at several earth stations
simultaneously. Therefore, it can transmit signals to many
stations simultaneously, such as TV. Actually, increasing of
submarine cable's capacity and distance between repeaters,
can make submarine cables competitive to satellite
communication specially when very large capacity is
required but for small traffic size countries, satellite
TC31009D10 137
communication is unavailable for the independent

communication services.
4.2 Advantages of Satellite Communications
Large coverage: Almost one–third of the earth with exception of

polar regions is visible from geostationary orbit. It is, thus, possible to
cover about 10,000 kms. distance irrespective of intervening terrain with
a single satellite.
High quality: Satellite links can be designed for high quality
performance. The link performance is highly stable since it is free from
ionospheric disturbances, multipath effects or fading.
High reliability: Reliability is high since there is only one repeater
in the link.
High capacity: With microwave frequencies, wide bandwidths are
available and large communication capacity can be obtained.
Flexibility: In a terrestrial system, communication is tied down to
the links installed. On the other hand, satellite communication is well
suited for changing traffic requirements, locations and channel
capacities.
Speed of installation: Installation of earth terminals can be
achieved in a short time as compared to laying of cables or radio relay
links.
Mobile, short–term or emergency communications: With
airliftable or road transportable terminals, short–term or emergency
communications can be quickly provided. Reliable long distance land
mobile, maritime mobile and aeronautical mobile services are feasible
only by means of satellite.
Satellite communication is ideally suited for point to multipoint
transmission on broadcasting over large areas. Application of satellites
for TV broadcasting, audio and video distribution and teleconferencing,
facsimile, data and news dissemination is, therefore, increasing rapidly.
All types of common services are possible.
4.3 Satellite Communication Network

Satellite Communication Network could be defined as an ensemble
of earth stations of pre–determined size spread over a pre–defined
coverage area, interconnected through a suitably designed satellite,
placed at a pre–determined location in properly chosen orbit around the
earth. Thus, two important elements of a satellite communication
network are:
• Space Segment
• Ground Segment
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Uplinks and Down Link

Uplink is the radio path from Ground segment, i.e. earth station to
the Space segment, i.e. satellite, whereas Downlink is the radio path
from space segment, i.e. satellite to the ground segment, i.e. earth
station.
4.4 Frequency Bands

Choice of Frequency band for space communication depends upon
• Band–width required.
• Noise consideration
• Propagation factors
• Technological developments with regard to component and
device.
As the signal levels from the satellite are expected to be very low,
any natural phenomenon to aid the reception of the incoming signals
must be exploited. Between the frequencies of 2 GHz to 10 GHz, the level
of the sky–noise reduces and this band of frequencies is known as the
'microwave window'.
The most of the communication satellites as on today are using a
frequency of 6 GHz for "Up link" and 4 GHz for "Down link" transmission.
These frequencies are preferred because of
• Less atmospheric absorption than higher frequency.

• Less noise both galactic and manmade.
• Less space loss compared to higher frequency.
• A well developed technology available at these frequencies.
• 6 GHz/4 GHz bands are shared with terrestrial services,
creating interference problem.
• As equatorial orbit is filling with geostationary satellites, RF
interference is increasing from one satellite system to
another is increasing.
• 14/11 and 30/20 GHz systems for telecommunication and
broadcasting satellite services are slowly coming.
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4.5 Frequency bands in use for satellite communication are:
"L" BAND 1830–2700 MHz

"S" BAND 2500–2700 MHz Insat is Using
5925–6425 MHz UP
"C" BAND 3700–4200 MHz Insat is Using
DOWN
7900–8400 UP
"X" BAND
7250–7750 DOWN
14.000–14.500 Hz.
UP
10950–11200
"KU" BAND
GHz/DN.
11450–11700
GHz/DN.
27.5–30 GHz UP
"K" BAND
17.7–21.2 GHz DOWN
6725–7025 UP
Extended C BAND Insat is Using
4500–4800 DOWN
40–51 GHz UP
V BAND
40–41 GHz DOWN
59–64 GHz
V Band Inter-satellite
54–58 GHz
4.6 Time Delay
The total earth–satellite–earth path length may be as much as

74,000 km thus giving a one–way propagation delay of 250 ms. The effect
of this delay on telephone conversations, where a 500 ms gap can arise
between one person asking a question and hearing the other person
reply, has been widely investigated, and was found to be less of a
problem than had been anticipated. With geostationary satellites, two–
hop operation sometimes unavoidable and gives rise to a delay of over
one second.
TC31009D10 140
4.7 Geographical Advantage

A station which is located closer to the sub–satellite point, as
demonstrated in Fig. 9.5 will have an advantage in received signal level
with respect to one at the edge of the service area of the satellite. For a
global coverage satellite, this can be as much as 4.3 dB.
Fig. 9.5 Example of Geographical Advantage
4.8 Kinds and Systems of Communication Satellite
Kinds of Communication Satellites – depends on type of orbit and

frequency band used. During the early experimental stage of
communication satellites, a passive satellite was used without any
amplifiers and it only reflected radio waves sent from the earth station.
But, later on active satellite with amplifiers was developed and put into
practical use. Communication Satellite can be classified by the orbit used
and also by frequency band used. Before discussing satellite orbits in a
more generalized manner, however, it is necessary to be aware of the
natural laws that control the movement of satellites.
These are based on Kepler's laws and basically stated are:

• The orbit plane of any earth satellite must bisect the Earth
centrally.
• The Earth must be at the centre of any orbit.
The choice of orbit is restricted to three basic types, namely: polar,

equatorial and inclined as illustrated in Fig. 9.6. The actual shape of the
orbit is limited to circular and elliptical. Any combination of type and
shape is possible but observations are made only of the circular polar,
elliptically inclined and the circular equatorial.
TC31009D10 141
Fig. 9.6 Three Basic Orbits
Circular polar orbit

This is the only orbit that can provide full global coverage by one
satellite, but requires a number of orbits to do so. In a communications
sense where instantaneous transfer of information is required, full global
coverage could be achieved with a series of satellites, where each satellite
is separated in time and angle of its orbit. However, this produces
economic, technical and operational disadvantages and is thus not used
for telecommunications though it is favored for some navigation,
meteorological and land resource satellite system.
Elliptically inclined orbit

An orbit of this type has unique properties that have been
successfully used for some communications satellite system, notably the
Russian domestic system. For this system, the elliptical orbit has an
angle of inclination of 63 degrees and a 12–hour orbit period. By design,
the satellite is made to be visible for eight of its 12–hour orbit period to
minimize the handover problem while providing substantial coverage of
the temperate and polar regions. By using three satellites, suitably
phased, continuous coverage of particular temperate region can be
provided that would not be covered by other orbits.
The elliptically inclined orbit is used exclusively by the Russians
for their Orbital and Molniya systems, but since coverage is limited to
particular areas (higher latitudes), it is, therefore, not suitable for a
global network.
Circular Equatorial Orbit

Circular orbits in the equatorial plane permit fewer satellites and
ground stations to be used, and satellites with long orbital periods (at
high altitudes) have greater mutual visibility. A satellite in a circular
orbit at 35,800 km has a period of 24 hours and consequently appears
TC31009D10 142
stationary over a fixed point on the earth's surface. The satellite is visible
from one third of the earth's surface, up to the Arctic circle, and this
orbit is almost universally preferred for satellite communications system.
Stabilization of the satellite is necessary since the earth is not truly
spherical, and the moon, sun and the earth's tidal motion have
gravitational effects on the satellite, tending to make it drift from its
correct position. Inclination to the equatorial plane produces a sinusoidal
variation in longitude, seen from earth as motion around an ellipse once
every 24 hours, with peak deviation equal to the inclination angle.
Incorrect velocity results incorrect altitude, and a drift to the east or to
the west. When a non re–usable launcher is utilized, injection of the
satellite into geostationary orbit requires two rocket burns: the first to get
the vehicle into a parking orbit, and the second via an elliptical transfer
orbit to geostationary altitude. The spacecraft's own apogee motor then
increases its velocity to about 10,000 fps to maintain the geostationary
orbit. When launched from the Space Transportation System (Shuttle), a
booster rocket is attached to the satellite to boost it to the geostationary
orbit.
The satellite must then be correctly positioned and held in position
for its required lifetime (typically 7 to 10 years). This is done by using
hydrazine (liquid nitrogen plus ammonia) and cold gas jets. About 40 lbs.
of hydrazine are required for corrections to maintain geostationary
position within q 0.1x for five years, but since hydrazine is also used for
initial positioning, the quantity available depends on the accuracy of the
launch. To extend the life of the satellites, less frequent corrections may
be made allowing the satellite to drift.
F = Noise Figure of Receiver. The antenna noise is expressed in
degrees kelvin and is called noise temperature of antenna. It can be
converted to familiar units of power, watts by multiplying it with
Boltzmann's constant K = 1.38 x 10–29 joule/kelvin and the bandwidth.
Noise temperature of an antenna is of the order of 20–50oK.
Geostationary Satellite
This satellite revolves above the equator round the earth at a
height of 35,790 km. Its period of revolving round the earth is same as
that of the earth rotation on its own axis. Therefore, it looks as if it is
stationary. This system was contributed to the "WIRELESS WORLD" by
Mr. A.C.Clark, Dr. Rosen (an American) and others. It launched a
Syncom communication satellite in 1963. Syncom No. 1 failed to launch
in February, 1963. But, Syncon No. 2 finally succeeded in July 1963.
This satellite centered the equator and moved like a figure eight (8). This
was not a complete geostationary satellite, but it came into practical use
(24 hours) as synchronous satellite.
TC31009D10 143
This satellite is advantageous because:
• Its large antenna at an earth station is easy to track.

• Twenty–four (24) hours communication can be made with
even only one satellite.
• The satellite looks at the earth as if it was stationary, and it
radiates highly effective wave power.
• Visibility from one (1) satellite is very wide, and global
communication can be made using only three (3) satellites.
Its drawback, however, is its delay caused in long distance

transmission. But, the system is economical and accordingly, it is widely
used for both international and regional domestic communications
5. VSAT System Components and Specifications
The C–200 Series Micro Earth Station is a part of two way data
communication system. The network consists of:
• Master Earth Station

• Micro Earth Station
• A Geosynchronous Satellite
It operates at C–band Microwave frequency which travels in

straight line. To cover large areas, modern satellite communication
systems use geosynchronous communication satellite as a repeater. The
satellite is positioned 36,000 km above the earth equator. The satellite
can 'see' a very large portion of the earth's surface and hence the signal
transmitted from it reaches the Micro Earth Stations located at far off
places.
The technology used is spread spectrum multiple access (SSMA)
method by which the size of the antenna is considerably reduced. This
permits C–200 series to transmit and receive signals regardless of its
location as long as the antenna physically points at the satellite and is
not blocked by any obstructions. Fig. shows a typical data
communication network.
TC31009D10 144
Fig. 9.7 Simplified Data Communication Network
5.1 Multi Channel per Carrier – VSAT (MCPC–VSAT)
Department of Telecommunications has decided to provide reliable

telecommunication services to sub divisional H.Q./Tahsil/Tourist
Centre/Pilgrimage places as one of the major objective of its eighth five
year plan, viz. 1992–1997. Most of such places are planned to be
provided with C–DOT 126 RAX or 64 lines – MLT Local Telephone
Exchange depending upon the local Telecommunication need. The local
Exchange is to be interconnected to either district HQ or State capital
through a reliable long distance trunk media.
A good number of stations under the category of such sub
divisional HQ Tahsil, tourist centre, pilgrimage places etc. are located in
remote hilly and in accessible terrain which are neither feasible nor
techno–economically viable on terrestrial media. Such station have been
planned and engineered to be connected on Satellite media using
Multichannel per carrier Very Small Aperture Terminal (MCPC–VSAT)
TC31009D10 145
technology. About 80 earth stations may be required to be linked with

their respective state capital through satellite media during 8th Five Year
Plan.
5.2 General Features of MCPC – VSAT
The MCPC–VSAT shall operate in normal C–frequency band as

given below:
• UPLINK 5.925–6.425 GHz

• DOWNLINK 3.7–4.2 GHz
The MCPC–VSAT shall operate in a star configuration and shall

provide connectivity on a pre assigned mode between a hub station
located at main station (11 Mtr. dia. Antenna and G/T > 31.7 dB/degree
K) of INSAT Network. A block schematic giving the system concept is
given in Fig.
The MCPC–VSAT shall be compact and it can be installed either in
a container along with Local telephone exchange or in a separate
building. A block diagram of MCPC–VSAT is given in Fig.2.
The MCPC–VSAT shall consist of 7 channels each with voice
digitised at the rate of 16 Kb/s and 1 low bit rate data channel. With the
rate of ½ FEC coding transmission would be at the rate of 256 Kb/s.
The MCPC–VSAT shall work with the power supply of –48V with
10% variation DC supply, available for local telephone exchanges at the
site.
5.3 MCPC-VSAT Parameters
1. G/T = 15.0 dB/degree K

2. Transmit Frequency = 5.925-6.425 GHz
3. Receive Frequency = 3.700-4.200 GHz
4. Network Architecture = Star
5. Modulation = QPSK
6. Access = PAMA – Pre assigned
7. Digital Transmission = 7 clear voice channels and
one data channel. Voice
digitised at 16 Kb/s rate.
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6. HVNET – DOT VSAT Network

This is the first High Speed Satellite based VSAT network of
Department of Telecom., Govt. of India. It provides for high speed data
transfers and voice communication covering the entire country.
HVNET SYSTEM
INSAT
NEWDELHI
VSAT
FAX
YEUR GPSS Personal

THANE HUB Earth
J&K
STATION Stations VSAT
PC
I-NET X.25
INTERNET
GUJARAT
VSAT
FEEDER X.28
TELEX PSTN
RABMN X.25
Fig. 8.8 HVNET Architecture
The DoT VSAT network consists of a HUB Station located at Yeur

Earth Station of Dept. of Telecommunication near Thane (about 40 kms
from Mumbai) and number of VSATs/Personal Earth Stations (PES)
located throughout the country.
The VSAT communicate to the Hub through the INSAT Satellite. All
VSATs are connected in STAR topology and VSAT to VSAT
communication is through HUB at Mumbai. The VSAT which is required
to be installed at subscriber’s premises consists of three units, namely
an Outdoor Unit, an Indoor Unit and Inter Facility Link (IFL). Cable
interconnecting the two Units along with a 2.4 meter diameter Antenna
assembly and can be installed easily in any open space and requires a
floor area of about 4 mt x 4 mt. The IFL cable, which carries the telecom
signals and power supply, the IFL cable can be up to 100 meters long.
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Module - 01
Chapter – 10
Introduction to Electronic Exchanges
Session Objectives:

• Development of Electronic Exchanges

• Influence of Electronics in Exchange Design.
• Facilities provided by Electronic Exchanges.
• Constraints of Electronic Exchanges
• Working principles of Electronic Exchanges
• Software for SPC Exchanges
TC31009D10 148
1. Development of Electronic Exchanges
To overcome the limitations of manual switching; automatic

exchanges, having Electro-mechanical components, were developed.
Strowger exchange, the first automatic exchange having direct control
feature, appeared in 1892. Though it improved upon the performance of
a manual exchange it still had a number of disadvantages, viz., a large
number of mechanical parts, limited availability, inflexibility, bulky in
size etc. As a result of further research and development, Crossbar
exchanges, having an indirect control system, appeared in 1926. The
Crossbar exchange improved upon many shortcomings of the Strowger
system. However, much more improvement was expected and the
revolutionary change in field of electronics provided it. A large number of
moving parts in Register, Marker, Translator, etc., were replaced en-
block by a single computer. This made the exchange smaller in size,
volume and weight, faster and reliable, highly flexible, noise-free, easily
manageable with no preventive maintenance etc.
The first electronic exchange employing Space-Division switching

(Analog switching) was commissioned in 1965. This exchange used one
physical path for one call and, hence, full availability could still not be
achieved. Further research resulted in development of Time-Division
switching (Digital Switching) which enabled sharing a single path by
several calls, thus providing full availability. The first digital exchange
was commissioned in 1970 in France.
Table 2: Development of Electronic Exchanges (Major exchanges)

MODEL Capacity (in thousands) Traffic Call
Digital Lines Trunks Erlangs Attempts
Exchanges per second
E-10B 30 4 2,400 25
Mentaconta 10-60 - 10,000 28-60
System X 100 60 25,000 800000
AXE -10 64 - 26,000 800000
FETEX-150L 290 60 24,000 1800000
OCB-283 200 60 25,000 800000
EWSD 250 60 25,200 1000000
NEAX-61E 100 60 27,000 1000000
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TABLE 5- Advantages of Electronic Exchange over

Electromechanical Exchanges
Electromechanical Exchanges Electronic Exchanges
• Category, Analysis, Routing, • Translation, speech path Sub’s
translation, etc; done by relays. Facilities, etc., managed by MAP and
• Any changes in facilities require other DATA.
addition of hardware and/or • Changes can be carried out by
large amount of wiring change. simple commands. A few changes can
Flexibility limited. be made by Subs himself. Hence,
• Testing is done manually highly flexible.
externally and is time • Testing carried out periodically
consuming. No logic analysis automatically and analysis printed
carried out. out.
• Partial full-availability, hence • Full availability, hence no blocking.
blocking. A large number of different types of
• Limited facilities to the services possible very easily.
subscribers. • Very fast. Dialing speed up to 11
• Slow in speed. Dialing speed is digits /sec possible. Switching is
max. 11 digits and switching achieved in a few microseconds.
speed is in l milliseconds. • Much lesser volume required floor
• Switch room occupies large space of switch room reduced to about
volume. one-sixth.
• Lot of switching noise. • Almost noiseless.
• Long installation and testing • Short installation and testing period.
time. • Remedial maintenance is very easy
• Large maintenance effort and due to plug-in type circuit boards.
preventive maintenance Preventive maintenance not required.
necessary.
2. Influence of Electronics in Exchange Design.

When electronic devices were introduced in the switching systems,
a new concept of switching evolved as a consequence of their extremely
high operating speed compared to their former counter-parts, i.e., the
Electro-mechanical systems, Relays, the logic elements in the
electromechanical systems, have operate and release times which are
roughly equal to the duration of telephone signals to maintain required
accuracy. However, to achieve the requisite simultaneous call processing
capacity, it became essential for such system to have number of such
electrical control units (Called registers in a Cross-bar Exchange), in
parallel, each handling one call at a time. In other words, it was
necessary to have an individual control system to process each call.
TC31009D10 150
Electronic logic components on the other hand, can operate a

thousand or ten thousand times during a telephone signal. This led to a
concept of using a single electronic control device to simultaneously
process a number of calls on time-sharing basis. Though such
centralisation of control is definitely more economical it has the
disadvantage of making the switching system more vulnerable to total
system failure. This can; however be overcome by having a stand by
control device.
Another major consequence of using electronics in control
subsystems of a telephone exchange was to make it technically and
economically feasible to realize powerful processing units employing
complex sequence of instructions. Part of the control equipment capacity
could then be employed for functions other than call processing, viz.,
exchange operation and maintenance. It resulted in greatly improved
system reliability without excessively increasing system cost. This
development led to a form of centralized control in which the same
processor handled all the functions, i.e., call processing, operation and
maintenance functions of the entire exchange.
In the earlier versions of electronic control equipment, the control
system was of a very large size, fixed cost unit. It lacked modularity. It
was economically competitive for very large capacity exchanges. Initially,
small capacity processors were costlier due to high cost per bit of
memory and logic gates. Therefore, for small exchanges, processor cost
per line was too high. However, with the progressive development of the
small size low cost processor using microprocessor, it became possible to
employ electronic controls for all capacities. In addition control
equipment could also be made modular aiding the future expansion. The
impact of electronics on exchanges is not static and it is still changing as
a function of advances in electronic technology.
3. Facilities provided by Electronic Exchanges.
Facilities offered by electronic exchanges can be categorised in

three parts.
a) Facilities to the Subscribers.
b) Facilities to the Administration.
c) Facilities to the Maintenance Personnel.
a) Facilities to the Subscribers.
MFC Push-button Dialing.

All subscribers in an electronic exchange can use push-button
telephones, which use Dual Tone Multi- frequency, for sending the dialed
digits. Sending of eleven digits per second is possible, thus increasing
the dialing speed.
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Priority Subscriber Lines

Priority Subscribers lines may be provided in electronic exchanges.
These subscribers are attended to, according to their priority level, by the
central processor, even during heavy congestion or emergency.
Toll (Outgoing Call) Restriction

All subscribers can avail of the facility of toll restriction or blocking
of subscriber line for specific types of outgoing traffic, viz., long distance
STD calls. This can be easily achieved by keying-in certain service codes.
Service Interception
Incoming calls to a subscriber can be automatically forwarded
during his absence, to a customer service position or a recorded
announcement. The customer service position answers the calls and
forwards any message meant for the subscriber.
Abbreviated Dialing
Most subscribers very often call only limited group of telephone
numbers. By dialing only prefix digit followed by two selection digits,
subscribers can call up to 100 predetermined subscribers connected to
any automatic exchange. This shortens the process of dialing all the
digits.
Call Forwarding
The subscriber having the call forwarding facility can keep his
telephone in the transfer condition in case he wishes his incoming calls
to be transferred to another telephone number during his absence.
Do Not Disturb
This service enables the subscriber to free himself from attending
to his incoming calls. In such a case, the incoming calls are routed to an
operator position or a talking machine. This position or machine informs
the caller that called subscriber is temporarily inaccessible.
Conference Calls
Subscribers can set up connections to more than one subscriber
and conduct telephone conferences under the provision of this facility.
Camp On Busy
Incoming call to a busy subscriber can be “Camped on” until the
called subscriber gets free. This avoids wastage of time in redialing a
busy telephone number.
TC31009D10 152
Call Waiting
The ‘Call Waiting’ service notifies the already busy subscriber of a
third party calling him. He is fed with a special tone during his
conversation. It is purely his choice either to ignore the third party or to
interrupt the existing connection and have a conversation with the third
party while holding the first party on the line.
Call Repetition
Instead of camp on busy a call can automatically be repeated. The
calling party can replace his hand set after receiving the busy tone. A
Periodic check is carried out on the called party’s status. When idle
status is ascertained, the connection is set up and ringing current fed to
both the parties.
Third party Inquiry

This system permits consultation and the transfer of call to other
subscribers. Consultation can be initiated by means of a special signal
from the subscriber telephone and by dialing the directory number of the
desired subscriber without disconnecting the previous connection.
Priority of calls to Emergency Positions

Emergency calls such as ambulance, fire, etc., are processed in
priority to other calls.
Subscriber charge Indicator

By placing a charge indicator at the subscriber’s premises the
charges of each call made can be ascertained by him.
Call Charge printout or immediate Billing

The subscriber can request automatic post call charge notification
in the printout form for individual calls or for all calls. The information
containing called number, date and time, and the charges can be had on
a Tele-type-write.
Malicious Call Identification

Malicious Call Identification is done immediately and the
information is obtained in the printout from either automatically or by
dialing an identification code.
Interception or Announcement.
In the following conditions, an announcement is automatically
conveyed to calling subscribers.
• Change of a particular number of transferred subscriber.
• Dialing of an unallocated cods.
• Dialing of an unobtainable number.
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• Route congested or out of order.

• Subscriber’s line temporarily out of order.
• Suspension of service due to non-payment.
Connection Without Dialing.

This allows the subscribers to have a specific connection set up,
after lifting the handset, without dialing. If the subscriber wishes to dial
another number, then he has to start dialing within a specified time
period, say 10 seconds, after lifting the handset.
Automatic Wake Up.

Automatic wake up service or morning alarm is possible, without
any human intervention.
Hot Line or Private Wire.

Hot line service enables the subscriber to talk to a specific
subscriber by only lifting the handset. This service cannot be used along
with normal dialing facility. The switching starts as soon as the receiver
is lifted.
Denied Incoming Call

A Subscriber may desire that no incoming call should come on a
particular line. He can ask for such a facility so that he can use the line
for making only outgoing calls.
Instrument Locking
A few subscribers may like to have their telephone sets locked up
against any misuse. Dialing of a secret code will extend such a facility to
them.
Free of charge Calls

Calls free of charge are possible on certain special services such as
booking of complaints, booking of telegrams, etc.
Collect call
If so desired, the incoming subscriber is billed for all the calls
made to him, instead of the calling subscriber.
b) Facilities to the Administration
Reduced Switch Room Accommodation

Reduction in switch room accommodation to about 1/6th to 1/4th
as compared to Crossbar system is possible.
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Faster installation and Easy Extension

The reduced volume of equipment, plug-in assemblies for
interconnecting cables, printed cards and automatic testing of exchange
equipment result in faster installation (about six months for a 10,000
line exchange) Due to modular structure, the expansion is also easier
and quicker
Economic Consideration
The switching speed being much faster as compared to Crossbar
system, the use of principle of full availability of trunk circuits and other
equipment makes the system economically superior to electromechanical
systems.
Automatic test of Subscriber line

Routine testing of subscriber lines for Insulation, capacitance,
foreign potential, etc., are automatically carried out during night. The
results of the testing can be obtained in the printout form, the next day.
c) Maintenance Facilities
Fault Processing
Automatic fault processing facility is available for checking all
hardware components and complete internal working of the exchange.
Changeover from a faulty sub-system to stand-by sub-system is
automatically affected without any human intervention. Only information
is given out so that the maintenance staff is able to attend to the faulty
sub-system.
Diagnostics
Once a fault is reported by the system, ‘on demand’ programs are
available which help the maintenance staff to localise the fault, who can
replace the defective printed card and restore the faulty sub-system. The
faulty card is attended at a centralised maintenance centre specifically
equipped for this purpose.
Statistical programs
Statistical programs are available to gather information about the
traffic conditions and trunks occupancy rate to assess and plan the
solutions in cases of anticipated problems. This facility helps the
maintenance and administration personnel to maintain a specified level
of grade of service.
Blocking
In case of congestion or breakdown of a specific route, facility of
blocking such routes is available in modes, such as
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(i) Blocking of a specified percentage of calls in such a route either

automatically or manually.
(ii) Blocking a specific category of subscribers.
Overloading Security
Overloading of central processor in an electronic exchange can lead
to disastrous results. To prevent this, central processor occupancy is
measured automatically periodically, when it exceeds a specified
percentage, audio-visual alarms are activated, in addition to printing out
the message. Maintenance personnel have the following options.
(i) Block some of the facilities temporarily, or
(ii) Reduce the load by blocking some of the congested routes.
4. Constraints of Electronic Exchanges
Though there are a number of definite advantages of Electronic

exchanges, over the electromechanical exchanges, there are certain
constraints, which should be considered, at the planning stage for
deciding between the two systems.
Traffic Handling Capacity

Apparently, the traffic handling capacity of an exchange is limited
by the number of subscriber lines and trunks connected to the switching
network, and the number of simultaneous paths available through the
switching network. However, in electronic exchanges, the prime
limitation is the number of simultaneous calls, which can be handled by
the control equipment, as it has to execute a number of instructions
depending on the type of the call. Therefore the extent of loading of the
exchange will be guided solely by the amount of processor loading.
Moreover, the facilities to the subscribers will also have to be limited
accordingly.
Power Supply
The power supply should be highly stable for trouble free operation
as the components are sensitive to variations beyond +10%. It is almost
essential to have a stand-by power supply arrangement.
Total Protection from Dust

All possible precautions should be observed for ensuring dust-free
environment.
Temperature and Humidity Control

Due to the presence of quiescent current in the components and
because of their compactness. Heat generated per unit volume is highest
in electronic exchanges. Moreover, as the component characteristics drift
substantially with the temperature and humidity, the air-conditioning
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load is higher. Obviously, the air-conditioning system should be highly

reliable and preferably there should be a stand-by arrangement. The
installation is also carried out in air-conditioned environment.
Static Electricity and Electromagnetic interference.

Due to the presence of static electricity on the body of persons
handling the equipment, the stored data may get vitiated. Handling of
PCB’s therefore, should be done with utmost care and should be
minimised care should also be taken to protect the cards from exposure
to stray electromagnetic fields.
PCB Repair
The repair of PCB’s is extremely complicated and sophisticated
equipments are required for diagnosing the faults. This results in having
costly inventory and a costly repair centre. With the frequent
improvement and changes in the cards, proper documentation of cards
becomes essential.
5. Working principles of Electronic Exchanges
a) Introduction
The prime purpose of an exchange is to provide a temporary path
for simultaneous and bi-directional transmission of speech between
(i) Subscriber lines connected to same exchange (local switching)
(ii) Subscriber lines and trunks to other exchange (outgoing trunk
call)
(iii) Subscriber lines and trunks from other exchanges (incoming
trunk calls) and
(iv) Pairs of trunks towards different exchanges (transit switching)
These are also called the switching functions of an exchange and

are implemented through the equipment called the switching network.
An exchange, which can setup just the first three types of connections. is
called a Subscriber or Local Exchange. If an exchange can setup only the
fourth type of connections, it is called a Transit or Tandem Exchange.
The other distinguished functions of an exchange are
i) Exchange of information with the external environment

(Subscriber lines or other exchanges) i.e. signaling.
ii) Processing the signaling information and controlling the
operation of signaling network, i.e. control, and
iii) Charging and billing
All these functions can be provided more efficiently using computer

controlled electronic exchange, than by the conventional
electromechanical exchanges.
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b) Stored Programme Controlled (SPC) Exchange
In electromechanical switching, the various functions of the

exchange are achieved by the operation and release of relays and switch
(rotary or crossbar) contacts, under the direction of a Control Sub-
System. These contacts are hard - wired in a predetermined way. The
exchange dependent data, such as, subscriber’s class of service,
translation and routing, combination signaling characteristics, are
achieved by hard-ware and logic, by a of relay sets, grouping of same
type of lines, strapping on Main or Intermediate Distribution Frame or
translation fields, etc. When the data is to be modified, for introduction
of a new service, or change in services already available to a subscriber,
the hardware change ranging from inconvenient to near impossible, are
involved.
In an SPC exchange, a processor similar to a general-purpose
computer is used to control the functions of the exchange. All the control
functions represented by a series of various instructions are stored in the
memory. Therefore the processor memories hold all exchange-dependent
data such as subscriber date, translation tables, and routing and
charging information and call records. For each call processing step. e.g.
for taking a decision according to class of service, the stored data is
referred to, Hence, this concept of switching. The memories are
modifiable and the control program can always be rewritten if the
behavior or the use of system is to be modified. This imparts and
enormous flexibility in overall working of the exchange.
Digital computers have the capability of handling many tens of
thousands of instructions every second, hence, in addition to controlling
the switching functions the same processor can handle other functions
also. The immediate effect of holding both the control programme and the
exchange data, in easily alterable memories, is that the administration
can become much more responsive to subscriber requirements, both in
terms of introducing new services and modifying general services, or in
responding to the demands of individual subscriber. For example, to
restore service on payment of an overdue bill or to permit change from a
dial instrument to a multi frequency sender, simply the appropriate
entries in the subscriber data-file are to be amended. This can be done
by typing- in simple instructions from a teletypewriter or visual display
unit. The ability of the administration to respond rapidly and effectively
to subscriber requirements is likely to become increasingly important in
the future.
The modifications and changes in services which were previously
impossible are achieved very simply in SPC exchange, by modifying the
stored data suitably. In some cased, subscribers can also be given the
facility to modify their own data entries for supplementary services, such
as on-demand call transfer, short code, (abbreviated) dialing, etc.
TC31009D10 158
The use of a central processor also makes possible the connection

of local and remote terminals to carry out man-machine dialogue with
each exchange. Thus, the maintenance and administrative operations of
all the SPC exchanges in a network can be performed from a single
centralised place. The processor sends the information on the
performance of the network, such as, traffic flow, billing information,
faults, to the centre, which carries out remedial measures with the help
of commands. Similarly, other modifications in services can also be
carried out from the remote centre. This allows a better control on the
overall performance of the network.
As the processor is capable of performing operations at a very high
speed, it has got sufficient time to run routine test programmes to detect
faults, automatically. Hence, there is no need to carry out time
consuming manual routine tests.
In an SPC exchange, a single processor can replace all control
equipment. The processor must, therefore, be quite powerful; typically, it
must process hundreds of calls per second, in addition to performing
other administrative and maintenance tasks. However, totally centralised
control has drawbacks. The software for such a central processor will be
voluminous, complex, and difficult to develop reliably. Moreover, it is not
a good arrangement from the point of view of system security, as the
entire system will collapse with the failure of the processor. These
difficulties can be overcome by decentralising the control. Some routine
functions, such as scanning, signal distributing, marking, which are
independent of call processing, can be delegated to auxiliary or
peripheral processors. These peripheral units, each with specialised
function, are often themselves controlled by small stored programmes
processors, thus reducing the size and complexity at central control level.
Since, they have to handle only one function, their programmes are less
voluminous and far less subjected to change than those at central.
Therefore, the associated programme memory need not be modifiable
(generally, semiconductors ROM's are used).
c) Block Schematic of SPC Exchange
Despite the many difference between the electronic switching

systems, and all over the world there is a general similarity between most
of the systems in terms of their functional subdivisions. In it’s simplest
form, an SPC exchange consists of five main sub-systems, as shown in
Fig 10.1.
• Terminal equipment provides on individual basis for each
subscriber line and for inter-exchange trunk.
• Switching network may be space- division or time-division, uni-
directional or bi-directional.
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• Switching processor, consisting mainly of processors and

memories.
• Switching peripherals (Scanner, Distributor and Marker), are
Interface Circuits between control system terminal equipment and
switching network.
• Signaling interfaces depending on type of signaling used, and
• Data Processing Peripherals (Tele - typewriters, Printers, etc.) for
man-machine dialogue for operation and maintenance of the
exchange.
Terminal Equipment.
In this equipment, line, trunk, and service circuits are terminated,
for detection, signaling, speech transmission, and supervision of calls.
The Line Circuits carry out the traditional functions of supervising and
providing battery feed to each subscriber line. The Trunk Circuits are
used on outgoing, incoming and transit calls for battery feed and
supervision. Service Circuits perform specific functions, like;
transmission and reception of Decadic dial pulses or MF signals, which
may be economically handled by a specialised common pool of circuits.
In contrast to electromechanical circuits, the Trunk and Service circuits
in SPC exchanges are considerably simpler because functions, like
counting, pulsing, timing charging, etc, are delegated to stored
programme.
Switching Network.
In an electronic exchange, the switching network is one of the
largest sub-systems in terms of size of the equipment. Its main functions
are
• Switching, i.e., setting up temporary connection between two or
more exchange terminations
• Transmission of speech and signals between these terminations,
with reliable accuracy.
There are two types of electronic switching system. viz. Space
division and Time Division.
d) Space Division switching System.

In a Space Division Switching system, a continuous physical path
is set up between input and output terminations. This path is separate
for each connection and is held for the entire duration of the call. Path
for different connections is independent of each other. Once a continuous
path has been established, signals are interchanged between the two
terminations. Such a switching network can employ either metallic or
electronic cross-points. Presently, usage of metallic cross-points, viz.,
reed relay, mini-cross bar derivative switches etc, is favored. They have
TC31009D10 160
the advantage of compatibility with the existing line and trunk signaling
conditions in the network.
e) Time Division Switching System.

In Time Division Switching, a number of connections (calls) share
the same path on time division sharing basis. The path is not separate
for each connection, rather, is shared sequentially for a fraction of a time
by different calls. This process is repeated periodically at a suitable high
rate. The repetition rate is 8 Khz, i.e. once every 125 microseconds for
transmitting speech on telephone network, without any appreciable
distortion. These samples are time multiplexed with staggered samples of
other speech channels, to enable sharing of one path by many calls.
Pulse Amplitude Modulation (PAM) switching initially accomplished the
Time Division Switching. However, it still could not overcome the
performance limitations of signal distortion noise, cross-talk etc. With
the advent of Pulse Code Modulation (PCM), the PAM signals were
converted into a digital format overcoming the limitations of analog and
PAM signals. PCM signals are suitable for both transmission and
switching. The PCM switching is popularly called Digital Switching.
Fig. 10.1 Functional sub-divisions of an SPC exchange
f) Compatibility with Existing Network

In this area, the application of electronic techniques has encountered the
greatest difficulty. To appreciate the reasons, let us consider the basic
requirements of a conventional switching network.
TC31009D10 161
• High OFF resistance and low ON resistance.

• Sufficient power handling capacity for transmitting ringing current,
battery feed etc., on subscriber lines.
• Good frequency response (300-3400 KHz)
• Bi-directional path (preferable)
• D.C. signaling path to work with existing junction equipment
preferable)
• Economy
• Easy to control.
• Low power consumption, and
• Immunity to extraneous noise, voltage surges.
The present day electronic devices cannot meet all these

requirements adequately. Electronic devices can easily meet most
requirements. These considerations show that substitutions of the
analog mode of electromechanical switching network by fully electronic
equipment are not, straight way practical. The main virtue of the existing
electromechanical devices is their immunity to extraneous noise voltage
surge, etc., which are frequently experienced in a telephone network.
Moreover, metal contact switches offer little restriction on the voltages
and currents to be carried. In the existing network and subscriber
handsets, typically, 80 volt peak to peak ringing current is required to be
transmitted on the line. This is difficult, if not impractical, for electronic
switches to handle. Therefore, to avail of the advantages of the electronic
exchanges, either of the two following alternatives may be adopted.
• Deploy a new range of peripherals/ equipments, suited to the
characteristics of the electronic switching devices, on one hand,
and requirements of telephone network on the other hand. i.e.
employ Time Division Switching systems, or
• Continue to use metal contact switches, while other sub-systems
may be changed to electronic. i.e. semi-electronic type of
exchanges rather than fully electronic exchanges, to employ Space
Division Switching Systems.
g) Switching Processor
The switching processor is a special purpose real time computer,
designed and optimised for dedicated applications of processing
telephone calls. It has to perform certain real time functions (which have
to be performed at the time of occurrence and cannot be deferred), such
as, reception of dialed digits, and sending of digits in case of transit
exchange. The block schematic of a switching processor, consisting of
central control programme store is shown in Fig. 10.2.
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To Switching
Central control Processor
Programme Translatio Data Store

Store n Store
Fig. 10.2 Switching Processor
Central Control (CC) is a high-speed data processing unit, which

controls the operation of the switching network. In Programme store, sets
of instructions called programmes are stored. The programmes are
interpreted and executed by the central control. Data Store provides for
the temporary storage of transient data, required in processing telephone
calls, such as digits dialed by the subscriber, busy / idle states of lines
and trunks etc. Translation Store contains information regarding lines.
e.g. category of calling and called line routing code, charging
information, etc. Data Stores is temporary memory, whereas Translation
and Programme Stores are of semi-permanent type. The information in
the Semi-permanent memories does not change during the processing of
the call, but the information in Data Store changes continuously with
origination and termination of each call.
h) Switching Peripheral Equipment
The time interval, in which the processor operates, is in the order

of microseconds, while the components in the telephone switching
section operate in milliseconds (if the switching network is of the analog
type). The equipment, known as the switching peripheral, is the interface
between these two equipments working at different speeds. The interface
equipment acts as speed buffer, as well as, enables conversion of digital
logic signals from the processor to the appropriate electrical signals to
operate relays and cross-points, etc. Scanners, Signal distributors and
Marker fall under this category of devices.
Scanner
Its purpose is to detect and inform CC of all significant events /
signals on subscriber lines and trunks connected to the exchange. These
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signals may either be continuous or discrete. The equipments at which

the events / signals must be detected are equally diverse.
• Terminal equipment for subscriber lines and inter-exchange
trunks and
• Common equipment such as DTMF (Dual - Tone Multi Frequency)
or MFC digit receivers and inter-exchange signaling senders /
receivers connected to the lines and trunks.
In view of this wide diversity in the types of lines trunks and
signaling, the scanning rate, i.e. the frequency at which scan points are
read, depends upon the maximum rate at which events / signals may
occur. For example, on a subscriber line, with Decadic pulse signaling
with 1:2 make -break ratio, the necessary precision, required for pulse
detection, is of the order of ten milliseconds, while other continues
signals (clear, off hook, etc.) on the same line are usually several
hundred milliseconds long and the same high precision is not required.
To detect new calls, while complying with the dial tone connection
specifications, each line must be scanned about every 300 milliseconds.
It means that in a 40,000 lines exchange (normal size electronic
exchange) 5000 orders are to be issued every 300 milliseconds, assuming
that eight lines are scanned simultaneously.
Marker
Marker performs physical setup and release of paths through the
switching network, under the control of CC. A path is physically operated
only when it has been reserved in the central control memory. Similarly,
paths are physically released before being cleared in memory, to keep the
memory information updated vis-a-vis switching network, depending
upon whether the switching is Time division or Space division, marker
either writes information in the control memory of time and space stages.
(Time Division Switching), or physical operates the cross - points (Space
Division Switching)
Distributor
It is a buffer between high - speed - low - power CC and relatively
slow-speed-high-power signaling terminal circuits. A signal distributor
operates or releases electrically latching relays in trunks and service
circuits, under the direction of central control.
Bus System
Various switching peripherals are connected to the central
processor by means of a common system. A bus is a group of wires on
which data and commands pulses are transmitted between the various
sub- units of a switching processor or between switching processor and
switching peripherals. The device to be activated is addressed by sending
TC31009D10 164
its address on the address bus. The common bus system avoids the
costly mesh type of interconnection among various devices.
Line Interface Circuits

To enable an electronic exchange to function with the existing
outdoor telephone network, certain interfaces are required between the
network and the electronic exchange.
Analogue Subscriber Line Interface

The functions of a Subscriber Line Interface, for each two-wire line,
are often known by the acronym: BORSHT
B : Battery feed O : Overload protection
R : Ringing S : Supervision of loop status
H : Hybrid T : Connection to test equipment
All these functions cannot be performed directly by the electronic circuits
and, therefore, suitable interfaces are required.
Transmission Interface
Transmission interface between analogue trunks and digital trunks
(individual or multiplexed) such as, A/D and D/A converters, are known
as CODEC, These may be provided on a per-line and per-trunk basis or
on the basis of one per 30 speech channels.
Signaling Interfaces
A typical telephone network may have various exchange systems

(Manual, Strowger, Cross bar, electronic) each having different signaling
schemes. In such an environment, an exchange must accommodate
several different signaling codes.
Signaling
Initially, all signaling between automatic exchanges was Decadic
i.e. telephone numbers were transmitted as trains of 1to 10 pulses, each
train representing one digit. To increase the speed at which the calls
could be set up, and to improve the reliability of signaling, compelled
sequence multi frequency signaling system was then introduced. In this
system, each signal is transmitted as a combination of 2 out of a group of
say 5 or 6 frequencies. In both Decadic and multi frequency methods,
the signals for each call are sent over a channel directly associated with
the inter-exchange speech transmission circuit used for that call. This is
termed as channel associated signaling. Recently, a different technique
has been developed, known as common channel signaling. In this
technique, all the signaling information for a number of calls is sent over
a signaling link independent of the inter-exchange speech circuits.
Higher transmission rate can be utilised to enable exchange of much
larger amount of information. This results in faster call setup,
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introduction of new services, e.g.. abbreviated dialing, and more retrials

ultimately accomplishing higher call completion rate, Moreover, it can
provided an efficient means of collecting information and transmitting
orders for network management and traffic engineering.
Data Processing Peripherals

Following basic categories of Data Processing Peripherals are used
in operation and maintenance of exchange.
• Man - machine dialogue terminals, like Tele-typewriter (TTY) and
Visual Display Units (VDU), are used to enter operator commands
and to give out low-volume date concerning the operation of the
switching system. These terminals may be local i.e. within a few
tense of meters of the exchange, or remotely located. These
peripherals have been adopted in the switching Systems for their
ease and flexibility of operation.
• Special purpose peripheral equipment is, sometimes employed for
carrying out repeated functions, such as, subscriber line testing,
where speed is more important than flexibility.
• High-speed large capacity data storage peripherals (Magnetic Tape
Drives, magnetic Disc Unit) are used for loading software in the
processor memory.
• Maintenance peripherals, such as, Alarm Annunciates and Special
Consoles, are used primarily to indicate that automatic
maintenance procedure have failed and manual attention is
necessary.
6. Software for SPC Exchanges
a) Introduction
The SPC software is classified, as per functions less than two
heads, viz., operational software and support software. The Operational
Software is the set of all programs necessary for operations of the
exchange. It may further be divided into two broad categories viz.,
System Programs and Application Programs. The System Programs are
more or less equivalent to the operating system of a conventional
computer. System software consists of programs that facilitate the
operation and use of the processor by Application programs. The
application Programs handle Call Processing, Administration and
Maintenance of the exchange.
The support Software comprising of assembler, loader and
simulation programs, etc., is located at a centralized place, known as
Software Centre, to serve a group of SPC exchanges.
TC31009D10 166
b) System Programs
System Programs work as interface between the Exchange

Hardware and the Application Programs. They manage and co-ordinate
the activities and features of the hardware and Application programs.
This is achieved by ensuring various functions.
Managing l/O Operations

The system programs for managing all input/output operations are
some-times called Device Handlers. They are a collection of routines,
which connect the system and the user programs with peripheral
devices. There is one device handler for each type of peripheral device in
the hardware configuration for transferring data between the device and
the main memory.
• Assigning memory and peripheral devices to active processes.
• Protecting the system from hardware and software faults and
errors.
• Managing man-machine communication with Exchange personnel.
• Providing access to the data describing the state of the system.
Application Programs
The application programs can be divided into three main categories.
Call Processing Programs

Call processing programs are meant for setting up, supervising,
releasing and charging calls in conformity with the telephone service
specifications.
Administration Programs
Administration programs perform traffic supervision and
measurement, line and trunk testing, modifying semi-permanent data
concerning subscriber lines and trunks, defining the exchange hardware
configuration, modifying data in translation and routing tables, e.g.,
changing the way a call is routed between two exchanges, or modifying a
subscriber’s service entitlements, or storing a full length number
corresponding to an abbreviated dialing code.
Maintenance programs
Maintenance programs carry out fault detection and fault
localisation by testing the exchange hardware, including the processor
itself.
TC31009D10 167
Operational Software Breakup
The basic purpose of an exchange is to set up and release

telephone calls. The most important function of the Operational Software
is; therefore, call processing, which includes detection of a call
origination, signal processing, path-finding through the switching
network, address and digit translation charging, supervision etc., and
finally releasing the call. Although call processing takes up the major
part of the processor time, the corresponding programs constitute only a
small fraction (around 1/5 th) of total instruction of the Operational
Software (Fig. 10.3.) Typical operational software of an exchange
comprises of 3 lakhs to 5 lakhs instructions. It may be seen that
administration and maintenance programs represent about two-thirds of
the software. Importance of administration and maintenance functions
and hence the size of the corresponding software is growing day by day
as the exchanges are becoming more sophisticated.
Fig. 10.3 Software for SPC Exchanges
Essential characteristics
SPC software must have a ‘real time’ operating system. It must be
capable of processing large number of calls, simultaneously, and should
have special features to ensure telephone services without interruption
even when maintenance of on-line capacity extension processes is in
progress.
Real time constraints

The software of an exchange must meet the specified traffic
handling Capacity and quality of service specifications. The traffic
handling capacities of control processors of the exchange are expressed
in terms of number of calls to be processed per second or per hour.
c) Quality of service
Quality of service is quantified mainly by two parameters.
TC31009D10 168
• Percentage of the calls failed, with respect to successful calls at a

specified traffic load, due to internal problems of the exchange,
viz., error in processing or congestion in the system, etc.,
• Percentage of calls, which wait for dial tone more than a specified
time, should be less than a prescribed value.
It means that an exchange must, not only process a certain

number of calls per second, but also ensure their processing within a
prescribed time limit determined by the exchange specifications. A
system, whose maximum response time is predefined in this way, is
called a Real time System. Hence, the software of an electronic exchange
must have this real time feature.
An electronic exchange has several real time processing stages,
each differing in terms of time constraints. For example, the most strict
real time constraints concerns signal and signal processing. The specified
accuracy for some signals is of the order of 10 ms, i.e., it should be able
to process a signaling event lasting for just 10ms.
The real time constraints for the remaining call processing
programs, e.g., connections of dial tone, etc., are less strict. The system
must respond to the subscriber action, say within a second.
Real time constraints for administration and maintenance
functions are least stringent. These may be between a few seconds
(answering an operator command from TTY) or a few minutes (starting
routine testing of some equipment). The corresponding programs are
therefore, run with low priority.
d) Multiprogramming
The control processors in an electronic exchange operate in the
multiprogramming mode, i.e., many tasks (mostly relating to call
processing) are active simultaneously. For example, in a 30,000 lines
exchanges there may be 3000 calls in the speech phase while another
500 calls being released or established at any instant of time. It means
that 3500 tasks are being carried out simultaneously. Moreover, the
system must monitor all the calls in the memory such that when any
change occurs in the external telephone environment, relating to the call,
its status is change accordingly.
In addition, a few administration and maintenance tasks may be
active, e.g., operator may issue a command through TTY for testing line,
or modifying routing table, etc. On other occasion, automatic testing and
traffic measuring may be simultaneously active. These requirements
warrant multiprogramming mode of operation.
It is not possible to set up a call in one continues processing
sequence because call establishing involves several elementary
processing actions, each lasting a few tens or hundreds of milliseconds,
separated by idle waiting time period. These waiting periods can last long
TC31009D10 169
up to a few seconds. If it were to process one call at a time, control

processor will also remain idle for the corresponding time period.
Instead, each time a call processing is waiting for an external event, the
corresponding call processing is de-activated and other tasks
corresponding to other calls are run. When the awaited event for the
preceding calls is detected, it is re-activated and processing of the call
resumes. Thus, many tasks are performed simultaneously. It may also be
noted that control processor is a sequential device, i.e., capable of
executing only one instruction at a time.
Continuity of Service
Telephone service must be available to subscribers round the clock

without any interruption. The cumulative duration of all the
interruptions of service must be less than a few hours, during the life of
the exchange. It is typically 2 hours, in 40 years. The system should
ensure continuity of service, even on occurrence of a hardware or
software fault.
The philosophy of fault handling is completely different for an
electronic exchange from a conventional scientific or data processing
system. In a data processing system a false result is far more serious
than a complete breakdown of the system, whereas in an electronic
exchange system some proportion of incorrect call processing (say, one in
500) is acceptable instead of total system failure. This influences the
design of SPC operational software in many ways, particularly the
programs relating to faults and on line capacity extensions.
Software design should enable maintenance of the processors, telephone
peripherals and other exchange equipment on-line without interrupting
service. To achieve this, a bit of degradation in the grade of service and
total traffic handling capacity may be accepted, it should also be possible
to extend the capacity of an operating exchange, i.e., add new telephone
equipment, processor, memory etc., without interrupting the call
processing. Similarly, correcting the software errors, adding new
programs or new version of existing programs and other software
maintenance and extension must not disturb telephone service.
e) Program storages
In most of the SPC systems, the total size (volume wise) of all the
programs added together is much bigger than the size of main memory.
It is therefore, not possible to make the entire programs main memory
resident. However, a program can be executed only when it is resident in
the main memory. In order to make best use of the limited resources,
i.e., the capacity of the main memory, only vital parts of the system
programs and application programs are kept permanently in main
memory. All the other programs that are not active may be stored outside
the main memory in auxiliary storage, also called mass memory.
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Hence, in any switching system, many programs, including the

large ones, which are not being used very frequently, are stored in low
cost mass memory (auxiliary storage) like magnetic tape or disk.
Generally most of administration and maintenance programs are stored
on magnetic disks and programs for system hardware and/or software
extension are stored on magnetic tape.
On-line and Off-line programs
On line Program
On line programs are run when the exchange is on-line, ie. while
performing call processing, testing and other administration functions.
The on-line programs can be either system programs or Application
Programs.
Off-line programs
The off-line programs are used during initial commissioning of the
SPC exchange, i.e., when it is off-line and has not started functioning. It
is not possible to use the off-line programs during the operation of an
SPC exchange because the main memory is normally made to
accommodate on-line programmes only. However, if the standby
processor, individual operational system, is put off-line programs can be
run on it. Examples of such programs are text editor, linker, etc.
f) Redundancy Methods
A telephone exchange must guarantee service to the subscribers
under all circumstances and as such it should never be at the mercy of
any fault in any equipment. Ideally not more than 2 hr. of total
interruption of service over a period of 40 years is admissible. Although
the reliability of electronic components is very high, but certainly not
absolute, Redundancy is always necessary at the processor level, to meet
the exchange availability specifications. In other words, duplication
must be provided, at least for all common control devices so that the
exchange must continue to function even if one of these units are taken
out of service, for any reason.
Regarding the central processor unit, redundancy, in general, can be
provided by any of the following techniques
• Load Sharing Method
• Hot Standby Method
• N+1 Redundancy Method
Load Sharing
Load sharing is the frequently used mode of redundancy methods.
Fig. 10.4 illustrates this concept.
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Fig.10.4 Load Sharing Redundancy Fig 10.5 Hot Standby Redundancy
In a load-sharing environment, each processor has access to all

inputs and all outputs, as in synchronous Replica Duplication. Calls are
randomly distributed for processing to either of the two processors When
a processor has accepted a call, it handles it right through to completion
Two processors share the resources dynamically, but operate
independently. They have to co-ordinate their operation for access to
subscriber lines, trunks and switching network. There is an inter-
processor link for exchanging the information for mutual co-ordination
and for hardware or software exclusion mechanism to prevent both the
processors accessing the same peripheral device.
The processors have separate memories for storing temporary data
for calls, because each processor handles its call, independently of the
other-Although, rest of the memory can be shared for storing
programmes and permanent data, it is preferable to have separate
storages for each processor. This provides greater protection against
software errors.
Moreover, there is an arrangement to ensure that each command,
sent from OBM, is executed by both the processors. This avoids issuance
of commands separately to each processor.
Each processor has access to all peripheral devices. During normal
operation, both processors are on-line and share half of the calls each,
on statistical basis. The exchange operator can however, send commands
to distribute the traffic unevenly between the two processors.
When a fault occurs, the fault processor is taken out of service
without interrupting the telephone services, but the calls currently being
established by the faulty processor are lost. The other processor while
establishing its own calls, also handles all new calls and takes all calls in
the ringing or speech phase, even though originally set up by the faulty
processor.
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In this mode, error detection is left to self detection or detection by

the other processor. This is one of the disadvantages of the load sharing.
The greatest advantage is that total processing capacity in terms of busy
hour call attempts is higher as compared to Synchronous Replica
Duplication operation. This arrangement has also got a high tolerance to
software faults because it is very rare that both the processors are
executing the same programme at the same time. If an error appears in
one of the software programmes that particular processor is taken out of
service and system is safeguarded against total traffic interruption.
Hot Standby System

This is the simplest mode of providing redundancy in a single
processor operation. Active processor handles the total traffic and other
processor is on standby. It is totally decoupled from active processor
both in hardware and in software. It is place on-line only if the active
processor fails. Unfortunately, updating of standby processor before it is
switched on-line is impossible. As the working processor has failed, it is
incapable of carrying out any operation. A fault can therefore cause an
interruption in service and requires total re-initialization, with release of
all calls already established. To avoid this problem, the processor must
have a common memory, as shown in the Fig. 10.5.
In such a system, current state of the active processor is copied
into, the back up storage, every 5 seconds. After switch over between the
two processors the new working processor loads the last stored state of
the environment into the main memory. This stored state differs only
slightly from the real state of calls in the ringing and speech phases and
most of these calls are saved.
However, because of common memory provision for both the
processors, many maintenance operations are unduly complicated,
restricting the popularity of such an arrangement.
g) N+1 Redundancy Methods

Some multiprocessor systems, as shown in Fig. 10.6 have the N+1
system configuration, in which one processor is added as a redundant
unit. This unit can, under normal conditions, take part of the traffic
handling so that the total N+1 configuration has a larger capacity than
the engineered value, based on N processors. However, in case of failure
of any processor, the redundant processor takes up the entire load of the
failed processor.
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Fig 10.6 N+1 Redundancy Method
In modern SPC exchanges, the control system is a combination of

various systems depending upon the type and complexity of the function
and requirements of the network. However, the load sharing redundancy
method is generally employed, to avoid a complete collapse of the
services, with the advantage of higher traffic handling capacity in normal
conditions.
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Module - 01
Chapter – 11
Switching Concepts
Session Objectives:

• Telecom Network Model

• Circuit Switching
• Packet Switching
• Telephone Switching Hierarchy
• Local Exchange
• Tandem Exchange
• Call flow
• Interconnections
TC31009D10 175
1. Telecom Network Model

The telecommunications network can be described by a layered
model consisting of the following layers:
• Switching and Services layer: This consists of all the switching

nodes, local as well as transit. It also consists of any other
equipment and like computers and software used to provide
services to the customers.
• Transport Layer: This represents the links among the nodes and
provides the medium and systems to carry the information from
one node to the other. These are junctions and trunks. Junctions
are links between the local switches and local and national
switches. Trunks are the links between the national switches, the
national and international switches and between the international
switches i.e. the long distance network. The long distance or trunk
network is composed of multiplexed channels of varying capacity
connecting the National Switches and the International Switches.
The trend has been to move from point-to-point links using
Plesiochronous Digital Hierarchy (PDH) towards advanced
networks with built in controllability based on Synchronous Digital
Hierarchy (SDH) technique. The two most important trends in the
long distance networks are digitization and introduction of fiber-
optic technologies. These developments have reduced the
transmission cost per channel-kilometer and improved the quality.
• Access Layer: This represents the access network that links the
customers to the local switch.
2. Switching concepts
In telecommunication, switching of voice or data calls are
established in two modes:
• Circuit Switching
• Packet Switching
3. Circuit Switching (CS)

In telecommunications, a CS network is one that establishes a
fixed circuit (or channel) between exchanges and terminals before the
users may communicate, as if the exchanges were physically connected
with an electrical circuit. The delay is constant during the connection.
Other callers cannot use each circuit until the circuit is released
and a new connection is set up. Even if no actual communication is
taking place in a dedicated circuit then, that channel still remains
unavailable to other users. Channels that are available for new calls to
be set up are said to be idle. The copper wire used for the connection
TC31009D10 176
could not be used to carry other calls at the same time, even if the
subscribers were in fact not talking and the line was silent.
Virtual CS is a packet switching technology that may emulate CS,
in the sense that the connection is established before any packets are
transferred, and that packets are delivered in order.
For call setup and control, it is possible to use a separate
dedicated signaling channel. The method of establishing the connection
and monitoring its progress and termination through the network may
also utilize a separate control channel to communicate the call setup and
control information and use TDM to transport the actual circuit data.
With CS, a route is reserved from source to destination. The entire
message is sent in order so that it does not have to be reassembled at the
destination. CS can be relatively inefficient because capacity is wasted on
connections, which are set up but are not in continuous use. On the
other hand, the connection is immediately available and capacity is
guaranteed until the call is disconnected.
Fig. 11.1 Circuit Switched Network
In circuit Switched Network- Path or pipe between two ends in

communication is opened and stays open for the duration of the call
4. Packet Switching (PS)

CS contrasts with PS, which splits traffic data (for instance, digital
representation of sound, or computer data) into packets that are routed
over a shared network.
PS is the process of segmenting a message/data to be transmitted
into several smaller packets. Each packet is labeled with its destination
and the number of the packet, precluding the need for a dedicated path
to help the packet find its way to its destination. Each is dispatched and
many may go via different routes. At the destination, the original
message is reassembled in the correct order, based on the packet
number. PS networks do not require a circuit to be established and allow
many pairs of nodes to communicate almost simultaneously over the
same channel.
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Telephone switches
A switch, in telephony jargon refers to a telephone switch or
exchange located at the local telephone company's central office, directly
serving subscribers. The switch services include basic dial tone, calling
features, and additional digital and data services to subscribers using
the local loop. These switches were slower to convert from circuit
switching technologies to time division multiplexing.
5. Telephone Switching Hierarchy

In order to organize Direct Dialing, switches in the Public Switched
Telephone Network (PSTN) are arranged into a hierarchy containing
different levels.
• International gateways - handing off and receiving traffic
from outside the country’s networks.
• TAX exchanges - exchanges connecting the various areas of
a city or towns in country.
• Tandem exchanges - which interconnect whole regions of the
network in a multi exchange environment or exchanges
connecting major population centres within particular region
of the network.
• Local exchanges – directly serves subscribers and end-users.
6. Local Exchange
The fundamental difference between a local exchange and the
other exchanges is that a local exchange provides telephone service to
customers, and as such is concerned with "subscriber type" activities:
generation of dial-tone and handling of network services such as advice
of duration and charge etc. Specifically, a local switch provides dial tone,
local switching and access to the rest of the network, Fig 11.2. Typically
a local switch will cover an area of a city, an individual town, or several
villages and could serve from several hundred to 100,000 subscribers.
Some form of remote switch often performs the function of a local
switch in rural areas or Remote Digital Terminal installed at the original
switch site to handle local switching or concentration, respectively. The
local switching infrastructure is then physically located in a larger
population center. Urban areas with extensive underground plant tend to
keep the classic local office architecture.
The telephone line from a subscriber runs underground or
overhead on poles to the local BT building. While this building is often
known as "the exchange", in actual fact it might well not be. All the lines
in an area terminate on a Main Distribution Frame (MDF). A second set
of wires run from the MDF to the switching unit in the building. For the
majority of lines this unit is a Remote Line Unit (RLU), while for the
remainder it is the Digital Local Exchange (DLE).
TC31009D10 178
Fig. 11.2 Direct Local Exchange connections
7. Tandem (TDM) Exchange

TDM switches do not provide dial tone - they simply route calls
between other switches, so they are more concerned with efficient
switching and signaling, Fig 11.3. Where a call is being made to a
subscriber on a different DLE from the caller, it needs to travel over some
kind of link between DLEs. In theory all DLEs could be connected to one
another, but this would be wasteful of links. Alternatively calls could be
relayed from DLE to DLE, but this would take up a lot of the processing
power of the DLEs. Instead there is a second network of switches, known
as the TDM Network. This network is made up of four kinds of switch:
In the following diagram, the lettered circles represent DLEs (and
their associated RLUs), the numbered squares tandem switches, and the
triangle a Direct Junction Switching Unit (DJSU). The tick lines show the
tandems all connected to one another, the thin lines are DLE-TDM links.
Fig. 11.3 Tandem Connections
TC31009D10 179
8. Call flow
There are four kinds of call:
• Own exchange calls, where the call remains within a single DLE.
• Linked exchange calls that go over an inter-DLE link, such as from
B to C (these two categories are together known as Local exchange
calls).
• Single tandem calls that go DLE-TDM-DLE, such as from A to B.
• Double TDM calls such as from A to E.
In fact, a call is normally described in terms of the most optimal
route, so B to C is linked exchange even if the call doesn't go over the
link, A to B is single TDM even if TDM1 has failed and the call is going
via TDM2 and TDM3, and A to E is double tandem even if congestion
means the call goes A-1-2-3-4-E. Each switch (both DLE and TDM) will
have a routing table giving up to four routes, in order of preference, for
each possible call.
9. Interconnections
There may be ranges of other licensed operators who also carry
telephone calls. In order that calls can be transferred from one operator
to the other or vice versa, it is obviously necessary for the two networks
to be connected. This is done at Points of Interconnection (POIs). POI can
be at any tandem switch or, in some circumstances, at a DLE. An
operator is not required to connect to every TDM switch. Instead, will
route calls through the network to a convenient POI.
In the above example, there are three POIs, two at tandems and
one at a DLE. Each DLE and tandem will have entries for the other
operators in its routing tables, just as with calls to other DLEs.
TC31009D10 180
Module - 01
Chapter – 12
Digital Switching
Session Objectives:

• Digital Switching
• Time and Space Switching
• Digital Space Switching Principle
• Practical Space Switch
• Digital Time Switching Principle
• Two Dimensional Switching
• TST Network
TC31009D10 181
1. Digital Switching
A Digital switching system, in general, is one in which signals are

switched in digital form. These signals may represent speech or data. The
digital signals of several speech samples are time multiplexed on a
common media before being switched through the system.
To connect any two subscribers, it is necessary to interconnect the
time-slots of the two speech samples, which may be on same or different
PCM highways. The digitalised speech samples are switched in two
modes, viz., Time Switching and Space Switching. This Time Division
Multiplex Digital Switching System is popularly known as Digital
Switching System.
2. Time and Space Switching
Generally, a digital switching system several time division

multiplexed (PCM) samples. These PCM samples are conveyed on PCM
highways (the common path over which many channels can pass with
separation achieved by time division.). Switching of calls in this
environment requires placing digital samples from one time-slot of a PCM
multiplex in the same or different time-slot of another PAM multiplex.
For example, PCM samples appearing in TS6 of I/C PCM HWY1 are
transferred to TS18 of O/G PCM HWY 2, via the digital switch, as
shown in Fig 12.1.
FIG 12.1 Digital Switch
The interconnection of time-slots, i.e., switching of digital signals

can be achieved using two different modes of operation. These
modes are:
• Space Switching
• Time switching
Usually, a combination of both the modes is used.
TC31009D10 182
In the space-switching mode, corresponding time-slots of I/C and

O/G PCM highways are interconnected. A sample, in a given time-slot,
TSi of an I/C HWY, say HWY1, is switched to same time-slot, TSi of an
O/G HWY, SAY HWY2. Obviously there is no delay in switching of the
sample from one highway to another highway since the sample transfer
takes place in the same time-slot of the PCM frame.
Time Switching, on the other hand, involves the interconnection of
different time-slots on the incoming and outgoing highways by re-
assigning the channel sequence. For example, a time-slot TSx of an I/C
Highway can be connected to a different time-slot. TSy, of the outgoing
highway. In other words, a time switch is, basically, a time-slot changer.
3 Digital Space Switching Principle
The Digital Space Switch consists of several input highways, X1,

X2,...Xn and several output highways, Y1, Y2,.............Ym, inter
connected by a cross point matrix of n rows and m columns. The
individual cross point consists of electronic AND gates. The operation of
an appropriate cross point connects any channel, a, of I/C PCM highway
to the same channel, a, of O/G PCM highway, during each appropriate
time-slot which occurs once per frame as shown in Fig 12.2. During
other time-slots, the same cross point may be used to connect other
channels. This cross point matrix works as a normal space divided
matrix with full availability between incoming and outgoing highways
during each time-slot.
Each cross point column, associated with one O/G highway, is
assigned a column of control memory. The control memory has as many
words as there is time-slot per frame in the PCM signal. In practice, this
number could range from 32 to 1024. Each cross point in the column is
assigned a binary address, so that only one cross point per column is
closed during each time-slot. The binary addresses are stored in the
control memory, in the order of time-slots. The word size of the control
memory is x bits, so that 2x = n, where n is the number of cross points in
each column.
A new word is read from the control memory during each time-slot,
in a cyclic order. Each word is read during its corresponding time-slot,
i.e., Word 0 (corresponding to TSO), followed by word 1 (corresponding to
TS1) and so on. The word contents are contained on the vertical address
lines for the duration of the time-slot, thus the cross point corresponding
to the address, is operated during a particular time-slot. This cross point
operates every time the particular time-slot appears at the inlet in
successive frames, normally, a call may last for around a million frames.
As the next time-slot follows, the control memory is also advanced
by one step, so that during each new time-slot new corresponding words
are read from the various control memory columns. This results in
operation of a completely different set of cross points being activated in
TC31009D10 183
different columns. Depending upon the number of time-slots in one

frame, this time division action increases the utilisation of cross point 32
to 1024 times compared with that of conventional space-divided switch
matrix.
Example
Consider the transfer of a sample arriving in TS7 of I/C HWY X1 to

O/G HWY Y3. Since this is a space switch, there will be no reordering of
time i.e., the sample will be transferred without any time delay, via the
appropriate cross point. In other words, the objective is to connect TS7 of
HWY X1 and TS7 of HWY Y3.
The central control (CC) selects the control memory column
corresponding output highway Y3. In this column, the memory location
corresponding to the TS7 is chosen. The address of the cross point is
written in this location, i.e., 1, in binary, is written in location 7. This
cross point remains operated for the duration of the time-slot TS7, in
each successive frame till the call lasts.
For disconnection of call, the CC erases the contents of the control
memory locations, corresponding to the concerned time-slots. The AND
gates, therefore, are disabled and transfer of samples is halted.
4. Practical Space Switch
In a practical switch, the digital bits are transmitted in parallel

rather than serially, through the switching matrix.
In a serial 32 time-slots PCM multiplex, 2048 Kb/s are carried on

a single wire sequentially, i.e., all the bits of the various time-slots follow
one another. This single wire stream of bits, when fed to Serial to Parallel
Converter is converted into 8-wire parallel output. For example, all 8 bits
corresponding to TS3 serial input are available simultaneously on eight
output wires (one bit on each output wire), during just one bit period, as
shown in fig. This parallel output on the eight wires is fed to the
switching matrix. It can be seen that during one full time-slot period,
only one bit is carried on the each output line, whereas 8 bits are carried
on the input line during this period. Therefore, bit rate on individual
output wires, is reduced to 1/8th of input bit rate=2048/8=256Kb/s
TC31009D10 184
Fig. 12.2 Space Switch

Due to reduced bit rate in parallel mode, the cross point is
required to be operated only for 1/8th of the time required for serial
working. Eight times more channels, i.e., 32 x 8 = 256 channels, in the
same frame, can thus, share it.
However, since the eight bits of one TS are carried on eight wires,
each cross point has eight switches to interconnect eight input wires to
eight output wires. Each cross point (all the eight switches) will remain
operated now for the duration of one bit only, i.e., only for 488 ns (1/8th
of the TS period of 3.9 µs)
Fig 12.3 Serial parallel converter
TC31009D10 185
For example, to connect 40 PCM I/C highways, a matrix of 40x 40

= 1600 cross points each having a single switch, is required in serial
mode working. Whereas in parallel mode working, a matrix of (40/8 x
40/8) = 25 cross point is sufficient. As eight switches are required at
each cross point 25 x 8 = 200 switches only are required. Thus, there is a
reduction of the matrix by 1/8th in parallel mode working, hence
reduction in size and cost of the switching matrix.
5 Digital Time Switching Principle

A Digital Time Switch consists of two memories, viz., a speech or
buffer memory to store the samples till destination time-slots arrive, and
a control or connection or address memory to control the writing and
reading of the samples in the buffer memory and directing them on to the
appropriate time-slots.
Speech memory has as many storage locations as the number of
time-slots in input PCM, e.g., 32 locations for 32 channel PCM system.
The Control Memory controls the writing/reading operation in the
speech memory. It has same number of memory locations as for speech
memory, i.e., 32 locations for 32-channel system. Each location contains
the address of one of the speech memory locations where the channel
sample is either written or read during a time-slot. These addresses are
written in the control memory of the CC of the exchange, depending upon
the connection objective.
A Time-Slot Counter, which usually is a synchronous binary
counter, is used to count the time-slots from 0 to 31, as they occur. At
the end of each frame, it gets reset and the counting starts again. It is
used to control the timing for writing/reading of the samples in the
speech memory.
Example
Consider the objective that TS4 of incoming PCM is to be
connected to TS6 of outgoing PCM. In other words, the sample arriving in
TS4 on the I/C PCM has to be delayed by 6 - 4 = 2 time-slots, till the
destination time-slot, viz., TS6 appears in the O/G PCM. The required
delay is given to the samples by storing it in the speech memory. The
I/C PCM samples are written cyclically i.e. sequentially time-slot wise, in
the speech memory locations. Thus, the sample in TS4 will be written in
location 4, as shown in Fig. 12.4.
The Control Memory controls the reading of the sample. The
Control Memory location corresponding to output time-slot TS6 is 6. In
this location, the CC writes the input time-slot number, viz., 4, in binary.
These contents give the read address for the speech memory, i.e., it
indicates the speech memory locations from which the sample is to be
read out, during read cycle.
TC31009D10 186
When the time-slot TS6 arrives, the control memory location 6 is

read. Its content addresses the location 4 of the speech memory in the
read mode and sample is read on to the O/G PCM.
In every frame, whenever time-slot 4 comes a new sample will be
written in location 4. This will be read when TS6 occurs. This process is
repeated till the call lasts.
For disconnection of the call, the CC erases the contents of the
control memory location to halt further transfer of samples.
Time switch can operate in two modes, viz.

• Output associated control
• Input associated control
Output associated control

In this mode of working, 2 samples of I/C PCM are written
cyclically in the speech memory locations in the order of time-slots of I/C
PCM, i.e., TS1 is written in location 1, TS2 is written in location 2, and so
on, as discussed in the example.
The contents of speech memory are read on output PCM in the
order specified by control memory. Each location of control memory is
rigidly associated with the corresponding time-slot of the O/G PCM and
contains the address of the TS of incoming PCM to be connected to. The
control memory is always read cyclically, in synchronism with the
occurrence of the time-slot. The entire process of writing and reading is
repeated in every frame, till the call is disconnected.
Fig 12.4 Output Associated Control Switch
TC31009D10 187
It may be noticed that the writing in the speech memory is

sequential and independent of the control memory, while reading is
controlled by the control memory, i.e., there is a sequential writing but
controlled reading.
Input associated control

Here, the samples of I/C PCM are written in a controlled way, i.e.,
in the order specified by control memory, and read sequentially.
Each location of control memory is rigidly associated with the
corresponding TS of I/C PCM and contains the address of TS of O/G
PCM to be connected to.
The previous example with the same connection objective of
connecting TS4 of I/C PCM to TS6 of O/G PCM may be considered for its
restoration. The location 4 of the control memory is associated with
incoming PCM TS4. Hence, it should contain the address of the location
where the contents of TS4 of I/C PCM are to be written in speech
memory. A CC writes the number of the destination TS, viz., 6 in this
case, in location 4 of the control memory. The contents of TS4 are
therefore, written in location of speech memory, as shown in Fig 12.5.
The contents of speech memory are read in the O/G PCM in a
sequential way, i.e., location 1 is read during TS1, location 2 is read
during TS2, and so on. In this case, the contents of location 6 will appear
in the output PCM at TS6. Thus the input PCM TS4 is switched to output
PCM TS6. In this switch, there is sequential reading but controlled
writing.
Fig 12.5 Input Associated Controlled Time Switch
TC31009D10 188
6 Two Dimensional Switching
Though the electronic cross points are not so expensive, the

cost of accessing and selecting them from external pins in a Space
Switch, become prohibitive as the switch size increases. Similarly, the
memory location requirements rapidly go up as a Time Switch is
expanded, making it uneconomical. Hence, it becomes necessary to
employ a number of stages, using small switches as building blocks to
build a large network. This would result in necessity of changing both
the time-slot and highway in such a network. Hence, the network,
usually, employs both types of switches viz., space switch and time
switch; and. therefore, is known as two Dimensional network. These
networks can have various combinations of the two types of switches and
are denoted as TS, STS, TSST, etc.
Though to ensure full availability, it may be desirable to use only T
stages. However, the networks having the architecture of TT, TTT, TTTT,
etc., are uneconomical, considering the acceptability of tolerable limits of
blocking, in a practical network. Similarly, a two-stage two-dimensional
network, TS or ST, is basically suitable for very low capacity networks
only. The most commonly used architecture has three stages, viz., STS or
TST. However, in certain cases, their derivatives, viz., TSST, TSSST, etc.,
may also be used.
An STS network has relatively simpler control requirements and
hence, is still being favored for low capacity networks, viz., PBX
exchanges. As the blocking depends mainly on the outer stages, which
are space stages, it becomes unsuitable for high capacity systems.
A TST network has lesser blocking constraints, as the outer stages
are time stages, which are essentially non-blocking, and the space stage
is relatively smaller. It is, therefore, most cost-effective for networks
handling high traffic, However, for still higher traffic handling capacity
networks, e.g., tandem exchanges, it may be desirable to use TSST or
TSSST architecture.
The choice of a particular architecture is dependent on other
factors also, viz., implementation complexity, modularity, testability,
expandability, etc. As a large number of factors favor TST structure, it is
most widely used.
7. TST Network
As the name suggests, in a TST network, there are two time
stages separated by a space stage. The former carry out the function of
time-slot changing, whereas the latter performs highway jumping. Let us
consider a network having n input and n output PCM highways. Each of
the input and output time stages will have n time switches and the space
stage will consist of an n x n cross point matrix. The speech memory as
well as the control memory of each time switch and each column of a
control memory of the space switch will have m locations, corresponding
TC31009D10 189
to m time-slots in each PCM. Thus, it is possible to connect any TS in

I/C PCM to any TS in O/G PCM.
In the case of a local exchange, the network will be of folded type,
i.e., the O/G PCM highways, via a suitable hybrid. Whereas, for a transit
exchange, the network will be non-folded, having complete isolation of
I/C and O/G PCM highways. However, a practical local exchange will
have a combination of both types of networks.
For the sake of explanation, let us assume that there are only four
I/C and O/G PCM highways in the network. Hence, there will be only
four time switches in each of the T-stages and the space switch will
consist of 4x4 matrixes. Let us consider an objective of connecting two
subscribers through this switching network of local exchange, assuming
that the CC assigns TS4 on HWY0 to the calling party and TS6 on HWY3
to the called party
The speech samples of the calling party have to be carried from
TS4 of I/C HWY 0 and to TS6 of O/G HWY3 and those of the called party
from TS6 of I/C HWY 3 to TS4 of O/G HWY 0, with the help of the
network. The cc establishes the path, through the network in three steps.
To introduce greater flexibility, it uses an intermediate time-slot, TSx,
which is also known as internal time-slot.
The three switching steps for transfer of speech sample of the
calling party to the called party are as under:
Step 1 Input Time Stage (IT) TS4 HWY0 to TSx HWY0

Step 2 Space stage (S) Tsx HWY0 to Tsx HWY3
Step 3 Output Time Stage (OT) Tsx HWY3 to TS6 HWY3
As the message can be conveyed only in one direction through this

path, another independent path, to carry the massage in the other
direction is also established by the CC, to complete the connection.
Assuming the internal time-slots to be TS10 and TS11, the connection
may be established as shown in Fig 12. 6.
FIG 12.6 T S T Switch
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Let us now consider the detailed switching procedure making some

more assumptions for the sake of simplicity. Though practical time
switches can handle 256 time-slots in parallel mode, let us assume serial
working and that there are only 32 time-slots in each PCM. Accordingly,
the speech and control memories in time switches and control memory
columns in space switch will contain 32 locations each.
To establish the connection, the CC searches for free internal time-

slots. Let us assume that the first available time-slots are TS10 and
TS11, as before. To reduce the complexity of control, the first time stage
is designed as output-controlled switch, whereas the second time stage is
input-controlled.
FIG 12.7 T S T Switch Structure
For transfer of speech samples from the calling party to the called
party of previous example, CC orders writing of various addresses in
TC31009D10 191
location 10 of control memories of IT-10, OT-3 and column 3 of CM-S of

corresponding to O/G highway, HWY3. Thus, 4 corresponding to I/C TS4
is written in CM-IT-0, 6 corresponding to O/G TS6 is written in CM-OT-3
and 0 corresponding to I/C HWY 0 is written in column 3 of CM-S, as
shown in Fig. 12.7.
As the first time switch is output-controlled, the writing is done
sequentially. Hence, a sample, arriving in TS4 of I/C HWY 0, is stored in
location 4 of SM-IT-0. It is readout on internal HWY 0 during TS10 as per
the control address sent by CM-IT-0. In the space switch, during this
internal TS10, the cross point 0 in column 3 is enabled, as per the
control address sent by column 3 of CM-S, thus, transferring the sample
to HWY3. The second time stage is input controlled and hence, the
sample, arriving in TS10, is stored in location 6 of SM-OT-3, as per the
address sent by the CM-OT-3. This sample is finally, readout during TS6
of the next frame, thus, achieving the connection objective.
Similarly, the speech samples in the other direction, i.e., from the
called party to the calling party, are transferred using internal TS11. As
soon as the call is over, the CC erases the contents in memory locations
10 and 11 of all the concerned switches, to stop further transfer of
message. These locations and time-slots are, then, available to handle
next call.
TC31009D10 192
Module - 01
Chapter – 13
Signaling in Telecom Systems
Session Objectives:

• Signaling in Telecommunication
• Types of signaling information
• Subscriber Line signaling
• Inter-exchange Signaling
• In-Band and Out-of-Band Signals
• E & M and R2 Signaling
• Channel Associated Signaling
• Common Channel Signaling System No 7
• Signaling Network Architecture
• Signaling Components
• SS7 Link Types
• Basic signaling Procedure
• Advantage of CCS
TC31009D10 193
1. Signaling in Telecommunication
Introduction
A telecommunication network establishes and realizes temporary
connections, in accordance with the instructions and information
received from subscriber lines and inter exchange trunks, in form of
various signals. Therefore, it is necessary to interchange information
between an exchange and it external environment i.e. between subscriber
lines and exchange, and between different exchanges. Though these
signals may differ widely in their implementation they are collectively
known as telephone signals.
A signaling system uses a language, which enables two switching
equipments to converse for the purpose of setting up calls. Like any
other language it possesses a vocabulary of varying size and varying
precision, ie. a list of signals, which may also vary in size and syntax in
the form of a complex set of rules governing the assembly of these
signals.
2. Types of signaling information

The signaling information can be categorized under four main
heads.
Call request and Release information
Call request information i.e. calling subscriber off hook or seizure
signal or an incoming trunk, indicates a new call. On its receipt, the
exchange connects appropriate equipment for receiving address
information (called number).
Release information i.e. on hook or release signal on a trunk
indicates that the call is over. The exchange releases all the equipment
held out for the call, and clears up any other information used for setting
up at including the call.
Selection (Address) information.

When the exchange is ready to receive the address information. It
sends back a request, which is known as proceed to send (PTS) signal in
trunk signaling and dial tone in subscriber signaling.
Address information essentially comprises of full or part of the called
subscribers number and possibly additional service data.
End of selection information

This information indicates the status of the called line, or the
reason for non-completion of the call attempt, essentially indicating
called line free or busy.
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Supervisory information
It specifies the on/off hook condition of a called subscriber after
the connection has been setup
Called subscriber off hook called subscriber has answered and
charging may commence.
Called subscriber on hook

Called subscriber has hung up to terminate the call, and the call is
disconnected after a time delay if the calling subscriber does not hang
up. The on/off-hook conditions of the calling subscriber are covered by
call request and release information.
Call connection
The interchange of signaling information can be illustrated with
the help of a typical call connection sequence.
• A request for originating a call is initiated when the calling
subscriber lifts the handset.
• The exchange sends dial tone to the calling subscriber to
indicate to him to start dialing.
• The called number is transmitted to the exchange, when the
calling subscriber dials the number.
• If the number is free, the exchange sends ringing current to him.
• Feedback is provided to the calling subscriber by the exchange
by sending.
• Ring-back tone, if the called subscriber is free
• Busy tone if the called subscriber is busy, or
• Recorded message, if provision exists, for non-completion of call
due to some other constraint.
• The called subscriber indicates acceptance of the incoming call
by lifting the handset
• The exchange recognizing the acceptance terminates the ringing
current and the ring-back tone, and establishes a connection
between the calling and called subscribers.
• The connection is released when either subscriber replaces the
handset. When the called subscriber is in a different exchange,
the following inter-exchange trunk signal functions are also
involved, before the call can be set up.
• The originating exchange seizes an idle inter exchange trunk,
connected to a digit register at the terminating exchange.
• The originating exchange sends the digit. The steps 4 to 8 are
then performed to set up the call.
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3. Subscriber Line signaling
Calling Subscriber Line Signaling

In automatic exchanges the power is fed over the subscriber’s loop
by the centralized battery at the exchange. Normally, it is 48 V. The
power is fed irrespective of the state of the subscriber, viz., idle, busy or
talking.
Call report
When the subscriber is idle, the line impedance is high. The line
impedance falls, as soon as, the subscriber lifts the handset, resulting in
increase of line current. This is detected as a new call signal and the
exchange after connecting an appropriate equipment to receive the
address information sends back dial-tone signal to the subscriber.
Address signal
After the receipt of the dial tone signal, the subscriber proceeds to
send the address digits. The digits may be transmitted either by decade
dialing or by multifrequency pushbutton dialing.
Decadic Dialing
The address digits may be transmitted as a sequence of
interruption of the DC loop by a rotary dial or a Decadic push-button
keypad. The number of interruption (breaks) indicates the digit, exept0,
for which there are 10 interruptions. The rate of such interruptions is 10
per second and the make/break ration is 1:2. There has to be an inter-
digital pause of a few hundred milliseconds to enable the exchange to
distinguish between consecutive digits. This method is, therefore,
relatively slow and signals cannot be transmitted during the speech
phase.
Multifrequency Push-button Dialing

This method overcomes the constraints of the Decadic dialing. It
uses two sets of four voice frequencies. Pressing a button (key) generates
a signal comprising of two frequencies, one from each group. Hence, it is
also called Dual-Tone Multi-frequency (DTMF) dialing. The signal is
transmitted as long as the key is kept pressed. This provides 16 different
combinations. As there are only 10 digits, at present the highest
frequency, viz., 1633 Hz is not used and only 7 frequencies are used, as
shown in Fig.13.1.
By this method, the dialing time is reduced and almost 10 digits
can be transmitted per second. As frequencies used lie in the speech
band, information may be transmitted during the speech phase also, and
hence, DTMF telephones can be used as access terminals to a variety of
systems, such as computers with voice output. The tones have been so
TC31009D10 196
selected as to minimize harmonic interference and probability of

simulation by human voice.
Figure 13.1 Tone-Dialing Frequency Groups.
End of selection signal

The address receiver is disconnected after the receipt of complete
address. After the connection is established or if the attempt has failed
the exchange sends any one of the following signals.
• Ring-back tone to the calling subscriber and ringing current to
the called subscriber, if the called line is free.
• Busy-tone to the calling subscriber, if the called line is busy or
otherwise inaccessible.
• Recorded announcement to the calling subscriber, if the
provision exists, to indicate reasons for call failure, other
than called line busy.
Ring back, tone and ringing current are always transmitted from
the called subscriber local exchange and busy tone and recorded
announcements, if any, by the equipment as close to the calling
subscriber as possible to avoid unnecessary busying of equipment and
trunks.
Answer Back Signal

As soon as the called subscriber lifts the handset, after ringing, a
battery reversal signal is transmitted on the line of the calling subscriber.
This may be used to operate special equipment attached to the calling
TC31009D10 197
subscriber, e.g., short-circuiting the transmitter of a CCB, till a proper

coin is inserted in the coin-slot.
Release signal
When the calling subscriber releases i.e., goes on hook, the line
impedance goes high. The exchange recognizing this signal, releases all
equipment involved in the call. This signal is normally of more than 500
milliseconds duration.
Permanent Line (PG) Signal

Permanent line or permanent glow (PG) signal is sent to the calling
subscriber if he fails to release the call even after the called subscriber
has gone on-hook and the call is released after a time delay. The PG
signal may also be sent, in case the subscriber takes too long to dial. It is
normally busy tone.
Called subscriber line signals.

• Ring Signal
• Answer Signal
• Release Signal
4. Inter-exchange Signaling
Inter-exchange signaling can be transmitted over each individual
inter exchange trunk. The signals may be transmitted using the same
frequency band as for speech signals (inband signaling), or using the
frequencies outside this band (out-of-band signaling). The signaling may
be
• Pulsed: The signal is transmitted in pulses. Change from idle
condition to one of active states for a particular duration
characterizes the signal, e.g., address information
• Continuous: The signal consists of transition from one condition
to another; a steady state condition does not characterize any
signal.
• Compelled: It is similar to the pulsed mode but the transmission
is not of fixed duration but condones till acknowledgement of the
receiving unit is received back at the sending unit. It is a highly
reliable mode of signal transmission of complex signals.
Line signals
The simplest cheapest, and most reliable system of signaling on
trunks, was DC signaling, also known as metallic loop signaling, exactly
the same as used between the subscriber and exchange, i.e. Circuit
seizure/release corresponding to off/on-hook signal of the subscriber.
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5. In-Band and Out-of-Band Signals

Exchanges separated by long distance cannot use any form of DC
line signaling. Suitable interfaces have to be interposed between them,
for conversion of the signals into certain frequencies, to enable them to
be carried over long distance. A signal frequency (SF) may be used to
carry the on/off hook information. Pulsing of the states can also transmit
the dialing pulses. The number of signals is small and they can be
transmitted in-band or out-of band. The states involved are shown in
Table 1.
Table 1. Single Frequency Signaling States Tone Signal Condition
State Forward Backward

Idle (On hook) On On
FORWARD
Seizure (off hook) Off On
Release (on hook) On Off/on
BACKWARD
Answer (off hook) Off Off
Clear Back (on hook) Off On
Blocking (off hook) On Off
For in band signaling the tone frequency is chosen to be 2600Hz.

or 2400 Hz. As the frequency lies within the speech band, simulation of
tone-on condition indicating end-of call signal by the speech has to be
guarded against, for pre-mature disconnection.
Out-of- Band signaling overcomes the problem of tone on condition
imitation by the speech by selecting a tone frequency of 3825 Hz which is
beyond the speech band. However, this adds up to the hardware costs.
6. E & M Signals
E & M lead signaling may be used for signaling on per-trunk basis.
An additional pair of circuit, reserved for signaling is employed. One wire
is dedicated to the forward signals ((M-Wire for transmit or mouth) which
corresponds to receive or R-lead of the destination exchange, and the
other wire dedicated to the backward signals (E-wire for receive or ear)
which corresponds transmit or send wire or S-Lead of the destination
exchange.
This type of signaling is normally used in conjunction with an interface
to change the E & M signals into frequency signal to be carried along
with the speech.
7. R2 Signaling
CCITT standardized the R2 signaling system to be used on national
and international routes. However, the Indian environment requires
TC31009D10 199
lesser number of signals and hence, a slightly modified version is being

used.
There is a provision for having 15 combinations using two out of
six frequencies viz., 1380, 1500, 1620, 1740, 1860 and 1980 Hz, for
forward signals and another 15 combination using two out of six
frequencies viz., 1140,1020, 900, 780, 660 and 540 Hz, for backward
signals. In India, the higher frequency in the forward group i.e., 1980 Hz,
and the lower frequency in the backward group, i.e., 540 Hz are not
used. Thus, there are 10 possible combinations in both the directions.
The weight codes for the combinations used are indicated in Table 3.
Table 3- Signal Frequency Index and Weight Code
Signal Frequency (Hz)
Forward 1380 1500 1620 1740 1860
Backward 1140 1020 900 780 660
Index f0 f1 f2 f3 f4
Weight 0 1 2 4 7
Code
8. Digital Signaling
All, the systems discussed so far, basically, are on per line or per
trunk basis, as the signals are carried on the same line or trunk. With
the emergence of PCM systems, it was possible to segregate the signaling
from the speech channel.
Inter exchange signaling can be transmitted over a channel directly
associated with the speech channel, channel-associated signaling (CAS),
or over a dedicated link common to a number of channels, common
channel signaling (CCS). The information transmitted for setting up and
release of calls is same in both the cases. Channel associated signaling
requires the exchanges, to have access to each trunk via the equipment
which may be decentralised, whereas, in common channel signaling, the
exchange is connected to only a limited number of signaling links
through a special terminal.
3. Channel- Associated Signaling (CAS)

CAS is composed of line signal and register signal. For CAS,
the signaling channel is combined with the bear information channel
(refers to register signal) or the two have fixed correspondence (refers to
line signal)
TC31009D10 200
Fig. 13.2 CAS
In the PCM systems the signaling information is conveyed on a

separate channel, which is rigidly associated with the speech channel.
Hence, this method is known as channel associated signaling (CAS).
Though the speech sampling rate is 8 KHz, the signals do not change as
rapidly as speech and hence, a lower sampling rate of 500 Hz, for
digitisation of signals can suffice. Based on this concept, TS 16 of each
frame of 125 microseconds is used to carry signals of 2 speech channels,
each using 4 bits.
Hence, for a 30 channel PCM system, 15 frames are required to
carry all the signals. To constitute a 2 millisecond multiframe of 16
frames. F 0 to F 15 TS 16 of the frame F 0 is used for multiframe
synchronisation. TS 16 of F1 contains signal for speech channels 1 and
16 being carried in TS 1 and TS 17, respectively, TS16 of F2 contains
signals of speech channels 2 and 17 being carried in TS2 and TS 18,
respectively and so on, Both line signals and address information can be
conveyed by this method.
Although four bits per channel are available for signaling only two
bits are used. As the transmission is separate in the forward and
backward direction, the bits in the forward link are called af and bf, and
those in the backward link are called ab and bb. Values for these bits are
assigned as shown in Table below.
As the dialing pulses are also conveyed by these conditions, the
line state recognition time is therefore, above a threshold value. The bit
bf is normally kept at 0, and the value 1 indicates a fault.
However, the utilisation of such a dedicated channel for signaling
for each speech channel is highly inefficient, as it remains idle during the
speech phase. Hence, another form of signaling known as common-
channel signaling evolved.
TC31009D10 201
Forward Backward
af bf ab
bb
Idle 1 0 1 0
Seizure 0 0 1 0
Seizure 0 0 1 1
acknowledge
Answer 0 0 0 1
Clear Forward 1 0 0/1 1
Clear Back 0 0 1 1
4. Common Channel Signaling System No. 7 (CCS#7)
History
Common Channel Signaling (CCS) is a technique that enables

stored program control exchanges, network databases, and other nodes
in a network to exchange messages related to call setup, call supervision,
call release (connection control information), information needed for
distributed application processing and network management
information.
From the point of view of plain old telephony service (POTS), CCS is
a technique that separates the physical channel used for signaling from
that, which is used to carry the end user’s telecommunication traffic.
One signaling channel is able to carry the signaling control information
of up to 3,000 trunks, depending on the implemented protocol; hence,
the term “common channel signaling” is used.
The CCITT has specified the common channel signaling system
no.7 (CCS-7). CCS-7 is optimised for application in digital networks. It is
characterised by the following main features:
• internationally standardized (national variations possible).
• suitable for the national, international and intercontinental
network level.
• suitable for various communication services such as telephony,
text services, data services digital network (ISDN).
• high performance and flexibility along with a future-oriented
concept, which well meet new requirements.
• high reliability for message transfer.
TC31009D10 202
• processor-friendly structure of messages (signal units of multiples

of 8 bits).
• signaling on separate signaling links; the bit rate of the circuits is,
therefore, exclusively for communication.
• signaling links always available, even during existing calls.
• use of the signaling links for transferring user data also.
• used on various transmission media
o cable (copper, optical fiber)
o radio relay
o satellite (up to 2 satellite links)
• use of the transfer rate of 64 kbit/s typical in digital networks.
• used also for lower bit rates and for analog signaling links if
necessary.
• automatic supervision and control of the signaling network.
Signaling Network
In contrast to channel-associated signaling, which has been
standard practice until now, in CCS7 the signaling messages are sent via
separate signaling links. One signaling link can convey the signaling
messages for many circuits.
Fig. 13.3 CCS
The CCS7 signaling links connect signaling points (SPs) in a

communication network. The signaling points and the signaling links
form an independent signaling network, which is overlaid over the circuit
network.
5. Definition
Signaling System 7 (SS7) is architecture for performing out-of-band

signaling in support of the call-establishment, billing, routing, and
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information-exchange functions of the public switched telephone network

(PSTN). It identifies functions to be performed by a signaling-system
network and a protocol to enable their performance.
SS7 is a means by which elements of the telephone network exchange

information is conveyed in the form of messages. SS7 messages can
convey information such as:
• Forwarding a call placed from XXXXXXXX to YYYYYYYY. Look for it

on trunk 067.
• The called subscriber for the call on trunk 11 is busy. Release the
call and play a busy tone.
• The route to ABC is congested. Don’t send any messages to ABC
unless whose priority is 2 or higher.
• Taking trunk 143 out of service for maintenance.
SS7 is characterized by high-speed packet data and out-of-band

signaling.
6. Out-of-Band Signaling
Out-of-band signaling is signaling that does not take place over the same
path as the conversation.
We are used to thinking of signaling as being in-band. We hear dial tone,

dial digits, and hear ringing over the same channel on the same pair of
wires. When the call completes, we talk over the same path that was
used for the signaling. Traditional telephony used to work in this way as
well. The signals to set up a call between one switch and another always
took place over the same trunk that would eventually carry the call.
Signaling took the form of a series of multi-frequency (MF) tones, much
like touch tone dialing between switches.
Out-of-band signaling establishes a separate digital channel for the

exchange of signaling information. This channel is called a signaling link.
Signaling links are used to carry all the necessary signaling messages
between nodes. Thus, when a call is placed, the dialed digits, trunk
selected, and other pertinent information are sent between switches
using their signaling links, rather than the trunks which will ultimately
carry the conversation. Today, signaling links carry information at a rate
of 56 or 64 kbps. It is interesting to note that while SS7 is used only for
signaling between network elements, the ISDN D channel extends the
concept of out-of-band signaling to the interface between the subscriber
and the switch. With ISDN service, signaling that must be conveyed
between the user station and the local switch is carried on a separate
TC31009D10 204
digital channel called the D channel. The voice or data which comprise
the call is carried on one or more B channels.
Out-of-band signaling has several advantages that make it more

desirable than traditional in-band signaling.
• It allows for the transport of more data at higher speeds (56 kbps
can carry data much faster than MF out pulsing).
• It allows for signaling at any time in the entire duration of the call,
not only at the beginning.
• It enables signaling to network elements to which there is no direct
trunk connection.
7. Signaling Network Architecture
If signaling is to be carried on a different path from the voice and data

traffic it supports, then what should that path look like? The simplest
design would be to allocate one of the paths between each interconnected
pair of switches as the signaling link. Subject to capacity constraints, all
signaling traffic between the two switches could traverse this link. This
type of signaling is known as associated signaling, and is shown below in
Figure.
Fig. 13.4 Associated Signaling
Associated signaling works well as long as a switch’s only signaling

requirements are between itself and other switches to which it has
trunks. If call setup and management was the only application of SS7,
associated signaling would meet that need simply and efficiently. In fact,
much of the out-of-band signaling deployed in Europe today uses
associated mode.
The North American implementers of SS7, however, wanted to design a

signaling network that would enable any node to exchange signaling with
TC31009D10 205
any other SS7–capable node. Clearly, associated signaling becomes much

more complicated when it is used to exchange signaling between nodes
which do not have a direct connection. From this need, the North
American SS7 architecture was born.
8. Signaling Components
The signaling architecture defines a completely new and separate

signaling network. The network is built out of the following three
essential components, interconnected by signaling links:
Signal Switching Points (SSPs)—SSPs are telephone switches (end

offices or tandems) equipped with SS7-capable software and terminating
signaling links. They generally originate, terminate, or switch calls.
Signal Transfer Points (STPs)—STPs are the packet switches of the SS7
network. They receive and route incoming signaling messages towards
the proper destination. They also perform specialized routing functions.
Signal Control Points (SCPs)—SCPs are databases that provide
information necessary for advanced call-processing capabilities.
Once deployed, the availability of SS7 network is critical to call

processing. Unless SSPs can exchange signaling, they cannot complete
any interswitch calls. For this reason, the SS7 network is built using a
highly redundant architecture. Each individual element also must meet
exacting requirements for availability. Finally, protocol has been defined
between interconnected elements to facilitate the routing of signaling
traffic around any difficulties that may arise in the signaling network.
To enable signaling network architectures to be easily communicated and

understood, a standard set of symbols was adopted for depicting SS7
networks. Figure shows the symbols that are used to depict these three
key elements of any SS7 network.
Fig. 13.5 Signaling Network Elements
STPs and SCPs are customarily deployed in pairs. While elements of a

pair are not generally co-located, they work redundantly to perform the
same logical function. When drawing complex network diagrams, these
pairs may be depicted as a single element for simplicity, as shown in
Figure.
TC31009D10 206
Fig. 13.6 STP and SCP Pairs
9. Basic Signaling Architecture
Figure shows a small example of how the basic elements of an SS7

network are deployed to form two interconnected networks.
Fig. 13.7 Sample Network
The following points should be noted:
1. STPs W and X perform identical functions. They are redundant.

Together, they are referred to as a mated pair of STPs. Similarly,
STPs Y and Z form a mated pair.
2. Each SSP has two links (or sets of links), one to each STP of a
mated pair. All SS7 signaling to the rest of the world is sent out
over these links. Because the STPs of a mated pair are redundant,
messages sent over either link (to either STP) will be treated
equivalently.
3. The STPs of a mated pair are joined by a link (or set of links).
TC31009D10 207
4. Two mated pairs of STPs are interconnected by four links (or sets
of links). These links are referred to as a quad.
5. SCPs are usually (though not always) deployed in pairs. As with
STPs, the SCPs of a pair are intended to function identically. Pairs
of SCPs are also referred to as mated pairs of SCPs. Note that they
are not directly joined by a pair of links.
6. Signaling architectures such as this, which provide indirect
signaling paths between network elements, are referred to as
providing quasi-associated signaling.
10. SS7 Link Types
SS7 signaling links are characterized according to their use in the

signaling network. Virtually all links are identical in that they are 56–
kbps or 64–kbps bidirectional data links that support the same lower
layers of the protocol; what is different is their use within a signaling
network. The defined link types are shown in Fig. 13.8 and defined as
follows:
Fig. 13.8 Link Types
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A Links
A links interconnect an STP and either an SSP or an SCP, which

are collectively referred to as signaling end points ("A" stands for access).
A links are used for the sole purpose of delivering signaling to or from the
signaling end points (they could just as well be referred to as signaling
beginning points). Examples of A links are 2–8, 3–7, and 5–12 as in
Figure.
Signaling that an SSP or SCP wishes to send to any other node is

sent on either of its A links to its home STP, which, in turn, processes or
routes the messages. Similarly, messages intended for an SSP or SCP will
be routed to one of its home STPs, which will forward them to the
addressed node over its A links.
C Links
C links are links that interconnect mated STPs. As will be seen

later, they are used to enhance the reliability of the signaling network in
instances where one or several links are unavailable. "C" stands for cross
(7–8, 9–10, and 11–12 are C links). B links, D links, and B/D links
interconnecting two mated pairs of STPs are referred to as either B links,
D links, or B/D links. Regardless of their name, their function is to carry
signaling messages beyond their initial point of entry to the signaling
network towards their intended destination. The "B" stands for bridge
and describes the quad of links interconnecting peer pairs of STPs. The
"D" denotes diagonal and describes the quad of links interconnecting
mated pairs of STPs at different hierarchical levels. Because there is no
clear hierarchy associated with a connection between networks,
interconnecting links are referred to as either B, D, or B/D links (7–11
and 7–12 are examples of B links; 8–9 and 7–10 are examples of D links;
10–13 and 9–14 are examples of interconnecting links and can be
referred to as B, D, or B/D links).
E Links
While an SSP is connected to its home STP pair by a set of A links,

enhanced reliability can be provided by deploying an additional set of
links to a second STP pair. These links, called E (extended) links provide
backup connectivity to the SS7 network in the event that the home STPs
cannot be reached via the A links. While all SS7 networks include A,
B/D, and C links, E links may or may not be deployed at the discretion
of the network provider. The decision of whether or not to deploy E links
can be made by comparing the cost of deployment with the improvement
in reliability. (1–11 and 1–12 are E links.)
TC31009D10 209
F Links
F (fully associated) links are links which directly connect two signaling
end points. F links allow associated signaling only. Because they bypass
the security features provided by an STP, F links are not generally
deployed between networks. Their use within an individual network is at
the discretion of the network provider. (1–2 is an F link.)
11. Basic Call Setup Example
Before going into much more detail, it might be helpful to look at

several basic calls and the way in which they use SS7 signaling.
Fig. 13.9 Call Setup Example
In this example, a subscriber on switch A places a call to a subscriber on

switch B.
1. Switch A analyzes the dialed digits and determines that it needs to

send the call to switch B.
2. Switch A selects an idle trunk between itself and switch B and
formulates an initial address message (IAM), the basic message
necessary to initiate a call. The IAM is addressed to switch B. It
identifies the initiating switch (switch A), the destination switch
(switch B), the trunk selected, the calling and called numbers, as
well as other information beyond the scope of this example.
3. Switch A picks one of its A links (e.g., AW) and transmits the
message over the link for routing to switch B.
4. STP W receives a message, inspects its routing label, and
determines that it is to be routed to switch B. It transmits the
message on link BW.
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5. Switch B receives the message. On analyzing the message, it

determines that it serves the called number and that the called
number is idle.
6. Switch B formulates an address complete message (ACM), which
indicates that the IAM has reached its proper destination. The
message identifies the recipient switch (A), the sending switch (B),
and the selected trunk.
7. Switch B picks one of its A links (e.g., BX) and transmits the ACM
over the link for routing to switch A. At the same time, it completes
the call path in the backwards direction (towards switch A), sends
a ringing tone over that trunk towards switch A, and rings the line
of the called subscriber.
8. STP X receives the message, inspects its routing label, and
determines that it is to be routed to switch A. It transmits the
message on link AX.
9. On receiving the ACM, switch A connects the calling subscriber
line to the selected trunk in the backwards direction (so that the
caller can hear the ringing sent by switch B).
10. When the called subscriber picks up the phone, switch B
formulates an answer message (ANM), identifying the intended
recipient switches (A), the sending switch (B), and the selected
trunk.
11. Switch B selects the same A link it used to transmit the ACM
(link BX) and sends the ANM. By this time, the trunk also must be
connected to the called line in both directions (to allow
conversation).
12. STP X recognizes that the ANM is addressed to switch A and
forwards it over link AX.
13. Switch A ensures that the calling subscriber is connected to
the outgoing trunk (in both directions) and that conversation can
take place.
14. If the calling subscriber hangs up first (following the
conversation), switch A will generate a release message (REL)
addressed to switch B, identifying the trunk associated with the
call. It sends the message on link AW.
15. STP W receives the REL, determines that it is addressed to
switch B, and forwards it using link WB.
16. Switch B receives the REL, disconnects the trunk from the
subscriber line, returns the trunk to idle status, generates a
release complete message (RLC) addressed back to switch A, and
transmits it on link BX. The RLC identifies the trunk used to carry
the call.
17. STP X receives the RLC, determines that it is addressed to
switch A, and forwards it over link AX.
18. On receiving the RLC, switch A idles the identified trunk.
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12. Database Query Example
People generally are familiar with the toll-free aspect of 800 (or
888) numbers, but these numbers have significant additional capabilities
made possible by the SS7 network. 800 numbers are virtual telephone
numbers. Although they are used to point to real telephone numbers,
they are not assigned to the subscriber line itself.
When a subscriber dials an 800 number, it is a signal to the switch

to suspend the call and seek further instructions from a database. The
database will provide either a real phone number to which the call
should be directed, or it will identify another network (e.g., a long-
distance carrier) to which the call should be routed for further
processing. While the response from the database could be the same for
every call (as, for example, if you have a personal 800 number), it can be
made to vary based on the calling number, the time of day, the day of the
week, or a number of other factors.
The following example shows how an 800 call is routed see Figure.
Fig. 13.10 Database Query Example
1. A subscriber served by switch A wants to reserve a rental car at a

company's nearest location. She dials the company's advertised
800 number.
2. When the subscriber has finished dialing, switch A recognizes that
this is an 800 call and that it requires assistance to handle it
properly.
3. Switch A formulates an 800 query message including the calling
and called number and forwards it to either of its STPs (e.g., X)
over its A link to that STP (AX).
TC31009D10 212
4. STP X determines that the received query is an 800 query and

selects a database suitable to respond to the query (e.g., M).
5. STP X forwards the query to SCP M over the appropriate A link
(MX). SCP M receives the query, extracts the passed information,
and (based on its stored records) selects either a real telephone
number or a network (or both) to which the call should be routed.
6. SCP M formulates a response message with the information
necessary to properly process the call, addresses it to switch A,
picks an STP and an A link to use (e.g., MW), and routes the
response.
7. STP W receives the response message, recognizes that it is
addressed to switch A, and routes it to A over AW.
8. Switch A receives the response and uses the information to
determine where the call should be routed. It then picks a trunk to
that destination, generates an IAM, and proceeds (as it did in the
previous example) to set up the call.
13. Layers of the SS7 Protocol
As the call-flow examples show, the SS7 network is an

interconnected set of network elements that is used to exchange
messages in support of telecommunications functions. The SS7 protocol
is designed to both facilitate these functions and to maintain the network
over which they are provided. Like most modern protocols, the SS7
protocol is layered.
Fig. 13.11 SS7 Protocol Stack
TC31009D10 213
Physical Layer
This defines the physical and electrical characteristics of the

signaling links of the SS7 network. Signaling links utilize DS–0 channels
and carry raw signaling data at a rate of 56 kbps or 64 kbps (56 kbps is
the more common implementation).
Message Transfer Part—Level 2
The level 2 portion of the message transfer part (MTP Level 2)

provides link-layer functionality. It ensures that the two end points of a
signaling link can reliably exchange signaling messages. It incorporates
such capabilities as error checking, flow control, and sequence checking.
Message Transfer Part—Level 3
The level 3 portion of the message transfer part (MTP Level 3)

extends the functionality provided by MTP level 2 to provide network
layer functionality. It ensures that messages can be delivered between
signaling points across the SS7 network regardless of whether they are
directly connected. It includes such capabilities as node addressing,
routing, alternate routing, and congestion control.
Collectively, MTP levels 2 and 3 are referred to as the message transfer

part (MTP).
Signaling Connection Control Part
The signaling connection control part (SCCP) provides two major

functions that are lacking in the MTP. The first of these is the capability
to address applications within a signaling point. The MTP can only
receive and deliver messages from a node as a whole; it does not deal
with software applications within a node.
While MTP network-management messages and basic call-setup

messages are addressed to a node as a whole, other messages are used
by separate applications (referred to as subsystems) within a node.
Examples of subsystems are 800 call processing, calling-card processing,
advanced intelligent network (AIN), and custom local-area signaling
services (CLASS) services (e.g., repeat dialing and call return). The SCCP
allows these subsystems to be addressed explicitly.
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Global Title Translation
The second function provided by the SCCP is the ability to perform

incremental routing using a capability called global title translation
(GTT). GTT frees originating signaling points from the burden of having to
know every potential destination to which they might have to route a
message. A switch can originate a query, for example, and address it to
an STP along with a request for GTT. The receiving STP can then
examine a portion of the message, make a determination as to where the
message should be routed, and then route it.
For example, calling-card queries (used to verify that a call can be

properly billed to a calling card) must be routed to an SCP designated by
the company that issued the calling card. Rather than maintaining a
nationwide database of where such queries should be routed (based on
the calling-card number), switches generate queries addressed to their
local STPs, which, using GTT, select the correct destination to which the
message should be routed. Note that there is no magic here; STPs must
maintain a database that enables them to determine where a query
should be routed. GTT effectively centralizes the problem and places it in
a node (the STP) that has been designed to perform this function.
In performing GTT, an STP does not need to know the exact final
destination of a message. It can, instead, perform intermediate GTT, in
which it uses its tables to find another STP further along the route to the
destination. That STP, in turn, can perform final GTT, routing the
message to its actual destination.
Intermediate GTT minimizes the need for STPs to maintain

extensive information about nodes that are far removed from them. GTT
also is used at the STP to share load among mated SCPs in both normal
and failure scenarios. In these instances, when messages arrive at an
STP for final GTT and routing to a database, the STP can select from
among available redundant SCPs. It can select an SCP on either a
priority basis (referred to as primary backup) or so as to equalize the load
across all available SCPs (referred to as load sharing).
ISDN User Part (ISUP)
ISUP user part defines the messages and protocol used in the
establishment and tear down of voice and data calls over the public
switched network (PSN), and to manage the trunk network on which they
rely. Despite its name, ISUP is used for both ISDN and non–ISDN calls.
TC31009D10 215
In the North American version of SS7, ISUP messages rely exclusively on

MTP to transport messages between concerned nodes.
Transaction Capabilities Application Part (TCAP)
TCAP defines the messages and protocol used to communicate

between applications (deployed as subsystems) in nodes. It is used for
database services such as calling card, 800, and AIN as well as switch-
to-switch services including repeat dialing and call return. Because TCAP
messages must be delivered to individual applications within the nodes
they address, they use the SCCP for transport.
Operations, Maintenance, and Administration Part (OMAP)
OMAP defines messages and protocol designed to assist

administrators of the SS7 network. To date, the most fully developed and
deployed of these capabilities are procedures for validating network
routing tables and for diagnosing link troubles. OMAP includes messages
that use both the MTP and SCCP for routing.
14. Information over the Signaling Link
Signaling information is passed over the signaling link in

messages, which are called signal units (SUs).
Three types of SUs are defined in the SS7 protocol.
1. Message signal units (MSUs)

2. Link status signal units (LSSUs)
3. Fill-in signal units (FISUs)
SUs are transmitted continuously in both directions on any link

that is in service. A signaling point that does not have MSUs or LSSUs to
send will send FISUs over the link. The FISUs perform the function
suggested by their name; they fill up the signaling link until there is a
need to send purposeful signaling. They also facilitate link transmission
monitoring and the acknowledgment of other SUs.
All transmission on the signaling link is broken up into 8-bit bytes,

referred to as octets. SUs on a link are delimited by a unique 8-bit
pattern known as a flag. The flag is defined as the 8-bit pattern
"01111110". Because of the possibility that data within an SU would
contain this pattern, bit manipulation techniques are used to ensure that
the pattern does not occur within the message as it is transmitted over
the link. (The SU is reconstructed once it has been taken off the link, and
any bit manipulation is reversed.) Thus, any occurrence of the flag on the
TC31009D10 216
link indicates the end of one SU and the beginning of another. While in
theory two flags could be placed between SUs (one to mark the end of the
current message and one to mark the start of the next message), in
practice a single flag is used for both purposes.
15. Addressing in the SS7 Network
Every network must have an addressing scheme, and the SS7

network is no different. Network addresses are required so that a node
can exchange signaling nodes to which it does not have a physical
signaling link. In SS7, addresses are assigned using a three-level
hierarchy. Individual signaling points are identified as belonging to a
cluster of signaling points. Within that cluster, each signaling point is
assigned a member number. Similarly, a cluster is defined as being part
of a network. Any node in the American SS7 network can be addressed
by a three-level number defined by its network, cluster, and member
numbers. Each of these numbers is an 8-bit number and can assume
values from 0 to 255. This three-level address is known as the point code
of the signaling point. A point code uniquely identifies a signaling point
within the American SS7 network and is used whenever it is necessary to
address that signaling point.
Network numbers are assigned on a nationwide basis by a neutral

party. Regional Bell operating companies (RBOCs), major independent
telephone companies, and interexchange carriers (IXCs) already have
network numbers assigned. Because network numbers are a relatively
scarce resource, companies' networks are expected to meet certain size
requirements in order to be assigned a network number. Smaller
networks can be assigned one or more cluster numbers within network
numbers 1, 2, 3, and 4. The smallest networks are assigned point codes
within network number 5. The cluster to which they are assigned is
determined by the state in which they are located. The network number
0 is not available for assignment and network number 255 is reserved for
future use.
TC31009D10 217
16. Basic Signaling Procedure
SPA & SPB – Signaling Points A & B
IAM – Initial Address Message ACM – Address Complete Message

ANC – Answer Signal Charge CLF – Clear Forward Message
RLG – Release Guard Signal
Fig 2.12 Signaling procedure
17. Advantages of CCS
The other advantages of CCS, in addition to space saving are: -

1. Faster call set up by cutting down the post dialer delay. In SPC
environment setting up a call via two transit centres takes just 0.8
second with CCS, compared to 3.5 seconds with MF signaling.
2. New services can be made available with a better quality. For
example, setting up a call with abbreviated dialing facility and routed
via two transit centres, takes just 3 seconds with CCS, as compared
to 12 seconds required by the network using CAS, moreover it is also
possible to use additional services, as it is possible to transmit signals
during speech phase also.
3. More call completion is possibly by re routing the call without
increasing the call set up time to an unacceptable level.
TC31009D10 218
4. In MF signaling system it is possible for a clever subscriber to access

the system by generating of generally used signaling tones. By
generating tones of the correct frequency and at the correct time can
make long distance calls without being charged thus resulting in loss
of revenue, however, such calls are not possible in CCS, as the
signaling link is totally separate from the speech link.
5. Unified signaling system is possible to provide all existing and
envisage services as required under the integrated services Digital
Network (ISDN).
6. Modern network management will be possible by provision of an
efficient means of collecting information and transmitting orders for
technical operation and maintenance of the network.
7. Traffic engineering becomes more efficient. The speech circuit’s
requirements will go down because of substantial reduction of
ineffective traffic. This advantage itself is sufficient to make
additional cost of signaling link cost effective. Moreover, as large
amount of data is available in shorter time span, the real time load on
the processor will come down resulting in increase in its efficiency by
almost 20%
Constraints of CCS
As in CCS more processing of the signaling is required, the cost of
hardware and software for the signaling interface will be more. In
addition to this, there would be following constraints of the network.
• As a single data link carries signaling information of a large
number of speech circuits, its failure would result in
immobilisation of all these speech circuits.
TC31009D10 219
Module - 01
Chapter – 14
ISDN
Session Objectives:

• Integrated Service Digital Network (ISDN)-

Introduction
• ISDN Service
• ISDN Channel Classification
• Interface Structures
TC31009D10 220
1. Integrated Services Digital Network (ISDN) - Introduction
The ISDN is an abbreviation of Integrated Services Digital Network.

The current communications networks vary with the type of service, such
as telephone network, telex network, and digital data transmission
network. On the other hand, the ISDN is an integrated network for
various types of communications services handling digitized voice
(telephone) and non-voice (data) information.
ISDN Definition
The CCITT defines the ISDN as follows:
“A complete, terminal-to-terminal digital network. Fig. shows the end-to-
end digital connectivity”.
Fig. 14.1 End-to-End Digital Connectivity
ISDN is a network that provides both telephone and non-telephone

services in the same network, which utilizes Signaling System No. 7
(SS7) for signaling between switching systems.
Fig. 14.2 The Signaling Connection between Switching Systems
A network offers standard user network interface. Fig. above shows

the standard user network interface.
TC31009D10 221
Fig. 14.3 Standard User Network Interface
2. ISDN Services
The ISDN provides the following functions, as shown in Fig.

• Packet switching service
• Circuit switching service
• Leased circuit service
Fig. 14,4 A Wide Range of Services
Circuit switching service includes both telephone and data circuit

switching. As shown in the Figure, ISDN can interface with various
terminals, such as a telephone set, FAX, Video terminal or personal
computer to provide a wide range of services.
Two statements can summarize the ISDN concept:
• ISDN offers a variety of services, such as telephone, data and
image transmission through one network.
• ISDN handles all information digitally.
TC31009D10 222
Standard user-network interface.
• The subscriber line is connected with an NT (Network Termination)

installed at the customer premises.
• Various terminals are connected to the NT. These terminals can
include digital telephones, multi media terminal, digital facsimile
machines, personal computers, etc. as shown in the figure.
• The NT and terminals are connected by S or T interface (S/T
interface), as recommended by the CCITT. Up to 8 terminals are
connected to one S/T interface. The NT and terminals are
connected using an 8-pin connector, which is also recommended
by the CCITT.
• As shown in this figure, the personal computer uses the RS232C
interface that is different from the ISDN S/T interfaces, so a TA
(Terminal Adapter) is provided to adapt the RS232C interface for
use with the ISDN interfaces.
The interface between the user and the network. Telephone service
makes use of two wires for the subscriber line between the switching
system and customer’s premises. These same two wires can be used by
ISDN to receive ISDN services. An NT (Network Termination) is installed
at the subscriber’s home and connected to the subscriber line.
Fig. 14.5 The Interface between the User and Network
TC31009D10 223
• The Interface between the NT and the ISDN exchange (switching

system) is called U interface. This interface has not been defined in
the CCITT Recommendations because circumstances are different
in each country. The point between the NT and the on-premises
terminals is called the S or T reference point. The ISDN
user/network interface refers to these S/T points, and is defined in
the CCITT Recommendations.
• The S/T interface uses four wires, two for sending and two for
receiving. Since U interface uses two wires, the NT provides a two-
wire/four-wire conversion function.
• CCITT recommends the use of AMI (Alternative Mark Inversion)
code at the S/T point. AMI code is a bipolar waveform.
• As shown in the figure, the ISDN Terminal provides S/T interface
that follows the CCITT Recommendations, and can be connected
directly to the NT. Since the personal computer and the analog
FAX utilize a different interface from S/T interface, they require
protocol conversion by a TA (Terminal Adapter).
3. ISDN Channel Classification
Various channels can be used to transmit information between a

terminal and the switching system. These include B, D and H channels.
Each channel has a different bit rate and information carrying attributes.
B- Channel
The B-channel carries user information such as voice and packet

data at a rate of 64 kbps. However, the B-channel does not carry
signaling information.
D- Channel
The D-channel interface carries mainly signaling information such

as originating or terminating subscriber number, call origination and
disconnect signals for circuit switching and packet switched user data at
16 kbps or 64 kbps. The D-channel also permits multiple logical
channels to be established for use in packet communications.
H-channel
The H-channel carries high-speed user information such as high-

speed facsimile, video, high-speed data, etc. H channels do not carry
TC31009D10 224
signaling information for circuit switching by the ISDN. Table outlines

these three channel types and characteristics.
Channel Bit Rate Function

Type
B 64 kbps To carry user information
Circuit switching mode and
packet switching mode
D 16 kbps To carry signaling information for
64 kbps circuit switching
H H0 : 384 kbps To carry high-speed packet data
H11 : 1536 kbps such as facsimile and video
H12 : 1920 kbps An H channel does not carry
signaling information for circuit
switching by the ISDN
Note: H0 : 64 X 6 = 384 kbps
K
H11 : 64K X 24 = 1536 kbps

H12 : 64K X 30 = 1920 kbps
4. Interface Structures
Basic Interface
This interface is primarily for home use. The basic interface is set
at a transmission speed of 144 kbps. This provides two (2) 64 kbps B-
channels for user information exchange and a 16 kbps D-channel for
signaling and control. The interface is thus referred to as 2B+D.
Primary Group Interface

These interfaces are primarily for business use. The primary group
interface for ATT system consists of a 1.544 Mbps line. This line can thus
provide up to 23 B-channels at 64 kbps and a single D-channel at 64
kbps.
In Europe and other countries using CEPT system standards, the
primary group is 2.048 Mbps and the interface is 30B-channels and
single 64 kbps D-channel (30B+D). This line is used for PABX etc.
TC31009D10 225
Module - 01
Chapter – 15
Telephone Traffic Engineering
Session Objectives:

• Basic Concepts of Telephone Traffic

• Parameter Definitions
• Traffic Load Measurement (TLM):
• Sampling Methods
• Call Arrival Patterns
• Traffic Models
• Measurement of Telephone Traffic.
• Grade of service.
• Erlangs formula
• Traffic offered
• Traffic Forecasting and Planning
• Demand Forecast Procedure
• Traffic Forecasting
• Traffic forecasting methods
TC31009D10 226
1. Basic Concepts of Telephone Traffic
Traffic Theory
Network designers need a way to properly size network capacity,
especially as networks grow. Traffic theory enables network designers to
make assumptions about their networks based on past experience.
Traffic is defined as either the amount of data or the number of
messages over a circuit during a given period of time. Traffic also
includes the relationship between call attempts on traffic-sensitive
equipment and the speed with which the calls are completed. Traffic
analysis enables you to determine the amount of bandwidth you need in
your circuits for data and for voice calls. Traffic engineering addresses
service issues by enabling you to define a grade of service or blocking
factor. A properly engineered network has low blocking and high circuit
utilization, which means that service, is increased and your costs are
reduced.
Telephone traffic is originated by the individual needs of different
subscribers and so it is beyond the control of telephone administration.
Any and every subscriber can originate a call at any and every moment
without giving any previous information and the duration of calls is also
not previously known. Although the individual telephone traffic
originates at random, the average telephone traffic for a particular
exchange follows the general pattern of activity in the exchange area.
Normally there is a peak in morning, a dip during lunch period followed
by an afternoon peak. In some localities the traffic has seasonal
characteristic, for example at a holiday resort.
Fig. 15.1 Switching links
TC31009D10 227
Fig. 15.2 Trunks or Circuits
Whatever be the nature of variation of traffic, a telephone engineer

is interested in maximum traffic that occurs in an exchange.
The hour in which maximum traffic usually occurs in an exchange
is known as Busy Hour.
Busy Hour Traffic is the average value of maximum traffic in the
busy hour. In computing Busy Hour Traffic the seasonal effects are also
taken into account. Sometimes it is convenient to refer to busy hour
calling rate. Busy hour calling rate is the number of calls originated per
subscriber in the busy hour. This provides a simple means for designing
the exchange with respect to the number of subscribers. It also provides
probable growth of traffic to the estimated growth in number of
subscribers. The busy hour calling rate may vary about 0.3 for a small
country exchange and 1.5 or more for a busy exchange in business area
in a city. In addition to Busy Hour calling rate day calling rate is also
useful for example to estimate the daily current drain on the exchange
battery.
When the volume of traffic is quoted in terms of number of calls
originated in a given time, this is insufficient to determine the
consequent occupancy of lines and equipment. Therefore, measurement
of traffic should not only consider number of calls but also their
duration. The duration during which equipments and circuits are held
when a call is made is called HOLDING TIME. Normally, it is average
holding time per call for the particular item of equipment that is taken
into account, so far as the caller is concerned the useful time is during
the conversation only. However, the total time during which equipments
and circuits are held when a call is made also includes, the period during
which call is being established and time taken to release the equipment
after the call has concluded.
2. Parameter Definitions
Trunk or Circuit: Trunk or Circuit describes the resources required, or

resources available, to carry traffic as shown in Fig. In wireless
communication, for analogue traffic, a trunk is usually equal to one radio
channel and for digital traffic; a circuit may be a dedicated radio channel
or one slot within multiple time slots or one code within a list of codes.
TC31009D10 228
Busy Hour Traffic (BHT):

Network traffic load is measured during the busiest hour because
this period represents the maximum traffic load that a network must
support. The result gives a traffic load measurement commonly referred
to as the Busy Hour Traffic (BHT). There are times, one cannot do a
thorough sampling or and can have only an estimate of how many calls
are handled daily. In such a circumstance, usually make assumptions
about environment, such as average number of calls per day and the
AHT. In the standard business environment, the busy hour of any given
day accounts for approximately 15 to 20 percent of the traffic for that
day. In computations, generally use 17 percent of the total daily traffic to
represent the peak hour traffic. In many business environments, an
acceptable AHT is generally assumed to be 180 to 210 seconds and used
to estimate trunking requirements without having more complete data.
Busy Hour Call Attempt (BHCA): BHCA is the number of times a

telephone call, or a telephony session request, is attempted or received
during the busiest hour of the day. BHCA refers to the transmitted
signaling requests per hour and it is the product of the number of users
(Nu) multiplied by the 2.4 Requests/hour. A call attempt means an
attempt to achieve a connection to one or more devices attached to a
telecommunications service.
HOLDING TIME (HT): HT is the period of time a trunk or circuit is busy

on a call. The holding time includes the channel request and seizes time,
the message length, and the hang time.
AVERAGE HOLDING TIME (AHT)
Total number of actual call seconds within an hour

Average holding time = -----------------------------------------------------------
Number of calls for same hour
AHT is the total time of all calls in a specified period divided by the
number of calls in that period.
Example:
(3976 total call sec.)/(23 calls) = 172.87 sec per call = AHT of 172.87
sec.
AVERAGE BUSY PERIOD: Average busy period is a continuous period of

time during which the highest usage occurs.
3. Traffic Load Measurement (TLM):

In traffic theory, you measure traffic load. Traffic load is the ratio
of call arrivals in a specified period of time to the average amount of time
TC31009D10 229
taken to service each call during that period. These measurement units
are based on Average Hold Time (AHT).
The two main measurement units used to measure traffic load are
Erlangs and Centum Call Seconds (CCS).
ERLANG: Unit commonly used to measure call intensity or telephone

traffic volumes, over a time period, normally 1 hour. It is generally
accepted that when referring to traffic usage, one Erlang represents a
single trunk or circuit occupied for 1 hour. One Erlang is 3600 seconds
of calls on the same circuit, or enough traffic loads to keep one circuit
busy for 1 hour.
(Number of hourly calls) (Average holding time in sec)

Traffic Load in Erlang = ---------------------------------------------------------
3600
Traffic in Erlang is the product of the number of calls times of the AHT
divided by 3600.
Example: If a group of user made 30 calls in 1 Hour, and each call had
an average call duration (AHT) of 5 minutes, then the number of Erlangs
this represents is worked out as follows:
Minutes of traffic in the hour = number of calls x duration

Minutes of traffic in the hour = 30 x (5 x 60 sec)
Minutes of traffic in the hour = 9000
Hours of traffic in the hour = 9000/3600
Hours of traffic in the hour = 2.5
Traffic figure = 2.5 Erlangs
Erlang traffic measurements are made in order to help

telecommunications network designers understand traffic patterns
within their voice networks. This is essential to successfully design their
network topology and establish the necessary trunk group sizes.
Erlang traffic measurements or estimates can be used to work out
how many lines are required between a telephone system and a central
office (PSTN exchange lines), or between multiple network locations.
CENTUM (HUNDRED) CALL SECONDS (CCS): One CCS is 100 seconds

of calls on the same circuit. Voice switches generally measure the
amount of traffic in CCS.
(Number of hourly calls) (Average holding time in sec)
Traffic Load (CCS)= ---------------------------------------------------------------
100
TC31009D10 230
Traffic in CCS is the product of the number of calls times of the AHT
divided by 100.
Example: (23 calls * 172.87 AHT)/100 = 39.76 CCS
CCS TO ERLANG: Erlang = CCS / 36
ACCESS DELAY (AD): AD is the amount of time it takes to get access to

the first available trunk. This time delay is expressed as multiples of the
average Holding Time (HT).
GRADE OF SERVICE (GOS)

GOS for systems with queues is the probability of a call being
delayed by busy servers and is associated to an access delay. It is
expressed as a decimal or percentage.
MESSAGE LENGTH (ML): ML is the time taken by the trunk to send the
information portion of a transmission associated with a call. This time
varies widely depending on the business type or radio service category.
TOTAL TRAFFIC VOLUME OR TRAFFIC INTENSITY (TI): TI is the

volume of traffic the radio system is expected to carry during peak
periods. This traffic is usually expressed in Erlangs.
TRAFFIC OFFERED (TO): TO per user is the volume of traffic each user
is expected to be offered by the system during peak periods, i.e. traffic
per phone / mobile. This traffic is usually expressed in Erlangs.
4. Sampling Methods
The accuracy of traffic analysis depends on the accuracy of
sampling methods. The following parameters will change the represented
traffic load:
• Weekdays versus weekends
• Holidays
• Type of traffic
• Apparent versus offered load
• Sample period
• Total number of samples taken
• Stability of the sample period
Probability theory states that to accurately assess voice network
traffic, at least 30 samples of the busiest hours of a voice network to be
taken during the sampling period. To get the most accurate results, we
need to take as many samples of the offered load as possible.
The ITU-T recommends that public switched telephone network
(PSTN) connections measurement or read-out periods are 60 minutes
TC31009D10 231
and/or 15-minute intervals. These intervals are important because they

summarize the traffic intensity over a period of time. If measurements
are taken throughout the day, one can find the peak hour of traffic in
any given day. There are two recommended ways to determine the peak
daily traffic, as follows:
• Daily Peak Period (DPP) records the highest traffic volume
measured during a day. This method requires continuous
measurement and is typically used in environments where the
peak hour may be different from day to day.
• Fixed Daily Measurement Interval (FDMI) requires measurements
only during the predetermined peak periods. It is used when traffic
patterns are somewhat predictable and peak periods occur at
regular intervals.
Business traffic usually peaks around 10:00 a.m. to 11:00 a.m.
and 2:00 p.m. to 3:00 p.m.
In the example in Table 1, using FDMI sampling, the hour with the
highest total traffic load is 10 a.m., with a total traffic load of 60.6
erlangs.
Table 1 Daily Peak Period Measurement
Hour Monday Tuesday Wednesday Thursday Friday Total

Load
9:00 12.7 11.5 10.8 11.0 8.6 54.6
a.m.
10:00 12.6 11.8 12.5 12.2 11.5 60.6
a.m.
11:00 11.1 11.3 11.6 12.0 12.3 58.3
a.m.
12:00 9.2 8.4 8.9 9.3 9.4 45.2
p.m.
1:00 10.1 10.3 10.2 10.6 9.8 51.0
p.m.
2:00 12.4 12.2 11.7 11.9 11.0 59.2
p.m.
3:00 9.8 11.2 12.6 10.5 11.6 55.7
p.m.
4:00 10.1 11.1 10.8 10.5 10.2 52.7
p.m.
TC31009D10 232
The example in Table 2 uses DPP to calculate total traffic load.
Peak 12.7 12.2 12.5 12.2 12.3 61.9

Traffic
Peak 9:00 2:00 10:00 a.m. 10:00 11:00
Time a.m. p.m. a.m. a.m.
It is also required to divide the daily measurements into groups

that have the same statistical behavior. The ITU-T defines these groups
as: Workdays, Weekend days, and Yearly exceptional days. Grouping
measurements that have the same statistical behavior becomes
important because exceptionally high call volume days (such as New
Year Day and Festival Day) might skew the results.
5. Call Arrival Patterns
The first step in choosing the proper traffic model is to determine

the call arrival pattern. Call arrival patterns are important in choosing a
traffic model because different arrival patterns affect traffic facilities
differently. The three main call arrival patterns are as follows:
• Smooth Call Arrival Pattern
• Peaked Call Arrival Pattern
• Random Call Arrival Pattern
Fig. 15.3 Smooth Call Arrival Pattern Fig. 15.4 Peaked Call Arrival
Pattern
6. Traffic Models
After the determination of the call arrival patterns and determined
the blocked calls, number of sources, and holding times of the calls, one
need to select the traffic model that most closely fits the environment.
Although no traffic model can exactly match real life situations, these
models assume the average in each situation.
TC31009D10 233
There are many different traffic models—the key is to find the

model that best suits the environment. The traffic models that have the
widest adoption are Erlang B, Extended Erlang B, and Erlang C.
ERLANG B
This is the most commonly used traffic model, and is used to work
out how many lines are required if the traffic figure (in Erlangs) during
the busiest hour is known. The model assumes that all blocked calls
are immediately cleared.
The Erlang B traffic model is based on the following assumptions:
• An infinite number of sources
• Random traffic arrival pattern
• Blocked calls cleared
• Hold times exponentially distributed
In Erlang B traffic model, three variables involved such as Busy
Hour Traffic (BHT), Blocking and Lines:
• BHT (in Erlangs) is the number of hours of call traffic there
are during the busiest hour of operation of a telephone
system.
• Blocking is the failure of calls due to an insufficient number
of lines being available. E.g. 0.02 mean 2 calls blocked per
100 calls attempted.
• Lines are the number of lines in a trunk group.
If two of the figures are known, the third can be obtained from
Erlang B Table (shown for 24 trunks only).
TC31009D10 234
The Erlang B model is used when blocked calls are rerouted, never
to come back to the original trunk group. This model assumes a random
call arrival pattern. The caller makes only one attempt; if the call is
blocked, then the call is rerouted. The Erlang B model is commonly used
for first-attempt trunk groups where you need not take into
consideration the retry rate because callers are rerouted, or expect to see
very little blockage.
Extended ERLANG B
The Extended Erlang B traffic model is used by telephone system

designers to estimate the number of lines required for PSTN connections
(CO trunks) or private wire connections and takes into account the
additional traffic load caused by blocked callers immediately trying to call
again if their calls are blocked. This traffic model may be used where no
overflow facilities are available from the trunk group being designed. The
four variables involved are Busy Hour Traffic (BHT), Blocking and Lines:
• Recall factor is the percentage of calls which immediately retry if
their calls are blocked.
• BHT (in Erlangs) is the number of hours of call traffic there are
during the busiest hour of operation of a telephone system.
• Blocking is the failure of calls due to an insufficient number of
lines being available. E.g. 0.03 mean 3 calls blocked per 100 calls
attempted.
• Lines are the number of lines in a trunk group.
Of the Busy Hour Traffic, Blocking and Lines values, if two of the
figures are known, the third can be worked out.
7. Measurement of Telephone Traffic.
The total cost of providing telephone service can be roughly

divided into those charges, which are constant and independent of
volume of traffic and those, which are determined by the amount of
traffic. The cost of subscriber’s line and instrument and certain
individual equipment in the exchange is totally independent of the
volume of traffic. The quantity of common switching equipment
required is almost entirely dependent by volume of traffic. The quantity
of such equipment is dependent not only on number of calls but also on
duration of calls. Therefore to determine the quantity of switching
equipment in automatic exchange or staffing in manual exchange
telephone traffic may be measured in terms of both the number of calls
and the duration of calls.
For certain purpose it is sufficient to specify a Traffic Volume
which is product of number of calls occurred during the time concerned
TC31009D10 235
by their average duration. However for the purposes of automatic

exchange a more precise unit of traffic flow is required. This is called
traffic intensity. Traffic intensity is the average number of calls
simultaneously in progress. The unit of traffic intensity has been named
the Erlang.
A traffic intensity of one Erlang is obtained in any specified period
when the average number of calls simultaneously in progress during that
period in unity. The specified period is always one hour and is taken as
being the busy hour unless some other period is indicated.
There is a more precise way to define traffic intensity. The average
Traffic Intensity during a specified period T carried by a group of circuits
or equipments is given by the sum of the holding times divided by T the
holding times and period T all being expressed in the same unit.
Sometime it is stated that the average traffic intensity is equal to
the average number of calls that originate during the average holding
time. All the above three definitions give the same numerical result.
The foregoing relationships may be expressed symbolically as
follows. Let S be sum of holding times during a given period T. both
expressed in hours. Then by definition
A = S/T
Where A is the average traffic intensity. Let C be the total number
of calls during the period T then the average holding time ‘t’ hours per
call, is given by
t=S/C
Then A = S/T Can also be written as
A = Ct/T
It also follows that when the average call duration is known, the
average call intensity can be obtained by determining the number of calls
occurring during the period T. Also because A is equal to average number
of calls simultaneously in progress, an approximate value of A can be
obtained by counting the number of occupied circuits or equipments at
uniform interval during the time T and finding the average value.
For example a group of selectors is examined at 5 minute intervals

for one hour with following result-
Period = 1 2 3 4 5 6 7 8 9 10 11 12
Number of selectors held = 324675962010
Total number of selectors held = 45
Average number of Selector held = 45/12 =3.75
The approximate traffic carried by the group was therefore 3.75 E.

This figure is not necessarily accurate because between actual counts
there may be greater or lesser number of simultaneous calls in progress.
Making the period not more than twice the average holding time
can reduce the error. A Commonly used interval is 3 minutes.
TC31009D10 236
8. Grade of service.
Owing to the fact that calls originated in a pure chance manner, it

is likely that during the busy hour some calls may fail to mature due to
insufficiency of switching equipment. To ensure that the number of calls
so lost is reasonably small, it is the standard practice switching
equipment such that on the average not more than one call out of every
500 in the busy hour is lost at each switching stage, with the provision
that loss does not fall below 1in 100 with a 10 percent increase of traffic.
This allowable loss is termed the grade of service and is usually
represented by the symbol ‘B’ with one lost call in 500 the grade of
service is written as
B = 1/500 or B = 0.002
The Grade of service is a factor employed for dimensions of the
exchange equipment.
A few typical problems are worked out below to illustrate how the
terms and definitions of telephone traffic are actually applied in practice.
Example 1
If the calling rate per line per day in an exchange of 5000 lines is
6.0 and proportion of the traffic that occurs in the busy hours is 12
percent, what is the busy hour’s traffic in Erlangs, assuming an average
holding time of 2.5 minutes per call?
Calling rate per line per day = 6.0
Capacity of the exchange = 5000 lines
Total number of calls made in a day = 5000 x 6 = 30000
Number of calls originated in busy hours = 30,000 x 12/100
Holding time of a call (t) = 2.5 minutes
Busy hour traffic = C x t/60
= 3600 x 2.5/60
= 150 Erlangs or TUs.
Example 2
A group of trunks observed for 10 busy hours carried an average of

20 Erlangs and the total number of calls lost was 12. The calls had an
average duration of 2 minutes. What grade of service was given?
Traffic carried by the selectors in one busy hour = 20 Erlangs

Average holding time = 2 minutes
Total number of calls carried in one busy hour = 20 x 60/2 = 600
Number of calls lost in ten busy hours = 12
Average number of calls lost in one busy hour = 12/10 = 1.2
Total number of calls offered in busy hour = 600 + 1.2 = 601.2
TC31009D10 237
Grade of service = Number of calls lost

Number of calls offered
Grade of Service = 1.2/601.2 = 0.001996, Say- 0.002
9. Erlangs formula
Erlangs formula, which is given below, furnishes a method of

computing the number of trunks (arranged in a full availability group)
required to carry a given volume of traffic with a specified grade of service
AN
N!
B= ..... (1)
1 + A + A + A .....+.....A
2 3 n
1! 2! 3! N!
When B = grade of service

N = Number of trunks in a full availability group
and A = average traffic offered.
The above formula is based on a number of assumptions, which

approximate nearly to the full availability conditions actually obtained in
practice. These assumptions are as follows...
(a) That calls occur individually and collectively at random.
(b) That full availability conditions exist.
(c) That the average traffic is the average of a large number of
busy hours.
(d) That calls which originate when all trunks are busy are lost
and that such lost calls have a zero holding time.
(e) That the condition known as statistical ‘equilibrium’ exists
i.e. there is no tendency for the traffic as a whole to rise or
fall.
10. Traffic offered
Traffic offered to each trunk in a full availability group

By definition, the grade of service (B) is the ratio of traffic lost to
traffic offered. Thus if “A” is the traffic offered and ‘a’ the traffic lost, then
B = a/A or a = A/B
Traffic lost = Traffic offered X grade of service
Substituting in Erlangs formula
AN
N !
Lost traffic (a) = A x

1 + A + A2--------------- + AN
1! 2! N!
Traffic offered to 1st trunk = total traffic = A
Traffic offered to 2nd trunk = Traffic lost off the first trunk
TC31009D10 238
= Traffic offered to 1st trunk X GoS

A1
1!
= A x
1 + A1
1!
A
A = A x
1 + A
= A2
1 + A .................... (2)
Traffic offered to 3rd trunk = Traffic lost off the second trunk
A2
2!
=A x
1 + A1 + A2
1! 2!
A3
= ----------------- (3)
2 (1 + A + A2 )
2
Similarly, the traffic offered to the Nth trunk
AN
= --------------- (4)
N-1 (1+ A + A2 +.......+ AN-1)
1! 2! N-1 !
Traffic carried by each trunk in a full availability group

Traffic carried by individual trunks is calculated by direct
application of Erlangs formula as follows.
(i) Traffic carried by the 1st Trunk
= Traffic offered to the 1st trunk-Traffic lost off the 1st trunk.
Now traffic offered = Total traffic originated to 1st trunk = A
and Traffic lost off 1st Trunk
A2
= (Vide equation ......2)
1 + A
Therefore Traffic carried by the 1st Trunk
TC31009D10 239
A
= A -
1 + A
A
= ......................... (5)
1 + A
(ii) Traffic carried by 2nd Trunk
= Traffic offered to 2nd Trunk - Traffic lost off 2nd Trunk

= Traffic offered to 2nd Trunk - Traffic offered to the 3rd
Trunk
A2 A3
=
1 + A 2 ( 1 + A + A2 /2 )
A2 ( 2 + A )
= ---------------- (6)
2 ( 1+A ) ( 1+A+A2/2 )
(iii) Similarly the Traffic carried by the Nth Trunk = Traffic offered
to the Nth trunk - Traffic lost off Nth Trunk
= Traffic lost off the (N-’) th Trunk - Traffic lost off the Nth
Trunk
= AB1 - AB --------------------- (7)
Where “A” is traffic offered, B and B1 are the grades of service with
N
and (N-1) Trunks respectively.
AN
N!
Now B = ----- (8)
1 + A + A + ----------------+ A
2 +A N-1 N
1! 2! N-1! N!
A N-1
N-1!
and B1 = ------- (9)
1+A +A 2 + .........................+ A N-1
1! 2! N-1!
Transposing equation (8)
TC31009D10 240
B (1 + A + A 2 + ................................+ A N-1 + A N ) = A N
1! 2! N-1! N! N!
i.e. B ( 1 + A + A2 +........................+ A N-1 + BAN-1 = A N

1! 2! N-1! N! N!
B ( 1 + A + A2 + A 3 --------- + A N-1 ) = A N ( 1-B ) ............. (10)
1! 2! 3! N-1! N!
Similarly
B1 ( 1 + A + A 2 + A 3 ................. A N-1) = A N-1 ............. (11)
1! 2! 3! N-1! N-1!
B1 = A N-1 x N! .
B N-1! AN (1-B)
N
=
A (1-B)
BN
or AB1 =
(1 - B )
Now last trunk Traffic = AB1- AB
BN - AB
= 1- B
B ( N A)
= 1-B
When the grade of service is reasonable good, (1-B) is nearly equal
to unit with this approximation.
Last Trunk traffic = B(N-A) -------------------------------------- (12)
Typical Example
Calculate (a) The Lost Traffic and (b) The Grade of Service given by
4 switches arranged in a full availability group, when offered 0.53 Erlang
With usual notation by Erlang formula

AN
N!
B =
1 + A + A 2 + ...................................+ A N
1! 2! N!
Substituting A = 0.53 Erlang and N = 4
TC31009D10 241
The GoS
(0.53) 4
4!
B=
1 + 0.53 + (0.53) 2 + (0.53) 3 + (0.53) 4
1! 2! 3! 4!
0.0789
24
=
1 + 0.53 + .1404 + .0245 + .003
0.003 = 1
= 1.698 566
= 0.00177
(a) Lost Traffic = traffic offered x grade of service

= 0.53 x 0.00177 = 0.000928 Erlang
11. Traffic Forecasting and Planning
In the development of a telecommunications network the time lag

between identification of the need to provide subscribers’ equipment,
lines and exchange plant, and the ability to meet those needs may be
quite considerable. To augment the network we must accurately forecast
these needs so that plant arrives and is installed before existing capacity
is exhausted. In an ideal telecommunications network, with no
restrictions, forecasting and planning would ensure that demand for
services are accurately foreseen and satisfied as they arise.
Types of Forecasts
Demand Forecast
To forecast the number of subscribers in a well defined area. For
access network planning this forecast may be done for each section first
and then this forecast may be combined to form forecasts for bigger
areas like blocks and exchange areas. For the purpose of planning the
junction network forecasts of subscribers are required in some of the
methods.
Traffic Forecast
For access network planning forecast of traffic per subscriber and
per block (originating and terminating) may be required. This would help
in deciding the number optical terminals and number of subscribers per
optical terminal when concentration is used. In case of design of ring
structures where a number of optical terminals would be put in a ring,
inter-block traffic may be of some importance.
TC31009D10 242
For junction network planning, total exchange originating and

terminating traffics, traffics for different traffic zones and traffic
dispersion (traffic interest) between different zones and exchanges will
have to be projected.
12. Demand Forecast Procedure
Period of Forecast
The demand forecast would usually be made annually in short
term, say for 3-5 years and then at an interval of 5 years such that the
last forecast is for 20 year period.
Method of Forecasting
Forecast made at the corporate level follows "Top-Down"
methodology using macro economic parameters and mathematical
models to arrive at a country level forecast which is then appropriated to
the regions, exchange areas etc. Forecast at the regional level or
exchange area level (or below) will involve surveying and field studies.
Information required for forecasting

Forecasting studies are facilitated by availability of good
geographical maps, network drawings and subscriber database.
Geographical maps that would be useful are
• A city area map showing all the exchange location and physical
features
• A detailed map of the exchange areas showing all the
plots/buildings. This is usually in the scale of 1:1000
• Network drawings showing exchange areas, exchange boundaries,
blocks, cabinet locations, cabinet/block boundaries would also be
required.
Zoning of the area

For ease of survey and correctness of forecasting a new area needs
to be subdivided into manageable units. A city would have a number of
exchange areas, an exchange area would have a number of blocks or
cabinet areas and each block would have a number of sections or DP
areas. To zone a new area, the area can be divided into survey units and
teams of surveyor could then survey the assigned survey units and
record information about the types of tenancies, residential or business
that may exist, under construction or planned. From this information,
tenancies/plots are grouped into sections. Sections should be
predominantly residential or business areas or they could be areas where
residential and business tenancies are uniformly mixed. Size of a section
in governed by the size of DPs normally used. This could be an area that
can be served by a DP of size 10 or 20 pairs. Multistoried buildings could
TC31009D10 243
make separate sections while in a low density area the size of section
could be large. The section would then be grouped into Blocks in such a
way that it should be possible to serve each block by a cabinet. In theory
at least the cabinet and block areas are synonymous but in practice
there may be times when one block has more than one cabinet. Section
becomes the smallest unit of area for which the forecast is made. Section
forecasts can be combined to make block forecast and the block forecasts
are grouped to make exchange forecast.
Classification of sections
Detailed survey would be carried out in each section to identify the
types of tenancies. Since all kind of tenancies do not have the same
growth potential, the tenancies would need to be classified on some
basis. A common method of classification is based on types of tenancies.
Broad classification of tenancies used is:
• Residential
• Business
Sub-classification is then done in each of the above categories. R1
could be detached houses, R2 could be Condominiums or luxury
apartments, R3 could be Low cost housing and so on. Similarly for
business, B1 could be big office complexes, B2 could be big shopping
malls, B3 could be detached shops, B4 could be factories/workshops, B5
could be hospitals, government offices, schools etc., B6 could be
restaurant, cinema, petrol station, parks, mosque/church/temple,
museums etc.
13. Traffic Forecasting

Traffic consideration has rarely been important for dimensioning
the access network. The reason for this has been exclusive rights of a
subscriber on the pair allocated to him. The maximum traffic that a pair
can carry is 1 Erlang. Also that is the maximum a subscriber can
generate. [Simply put, a circuit continuously busy during the
observation period is said to be carrying 1 Erlang traffic]. With the advent
of new technologies, use of concentration in the access network (a la
V5.2 interface), subscriber will no longer have exclusive right over a
channel from his phone to the exchange. A subscriber does not use his
phone all 24 hours a day and channels can be more efficiently utilized if
shared among many subscribers. This would need measurement and
forecasting of traffic.
Traffic data
The production of traffic forecasts and the subsequent application
of traffic theory to the dimensioning and administration of a telephone
network depend on the availability and quantity of reliable reference
TC31009D10 244
data. These data must be systematically checked during collection and

processing to ensure their integrity.
Traffic data for planning purposes are of three main types:
a) Exchange Data
These are the general statistics which specify the traffic generating
capabilities of part or all of the exchange and include measured data as
well as data supplied from other sources, together with information
derived from these data which includes:
• Number of subscribers of various categories.
• Total originating and terminating traffics from groups of
subscribers within the exchange
• Call holding times.
• Usage rates (erlangs per subscribers)
b) Traffic Route Data

These data usually include established routes which are
dimensioned for a good grade of service and contain information about
route size, identity, and the traffic carried by it. This would be required
for dimensioning junction network
c) Dispersion Data
These data are held in the form of a set of row vectors for each
originating exchange. They may contain call dispersion and/or traffic
dispersion and associated mean holding time statistics. This would be
required for dimensioning junction network
14. Traffic forecasting methods
1. Intuitive forecasting is the systematic assessment of informed

opinion and is often the basis of subscriber surveys which are developed
to produce forecasts of subscriber development.
2. Trend methods assume that the future will have a predictable

relationship with past performance. Their application depends on the
existence of a database of past statistics which can be analysed to
determine past trends. Trend projection is a frequently used traffic
forecasting method for the short to medium term.
3. Goal-oriented (Normative) forecasting assumes that there will exist

needs in the future which will have some effect on the parameter being
studied. This is of importance in longer term traffic forecasting, (typically
in excess of 5 years), since the total traffic is dependent on a variety of
other parameter such as population, subscribers and calling rates, etc.
TC31009D10 245
4. Comparison methods, where traffic patterns in a particular area are

forecasted on the basis of known historical developments in another
area.
Economic Planning Periods

When cable is laid it is necessary to provide some spare capacity to
meet the future requirements. The number of years for which provision is
made in advance is called the planning period. If a cable is provided for a
long planning period, the initial cost per pair becomes less but
considerable cable capacity remains idle for a long time period and such
provision becomes uneconomical. On the other hand, if the cable is
provided for too short a planning period, the cost per pair becomes very
high and the cable provision again becomes uneconomical. There is an
optimum period of planning for provision of cables. The economical
planning periods for the primary network is short to medium term while
that for distribution network is long term.
Generally for bigger systems where the demand/rate of growth is
high, it is not practicable to have long planning periods in view of
uncertainty in forecast, the high capital cost involved, large scale
introduction of fiber in the network and technological advances in the
access network. It is therefore becoming increasingly common to plan the
primary network for 3 to 5 years and distribution network for a 5-10 year
forecast.
Duct Planning
The system of laying cables in pipes laid underground with
provision of manholes/ joint boxes at specified distances so that, without
resorting to repeated digging, these cables may be operated for
rectification of faults or for joining with other cables. Additional cables
may be laid subsequently on the same route in near future. Ducts are
planned for long term.
Planning New Technologies in the Access Network
The term access network refers to the network between the local
exchange and the subscriber. This network is still predominantly made
up of the copper cable based point-to-point connections. This has kept
the network in large proportions passive, inflexible and relatively
unreliable. With the advent of digital technology, the process of
installation, maintenance has become less cumbersome and quality of
services has improved. It is therefore felt that the any cause for
dissatisfaction, among customers about present services, is
predominantly due to the frequent failures in the access network and the
time taken for restoring them. One of the most fundamental and
remarkable of the driving technologies in access network is the optical
fiber.
TC31009D10 246
Telephone operators keep pace with the changing technologies and

take the following steps:
• Provide infrastructure prepared for higher bandwidth, like
fiber to the curb (FTTC) solutions.
• Move from a passive to a very active access network.
• Provide network units, which will enable them to flexibly
provision a mix of services with minimum impact on network
management and installed equipment base.
• Develop ring structures within the network to increase the
subscriber loop reliability.
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Module - 01
Chapter – 16
Long Distance Switching Plans
Session Objectives:

• Charging Plan
• National Switching Plan.
• National Routing Plan
• Transmission Plan
• National Numbering Plan
• Synchronous Plan
• Signaling Plan
TC31009D10 248
1. Charging Plan
The Charging plan was introduced on 1.1.71. The whole country
was divided into 278 areas called as “Long Distance Charging Areas”
(LDCAs). Each charging area comprised one or more revenue district
with a nominated “Long Distance Charging Centre” (LDCC), which was
an important town, or headquarters of one of the revenue district.
The trunk calls were categorised into two types- Short Distance
Trunk Calls and Long Distance Trunk Calls. The trunk calls between
two telephone exchange systems in the same LDCA or the contiguous
LDCAs were termed as Short Distance Trunk Calls. And the trunk calls
between two telephone exchange systems in the two noncontiguous
LDCAs were termed as Long Distance Trunk Calls.
The Short distance Trunk Calls were charged on point to point
basis as per the radial distance between the calling and called exchange
system while the long distance trunk calls were charged on area to area
basis as per the radial distance between the LDCC of the calling and
called exchange systems LDCAs
1.1 Need for the provision of the this charging plan

In this method of charging for each new exchange installation (i.e.
new exchange system) calculation of radial distance of this exchange
system from other exchange systems in the same LDCA as well as
contiguous LDCAs is required for the determination of trunk call rates.
Then this information has to be sent to all the concerned exchange
systems.
The country has now been divided into 321 Secondary Switching
Areas. Therefore, the existing LDCAs have to be revised to meet this new
concept.
To further simplify the charging plan so that minimum calculation
works is involved whenever a new exchange is installed.
1.2 Revised Charging Plan
1.3 Revision of Long Distance Charging Areas

The country has been divided into 321 Secondary Switching Areas
(SSAs), which are generally co-terminus with a revenue district. The
technical and administrative infrastructure of the department has been
changed on the basis of these SSAs. Therefore, the 278 Long Distance
Charging Areas (LDCAs) have now been increased to 321 LDCAs to be co-
terminus with the 321 SSAS. An important town in each LDCA has been
nominated as Long Distance Charging Centre (LDCC)
1.4 Identification of Short Distance Charging Areas
Each LDCA has been further divided into short Distance Charging
Areas (SDCAs). One important town in each SDCA has been nominated
as Short Distance Charging Centre (SDCC). Generally each SDCA
TC31009D10 249
comprise of one tehsil or revenue sub-division of a district. This has been

done as per the following guidelines:
• As far as possible a whole Tehsil is made as a SOCA.
• The area of SDCA is generally between 800 and 2000 Sq. Kms.
• Where the area of a Tehsil has been below 800 Sq. Kms, it has
been combined with an adjacent Tehsil to form a single SDCA
provided the distance between the farthest exchanges is up to 40
Kms.
• Where the area of a Tehsil has been above 2000 Sq. Kms. or where
the distance between any two exchanges exceeds 40 Kms, it has
been split into two more SDCAs as warranted by the existing or
anticipated telephone facilities. There are a few exceptions where
the splitting has not been done and the respective heads of circles
have been requested to reconsider them for splitting then into
smaller SDCAs.
• In the case of hilly areas the distance between the farthest
exchanges in an SDCA can be up to 50 Kms. instead of 40 Kms.
Fig. 16.1 Charging areas
TC31009D10 250
1.5 Charging:
The charging of calls is to be done on the basis of the following

guidelines: -
• Call between the exchanges within the same SDCA:
The STD call within the same SDCA is to be charged at the
rate of one pulse every 180 seconds.
• Calls between the Exchanges in SDCAs Belonging to the same

LDCA or Adjacent LDCAs:
The trunk calls (manual or STD) between exchanges in
different SDCAs situated in the same LDCA or adjacent LDCAs are to be
charged on the basis of radial distance between nominated Short
Distance Charging Centre appropriate to the calling and called
exchanges. In case of two exchanges systems whose local areas have a
common border, one call unit per 180 seconds will apply for STD calls.
• Calls between the Exchanges in Non-Adjacent LDCAs:

Trunk calls (manual or STD) between exchanges in non-
adjacent LDCAs are to be charged on the basis of radial distance between
the Long Distance Charging Centres appropriate to each exchange
subject to a minimum of 51 Kms.
• Charging of Local Calls.

The local calls emanating from electronic exchange in all
telephone systems with an equipped capacity of 30,000 Lines or more are
to be metered at the rate of one call unit every five minutes. The local
calls in non-electronic exchanges of such systems or Local calls in
telephone systems less than 30,000 lines will continue to be unit fee
untimed.
1.6 Advantage of the Revised Charging Plan.

• The revised charging plan will once for all define the charging areas
in the country, which will not undergo any change thereafter.
• Whenever a new exchange will be opened, for the calculation of
charges from this exchange to other exchanges, only identification
of the SDCA of this new exchange will be required.
• The telephone exchanges located in the periphery of bigger cities
like Delhi, Mumbai, Chennai, and Pune etc. demand for the
inclusion of their exchanges within the local area of the bigger
systems on account of present high tariff of one unit for 36
seconds. The introduction of three minutes pulse per unit call will
benefit these peripheral exchanges also and their demand for
inclusion of these exchanges within the local area of the bigger
systems will reduce.
TC31009D10 251
2. National Switching Plan.
2.1 General
The objective of the National Traffic Routing and Switching Plan is
to decide the structure of the network and also to define the manner of
traffic routing. The first Plan was defined in 1965, which was
subsequently revised in 1972 and 1987.
2.2 Switching Plan (PSTN).

There will be 3-level TAX hierarchy for 321 TAX switches located at
each Secondary Switch Centre (SSC). Level of a particular TAX is
decided based on the hierarchy plan as well as the traffic emanating from
the area served by TAX concerned. A terminal exchange will preferably
be directly connected to the parent TAX switch. But in some cases it
may be connected to the TAX through a Tandem exchange.
• 21 level I TAXs will be completely mesh connected. Overflow of
traffic through other Level I TAXs (without circulating traffic) is to
be provided.
• 160 Level II TAXs will be connected to the respective Level I TAXs
as a backbone route. The level III TAXs, which were shown directly,
parented to level I TAX stations and is being redesignated as Level
II TAXs and hence the number of Level II TAXs has increased.
• 140 Level II TAXs will be connected to the respective Level II TAXs
as a backbone route.
• Direct (or High usage) route will be opened from one TAX to
another TAX (other than the backbone route) if the total traffic
(outgoing and incoming) exceeds 12 Erlang for digital route and 8
erlangs for analog route.
• Tandem exchange may be provided in a large multi-exchange area
to cater to mostly local calls. Terminal exchange preferably will be
directly connected to the TAX for STD calls. However, small
terminal exchanges any be connected to the TAX through a
Tandem exchange.
• Rural Tandem exchanges may be provided in SDCAs with more
than four exchanges to cater to intra SDCA calls and also STD
calls via TAX. In other SDCA, terminal exchanges may be parented
to the TAX directly. Exchanges of more than 512 lines may be
directly parented to the TAX switch in the SSA.
• Terminal exchanges will be parented to Level I.II or level III TAXs.
RLU / RSU/ Concentrator are to be treated as part of the terminal
exchange.
• Parenting of Local Exchanges to more than 1 TAX would purely be
governed by traffic and reliability consideration.
TC31009D10 252
• For international traffic each Level I TAX will be connected to at

least one international Gateway Switch. For that Level I TAX
where (outgoing and incoming) traffic is high (at least 30 erlangs),
these will be connected to two Gateway Switch.
• Level II/III will be connected to Gateway Switch if the traffic
(outgoing and incoming) is more than 15 erlangs.’
• All TAXs and Tandem exchanges should be 4 W switches.
At some locations, the requirement of TAX capacity makes it

necessary to use two or more TAXs due to limitations of the TAX switches
presently available. It is desirable to have larger capacity TAX so that
multiple TAXs in same location are avoided. In the interim period if more
that one TAX is used in a location, their configuration and use (e.g.
functional separation, geographical usage separation or integrated
approach) may have to be decided on case by case basis with due
consideration of economics and security.
Merits and demerits of an integrated TAX vis a vis the functionally

separate TAXs of various levels at a particular location was examined
and it was found that this is an involved exercise and deserves a
separate study on the subject, hence, it was decided that an additional
study can be taken up for studying the merits of an integrated TAX
completely meshed with other level I TAX vis a vis the functionally
separator TAX of various categories of various level at a particular
location. The question of merger of level II and level III TAX can also
similarly be referred as terms of reference to the future study indicated
above.
3. National Routing Plan (PSTN)
The principle of traffic routing showing choice of direct (or high

usage) route, alternate route and backbone route is shown in Fig 16.2. In
outgoing chain, direct route to the lowest chain where available is the
first choice. This is followed by direct route to the next higher level in the
terminating chain and so forth. And the final choice is the backbone
route.
TC31009D10 253
Fig: 16.2 Traffic

Routing
LEGEND: I - Level 1 TAX : II - Level 2 TAX : III - Level 3 TAX
TE --- Terminal Exchange T --- Tandem Exchange

Backbone Routes ______ Direct (or High Usage Route)
Example of Call between A4 and B4

First A4-A3-B3-B4
Choice
Second A4-A3-B2-B2-B4
Choice
Third Choice A4-A3-B1-B2-B3-B4
Fourth A4-A3-A2-B3-B4
Choice
Fifth Choice A4-A3-A2-B2-B3-B4
Sixth Choice A4-A3-A2-B1-B2-B3-B4
Seventh A4-A3-A2-A1-B3-B4
Choice
Eighth Choice A4-A3-A2-A1-B2-B3-B4
Final Choice A4-A3-A2-A1-B1-B2-B3-
B4
(Backbone Route)
Note: Many of these routes may not be
available.
TC31009D10 254
The grade of service will be 1: 200 for backbone route for each
switching stage. High usage routes will be dimensioned according to
economic criterion.
Calls from a Terminal exchange should be routed to the TAX
directly or through a Tandem exchange it is preferable to have direct
route to the TAX except in
• Small exchanges in large multi-exchange area.
• SDCA with large number of small exchanges.
The grade of service for connection between a terminal exchange
and TAX/ Tandem will be 1: 200.
The grade of service for connection between a Tandem and TAX will
be 1:200. Traffic between terminal exchanges in a multi-exchange area
will be routed direct where economically possible. In other cases, the
traffic will be routed through a Tandem Exchange. Traffic between small
terminal exchanges in SDCA will be routed through the Rural Tandem
Exchange located at SDCC or the TAX located at SCC. Traffic from the
Cellular will be routed via Level TAXs.
International traffic to neighboring countries will flow through
designated Level I TAX. International traffic to other countries will be
sent from Level I and other TAXs connected to the International Gateway
Switch. International traffic from other TAXs will flow through the
backbone route to Level I (or Level 2 TAX if connected to Gateway Switch)
and then to the International Gateway Switch. Similarly, incoming
international traffic will also flow to level and other TAXs connected to
Gateway Switch and through these to other TAXs. For high traffic
countries international traffic from those TAXs, which are connected to
two Gateway switches, may be routed to a designated Gateway Switch in
consultation with VSNL. Overflow of such traffic will go to the other
Gateway Switch.
3.1 Switching / Routing for Mobile Communication

For Routing of calls meant for Mobile Communication Digital
Tandem switches will be receiving the calls for Mobile Switching Centre
(MSC) from either the local exchanges or distant TAXs. All incoming
trunk groups from MSC and PSTN in the Digital Tandem switches will be
required to be put under detailed observation.
Following routes are identified for interworking between PSTN and
MSC (PLMN.)
1. Traffic PSTN (Local) to PLMN (Local)
Fig. 16.3 indicates that after analysis of ’98 PQR from S x S,
Xbar and Digital Exchanges, call is routed to a Digital Tandem switch
and then it routes the same with all 9 digits to the local MSC provided
PQR is the valid code for the local network under consideration. For
other invalid PQRs, NU tone will be fed.
TC31009D10 255
2. Traffic between PSTN (Local) and PLMN (Distant)

Fig. 16.4 indicates that after analysis of ‘0’ in the local
exchanges, calls are routed to the superior TAX. The same after
analyzing 98PQR will route the call to distant TAX and the same will
switch it to the desired MSC through distant Digital Tandem switch.
3. Traffic between PLMN and PLMN Subscribers (Same City)

PLMN to PLMN subscribers call gets routed through MSC
itself if more than 1 operator is there in the same city both MSCs are
required to be linked up Digital Tandem will not be required for routing
purpose.
4. Traffic between PLMN and PSTN (Local)

Fig. 16.5 indicates that Local tandem on analysis of dialed
digits other than ‘0’ and 98 routes the call to the desire PSTN exchange.
5. Traffic between PLMN and PSTN (Distant)

Fig. 16.5 indicates that MSC (PLMN Exchange) traffic for
distant PSTN station gets routed through Local Digital Tandem and Local
TAX for all codes starting with ‘0’ Appropriate distant TAX and distant
PSTN exchange are also then involved in the switching of the call.
6. Traffic between PLMN (Local) and PLMN (Distant)

Fig. 16.6 indicates that MSC traffic for distant gets MSC
routed through Digital Tandem Switches at both ends and also Digital
TAXs if necessary for all codes starting with ‘098PQR’.
Fig: 16.3 Traffic between PSTN (Local) and PLMN (Local)
TC31009D10 256
Fig: 16.4 Traffic between PSTN (Local) and PLMN (Distant)
Fig: 16.5 Traffic between PLMN and PSTN (Local) & PSTN (Distant)
TC31009D10 257
Fig: 16.6 Traffic from PLMN (Local) and PLMN (Distant)
4. Transmission Plan
4.1 General
The objective of the Transmission Plan is to define the required
transmission quality of the telecommunication network. The National
Transmission Plan was earlier issued in 1971-72. A revised
Transmission Plan was issued in 1988. The transmission plan (1988) is
not very explicit on some points. Switching Plan has also been changed
which affects some aspects of Transmission Plan. In addition, it is also
possible to improve upon some of the parameters so that the
Transmission Plan can meet the CCITT objectives. The new
Transmission Plan is given the following paragraphs.
The Loss Plan, Noise Plan and Error Performance Plan are the
most critical elements of the transmission Plan.
4.2 Transmission Loss Plan

Telephone Instrument should meet the following Reference
Equivalents.
Minimum SRE (send reference equivalent) 3 dB
Minimum ORE (Overall reference equivalent) 2 dB
Maximum ORE 13 dB
Junction from local Exchange to TAXs should be on digital

transmission media (4W). Junction loss (Loss from output 2W local
Exchange analog switches or loss from 2W analog subscriber interface in
case of digital local exchange to input of TAX switch) should be 3.5 dB.
TC31009D10 258
Junction from tandem exchange to TAX / Tandem should be on

transmission media (4W) with 0 dB loss. The maximum Junction loss
between two local exchanges (for local calls) should not exceed 9 dB. The
maximum Junction loss between local exchange and tandem exchange
(including hybrid loss) should not exceed 4.5 dB.
Local PCM system should be lined for following losses.
2W switch to 2 switch 3 dB
2W switch to 4 W switch 3.5 dB
Analog long distance circuits should be lined up for 4W to 4W with
0 dB loss.
The loss between MUX center and local exchange / TAX should not
exceed 1 dB (preferably they should be collocated). Relative level of
channel at 4W points is – 3.5 dB. The traffic weighted mean ORE for the
connection with these will be 14-16 dB and will meet the CCITT
objectives.
Fig. 16.7 Transmission Loss Plans
TC31009D10 259
4.3 Transmission Noise Plan

The following noise limits for analog circuits should be used:
Distance Commissioning Maintenance Usable

Limit (dBmop) Limit (dBmop) Limit (dBmop)
a) Terrestrial Circuits
1 – 160 Km -68 -55 -52
161 – 320 km -55 -52 -49
321 – 640 km -53 -50 -47
641 – 1000 km -51 -48 -45
1001 – 2500 km -49 -46 -43
2501 – 5000 km -46 -43 -41
b) Satellite Circuit
Maintenance -50 -48 -46
Others -47 -45 -43
4.4 Error Performance Plan (Digital System)

The digital transmission will have following three grades for
different applications.
• High grade – Above level TAXs (Digital Microwave, Coaxial and
Optical Fiber)
• Medium grade – Between level I TAX and local exchanges (Digital
Microwave Coaxial, Optical Fiber, cable PCM and UHF).
• Local grade – Between local exchange and subscriber (Digital
subscriber, VHF and subscriber PCM)
The reference performance objective, commissioning limits and
maintenance limits 64 kb/s digital section should be met for different
grade of circuits. At other measuring bit rates DM and SES are almost
same but ES may be correction as given by CCITT Recommendation G.
821 (Annexure 3.3).
4.5 Transmission Loss Plan

The CCITT now has given recommendations for the loss plan in
terms of loudness rating (LR). The corresponding reference equivalent
(RE) is also indicated by CCITT. For telephones used in our network,
information is available in terms of RE only and information in terms of
LR is not available. Hence, the present transmission plan is with
reference to RE.
The considerations for required assignment of loss values in
different parts of network are:
Over all reference equivalent of the connection should not be very
high or should not be very low CCITTs recommendations for ORE are:
• Long term objectives of 13-18 dB traffic weighted mean and
• Short term objectives 13-23 dB weighted mean
TC31009D10 260
The maximum value of ORE is recommended as 34 dB. CCITT has

recommendations that the maximum and minimum of SRE (send
reference equivalent) up to send virtual switching point (-3.5dB) should
not be lower than 9.6 dB and 6.dB respectively. This is to prevent
overloading of the transmission systems. A minimum of 3 dB losses can
be accepted between local exchange and 3-5 dBr points. Thus the
minimum SRE of telephone instrument should be at least 3dB. For cross
talk and echo consideration the minimum ORE of the subscriber circuit
should not be less than 2 dB.
The loss assignment should be such that the probability of singing
or hollowness is very low (S…. occurs when loop loss gets reduce below
0.dB. Hollowness occurs when loop loss gets 3 dB.) Singing with one end
open occurs during switching and fault conditions and its probability is
much less). The stability of the channel and the return loss at the 2/4W
point contributes to this stability. For long connection echo level to be
controlled. The CCITT has given recommendations in terms of echo
reference equivalent for different round trip delay. In mixed analog and
digital network the stability of analog and digital channels are different
and needs to be taken in consideration.
The following points may be noted.
Super Group Regulating Equipment (SGPR) has been introduced in
the Indian network. With use of SGPR the stability of the channel is
likely to meet the CCITTs recommendations of 1 dB standard deviation.
The stability of channel derived from the digital medium will be
much better. Each A to D converter is likely to give stability with a
standard deviation of 0.4dB.
Very few measurements on balance return loss has been done at
the 2W/4W point. Limited measurements indicate that the balance
return loss can be improved by changing balancing network from 600
ohms to 900 ohms. The existing value may have a mean of dB with
standard deviation of 1.5dB.
The above ranges of permissible noise in a connection when
converted to pwop/Km may appear too large compared to the 2 to 3
pwop/Km quoted as design limits for the line noise of coaxial and
microwave systems. However, it is interesting to note that the results of
measurements made by other administrations on actual international
connections reveal similar figures.
The margin against misoperation of signaling receivers with these
noise limits would be better than 18 dB (with MFC receiver sensitivity
changed to 29 dBm against 36 dBm originally). The domestic
communication satellite is planned to have commissioning limits of -50
dBmop for circuits between level I Tax and -47 dBmop for other circuits
(e.g. main to primary and main to remote). The variations on such
circuits are much smaller and these would give noise in connections
similar to those of terrestrial circuits.
TC31009D10 261
Fixed services to remote and rural areas (mostly VHF and MARR)
are being planned with noise values ranging from -46 to -50 dBmo.
4.6 Error Performance

The Hypothetical Reference Circuit (HRX) of 27500 km is divided
into 25000 km of High Grade and 1250 km of Medium Grade at each end
and two local grades at each and International Model link. The
performance objective of HRX (in percentage) is:
DM 10
ES 8
SES 0.1 + 0.1 for propagation.
Note: Errored Seconds (ES) is a second with at least one defect or
anomaly.
Severely Errored Seconds (SES) is a second with Binary Error Ratio
(BER) of 10-3 or worse.
Degraded Minutes (DM) is a group of 60 consecutive seconds after
excluding SES with a BER of 10-6 or worse.
The objectives are allocated as:
High Grade 40%
Medium Grade 15%
(Each end)
Local Grade 15%
(Each end) (Propagation margin is not given for Local Grade
System as per CCIR)
5. National Numbering Plan
5.1 Introduction
The last few years have seen tremendous growth all around and
particularly in the field of cellular mobile services. In some of the
countries, these services have already exceeded the traditional basic
services. In India too, the cellular mobile services have seen a growth of
almost 100% during the last one year. Further, the existing Numbering
Plan was meant to address monopolistic environment in national and
international long distance dialing. The Government of India has since
introduced unlimited competition in basic, National Long Distance (NLD)
and International Long Distance (ILD) Services and licensed four service
providers in respect of cellular mobile services in most of the licensed
service areas. As such, it was felt to review the existing Numbering Plan
and to formulate a plan, which will be futuristic, flexible and could cater
to the numbering needs for about next 30 years in respect of the existing
and likely new services. Keeping this in view, the new Numbering Plan
has been formulated for a projected forecast of 50% tele-density by the
year 2030 and thus making numbering space available for 75 crore
telephone connections in the country comprising of 30 crore basic & 45
crore cellular mobile connections.
TC31009D10 262
5.2 Objectives of the plan

• To plan in conformity with relevant and applicable ITU standards
to the extent possible.
• To meet the challenges of the changing telecom environment.
• To reserve numbering capacity to meet the undefined future needs.
• To support effective competition by fair access to numbering
resources.
• To meet subscriber needs for a meaningful and user-friendly
scheme.
• To standardise number length wherever practical.
• To keep the changes in the existing scheme to the minimum.
Only the decimal character set 0-9 has been used for all number
allocations. Letters and other non-decimal characters shall not form part
of the National (Significant) Number [N(S)N]. Dialing procedure as per ITU
Recommendation E.164 has been followed. The Short Distance Charging
Area (SDCA) based linked numbering scheme with 10-digit N(S)N has
been followed. This would expand the existing numbering capacity to ten
times.
5.3 Salient features of the National Numbering Plan

• It is a SDCA based linked numbering scheme.
• N(S)N is 10-digit for both the basic as well as cellular mobile
services.
• The Subscriber Number (SN) for basic services will be of 6, 7 or 8
digits depending upon the length of SDCA code.
• Basic to cellular mobile service calls shall use prefix ‘0’ only if Point
of Interconnect (POI) is not available in the same Long Distance
Charging Area (LDCA) from where the call is originated.
• Cellular mobile using ‘0’ shall access basic services.
• There is no change in the cellular mobile numbering structure.
• Levels 0, 1, 7, 8 and 9 shall not be used as first digit for telephone
exchange codes in basic services.
• There is no change in the numbering structure for paging services.
• Carrier Access Code (CAC) for NLD and ILD has been defined as
‘10’.
• Separate Carrier Identification Codes (CIC) has been earmarked for
toll and non-toll quality NLD and ILD services.
• All the service providers shall use ‘100’, ‘101’ and ‘102’ for Police,
Fire and Ambulance services respectively.
• ‘107X’ has been defined for emergency information services like
earthquake, floods, air and train accident etc.
TC31009D10 263
• Intelligent Network service access codes on ‘16XX’ are shifted to

‘18XX’.
• Trunk services codes are shifted from ‘18X’ to ‘150X’.
• Certain level ‘1’ codes are earmarked for all service providers to
offer various subscriber related services, as per their choice, within
their network.
• Voice Mail Service (VMS) access code is shifted from ‘93’ to ‘170’.
• Enough spare levels/ codes are reserved for future needs.
5.4 Planning Period

A Planning period of 50 years has been assumed for the National
Number Plan presented here.
5.5 Population Growth

Based on past growth, the population is expected to increase at a
rate of about 2% every year. The National Numbering Plan is based on a
Growth Factor of 2.5% for population growth during the 50-year period.
5.6 Telephone Growth

It has been assumed that the National Numbering Plan should
permit minimum telephone densities of 50% in large urban areas and
10% in non-urban areas, though in practice all areas may not develop
equally or to those limits.
5.7 Local Charge Area

As per present practice, charging distance for unit fee is 20
kilometer and Local Charge (Unit Fee) Area approximately 400 Sq. Kms.
Each such Local Charge Areas of 400 Sq. Kms. has to be distinguished
by a unique National Code.
5.8 Number Length

The theoretical capacity of an 8 digit-numbering scheme is 100
million telephones and that of a 9 digit numbering scheme 1,000 million
telephones. In actual practice, because of normal wastage in numbering
scheme, necessary for leaving spare codes for specific purpose (such as
digit 00’ for international trunk working, digit 01’ in local numbering
scheme for special services, etc) the utilization of the total capacity of any
Numbering Plan would be below 50%. Therefore a nine digit-numbering
scheme has been adopted.
5.9 National Numbering Plan

In brief, the National Numbering Plan allots.
(a) 4 two digit (AB) Codes for the major telephone systems at
Mumbai, Kolkata, Delhi and Chennai.
TC31009D10 264
(b) 137 three-digit (ABC) codes for systems with present day
telephone potential (working connections PLUS waiting list) of
500 and over.
(c) 314 three-digit (ABC) codes for Group areas up to 25,000 sq.
Kms. areas and or two million populations, whichever is higher.
5.10 2-Digit (AB) Codes

2- digit (AB) codes have been allotted to Mumbai, Kolkata, Delhi
and Chennai, which would enable the telephone systems in each of these
large metropolitan areas to develop up to 5 million, telephones each.
“ABC” codes with capacity of only 0.5 million would not be sufficient for
these areas. A-digit codes “1” “2” “3” and “4” have been allotted to Delhi,
Mumbai Kolkata and Chennai regions respectively, and accordingly
following “AB” codes have been allotted to these metropolis for simplicity
in remember double-digit numbers.
11 Delhi
22 Mumbai
33 Kolkata
44 Chennai
5.11 3-digit (ABC) codes for other Large telephone system

To other large telephone systems with over 500 telephone potential
today, i.e., working connections PLUS waiting list of over 500, one more
3-digit (ABC) codes have been allotted, ensuring that the ultimate date,
the codes now allotted should permit a telephone density of at least 50%.
In cases where with one 3-digit (ABC) code, the number of telephone
would be restricted, to below 50% telephone density at the end of plan
period, a second (or even a third 3 digit (ABC) code has been allotted,
thus doubling (or trebling) the telephone potential of the area. Such
exchange areas would be of the National code at a future data in the
sense that the 3 digit (ABC) national code would become a 2 digit (AB)
national code by absorption of the last digit, i.e., the ‘C’ digit, of the
national code into the local numbering scheme.
5.12 3-Digit (ABC) Code for group Areas

One 3-digit ABC code has been allotted to each Group area with up
to 25,000 sq. Kms. areas or 2,000,000 populations at present, whichever
is higher.
In a group Area of up to 25,000 Sq. Kms. there can be up to 64 charging
areas of 400 Sq. Kms. Theoretically, on each ABC code there can be 100
ABCDE codes. Therefore, even 64% utilization of such codes would cover
all the charging areas. In practice, the Group areas are generally much
smaller than 25,000 Sq. Kms. and therefore, the availability of National
codes for the individual charging areas would be more than adequate.
In some very special cases, such as the desert districts of Rajastan
and Ladakk, Kashmir and NEFA, one 3-digit ABC code has been allotted
TC31009D10 265
for an area of more than 25,000 Sq. Kms. It has been assumed that in
cases of such very sparsely populated areas, the limit of 20 Kms. for unit
fee calls would itself need relaxation and a much higher distance limit for
unit fee areas would be adopted, thus enabling 65 codes or less to be
sufficient even for that Group Area.
The limit of 2 million population for a Group Area (which excludes
population of any large station within the Group Area which has been
allotted separate National Code) enables each Group Area to expand up
to 500, 000 telephones giving an ultimate date telephone density of 10%
or over. Where the present population in the proposed Group Area is
beyond 2 million or the area is more than 25,000 Sq. Kms., a second
ABC code has also been allotted. In such cases, when doing detailed
allocation of ABCDE National Codes to individual local charging areas
within each Group Area, Group Area would be first divided into two sub-
Group Areas, each with a separate 3-digit (ABC) code of its own.
5.13 Allocation of Codes

With 100% utilization of all 10 digits ‘1’ to ‘0’ as ‘A’ ‘B’ and ‘C’ code
digits, it would be possible to make available 1000 ABC codes. The
actual requirements for ABC codes are much less and therefore only the
following have been used as ‘A’ ‘B’ and ‘C’ digits.
‘A’ digit : 2,3,4,5,6,7 and 8
‘B’ digit : 2,3,4,5,6,7,8 and 9
‘C’ digit : 2,3,4,5,6,7,8 and 9
This leaves ‘9’ as spare ‘A’ digit for future allocation either within
India by rearrangement or to adjacent countries, if this becomes feasible
at a future date ‘0’ is also available spare as ‘B’ and ‘C’ digit for future
allotment.
The Country has been divided into 8 ‘A’ digit areas or region as follows.
Digit ‘1’ Punjab, J & K., U.P.(Part) and Rajasthan (Part).
Digit ‘2’ Maharashtra (Part), Gujarat and Rajasthan (Part)
Digit ‘3’ West Bengal, Assam, NEFA, Sikkim and A & N Islands.
Digit ‘4’ Chennai, Kerala and Lacadive and Minicoy Islands.
Digit ‘5’ U.P (Part)
Digit ‘6’ Bihar and Orissa.
Digit ‘7’ Madhya Pradesh, Maharashtra (Part) and Rajasthan
(Part).
Digit ‘8’ Mysore and Andhra Pradesh.
Nine Regions
1. Delhi 4. Chennai 7. Gauhati
2. Mumbai 5. Jullender 8. Bangalore
3. Kolkata 6. Ahmedabad 9. Nagpur
Each ‘A’ digit area is sub-divided into nine ‘AB’ areas each ‘AB’
area comprising up to 9 continuous ‘ABC’ Group Areas. In grouping of
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‘ABC’ areas into ‘AB’ areas, Telephonic community of interest as well as

present and proposed major trunk routing have been kept in view.
The delineation of the Group area is present based on the political
district boundaries and are tentative at this stage. They would be
confirmed after delineating individual charging areas within each Group
are for allocation of ‘DE’ codes.
5.14 National Numbering Scheme
The prefix ‘000’ shall be used for home country direct service
(Bilateral) and international toll free service (Bilateral).
The format used is: ‘000 + Country Code + Operator Code’ except
‘000800’ which is used for bilateral international toll free service.
5.15 Sub level ‘00’ - International Prefix:
The prefix ‘00’ shall be used for International dialing. It will be

followed by country code and the N(S)N of the country to which that call
is attempted. The format is as per ITU Recommendation E.164:
Prefix Country Code National (Significant) Number

00 CC N(S)N
5.16 Sub level ‘010’ - National Carrier Access (Prefix) Code:
The prefix ‘010’ shall be used for selection of national long distance
carrier. It will be followed by (National) Carrier Identification Code (CIC)
and N(S)N. The format shall be as under:
Prefix Carrier Identification National (Significant)

Code Number
010 CIC N(S)N
Initially CIC shall be a two-digit code. This will be sufficient for

allotment to 40 NLDOs (including NLDOs licensed for basic services) and
10 BSOs licensed only for basic services, considering that maximum of
two codes may be allotted to each service provider depending upon toll
quality and non-toll quality network. However to take care of all possible
future requirements, length of CIC may be reviewed and changed to 3-
digit code in future. The allotment of CIC may start from ‘10’ and codes
‘00’ to ‘09’ may be kept reserved.
For intra circle long distance service, the carrier access code shall
be the same as applicable for NLD service. The CIC from ‘10’ to ‘79’ shall
be allotted to NLD service providers. For the NLD service providers, who
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are also Basic Service Operators (BSOs), same CIC shall be applicable for
intra circle (service area) calls.
CIC from ‘80’ to ‘99’ shall be allocated to the BSOs who are not
licensed to provide NLD service.
5.17 Sub level ‘0’ - National Prefix:
The prefix ‘0’ shall be used for national long distance calls (cellular
mobile as well as basic services), intra service area (Circle) long distance
calls of basic services, cellular mobile to basic services calls and calls
from basic services to cellular mobile (depending upon point of
interconnect). The format shall be as under:
• For basic services (PSTN) long distance calls:
Prefix National Destination Code Subscriber Number

0 2/ 3/ 4-digit trunk (SDCA) 8/ 7/ 6-digit
code
• For basic services to cellular mobile calls if Point of Interconnect is

not available in the same LDCA from where the call is originated:
Prefix PLMN Access MSC Code Subscriber Number

Code
0 2-digit 3-digit 5-digit
e.g. 98, 94 etc.
• For cellular mobile to cellular mobile calls outside the service area
from where the call is originated:
Prefix PLMN Access MSC Code Subscriber Number

Code
0 2-digit 3-digit 5-digit

e.g. 98, 94 etc
• For cellular mobile to basic services calls:
Prefix National Destination Code Subscriber Number

0 2/ 3/ 4-digit trunk (SDCA) 8/ 7/ 6-digit
code
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5.18 Sub level ‘011’ to ‘089’ - Geographical Number Range:
These codes are also called trunk codes and identify a specified
geographical area where a call is to be terminated. The national telecom
network in India has been divided as under:
• SDCA: Short Distance Charging Area also called local area
• LDCA: Long Distance Charging Area comprising of one or
several SDCAs
ITU Recommendation E.164 provides four options for National
Destination Code (NDC) structure. India has adopted type-2 structure for
PSTN where NDC is the trunk (Area) code assigned to each SDCA. Each
SDCA is allotted a unique trunk code. There are at present 2645 SDCAs
distributed in 322 LDCAs.
Accordingly, 2645 codes are required to identify the complete
country based on SDCA linked numbering scheme. The length of the
Trunk Code (TC) shall vary from 2 to 4 digits depending upon the size
and telephone density requirement of the SDCA. Details regarding SDCA
linked numbering scheme for PSTN, SDCA trunk codes (geographical)
and the spare 2/ 3/ 4-digit codes are given at Annex-I, II and III
respectively.
5.19 Level ‘1’ Special Services:

Level ‘1’ is used for accessing special services like emergency
services, supplementary services, inquiry and operator-assisted services.
Some sub levels have been allocated for use by access providers
(operators). These levels can be used for providing the services within
their network.
5.20 Level ‘2’ to ‘8’ - PSTN Number:

The numbers starting from ‘2’ to ‘8’ are reserved for PSTN within
SDCA. At present the PSTN numbers in SDCAs vary between 5 to 7 digits
so that the total N(S)N is 9-digit, except in a few cases where it is already
10-digit. It has been observed that the total telephone number
requirement in some SDCAs with 5 digits numbers would be more than
50,000 in next 25 to 30 years. Similarly, in many SDCAs with 6 digits
local number length, projected demand is likely to go beyond 5,00,000.
Therefore, the subscriber number in SDCAs is increased by one digit so
that the N(S)N is uniform and of 10-digit length.
5.21 Level ‘9’ Services:
The range of numbers in level ‘9’ except ‘90’, ‘95’ and ‘96’ are
reserved for cellular mobile services. These will be allocated in 2-digit
format.
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5.22 Access to adjacent Area

‘95’ shall be used for accessing adjacent areas in the following
format:
95 SDCA Code Subscriber Number
Table of National Numbering Scheme

Number
/Prefix Services Structure Remarks
Sub
level
0 0 National Long 11-digit number As per ITU
Distance Service ‘0+NDC+SN’ Rec. E.164
NDC+SN is also called as
N(S)N
11XX National NDC=2 to 4-digit
to Destination Code Also known as SDCA Code
89XX
range
0900 IN Service - ‘0900+XXX + IN Number’ ‘xxx’ to be
Premium Rate Where ‘XXX’ is SCP code allocated to IN
from ‘000’ to ‘999’ provider
0901 IN Service - ‘0901+XXX + IN Number’ ‘xxx’ to be

Universal where ‘XXX’ is SCP code allocated to IN
Number (Long from ‘000’ to ‘999’ provider
Distance)
0902 UPT Service Not allocated1

0903 Reserved for IN Not allocated1
to Services
091 to Reserved for Not allocated1
093 PLMN
094 For dialling 11-digit number .
PLMN ‘0+94+MSC Code+SN'
MSC Code=’000’ to ‘999’,
SN=5 digits
0950 Reserved for ‘0950+XX+SN’
Mobile Satellite ‘XX’= service provider
Services
0951 Reserved for Not allocated1
Mobile Satellite
Service
0952 HVNET ‘0952+X+SN’
X is service provider
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Number
/Prefix
0953 INMARSAT Mini- ‘0953+X+SN’
M ‘X’ is service provider
0954 Digital Satellite ‘0954+X+SN’
Phone Terminal ‘X’ is service provider
0955 Reserved Not allocated1
~
096 Spare Not allocated2
PLMN
098 For dialling 11-digit Number
PLMN ‘0+98+MSC code+ SN
MSC code =’000’ to ‘999’
SN= 5 digits
PLMN
1 Special Services3 to N digits depending on
service.
2 to 8 PSTN Subscriber SN= 6 to 8 digits
Number
9 Services
90 Spare Not allocated2
91 to Reserved for Not allocated1
93 PLMN
PLMN ‘94+MSC code+ SN’
MSC code =’000’ to ‘999’
SN= 5 digits
95 Access to ‘95+SDCA code+ SN’
adjacent SDCA
96 Paging Service 10-digit ‘XY’= ‘00’ to
‘96+XY+ Pager Number’ ‘99’ to be
‘XY’ is service provider allocated.
code Accessible
Where ‘XY’= ‘00’ to ‘99’ from Outside
Service area.
PLMN
PLMN ‘98+MSC code+ SN’
MSC code =’000’ to ‘999’
SN= 5 digits
PLMN
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5.23 Numbering Format
The PSTN Numbering format shall be as per the table 1:
Trunk Code + Telephone + Last n digits of

(SDCA code) Exchange Code Subscriber Number
ABCD + EF + PQRS
ABCD + EFG + PQR
ABC + EF + PQRST
ABC + EFG + PQRS
AB + EFG + PQRST
AB + EFGH + PQRS
• Digit A can have any value from 1 to 8.

• Digit B, C and D can have any value between 0 and 9.
• Digit E can have any value between 2 to 8. [0, 1, 9 are NOT
allowed, since level ‘0’ is used as trunk prefix, level ‘1’ is used for
special services, levels ‘7’ & ‘8’ are kept spare for future services
and level ’9’ is used for cellular mobile services, paging services &
access to adjacent areas.]
• Digit F, G and H can have any value from 0 to 9.
• Digit P, Q, R, S and T can have any value from 0 to 9.
Dialing
• Dialing within SDCA

For a call within a local area i.e. SDCA, subscriber number only
will need to be dialed. The number of dialed digits will thus be 6, 7 or 8.
• Dialing outside SDCA

For calls outside the SDCA, ‘0+N(S)N’ as per ITU E.164 or
‘0+10+CIC+N(S)N’ needs to be dialed. However, access to adjacent areas
can also be on level ‘95’ followed by N(S)N. All such adjacent areas shall
have to be accessible by dialing N(S)N with ‘0’ prefix or through carrier
selection procedure.
5.24 General Guidelines

• The allocation of SDCA codes in the SDCA based linked numbering
scheme shall be as per Annex-II.
• Spare codes, which are not allocated to any SDCA at present, are
listed at Annex-III.
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• Two digits spare codes are reserved for allocation to those SDCAs
in which numbering requirements may become more than 40
Lakhs in the long run.
• Certain three digit spare codes like 555, 666 and 888 are not to be
used as SDCA codes. These are reserved for future services as
indicated in Annex-III.
• 10-digit N(S)N is considered sufficient for more than 30 years.
5.25 General requirements for migration

• First digit of the telephone exchange codes can have any value
between ‘2’ to ‘6’. Presently ‘2’, ‘3’ and ‘5’ are being used. ‘4’ and ‘6’
have been reserved for future use.
• Digit ‘2’ has been allocated as the first digit for BSNL/ MTNL
numbers.
• For the private BSOs, digit ‘3’ or ‘5’ is to be used as first digit.
• First digit in subscriber number should not be ‘0’, ‘1’, ‘8’ and ‘9’.
5.26 Numbers for Special Services
(10) Emergency Services

100 POLICE 3-digit Restricted
number
101 FIRE 3-digit Restricted
number
102 AMBULANCE 3-digit Restricted
number
103 SPARE Not Allocated1
104 SPARE Not Allocated1
116 Wakeup call 3-digit
Registration number
117 Wakeup call 3-digit Restricted
Cancellation number
118 Call waiting 3-digit Restricted
Registration number
119 Call Wailing 3-digit Restricted
Cancellation number
123 Dynamic STD Password 3-digit Restricted
Setting number
124 Class of Service 3-digit Restricted
Registration number
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180 IN Services 180X+YYY+ ‘YYY’ code to

IN number, be allocated
Where
‘X’=IN
service ‘0’ to
‘9’
‘YYY’= SCP
code
5.27 Numbering for Cellular Mobile Network
Allotment of Mobile Switching Centre (MSC), Signaling Point (SP)

and Mobile Network Codes (MNC) for Cellular Operators
The cellular mobile network has been divided into 19 service areas
and 4 metropolitan cities for which licenses were issued. Accordingly
Numbering Plan was formulated for allotment of MSC code for each
operator with spare codes for future allotment.
The format of cellular mobile service Numbering Plan is given below:
ACCESS CODE MOBILE SWITCHING SUBSCRIBER

CENTRE (MSC) CODE NUMBER
2-digit 3-digit 5-digit
98/94 etc. ABC XXXXX
• Presently level ‘98’/’94’ have been allocated as access code for

mobile networks.
• ABC=’000’ to ‘999’. However, ABC=’000’ to ‘099' have not been
allocated and are reserved.
• XXXXX= 00000 to 99999.
Initially in each service area/metro, two operators were given

license to operate. With further opening up of mobile services, two more
operators are permitted to operate in each service area/metro areas.
The Signaling Point (SP) codes for mobile operators are given in
9000/10000 series. The MSC codes and SP codes allotted to the cellular
mobile operators in different circles/metros.
TC31009D10 274
5.28 Guidelines for allotment of MSC codes
• MSC codes shall be allotted to the service providers from the series
from which they are already issued the MSC codes.
• Additional MSC codes shall be allotted only when the subscriber
base of 60% has been achieved with the allotted codes.
• When all the codes are exhausted from the same series, MSC codes
from other series, which are not used or expected to be less used,
may be allocated.
5.29 Mobile Number allotment
All mobile numbers in India have the prefix 9 (This includes pager
services, but the use of pagers is on the decline). Each zone is allowed to
have multiple private operators (earlier it was 2 private + BSNL,
subsequently it was changed to 3 private + BSNL in GSM 900/1800, now
it also includes 2 private + BSNL in CDMA).
All cell phone numbers are 10 digits long, (normally) split up as
OO-AA-NNNNNN where OO is the operator code, AA is the zone code
assigned to the operator, and NNNNNN is the subscriber number.
5.30 Mobile Numbering plan
• 92-xx-yyyyyy - TATA Indicom Numbers

• 93-xx-yyyyyy - Reliance Mobile Numbers
• 94-xx-yyyyyy - BSNL CellOne Numbers
• 97-xx-yyyyyy - Various operators except Reliance, TATA and BSNL
Example: Mumbai
Bharti Airtel - Pre fix: 98 , 99 MNC: 92 , 67 , 87
Vodafone Essar - Pre fix: 98 , 99 MNC : 20 , 19 , 33 , 30
MTNL - Pre fix : 98 , 99 MNC : 69
BPL Mobile - Pre fix: 98 MNC : 21 , 70
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6. Synchronisation Plan
A synchronization plan is the determination of the distribution of
synchronization in a network. It involves the selection and location of a
master clock or clocks, the distribution of primary and secondary timing,
and the selection of the clocks and reference facilities. To achieve the
best performance and most robustness from a synchronization network,
several rules and procedures must be followed when developing a
synchronization plan. Some of the most important rules are avoiding
timing loops, maintaining a hierarchy, following the BITS concept, using
the best facilities for synchronization reference transport, and minimizing
the cascading of the timing reference. Timing loops occur when a clock
uses a timing reference that is traceable to it. When such loops occur,
the reference frequency becomes unstable. The clocks in a timing loop
will swiftly begin to operate at the accuracy of the clock’s pull-in range.
This will result in the clock exhibiting performance many times worse
than it does in free-run or holdover mode. Therefore, it is important that
the flow of timing references in a network be designed such that timing
loops cannot form under any circumstance. No combination of primary
and/or secondary references should result in a timing loop. Timing loops
can always be avoided in a properly planned network.
Slips
If the node clocks in a telecommunication network operate
asynchronously then, transmit and receive rates of telecommunication
systems in each node would be different to the other nodes. In this case,
the input buffers of the telecommunication systems would frequently
overflow or underflow, causing data errors commonly referred to as slips.
The object of network synchronisation is therefore to avoid and to
minimise slips. This can only be achieved by synchronising all the node
clocks, and hence all the telecommunication systems, to the same
master clock or to a number of pseudosynchronous (very closely
matched, nearly synchronous) master clocks. In practice, master clocks
or Primary Reference Clocks (PRCs) are Cesium beam oscillators, and
slave node clocks are usually Ovenised Crystal Quartz Oscillators
(OCXOs).
Table 1: Effect of slips on services
Service Effect of slips
Voice (uncompressed) Only 5% of slips will lead to audible clicks
Voice (compressed) A slip will cause a click
Facsimile A slip can wipe out several lines
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Modem A slip can cause several seconds of drop

out
Compressed video A slip can wipe out several lines. More slips
can freeze frames for several seconds
Encrypted data Slips will reduce transmission throughput

protocol
The slip rate between systems can be calculated by equation

below, and slip rates for 8k frames per second signals under various
frequency differences are shown in Table 2.
Equation 1: Slips per day =
Frequency difference x Traffic frames/second x seconds/day
(86400)
Table 2: Frequency differences and slip rates
Frequency difference Slip rate for 8k frames per

between systems second signals
0 0
10-11 1 slip in 4.8 months
10-10 1 slip in 14.5 days
10-9 1 slip in 1.45 days
10-8 6.9 slips per day
10-7 2.9 slips per hour
10-6 28.8 slips per hour
10-5 4.8 slips per minute
Apart from frequency difference, wander levels that exceed the

input tolerance of telecommunication systems would also cause slips.
Wander is slow modulation of the clock or traffic signals from their ideal
positions in time and very low frequency (mHz) wander is impossible to
filter out in a synchronisation network. Contraction and expansion of
transmission cables under varying temperatures generate very low
frequency wander on the traffic/ synchronisation signals. The levels of
wander generated by optical fiber are lower than the copper cables.
TC31009D10 277
6.1 Maintaining a hierarchy

It is the policy of our department to develop a national integrated
digital network (IDN). To ensure satisfactory performance of the IDN, it
is necessary to accurately control the rate at which digital signals are
transmitted throughout the network. This can be achieved by
synchronizing the clocks controlling digital switches and transmission
systems. Further, Synchronisation of the digital network is an essential
prerequisite to the introduction of high-speed data services and
integrated services digital network (ISDN).
The objective of the Synchronisation plan is to create a wholly
synchronised national network meeting the CCITT recommendations on
controlled slip rate (Recommendation G822).
6.2 Methods of Controlling Slips

(a) By Plesynchronous Operation
(b) By Synchronous Operation
6.3 Plesynchronous Operation

In this method of operation, each exchange has its independent
clock with very high stability i.e. the long-term frequency departure of
1x10E- 11 or more per year. There is no need to establish a
synchronous network of the exchange clocks. The cost of such accurate
clocks is very high and hence this method is no suitable for controlling
slips in the national digital network.
6.4 Synchronous Operation

In this method of operation the synchronisation of nodal clock is
done in a continuous manner. There are two ways of synchronisation:
Master-Slave Method: In this method, there is one master or Primary

Reference Clock (PRC) that supplies the timing reference for the entire
network. Almost all carriers with digital networks rely on Primary
Reference Source (PRS) for timing since ITU recommends that networks
operate with a long-term accuracy of 1×10–11. The PRS times all the
equipment in the location in which it resides. This equipment, in turn,
will time the rest of the network.
The slip rate contribution of a PRS is usually negligible. A network,
which derives timing from two PRS clocks, will experience at most, five
slips per year, caused by the inaccuracy of the two clocks. This is
negligible compared to the performance of receiver clocks. Receiver
clocks typically operate with a daily performance that is 10 to 100 times
worse than the PRS to which they are slaved. Therefore, it has been the
trend of telecommunication network operators to rely more heavily on
PRS clocks and to use multiple PRS clocks to time their network. This
reduces the cascading of timing in the synchronization network.
TC31009D10 278
The locations in which the PRSs are used are determined by

network topology. PRSs are usually placed in locations that will minimize
the cascading of timing in the network. In this manner, the best
performance can be achieved in the network. Additional sites that may
require the use of PRSs are international switching locations. It is at
these locations that one administration interfaces with another and all
signals are transferred plesiochronously. It is important, therefore, to
guarantee that these locations operate with the 1×10–11-frequency
accuracy necessary for plesiochronous operation.
Fig. 16.8
Synchronous operation
through PRS
The digital network is conceived as a hierarchy of digital nodes

arranged in layers or levels. The PRS (Master Clock) supplies timing
reference to nodes at the highest level (i.e. level I TAX), which uses this
reference to phase lock, its own clock. Each node in the network receives
the timing reference from a higher order node. The master clock is of
high standard of accuracy and reliability say with long term frequency
departure of 1x10E-11 per year. This method is shown in Fig. 16.9.
Mutual Synchronisation Method: This is a concept for achieving a

synchronous interconnected digital network without a master. When
mutually synchronised, every exchange clock in the network is locked to
the average of all incoming clock rates. A common system frequency is
thus obtained by forcing a number of clocks to be interdependent on
each other. This method of synchronisation is shown in Fig. 16.10.
TC31009D10 279
Fig. 16.9 Master-Slave Method Fig. 16.10 Mutual Synchronisation
6.5 Proposed Synchronisation Plan

On the basis of technical and economic considerations, the
Committee set up for this purpose has recommended the Master-Slave
method on a hierarchical basis. The details of their recommendations
are given in the following paragraphs.
• National reference clock centres

There will be two National reference clock centres (NRC), one to be
designated as Main National reference clock centre (MNRC) and the
other as the backup NRC (BNRC). The MNRC with normally
control the entire network. In the event of failure of MNRC, the
Back-up National reference clock will take over the
synchronisation function. The MNRC will be at Bombay
established by VSNL. The BNRC will be installed at Delhi,
dedicated and duplicated links will be provided between MNRC and
BNRC.
• Clock Hierarchy
The clock hierarchy for the proposed synchronisation network
closely follows the natural topology of the telecom traffic hierarchy.
This is shown in Fig. 16.11. The MNRC/BNRC forms layer 1, the
level 1 TAXs or layer 2, the level II TAXs form layer 3, the level III
TAXs from layer 4 and local exchanges from layer 5.
TC31009D10 280
Fig. 16.11 Synchronous Hierarchy vs. Traffic Hierarchy
• Synchronisation Equipment
Digital Network Synchronisation Equipment (DNSE) will be
provided at each node i.e. each digital exchange in the network.
The DNSE will be of the same type for all level of exchanges i.e.
layer 2 to 5, to provide economies of scale and permit uniform
maintenance practice. The synchronisation equipment will accept
timing reference over 2 Mb/s PCM links. To account for possible
link failures, a number of links (typically three) can be terminated
with a pre-assigned priority, so that if the priority link 1 fails
reference timing is derived from priority 2 link and so on. The
equipment will phase lock the clock of its node to a timing
reference supplied from another predetermined node. The timing
output obtained from the DNSE will then be supplied to the digital
exchange.
• Synchronisation links
Normally the synchronisation links will from part of the traffic
circuits except for links from the two NRCs to the four metro
centres. These will be dedicated links. The links from the MNRC
will be the main links ‘M’ and those from BNRC the standby links
‘S’.
TC31009D10 281
Both the NRCs will monitor each other’s timing references received
on the duplicated dedicated links between them. In normal operation,
the BNRC will not utilise its own master clock output; instead, it will
divert the received reference from the MNRC on the standby links. Thus
both links ‘M’ and ‘S’ will carry timing reference from MNRC. In the
event of failure of the timing reference of the MNRC the BNRC will feed
its own timing reference on the ‘S’ links. In this case, the MNRC since it
is faulty diverts the received reference from BNRC on the ‘M’ links. This
scheme provides an extremely secure arrangement of feeding primary
timing reference, adequately taking care of both link failures and failure
of an entire NRC.
The synchronisation information shall be passed between clocks in

a strict hierarchical manner. All synchronisation links are unidirectional
i.e. timing information flows in only one direction and only form a higher
level to a lower level or at the same level.
Each exchange in the network will have up to three independent

inputs for improved reliability. These will be independent with respect to
availability.
The layer 2 exchanges will have timing inputs through ‘M’ and ‘S’
links. In addition they may also receive a third reference input from
other layer 2 exchanges. This arrangement is shown in Fig. 16.12.
Note: Only one layer 2 exchange shown in figure

Fig. 16.12 Reference Timing inputs to Layer 2
TC31009D10 282
Each exchange at layer 3 will have up to three reference inputs.

One will be from its own parent layer 2 nodes, second from one of the
non-parent layer 2 node and third will be from one of the designated
layer 3 node. Links will be provided taking diversity of the media into
account. A typical connection is shown in Fig. 16.13.
Ref. timing is from layer 2 to layer 3 and links are provided with media
diversity
Fig. 16.13 Reference inputs to Layer 3 (Example)
Each exchange at layer 4 will have up to three reference inputs –

one form its own parent layer 3 nodes, second form one of the non-
parent layer 3 node and third form one of the layer 4 node designated for
this purpose or from layer 2 node if possible. Links will be provided
taking diversity of the media into account. A typical connection is shown
in Fig. 16.14.
Media Diversity to be ensures

Fig. 16.14 Reference inputs to Layer 4 (Example)
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The layer 5 exchanges (local exchanges) will also have three

reference inputs from exchanges from higher layers or the same layer.
In a multi exchange building, the distribution of synchronisation
information to the exchanges within the building may be by coaxial or
similar cable. One Digital Network Synchronisation equipment can be
used to supply timing to several exchanges.
6.6 Clock Quality
The clock characteristics at different layers as recommended by the

Committee are:
• Layer 1: Primary timing reference standard with a minimum
stability of
1x10E-11 (for the life of caesium beam tube).
• Layer 2: Clock with a minimum stability of 2x10E-10 per day.
• Layer 5: Clock with a minimum stability of 10E-6 per day.
Stringent Synchronization planning must be done for all networks

if performance objectives and service needs are to be met. In carrier
networks, the major focus of synchronization planning lies in the
determination of timing distribution and the selection of clocks and
facilities used to time the network. Careful attention must be made to the
selection of clocks and reference facilities, and to the minimization of
reference cascading. In private networks, the major goal of
synchronization planning is to reduce errors caused and propagated by
poor CPE clocks. This requires limited use of stratum 4 CPE clocks,
limited cascading, use of as many carrier-timing sources as possible, and
the use of BITS architectures.
7. Signaling Plan
The worldwide signaling network has two functionally independent

levels: the international level and the national level. This provides for a
clear division of responsibility for signaling network management and
allows identification plans of signaling points in the international
network and the different national networks to be independent of one
another. Within the International Signaling System No. 7 network, an
International Signaling Point Code (ISPC) identifies a signaling point
while within the National signaling system No.7 network, a signaling
point is identified by a National Signaling Point Code (NSPC).
The Point Code Management function ensures that the current and
future signaling address requirements for the telecommunications
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industry are properly met. The detailed texts and definitions are available
at ITU-T recommendations in Q.70x series recommendations.
7.1 Definitions
• Signaling Point (SP): A node in a signaling network that originates

and receives signaling messages, or transfers signaling messages
from one signaling link to another, or both.
• Signaling relation: An association between two signaling points

that allows interexchange of Signaling System No. 7 messages.
• Signaling Point Code (SPC): A code used to identify a signaling

point and processed within the Message Transfer Part (MTP) of
each signaling point and within users of the MTP.
• International Signaling Point Code (ISPC): A signaling point code

with a unique 14-bit format used at the international level for
signaling message routing and identification of signaling points
involved. The ISPC is used in signaling messages containing the
Network Indicator NI=00.
• Member State: Country" (or geographical area), and/or "Regulator"

shall be considered as Member State.
• Signaling links: Signaling links are basic components in a

signaling network connecting together signaling points. The
signaling links encompass the level two functions, which provide
for message error control (detection and subsequent correction).
7.2 Signaling Links

CCS7 messages are exchanged between network elements over
64kbps or higher speed bi-directional channels called Signaling links.
Signaling occurs out-of-band on dedicated channels rather than in-band
on voice channels. Compared to in-band signaling, out-of-band signaling
provides:
• Faster call setup times (compared to in-band signaling using

multi-frequency (MF) signaling tones)
• More efficient use of voice circuits
• Support for Intelligent Network (IN) services which require
signaling to network elements without voice trunks (e.g., database
systems)
• Improved control over fraudulent network usage
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7.3 Signaling Points

Each signaling point in the CCS7 network is uniquely identified by
a numeric point code. Point codes are carried in signaling messages
exchanged between signaling points to identify the source and
destination of each message. Each signaling point uses a routing table to
select the appropriate signaling path for each message.
There are three kinds of signaling points in the CCS7 network as

shown in Fig. 5.14.
• SSP (Service Switching Point)
• STP (Signal Transfer Point)
• SCP (Service Control Point)
Fig. 16.15 SS7 Signaling Points
SSPs are switches that originate, terminate, or tandem calls. An

SSP sends signaling messages to other SSPs to setup, manage, and
release voice circuits required to complete a call. An SSP may also send a
query message to a centralized database (an SCP) to determine how to
route a call (e.g., a free 800 call). An SCP sends a response to the
originating SSP containing the routing number(s) associated with the
dialed number. An alternate routing number may be used by the SSP if
the primary number is busy or the call is unanswered within a specified
time. Actual call features vary from network to network and from service
to service.
Network traffic between signaling points may be routed via a
packet switch called an STP. An STP routes each incoming message to an
outgoing signaling link based on routing information contained in the
SS7 message. Because it acts as a network hub, an STP provides
improved utilization of the SS7 network by eliminating the need for direct
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links between signaling points. An STP may perform global title

translation, a procedure by which the destination signaling point is
determined from digits present in the signaling message (e.g., the dialed
800 numbers, calling card number, or mobile subscriber identification
number). An STP can also act as a "firewall" to screen SS7 messages
exchanged with other networks.
Because the SS7 network is critical to call processing, SCPs and
STPs are usually deployed in mated pair configurations in separate
physical locations to ensure network-wide service in the event of an
isolated failure. Links between signaling points are also provisioned in
pairs. Traffic is shared across all links in the link set. If one of the links
fails, the signaling traffic is rerouted over another link in the link set. The
SS7 protocol provides both error correction and retransmission
capabilities to allow continued service in the event of signaling point or
link failures.
7.5 Structure of international and national signaling networks
The national and international networks are considered

structurally independent and, although a particular, signaling point may
belong to both networks, signaling points are allocated signaling point
codes according to the rules of each network.
The most elementary signaling network consists of originating and

destination signaling points connected by a single signaling link. To meet
availability requirements this may supplement by additional links in
parallel, which may share the signaling load between them. If, for all
signaling relations, the originating and destination signaling points (OSP
and DSP) are directly connected in this way in a network, then the
network operates in the associated mode.
The worldwide signaling network is structured into two

functionally independent levels, namely the international and national
levels, as illustrated in Fig. 16.17. This structure makes possible a clear
division of responsibility for signaling network management and allows
numbering plans of signaling points of the international network and the
different national networks to be independent of one another.
A signaling point (SP), including a signaling transfer point (STP),

may assign to one of three categories:
• National signaling point (NSP) (signaling transfer point) which

belongs to the national signaling network only (e.g. NSP1) and is
identified by a signaling point code (Originating Point Code –OPC
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or Destination Point Code-DPC) according to the national

numbering plan of signaling points;
• International signaling point (ISP) (signaling transfer point) which

belongs to the international signaling network only (e.g. ISP3) and
is identified by a signaling point code (OPC or DPC) according to
the international numbering plan of signaling points;
• A node that functions both as an international signaling point

(signaling transfer point) and a national signaling point (signaling
transfer point) and therefore belongs to both the international
signaling network and a national signaling network and
accordingly is identified by a specific signaling point code (OPC or
DPC) in each of the signaling networks.
Fig. 16.17 Structural of international and national signaling networks
7.6 International and national signaling networks Criteria
The signaling network structure must be selected to meet the most

stringent availability requirements of any User Part served by a specific
network. The availability of the individual components of the network
signaling links (signaling points and signaling transfer points) must be
considered in determining the network structure.
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In order to take account of signaling message delay considerations,

regard should be given, in the structuring of a particular signaling
network, to the overall number of signaling links (where there are a
number of signaling relations in tandem) related to a particular user
transaction (e.g. to a specific call in the telephone application).
For all messages for the same transaction (e.g. a telephone call),
the MTP will maintain the same routing if the same signaling link
selection code is used in the absence of failure. However, a transaction
does not necessarily have to use the same signaling route for both
forward and backward messages.
The number of signaling links used to share the load of a given

flow of signaling traffic typically depends on
• The total traffic load;
• The availability of the links;
• The required availability of the path between the two signaling
points concerned; and
• The bit rate of the signaling links.
Load sharing requires at least two signaling links for all bit rates,
but more may be needed at lower bit rates. When two links are used,
each of them should be able to carry the total signaling traffic in case of
failure of the other link.
In the international signaling network, the number of signaling
transfer points between an originating and a destination signaling point
should not exceed two in a normal situation. In failure situations, this
number may become three or even four for a short period. This
constraint is intended to limit the complexity of the administration of the
international signaling network
A 14-bit code shall be used for the identification of signaling
points.
For National Signaling Networks, no specific structures are
required; however, Administrations should cater for the requirements
imposed on a national network for the protection of international services
in terms of network related user requirements, such as availability and
performance of the network perceived by users.
The signaling points and the signaling transfer points, which are
involved in a signaling of cross–border traffic, should belong to the
international hierarchical level. When those signaling points or signaling
transfer points are also involved in signaling of national traffic, they
should belong to their national hierarchical level as well. Therefore, the
double numbering of signaling point codes based on both the
international and national numbering schemes should be required.
The Network Indicator in the service information octet as described
in section 6 makes discrimination between international and national
point codes.
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7.7 National Signaling Point Code (NSPC) Format

This section describes the format of the code used to identify
National Signaling Points Codes in the national Signaling System No. 7
network, which is identified within the signaling system by the Network
Indicator (NI) as follows:
• NI=10 is national network (currently used)
• NI=11 is national network (Reserved for other national networks, if
needed.
The National Signaling Point Codes are the numbers that uniquely
identify a network (NE) in a CCS7 network.
The format of the 14-bit binary code used for the identification of
national signaling points. The 14 bits of NSPC are first converted to a
five- digit decimal number denoted ABCDE. That will be range from
00000 to 16383. The NSPC (ABCDE) decimal number will be divided into
two fields. The first field will be consisting of three decimal digits (ABC)
representing the Network Identity. The network identity will have 164
blocks, 163 of which have the capacity of 100 codes and one (#164) with
a capacity of 83 codes.
The second field will be consisting of two decimal digits (DE)
representing the signaling point code. Each block of (DE) will have a
capacity of 100 signaling point codes.
The NSPC structure is illustrated below:
Network Identity Signaling Point Code

ABC DE
3 digits 2 digits
Where ABC= 000 to 163 Where DE has a range from 00 to 99 for all
ABC values except for ABC=163, DE has a
range 00 to 83.
7.8 International Signaling Point Code (ISPC) Format

This section describes the format of the code used to identify
international signaling points in the international Signaling System No. 7
network which is identified by the Network Indicator NI=00.
The format of the 14-bit binary code used for the identification of
international signaling points is illustrated below.
Three (3) decimal numbers represent the binary code as follows:
• The first indicating the three (3) most significant bits (NML), with a
range of 0 to 7;
• The second indicating the following eight (8) bits (K-D), with a
range of 000 to 255; and.
• The third consisting of the three (3) least significant bits (CBA),
with a range of 0 to 7.
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The combination of the fields containing bits NML and bits K-D is
regarded as the Signaling Area/Network Code (SANC). The three (3) bits
(CBA) identify a specific signaling point which when combined with the
SANC forms the 14-bit ISPC (e.g. 2-068-1).
7.9 Standards
SPCs shall conform to relevant and applicable international
standards. Particular attention is drawn to the following ITU-T
Recommendations:
• Q.704: Signaling Network Functions and Messages
• Q.705: Signaling Network Structure
• Q.708: Numbering of International Signaling Point Codes
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Module - 01
Chapter – 17
Network Management
Session Objectives:

• Introduction
• Objectives of Network Management
• Present Scenario
• Success Factors for Network Management
• Network Management Process and Procedures
• Network Management Strategy
• Network Traffic Management
• Functions within the NMC
• NTM Principles and Objectives
• NTM measurements and Parameters
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1. Introduction
Network management means deploying and coordinating resources

in order to plan, operate, administer, analyze, evaluate, design and
expand communication networks to meet service level objectives at all
times at a reasonable cost and with optimum capacity.
Most organizations have recognized the strategic importance of
their communication network and its management. In most cases, better
control ensures a higher level of performance and this performance
corresponds with higher productivity. In addition, higher productivity
often translates into bottom line financial improvements. This leads us
to consider what the principal driving forces are for investing and
spending more on network management.
• Controlling strategic assets

• Controlling complexity
• Improving service
• Balancing various needs
• Reducing downtime
• Controlling costs
2. Objectives of Network Management

Based on surveys with a number of users including large, medium
and small environments, what they required and want, may be listed as
follows:
• Assurance of continued end-user service, characterized by
availability and quick response time, despite growth and change: End-
users are interested in maintaining a certain agreed-upon level of
services, which may necessitate connecting end-user devices to local and
or remote computing facilities by point-to-point, multipoint
communication links, or by using LAN's, MANs, or specific
communication solutions. Manageability at this level requires powerful
measurement techniques in both logical and physical components of the
networks. Changing technology and growth rates must not impair the
level of service.
• Capability to heal bypass, or circumvent failed network elements
as automatically as possible by integrating physical and logical network
management. Early detection and powerful alarm correlation techniques
should help diagnose problems quickly. As a result, the right strategy to
repair, bypass or circumvent failed components may be rapidly selected
and implemented. Artificial intelligence will pay a key role in this
activity. Constant availability of individual diagnostic tests, preferably
from any point in the network is desired as well.
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• Capability to operate fully even when important network

elements have failed: Powerful backup components and procedures for
both the physical and logical segments of the network are expected to
help in resuming service with no or minimal performance impairment
while trouble shooting failed components. Network management is
expected to guide and supervise this activity as part of fault
management.
• Capability to monitor and diagnose unsatisfactory conditions in
the entire network, including systems software and applications:
Performance monitoring is expected to embrace a broader scope than
ever before. Not only physical and logical network components, but also
a portion of the server's software, database activities and applications
need to be included. By doing so, real end-to-end network management
may be offered to users.
• Real-time or near-real-time analysis of network performance:
users request not only historical data for performance trending and
thresholding, but also real-time or near-real-time information. Early
recognition of performance bottlenecks, and selecting alternate routes,
may help to return the level of service to the expected range. Powerful
measurement techniques with low overhead data collection combined
with efficient real-time processing are the key prerequisites.
• Statistics and historical data automatically saved in a standard
data base with vendor-provided formats for analysis and also provision
for users to support their own analysis formats, first, to offer portability,
a general purpose data base is recommended for use as a repository;
second for satisfying specific needs, flexibility is requested in terms of
processing, presentation services and operational procedures. Users
expect a general-purpose platform, supported by the majority of vendors;
first, in further steps, event and alarm consolidation, correlation and
finally integration across different systems are on users request lists.
Increasing operations productivity by controlling quantity of staff,
number of operations sites, and skill levels: Due to a serious shortage of
human resources, users are looking for solutions that can help to
stabilize the number of employees required despite growth and new
applications support. Staff reduction is high priority in some voice-only
networking environments. People are also looking to artificial intelligence
as a significant means of supporting human work that can provide some
consistent assistance in problem determination. One of the scenarios for
the next century is a lights-out network control center. The path to such
a reality is via automated and unattended operations.
• Providing a powerful network management database for
supporting operations, administration, analysis, and planning: Users
are looking for a central information repository as an ultimate solution.
This repository or management information base (MIB) is expected to
store all relevant information about components, procedures, operational
rules, projects, and so on. Users expect this to be a relational or object-
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oriented database. They expect this database to store objects instances,

conductivity data, and status. The database then forms the basis of the
documentation system.
• Rapid continual response to changing network applications,
subscribers, devices, tariffs, and services: Dynamic adoption to the ever-
changing environment is a key item that drives network management
expenditures. While awaiting the development of a central information
repository, users will very likely compromise on a directory device, with
other word; acting as an umbrella connecting all existing databases and
files. Performance expectations are modest all the beginning. X.500
seems to be an interesting alternative in this respect.
• Dynamic expansion and reconfiguration of network capacity
using bandwidth management techniques: Co-operation between users
and suppliers is expected to improve substantially. Users want to gain
insight into networking segments they do not own or control. To remain
competitive, leading suppliers will very likely offer the opportunity in
other form of importing/exporting data. Electronic data interchange is
considered the real target.
• Use of one network-generation source: By consistent naming
and network addressing conventions, there is no need to have too many
'views' of a network. Different views have grown historically as different
management products have set other specific definitions of management
depth and scope. If the configuration database is considered the single
and only source of generating the network, the development of
unnecessary contradictory names and addresses can be avoided.
• Increasingly accurate and simplified accounting data: To
supervise spending and control costs, users request timely access to
accounting information for both data and voice networks. It is
envisioned that suppliers on a near real-time basis will provide station
detail maintenance records to users.
• Implementation of generic applications: Users are actually
looking for generic applications across single functions, groups of
functions, and instrument. Such application packages may be
purchased or leased by the user.
• Integrated network management: Users are requesting
increasing technical levels of integration. This means integrated
solutions across consolidated forms, multiple network architecture's
processing systems and network elements, geographical areas, multiple
vendors and private, virtual, and public network management
instruments.
• Centralization with distributed implementation: Users tend to
prefer a central network-management solution, with implementation of
certain "local" functions controlled by the central facility. In particular,
users generally request a solution for managing remote local area
networks by means of some distributed monitoring features.
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• Practical implementation of solutions based on international

standards: To avoid the confusion of using too many proprietary
solutions, international standards are preferred, but not at any price.
Users generally request the co-existence of leading de facto standard
solutions, such as IBM's Net View with OSI-based solutions or AT&Ts
Accumaster. In addition, SNMP may be considered for peer-to-peer
connections.
• Integrated reporting of data flows and capacity trends, as well as
standard performance metrics: Users are expecting powerful reporting
features with a minimal number of periodic paper-based reports, but
with the opportunity of accessing data elements and generating adhoc
reports on indicators needed for a specific application area. Standard
database and access features are highly ranked.
3. Present Scenario
There is a serious gap between what users require or want and the
present status of network-management applications, instruments, and
use of human resources. When analyzing data and voice networks costs,
the results show a continuous decrease in equipment cost and an
increase in communication costs. The equipment cost decrease is due
partially to large-scale integration at the hardware level and partially to
standardization results at the software level. The increase in
communication costs is explainable as part of the progress of
distributing computing power and databases with the consequent push
to connect surprisingly high numbers of stand-alone user devices.
Surveys confirm annual growth rates at the end-user device level in the
range of 35% to 45%.
The trend of ever-increasing communication costs is expected to
continue as non-networked personal computers and local area networks
continue being hooked together with the increasing variety of networking
options. The alarming fact, however, is that people costs are rapidly
increasing. The reasons may be found in one or more of the following:
networks never get smaller or less complex, resulting in a demand for
more human resources: higher skill levels of analysts and planners are
paving the way for higher salaries; expanding the scope and depth of
service to users requires still more people, especially in the operating
area. There is a push from company’s management for budgets that are
more evenly distributed between data processing and communication
resources.
In terms of the basic communication forms, the current situation
may be characterized as follows:
• Voice is strictly centrally managed and not yet ready for
integration.
• Data is both centrally and decent rally managed: there is interest
to include voice and image management. Image including word is
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strictly decent rally managed: image is beginning to be ready for

electronic mail.
• Video is strictly centrally managed: at the moment this
communication form is limited to innovators.
In summary, the integration process over multiple communication
forms will very likely take many years. Data and voice integration will
happen first, driven by shared communication form facilities and
integrated equipment.
4. Critical success Factors for Network Management

Critical success factors are those few key areas of activity in which
favorable results are absolutely necessary for an organization to reach its
goals. The goal for managing networks is to maintain end-user service
levels and thus ensure that the network is operating effectively and
efficiently at all times in order not to cause any problems in the
corporation's short-middle and long-range operations.
Critical success factors for network management are:
Processes and procedures: Sequence of application steps including

guidelines for how to use tools necessary to execute network-
management functions.
Instruments: Hardware and software or both, for collecting,

compressing data basing information and predicting future performance
of network components.
Human resource addresses the first of these factors i.e. process &
procedure.
The following sections will address each of these critical success

factors individually.
5. Network Management Process and Procedures
Fig. shows the most important network-management subsystems

based on the recommendations of international standard organizations.
The subsystems consist of a number of well-defined functions that are
supported by many processes and procedures in practical
implementations.
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Fig. 17.1 Network Management Sub-systems
Configuration management is a set of middle and long-range

activities for controlling physical, electrical, and logical inventories,
maintaining vendor files and trouble tickets, supporting provisioning and
order processing managing changes and distributing software. Directory
service and help for generating different network generations are also
provided.
Fault management is a collection of activities required to
dynamically maintain the network service level. These activities ensure
high availability by quickly recognizing problems and performance
degradation, and by initiating controlling functions when necessary,
which may include diagnosis, repair test, recovery, work around, and
backup. Log control and information distribution techniques are
supported as well.
Performance management defines the ongoing evaluation of the
network in order to verify that service levels are maintained, identify
actual and potential bottlenecks and establish and report on trends for
management decision making and planning. Building and maintaining
the performance database and automation procedures for operational
control are also included.
Security management is a set of functions to ensure the ongoing
protection of the network by analyzing risks, minimizing risks,
implementing a network security plan, and by monitoring success of the
strategy. Special functions include the surveillance of security
indicators, partitioning password administration, and warning or alarm
messages on violations.
Accounting management is the process of collecting, interpreting
processing and reporting counting-and charging-oriented information of
resource usage. In particular, processing of SMDRs, (Stations Message
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Detailed Record) bill notification, and charge-back procedures are

included for voice and data.
Network planning is the process of determining the optimal
network, based on data for network performance, traffic flow, resource
utilization, networking requirements, technological trade-offs and
estimated growth of present and future applications. Sizing rules and
interfaces to modeling devices are also parts of the planning process.
6. Network Management Strategic Directions and Benefits

The principal directions of network management are integration,
centralization, automation and repository support. Integration has to be
accomplished across multiple communication forms, multiple vendors,
multiple network architecture's, private, public, and virtual networks,
LANs, MANs, and WANs across multiple processors, applications
databases, and network-management products. Centralization offers the
opportunity of central control supported by shared or dedicated
processors in combination with distributed implementation of certain
network-management functions, such as flittering, problem detection,
data compression, and change management. Automation aims for
simplification of the operator’s tasks by improving productivity, error
minimization, problem prediction and prevention and speeding up
recovery using various tools techniques, and facilities. Artificial
intelligence is expected to play an important role in future automation.
Repositories are expected to be everything to everybody. Besides
configuration databases, performance data, vendor data, and trouble
tickets may also be integrated. In addition to inventory data,
connectivity information and dynamic indicators are expected to be
included.
If network-management function, procedures, and instruments are
properly implemented, and human resources are properly assigned
responsibilities, the Information Service Organization will benefit in
several ways:
Visibility of the networking topology

• Correlated alarm management
• Facts for service and utilization indicators
• Visibility of cost elements
• Sizing facilities and equipment on the basis of quantified
demand
By doing so, unnecessary costs, excess capacity, inefficient
network management applications, inappropriate topologies, and
unsatisfied users will be avoided.
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7. Network Traffic Management
Network Traffic Management (NTM) is the function of supervising

the performance of the network and taking actions to control the flow of
traffic, when necessary, to ensure the maximum utilization of network
capacity in all situations.
This function is performed in 'real time' that is the time between
the occurrence of an event and the reporting of it is almost negligible.
The aim is to control unusual traffic flow in the network, almost as it
occurs.
A total network traffic management system consists of four
component parts. These are:
1. Monitoring: means of measuring the network data
2. Controlling: means of changing the network configuration
3. Transport : means of transporting, monitoring & controlling
information
4. Support: means through which decision making at the man/machine
interfaces are aided.
General
Network Traffic Management is the function of supervising the
performance of the network and taking action to control the flow of
traffic, when necessary, to insure the maximum utilisation of network
capacity in all situations. A telecom/PTT faces mounting congestion
problems as the network grows in size and complexity.
The factors that were influential in considering adoption of network

traffic management are
• A growing concern with the consequences of total failure of SPC
exchanges.
• Evidence that the network was less robust, as efficiency
improved.
• Studies showing that the congestion effects of plant shortfalls
due to various practical problems, and traffic surges due to major events,
either man-made or natural, would be partly offset by NTM.
• Increasing recognition of the network as a single entity.
It is for all the above reasons and much more, that network traffic
Management (NTM) was formed as a discipline within the Network
Management Operations.
Other disciplines within the Network Management Operations are
• Common Channel Signaling
• Network restoration coordination
• Planned Event coordination
• Real Time Outage Reporting
• Major Outage Management
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• Customer Traffic Management
8. Functions within the NMC

The Network Management Center can be broadly categorized into
two main areas:
Network Management Operations
Network Management Support
The function of the operational group is to take action as close to
real time as possible to reduce the effect of any problems in the network
using the facilities available in each discipline, and to report to
management on any major unplanned outage.
The primary function of the support group is to investigate and
provide all the information and necessary tools for the operators to carry
out their function efficiently, while, the service restoration Planning
section of the support group develops all the strategies and controls for
the operators to use.
Operator Functions:
The action taken by the NTM operator is based on a NTM Service
Restoration Plan which consists of a set of NTM strategies, each relating
to a network overload or failure situation and each consisting of a set of
NTM controls which are available for use depending on the prevailing
conditions.
The NTM staff within the Support SRP section prepares these
strategies and controls after considerable investigations into exchange
data trunking and charging.
Once they are prepared, the control exchange data is loaded into
exchanges and tested. If tested satisfactory the operator then approves
them for use.
The specification and testing of NTM support systems and the
management of the network database of these systems are functions of
the support section.
The NTM staff within the Support section also performs the
following functions:
• Develop and Maintain operational guidelines
• Coordinate all adhoc investigations, reports and system software
and hardware modifications.
• Develop and monitor NTM performance parameters
• Provide high-level support to operations staff
9. NTM Principles and Objectives

The objective of NTM is to ensure that as many calls as possible
are successfully completed in all situations.
This objective is reached by using the following principles.
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NTM Principles
If one or more of the following NTM principles are not used in a
strategy plan then it is not a valid NTM strategy
• Keep all circuits fill with successful calls.
• Utilise all available circuits.
• Give priority to calls requiring a minimum number of links to
form a connection when all available circuits are in use.
• Inhibit switching congestion and prevent its spread.
The SRP section takes all these objectives and principles into
account when developing strategies and controls for use by the operator,
the operator must also consider the following basic operational
principles.
Operational Principles
• An NTM 'hot spot' must exist before a control action is
considered.
• Solve problem in local area before involving distant areas.
• Use expansive controls before protective controls.
10. NTM measurements and Parameters

The following measurements are used by NTM:
Bid: A signal attempt to obtain the service of a resource i.e. a call
attempt on a circuit is a bid.
Seizure: A successful bid, i.e. a call carried in a circuit.
Overflow: An unsuccessful bid to obtain a resource (Circuit) on the
selected route.
Usage
(Occupancy): A measurement of the load carried by a server
(Circuit or item of equipment) or by a group of servers. Expressed as a
percentage in use.
(Traffic intensity): A measurement of the load carried by a server
(circuit or item or equipment) or by a group of servers expressed in
Erlangs.
(Time): A measurement of the duration of a successful seizure.
Timeouts: A measurement of delay in obtaining a resource.
Rejected: A seizure on incoming circuits, which is refused by the
central processor.
Answer: A signal in a backward direction indicating a call is answered.
Terms and Definitions
Busy Condition: of a resource which is in use following a seizure.

Normally applied to a customer’s service.
Congestion: The state when the immediate establishment of a new
connection is impossible owing to inaccessibility of any the resources.
TC31009D10 302
Normally applies to the failure to successfully seizure a circuit from an

exchange towards the destination exchange.
Destination: A destination with a LOW Answer Bid Ratio

Answer Bid Ratio: Number of answered calls per number of bids
ABR = Answered Calls / Bids
Answer Seizure Ratio: Number of answered calls per numbers seizures
ASR = Answered Calls / Seizures
Seizures Per Circuit per Hour: Number of seizures per circuit
expressed per hour
Mean Holding Time: Average holding time of equipment in second
MHTS = Total Usage in Seconds
Percentage Overflow: Number of bids overflowing expressed as a
percentage.
%OFL = Bids - Seizures X 100
Bids
These measurements and calculated parameters provide the following:
Exchange Parameters
Rejected Calls
Central Processor % Occupancy
Common Control Equipment % Occupancy
Common Control Equipment Timeouts
Exchange Status Signal
*****************
TC31009D10 303

M01 - Communication Basics

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TC31009

Certified Wireless Maintenance Professional

Chapter Topic Pages

01 Transmission Units and Impairments 4 - 18

10 Introduction to Electronic Exchanges 148 – 174

Teleman Certified Wireless

Module No. Topic No. of Chapters

Issue No. Revision No. Issue Date Effective Date

Teleman (A Division of Plug-n-Work Directions)

Transmission Units and Impairments

On completion of this session, you will be able to understand the

Telecommunication systems started with the transmission of

In 1876, Alexander Graham Bell invented the Telephone and as

However, interest in the digital transmission received an impetus

Digital transmission systems are gaining more acceptances in view

The study of transmission units has a unique importance for

With analog transmission systems using copper cable there are

• Attenuation: There are two commonly used processes to

(a) Repeaters are the most commonly used devices to compensate

• Noise: Any random disturbance or unwanted signal on a

The higher voice frequencies attenuate at a higher rate than the

4. Decibel and Neper

One tenth of the common logarithm of the ratio of relative powers,

X = log P2/P1 B i.e. = 10 log P2/P1 dB

X = 20 log V2/ V1 dB. (When Z 1 =Z 2 )

Note: The dB is used rather than arithmetic ratios or percentages

The input is 1W and its output 2W, therefore,

Loss = 10 log Input/Output =10 log 1000/1 dB =10 log 103

0.1W Network 13 db gain ?

Gain = 10 log P2/P1 dB = 10 log P2/0.1 dB =13db

What is the power output of this network?

When the loss (gain) is X Neper, V1 and V2 are voltages, then

Comparing powers, X= 1/2 loge P1/P2 (N) or

Other transmission units 1 deciNeper (dN) = 0.1 N

5. Basic derived decibel units

common of these. By definition dBm is a power level related to 1 mw. The

The dBm formula may then be written as:

An amplifier has an output of 20 W; what is its output in dBm?

Power (dBm) = 10 log 20 W/1 mW = 10 log 20x103 mW/1mW = +43 dBm.

Conversion from Neper to decibel and Vice Versa

We know that decibel is fundamentally a unit of power ratio but it

Where Z1 and Z2 are characteristic impedances

Atténuation in decibels = 10 log10 |PS/PR |.................. (i)

Attenuation in dB = 20 log10 IS/IR if R1= R2 (iii)

If the resistive components of the impedance at the input and

In Analogue Transmission system, the quality of communication is

It is popularly known as SNR. SNR is the ratio of signal power to

In Digital Transmission system, the quality of communication is

1. BER (Bit Error Ratio)

Bit Error Rate (BER)

measuring instrument measures the number of bits transmitted in a

7. Digital Transmission - Performance Criteria (General)

1 in 106 (1.OE – 6) : Better

Bit errors greatly affect data service.

For data channels 1 in 109 (1.OE – 9) is normally realizable.

The quality parameters are:

1. Error Seconds (ES)

Error Seconds (ES):

Severely Error Seconds (SES):

Non-Severely Error Seconds (NSES):

Degraded Minutes (DM):

Available and non-available time