L-1

PCM Principles

PCM PRINCIPLES
1.0 INTRODUCTION
1.1

A long distance or local telephone conversation between two persons

could be provided by using a pair of open wire lines or underground
cable as early as early as mid of 19th century. However, due to fast
industrial

development

and

an

increased

telephone

awareness,

demand for trunk and local traffic went on increasing at a rapid rate. To
cater to the

increased demand of traffic between two stations or

between two subscribers at the same station we resorted to the use of an

increased number of pairs on either the open wire alignment, or in
underground cable. This could solve the problem for some time only as
there is a limit to the number of open wire pairs that can be installed on one
alignment

due

to

headway

consideration

and

maintenance

problems. Similarly increasing the number of open wire pairs that can be
installed on one alignment due to headway consideration and
maintenance problems. Similarly increasing the number of pairs to the
underground

cable

is

uneconomical

and

leads

to

maintenance

problems.
1.2

It, therefore, became imperative to think of new technical innovations

which could exploit the available bandwidth of transmission media such as
open wire lines or underground cables to provide more number of circuits
on one pair. The technique used to provide a number of circuits using a
single transmission link is called Multiplexing.

2.0 MULTIPLEXING TECHNIQUES

2.1

There are basically two types of multiplexing techniques

i.
ii

Frequency Division Multiplexing (FDM)

Time Division Multiplexing (TDM)

2.2

Frequency Division Multiplexing Techniques (FDM)

The FDM techniques is the process of translating individual speech circuits
(300-3400 Hz) into pre-assigned frequency slots within the bandwidth of
the transmission medium. The frequency translation is done by amplitude
modulation of the audio frequency with an appropriate carrier frequency.
At the output of the modulator a filter network is connected to select either
a lower or an upper side band. Since the intelligence is carried in either side
band, single side band suppressed carrier mode of AM is used. This
results in substantial saving of bandwidth mid also permits the use of low
power amplifiers. Please refer Fig. 1.

FDM techniques usually find their application in analogue transmission systems. An

analogue transmission system is one which is used for transmitting continuously
varying signals.
2.3

Time Division Multiplexing

2.3.1 Basically, time division multiplexing involves nothing more than sharing
a transmission medium by a number of circuits in time domain by establishing

a sequence of time slots during which individual channels

(circuits) can be

transmitted. Thus the entire bandwidth is periodically available to each channel.

Normally all time slots1 are equal in length. Each channel is assigned a time slot
with a specific common repetition period called a frame interval. This is illustrated
in Fig. 2.
2.3.2 Each channel is sampled at a specified rate and transmitted for a fixed
duration. All channels are sampled one by, the cycle is repeated again
and again. 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 unision with the gates at the transmitting
end.
2.3.3 The signal received at the receiving end will be in the form of discrete
samples and these are combined to reproduce the original signal. Thus, at a
given instant of time, onty one channel is transmitted through the medium, and by
sequential sampling a number of channels can be staggered in time as opposed
to transmitting all the channel at the same time as in EDM systems. This
staggering of channels in time sequence for transmission over a common
medium is called Time Division Multiplexing (TDM).

3.0 PULSE CODE MODULATION SYSTEM

3.1

It was only in 1938, Mr. A.M. Reaves (USA) developed a Pulse Code
Modulation (PCM) system to transmit the spoken word in digital form.
Since then digital speech transmission has become an alternative to
the analogue systems.

3.2

PCM systems use TDM technique to provide a number of circuits on

the same transmission medium viz open wire or underground cable pair or
a channel provided by carrier, coaxial, microwave or satellite system.

3.3

Basic Requirements For PCM System

To develop a PCM signal from several analogue signals, the following
processing steps are required

Filtering

Sampling

Quantisation

Encoding

Line Coding

4.0 FILTERING
4.1

Filters are used to limit the speech signal to the frequency band 300-3400
Hz.

5.0 SAMPLING
5.1

It is the most basic requirement for TDM. Suppose we have an

analogue signal Fig. 3 (b), which is applied across a resistor R through a
switch S as shown in Fig. 3 (a) . Whenever switch S is closed, an output
appears across R. The rate at which S is closed is called the sampling
frequency because during the make periods of S, the samples of the
analogue modulating signal appear across R. Fig. 3(d) is a stream of
samples of the input signal which appear across R. The amplitude of the
sample is depend upon the amplitude of the input signal at the instant of
sampling. The duration of these sampled pulses is equal to the duration for
which the switch S is closed. Minimum number of samples are to be sent for
any band limited signal to get a good approximation of the original
analogue signal and the same is defined by the sampling Theorem.

FIG. 3 : SAMPLING PROCESS

5.3

Sampling Theorem

5.3.1 A complex signal such as human speech has a wide

range of

frequency components with the amplitude of the signal being different at

different frequencies. To put it in a different way, a complex signal will have
certain amplitudes for all frequency components of which the signal is
made. Let us say that these frequency components occupy a certain
bandwidth B. If a signal does not have any value beyond this bandwidth B,
then it is said to be band limited. The extent of B is determined by the
highest frequency components of the signal.
5.3.2 Sampling Theorem States
"If a band limited signal is sampled at regular intervals of time and at a rate
equal to or more than twice the highest signal frequency in the band, then
the sample contains all the information of the original signal."
Mathematically, if fH is the highest frequency in the signal to be sampled then
the sampling frequency Fs needs to be greater than 2 fH.
i.e. Fs>2fH

5.3.3 Let us say our voice signals are band limited to 4 KHz and let sampling
frequency be 8 KHz.
Time period of sampling Ts =

1 sec
8000

or Ts = 125 micro seconds

If we have just one channel, then this can be sampled every 125 microseconds
and the resultant samples will represent the original signal. But, if we are to
sample N channels one by one at the rate specified by the sampling theorem,
then the time available for sampling each channel would be equal to Ts/N
microseconds.
5.3.4 Fig. .4 shows how a number of channels can be sampled and combined.
The channel gates (a, b ... n) correspond to the switch S in Fig. 3. These gates
are opened by a series of pulses called "Clock pulses". These are called gates
because, when closed these actually connect the channels to the transmission
medium during the clock period and isolate them during the OFF periods of the
clock pulses. The clock pulses are staggered so that only one pair of gates is open
at any given instant and, therefore, only one channel is connected to the
transmission medium. The time intervals during which the common transmission
medium is allocated to a particular channel is called the Time Slot for that channel.
The width of.this time slot will depend, as stated above, upon the number of
channels to be combined and the clock pulse frequency i.e. the sampling
frequency.

FIG. 4: SAMPLING & COMBINING CHANNELS

5.3

In a 30

channel PCM system. TS i.e. 125 microseconds are divided into

32 parts. That is 30 time slots are used for 30 speech signals, one time
slot for signalling

of

all the

30

chls,

and

one

time

slot for

synchronization between Transmitter & Receiver.

The time available per channel would be Ts/N = 125/32
= 3.9 microseconds
Thus in a 30 channel PCM system, time slot is 3.9 microseconds and time period
of sampling i.e..the interval between 2 consecutive samples of a channel is 125
microseconds. This duration i.e. 125 microseconds is called Time Frame.
5.4 The signals on the common medium (also called the common highway)
of a TDM system will consist of a series of pulses, the amplitudes of
which are proportional to the amplitudes of the individual channels at
their respective sampling instants. This is illustrated in Fig. 5

i
FIG 5 : PAM OUTPUT SIGNALS
5.5 The original signal for each channel can be recovered at the receive end by
applying gate pulses at appropriate instants and passing the signals through low
pass filters. (Refer Fig. 6)

Fig. 6 : RECONSTRUCTION OF ORIGINAL SIGNAL

6.0 QUANTISATION
6.1

In FDM systems we convey the speech signals in their analogue

electrical form. But in PCM, we convey the speech in discrete form. The
sampler selects a number of points on the analogue speech signal (by
sampling process) and measures their instant values. The output of the
sampler is a PAM signal as shown in Fig. 3; The transmission of PAM signal
will require linear amplifiers at trans and receive ends to recover distortion
less signals. This type of transmission is susceptible to all the disadvantages
of AM signal transmission. Therefore, in PCM systems, PAM signals are
converted into digital form by using Quantization Principles. The discrete
level of each sampled signal is quantified with reference to a certain
specified level on an amplitude scale.

6.2

The process of measuring the numerical values of the samples and

giving them a table value in a suitable scale is called "Quantising". Of
course, the scales and the number of points should be so chosen that the
signal could be effectively reconstructed after demodulation.

6.3

Quantising, in other words, can be defined as a process of breaking

down a continuous amplitude range into a finite number of amplitude
values or steps.

6.4

A sampled signal exists only at discrete times but its amplitude is drawn from
a continuous range of amplitudes of an analogue signal. On this basis, an
infinite number of amplitude values is possible. A suitable finite number of
discrete values can be used to get an. approximation of the infinite set.
The discrete value of a sample is measured by comparing it with a
scale having a finite number of intervals and identifying the interval in
which the sample falls. The finite number of amplitude intervals is called
the "quantizing interval". Thus, quantizing means to divide the analogue
signal's total amplitude range into a number of quantizing intervals and
assigning

level

to

each

intervals.

For example, a 1 volt signal can be divided into 10mV ranges like 10-

20mV, 30-40mV and so on. The interval 10-20 mV, may be designated as
level 1, 20-30 mV as level 2 etc. For the purpose of transmission, these
levels are given a binary code. This is called encoding. In practical systemsquantizing and encoding are a combined process. For the sake of
understanding, these are treated separately.
6.5

Quantizing Process

6.5.1 Suppose we have a signal as shown in Fig. 7 which is sampled at instants a, b,

c, d and e. For the sake of explanation, let us suppose that the signal has
maximum amplitude of 7 volts.
In order to quantize these five samples taken of the signal, let us say the total
amplitude is divided into eight ranges or intervals as shown in Fig. 7. Sample (a)
lies in the 5th range. Accordingly, the quantizing process will assign a binary
code corresponding to this i.e. 101, Similarly codes are assigned for other
samples also. Here the quantizing intervals are of the same size. This is
called Linear Quantizing.

FIG. 7 : QUANTIZING-POSITIVE SIGNAL

6.5.2

Assigning an interval of 5 for sample 1, 7 for 2 etc. is the quantizing

process. Giving, the assigned levels of samples, the binary code is

called coding of the quantized samples.

6.5.3
Fig.

Quantizing is done for both positive and negative swings. As shown in

6,

analogue

eight quantizing levels are used for each

signal.

To

indicate

whether

sample

direction
is

of the

negative

with

reference to zero or is positive with reference zero, an extra digit is

added to the binary code. This extra digit is called the "sign bit". In Fig.
8. positive values have a sign bit of ' 1 ' and negative values have sign
bit of'0'.

FIG. 8 : QUANTIZING - SIGNAL WITH + Ve & - Ve VALUES

6.6

Relation between Binary Codes and Number of levels.

6.1 Because the quantized samples are coded in binary form, the quantization
intervals will be in powers of 2. If we have a 4 bit code, then we can have 2" = 16
levels. Practical PCM systems use an eight bit code with the first bit as sign bit. It
means we can have 2" = 256 (128 levels in the positive direction and 128 levels
in the negative direction) intervals for quantizing.

6.7

Quantization Distortion
Practically in quantization we assign lower value of each interval to a sample
falling in any particular interval and this value is given as
Table-1 : Illustration of Quantization Distortion

Analogue

SignalQuantizing Interval Quantizing Level

Amplitude Range (mid value)

0-10 mv
5 mv
10-20mv
15mv
20-30 mv
25 mv
30-40 mv
35 mv
40-50 mv
45 mv

0
1
2
3
4

Binary Code
1000
1001
1010
1011
1100

If a sample has an amplitude of say 23 mv or 28 mv, in either case it will be assigned

\he \eve\ "2". This Is represented in binary code 1010. When this is decoded at the
receiving end, the decoder circuit on receiving a 1010 code will convert this into an
analogue signal of amplitude 25 mv only. Thus the process' of quantization leads to an
approximation of the input signal with the detected signal having some deviations in
amplitude from the actual values. This deviation between the amplitude of samples at
the transmitter and receiving ends (i.e. the difference between the actual value & the
reconstructed value) gives rise to quantization distortion.
6.7.2

If V represent the step size and 'e' represents the difference in amplitude

fe' must exists between - V/2 & + V/2) between the actual signal level and its quantized
equivalent then it can be proved that mean square quantizing error is equal to (V 2).
Thus, we see that the error depends upon the size of the step.
6.7.3

12

In linear quantization, equal step means equal degree of error for all input

amplitudes. In other words, the signal to noise ratio for weaker signals will be poorer.
6.7.4

To reduce error, we, therefore, need to reduce step size or in other words,

increase th,e number of steps in the given amplitude range. This would

however,

increase the transmission bandwidth because bandwidth B = fm log L. where L is

the number of quantum steps and fm is the highest signal frequency. But as we knows
from speech statistics that the probability of occurrence of a small amplitude is much
greater than large one, it seems appropriate to provide more quantum levels (V = low

value) in the small amplitude region and only a few (V = high value) in the region of
higher amplitudes. In this case, provided the total number of specified levels remains
unchanged, no increase in transmission bandwidth will be required. This will also try to
bring about uniformity in signal to noise ratio at all levels of input signal. This type of
quantization is called non-uniform quantization.
6.7.5 In practice, non-uniform quantization is achieved using segmented quantization
(also called companding). This is shown in Fig. 9 (a). In fact, there are equal number of
segments for both positive and negative excursions. In order to specify the location of a
sample value it is necessary to know the following :
1. The sign of the sample (positive or negative excursion)
2. The segment number
3. The quantum level within the segment

As seen in Fig. 9 (b), the first two segment in each polarity are collinear, (i.e. the slope
is the same in the central region) they are considered as one segment. Thus the total

number of segment appear to be 13. However, for purpose of analysis all the 16
segments will be taken into account.
7.0 ENCODING
7.1 Conversion of quantised analogue levels to binary signal is called encoding. To
represent 256 steps, 8 level code is required. The eight bit code is also called an eight
bit "word".
The 8 bit word appears in the form
P

ABC

WXYZ

Polarity bit 1
Segment Code
Linear encoding
for + ve 'O' for - ve.
in the segment
The first bit gives the sign of the voltage to be coded. Next 3 bits gives the segment
number. There are 8 segments for the positive voltages and 8 for negative voltages.
Last 4 bits give the position in the segment. Each segment contains 16 positions.
Referring to Fig. 9(b), voltage Vc will be encoded as 1 1 1 1 0101.

FIG. 9 (b) : ENCODING CURVE WITH COMPRESSION 8 BIT CODE

7.2

The quantization and encoding are done by a circuit called coder. The coder

converts PAM signals (i.e. after sampling) into a 8 bit binary signal. The coding is done
as per Fig. 9 which shows a relationship between voltage V to be coded and equivalent
binary number N. The function N = f(v) is not linear.
The curve has the following characteristics.
It is symmetrical about the origins. Zero level corresponds to zero voltage to be
encoded.
It is logarithmatic function approximated by 13 straight segments numbered 0 to 7 in
positive direction and 'O' to 7 in the negative direction. However 4 segments 0, 1, 0, 1
lying between levels + vm/64 -vm/64 being colinear are taken as one segment.
The voltage to be encoded corresponding to 2 ends of successive segments are in the
ratio of 2. That is vm, vm/2, vm/4, vm/8, vm/16, vm/32, vm/64, vm/128 (vm being the
maximum voltage).
There are 128 quantification levels in the positive part of the curve and 128 in the
negative part of the curve.
7.3 In a PCM system the channels are sampled one by one by applying the sampling
pulsqs to the sampling gates. Refer Fig. 10. The gates open only when a pulse is applied
to them and pass the analogue signals through them for the duration for which the gates
remain open. Since only one gate will be activated at a given instant, a common
encoding circuit is used for all channels. Here the samples are quantized and encoded.
The encoded samples of all the channels and signals etc are combined in the digital
combiner and transmitted.

7.4 The reverse process is carried out at the receiving end to retreive the original
analogue signals. The digital combiner combines the encoded samples in the form of
"frames". The digital separator decombines the incoming digital streams into individual
frames. These frames are decoded to give the PAM (Pulse Amplitude Modulated)
samples. The samples corresponding to individual channels are separated by

operating the receive sample gates in the same sequence i.e. in synchronism with the
transmit sample gates.

8.0 CONCEPT OF FRAME

8.1 In Fig. 10, the sampling pulse has a repetition rate of Ts sees and a pulse width of
"St". When a sampling pulse arrives, the sampling gate remains opened during the time
"St" and remains closed till the next pulse arrives. It means that a channel is activated for
the duration "St". This duration, which is the width of the sampling puse, is called the
"time slot" for a given channel.
8.2. Since Ts is much larger as compared to St. a number of channels can be sampled
each for a duration of St within the time Ts. With reference to Fig. 10, the first sample of
the first channel is taken by pulse 'a', encoded and is passed on the combiner. Then the
first sample of the second channel is taken by pulse 'b' which is also encoded and passed
on to the combiner, Likewise the remaining channels are also sampled sequentially and
are encoded before being fed to the combiner. After the first sample of the Nth channel is
taken and processed, the second sample of the first channel is taken, this process is
repeated for all channels. One full set of samples for all channel taken within the
duration Ts is called a "frame". Thus the set of all first samples of all channels is one
frame; the set of all second samples is another frame and so on.
8.3

As already said in para 5.3.5, Ts in a 30 channel PCM system is 125

microseconds and the signalling information of all the channels is transmitted

through a separate time slot. To maintain synchronization between
receive

ends,

the

synchronization

data

Thus the time available per channel would be 3.9 microsecs.

Frame = 125 microseconds

and

is transmitted through another time slot.

Thus for a 30 chl PCM system, we have 32 time slots.

Thus for a 30 chl PCM system,

transmit

Time slot per chl = 3.9 microseconds.

8.4

Structure of Frame

8.4.1 A frame of 125 microseconds duration has 32 time slots. These slots are
numbered Ts 0 to Ts 31.
Information for providing synchronization between trans and receive ends is passed
through a separate time slot. Usually the slot Ts 0 caries the synchronizsation signals.
This slot is also called Frame alignment word (FAW).
The signalling informatiori is transmitted through time slot Ts 16.
Ts 1 to Ts 15 are utilized for voltage signal of channels 1 to 15 respectively.
Ts 17 to Ts 31 are utilized for voltage signal of channels 16 to 30 respectively.
9.0 SYNCHRONIZATION
9.1

The output of a PCM terminal will be a continuous stream of bits. At the

receiving end, the receiver has to receive the incoming stream of bits and
discriminate between frames and separate channels from these. That is, the
receiver has to recognise the start of each frame correctly. This operation is called
frame alignment or Synchronization and is achieved by inserting a fixed digital
pattern called a "Frame Alignment Word (FAW)" into the transmitted bit stream at
regular intervals. The receiver looks for FAW and once it is detected, it knows that
in next time slot, information for channel one will be there and so on.
9.2

The digits or bits of FAW occupy seven out of eight bits of Ts 0 in the

following pattern.
Bit position of Ts 0
FAW digit value
9.3

B1
X

B2
0

B3
1

B4
1

B5
0

B6

B7

B8

The bit position B1 can be either ' 1 ' or '0'. However, when the PCM

system is to be linked to an international network, the B1 position is fixed at '1 ' .

The FAW is transmitted in the Ts O of every alternate frame.
Frame which do not contain the FAW, are used for transmitting supervisory
and alarm signals.

To distinguish the Ts 0 of frame carrying supervisory/alarm signals from those carrying

the FAW, the B2 bit position of the former are fixed at T. The FAW and alarm signals
are transmitted alternatively as shown in Table - 2.

TAB L E -2
Frame
Numbers
FO
F1
F2
F3 etc

Remark
B1
X
X
X
X

B2
0
1
0
1

B3
0
Y
0
Y

B4
1
Y
1
Y

B5
1
Y
1
Y

B6
0
1
0
1

B7
1
1
1
1

B8
1
1
1
1

FAW
ALARM
FAW
ALARM

In frames 1, 3, 5, etc, the bits B3, B4, B5 denote various types of alarms. For
example, in B3 position, if Y = 1, it indicate Frame synchronisation alarm. If Y = 1 in B4,
it indicates high error density alarm. When there is no alarm condition, bits B3 B4
B5 are set 0. An urgent alarm is indicated by transmitting "all ones". The code
word for an urgent alarm would be of the form.
X
111
1111
10.0 SIGNALLING IN PGM SYSTEMS
10.1 In a telephone network,-the signalling information is used for proper routing of a
call between two subscribers, for providing certain status information like dial tone,
busy tone, ring back. NU tone, metering pulses, trunk offering signal etc. All these
functions are grouped under the

general

terms

"signalling"

in

PCM

systems.

The signaling information can be transmitted in the form of DC pulses (as in step by
step exchange) or multifrequency pulses (as in cross bar systems) etc.
10.2 The signalling pulses retain their amplitude for a much longer period than

the

pulses carrying speech information. It means that the signalling information is

a slow varying signal in time compared to the speech signal which is fast changing
in the time domain. Therefore, a signalling channel can be digitized with less number
of bits than a voice channel.
10.3 In a 30 chl PCM system, time slot Ts 16 in each frame is allocated for carrying
signalling information.

10.4 The

time

slot

16

of

each

frame

carries

the

signalling

data

corresponding to two VF channels only. Therefore, to cater for 30 channels, we

must transmit 15 frames, each having 125 microseconds duration.
synchronization

data

for

all

frames,

For

carrying

one additional frame is used. Thus a

group of 16 frames (each of 125 microseconds) is formed to make a

"multiframe". The duration of a multiframe is 2 milliseconds. The multiframe has 16
major time slots of 125 microseconds duration. Each of these (slots) frames has 32
time slots carrying, the encoded samples of all channels plus the signaling and
synchronization data. Each sample has eight bits of duration 0.400 microseconds
(3.9/8 = 0.488) each. The relationship between the bit duration frame and multiframe
is illustrated in Fig. 11 (a) & 11 (b).

FIG. 11 (B) 2.048 Mb/s PCM MULTIFRAME

10.4 We have 32 time slots in a frame, each slot carries an 8 bit word.
The total number of bits per frame = 32 x 8 = 256
The total number of frames per seconds is 8000
The total number of bits per second are 256 x 8000 = 2048 K/bits.
Thus, a 30 chP PCM system has 2048 K bits.
10.6

Multiframe Structure

10.6.1 In the time slot 16 of FO, the first four bits (positions 1 to 4) contain the multiframe
alignment signal which enables the receiver to identify a multiframe.
The other four bits (no. 5 to 8) are spare. These may be used for carrying alarm
signals.
Time slots 16 of frames F1 to FT5 are used for carrying the signalling information. Each
frame carries signalling, data for two VF channels. For instance, time slot Ts 16 of frame
F1 carries the signal data for VF channel 1 in the first four bits. The next four bits are used
for carrying signalling information

for channel 16. Similarly, time slot Ts16 of F2 carries signalling data of chls 2 .and 17.
Thus in multiframe structure, four signalling bits are provided for each VF channels.
As each multiframe includes 16 frames, each with a sacnqtoq per sec.,.the.signalling of each channel will occur at a rate of 500 per sec.

Chapter -1

FUNDAMENTALS OF ELECTRONIC
EXCHANGES
1.0

Introduction

1.1
To overcome the limitations of manual switching; automatic exchanges, having Electromechanical components, were developed. Strowger exchange, the first automatic exchange
having direct control feature, appeared in 1892 in La Porte (Indiana). 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 in Sundsvall, Sweden. 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.
1.2
The first electronic exchange employing Space-Division switching (Analog switching)
was commissioned in 1965 at Succasunna, New Jersey. 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 Brittany, France.
1.3
This handout reviews the evolution of the electronic exchanges, lists the chronological
developments in this field and briefly describes the facilities provided to subscribers,
administration and maintenance personnel.

1965
1972
1973
1974
1975

Table 1 Chronological Development of Electronic Exchanges.

ANALOG
No.1 ESS
Local
Bell Labs, USA
D 10
Local and Transit NEC. Japan.
Metaconta
Local
LMT. France
No. 1 ESS Centrex
Local and Transit Bell Labs. USA
Proteo
Local & Transit
Proteo, Italy

1976
1976
1978

AXE
No.4 ESS
AXE

MODEL
Analog
No. 1 ESS
No. 1 ESS
NO. 4 A XB ETS
No. 4 ESS
D 10
XE 1
EWSD
EWSP
TXE-4
Proteo
AXE
PRX-205
Digital
Exchange
E-10B
Mentaconta
MT 20
E 12
System X
AXE -10
FETEX-150L
OCB-283
EWSD
No.5ESS
NEAX-61E

Local
Transit
Local

PTT & LM Ericsson, Sweden

Bell Labs, USA
LM Eiricsson, Sweden.

Table 2: Development of Electronic Exchanges

Capacity (in thousands)
Traffic
lines
Trunks
Erlangs
Call Attempts
per second
10-65
6,000
30
20-128
32
10,000
65
22.4
6,200
35
107
47,500
150
98
14.3
4,400
30
13
2,500
3.6
30
2,000
11-16
13
5,000
40
5,000
50
30
15
64
6,000
35
10
1,000
10-15
30
10-60
100
64
290
200
250
100

4
64
65
60
60
60
60
60

2,400
10,000
20,000
15,000
25,000
26,000
24,000
25,000
25,200
27,000

25
28-60
83-110
86
800000
800000
1800000
800000
1000000
1000000

TABLE 5- ADVANTAGES OF ELECTRONIC EXCHANGE OVER

ELECTROMECHANICAL EXCHANGES
Electromechanical Exchanges Electronic Exchanges
Category, Analysis, Routing, translation, etc;,
Translation, speech path Subs Facilities, etc.,
done by relays.
managed by MAP and other DATA.
Any changes in facilities require addition of
Changes can be carried out by simple
hardware and/or large amount of wiring
commands. A few changes can be made by
change. Flexibility limited.
Subs himself. Hence, highly flexible.
Testing is done manually externally and is time
consuming. No logic analysis carried out.
Testing carried out periodically automatically

Partial full-availability, hence blocking.

limited facilities to the subscribers.
Slow in speed. Dialing speed is max. 11 Ips
and switching speed is in l milliseconds.
Switch room occupies large volume.
Lot of switching noise.
Long installation and testing time.
Large maintenance effort and preventive
maintenance necessary.

3.2

and analysis printed out.

Full availability, hence no blocking. A large
number of different types of services possible
very easily.
Very fast. Dialing speed up to 11 digits /sec
possible. Switching is achieved in a few
microseconds.
Much lesser volume required floor space of
switch room reduced to about one-sixth.
Almost noiseless.
Short installation and testing period.
Remedial maintenance is very easy due to
plug-in type circuit boards. Preventive
maintenance not required.

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.
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.
3.3 Phased Developments
Many electronic switching systems, including the recent ones, had an electromechanical
switching network and used miniature electromagnetic relays in junctors and subscriber
line equipments None-the-less the trend is towards all electronic equipments for both
public and private switching and the switching network has already been made fully
electronic with the advent of digital switching.
However, very recently, several countries have developed or specified stored program
equipment for upgrading electromechanical exchanges. This typically involves replacing
the registers and translators of crossbar exchanges by processor-based facilities. These
allow the exchange subscribers to benefit from new services like abbreviated dialing call
forwarding automatic alarm call, and detailed billing. They, very significantly, enhance
exchange administration and maintenance capabilities for day-to-day operations, such as,
modifying a subscribers class of service, changing the way traffic is routed, collecting
traffic and load data, call charging, etc,
4.0

Facilities provided by Electronic Exchanges.

4.1

Facilities offered by electronic exchanges can be categorised in three arts.

(I) Facilities to the Subscribers.
(ii) Facilities to the Administration.
(iii) Facilities to the Maintenance Personnel.

4.2 Facilities to the Subscribers.

4.2.1

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.

4.2.2 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.

4.2.3

Toll (Outgoing Call) Restriction

The facility of toll restriction or blocking of subscriber line for specific types of outgoing
traffic, viz., long distance STD calls, can be availed of by all subscribers. This can be
easily achieved by keying-in certain service codes.

4.2.4

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.

4.2.5 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.
4.2.6

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.

4.2.7 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.
4.2.8 Conference Calls
Subscribers can set up connections to more than one subscriber and conduct telephone
conferences under the provision of this facility.
4.2.9

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.

4.2.10 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.
4.2.11 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 partys status. When idle status is ascertained, the connection is set up and ringing
current fed to both the parties.

4.2.12 Third party Inquiry

This system permits consulation and the transfer of call to other subscribers. Consulation
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.
4.2.13 Priority of calls to Emergency Positions
Emergency calls such as ambulance, fire, etc., are processed in priority to other calls.
4.2.14 Subscriber charge Indicator
By placing a charge indicator at the subscribers premises the charges of
can be ascertained by him.

each call made

4.2.15 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.
4.2.16 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.
4.2.17 Interception or Announcement.
In the following conditions, an announcement is automatically conveyed to calling
subscribers.
(i) Change of a particular number of transferred subscriber.
(ii) Dialing of an unallocated cods.
(ii) Dialing of an unobtainable number.
(iv)
Route congested or out of order.
(v)
Subscribers line temporarily out of order.
(vi) Suspension of sevice due to non-payment.
4.2.18 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.
4.2.20 Automatic Wake Up.
Automatic wake up service or morning alarm is possible, without any human
intervention.
4.2.21 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.
4.2..22 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.
4.2.23 Instrument Locking
A few subscrbers may like to have their telephone sets locked up against any misuse.
Dialing of a secret code will extend such a facility to them.
4.2.24 Free of charge Calls
Calls free of charge are possible on certain special services such as booking of
complaints , booking of telegrams, etc.
4.2.25 Collect call
If so desired, the incoming subscriber is billed for all the calls made to him, instead of the
calling subscriber.
4.3

Facilities to the Administration

4.3.1 Reduced Switch Room Accommodation

Reduction in switch room accommodation to about 1/6th to 1/4th as compared to Crossbar system is possible.
4.3.2 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.
4.3.4 Economic Consideration
The switching speed being much faster as compared to Cross-bar system, the use of
principle of full availability of trunk circuits and other equipment makes the system
economically superior to electromechanical systems.
4.3.5 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.
4.4

Maintenance Facilities

4.4.1 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 subsystem.
4.4.2 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.
4.4.3 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.
4.4.4 Blocking
In case of congestion or breakdown of a specific route, facility of blocking such routes is
available in modes, such as
(I)
Blocking of a specified percentage of calls in such a route either
automatically or manually.
(ii)
Blocking a specific category of subscribers.
4.4.5 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.
5.0

Constraints of Electronic Exchanges

5.1

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

5.2

guided solely by the amount of processor loading. Moreover, the facilities to the
subscribers will also have to be limited accordingly.
5.3 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.
5.4 Total Protection from Dust
All possible precautions should be observed for ensuring dust-free environment.
5.5 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 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.
5.6

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 PCBs 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.

5.7

PCB Repair
The repair of PCBs 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.

5.8 Faster Obsolescence

The changes in the field of electronics are almost revolutionary with the very fast
improvements. Hence, the current technology becomes obsolete at a very fast rate. The
equipment becomes obsolete before it can possibly complete one third of its life and it
might be impossible to get spare parts for the entire currency of the life of the system.
6.0 Conclusion
6.1

After 1950, the development in the field of electronic devices

induced the
telephone system designers to make use of innumerable
advantages offered by their
inventions. Therefore, telephone switching system with both electronic and
electromechanical components was evolved.

Later on, Stored Program Control concept was evolved and adapted to the
electromechanical exchanges. This developmental step opened a new era of innumerable
additional facilities to the subscribers, administration and maintenance personnel.

CHAPTER-2
DIGITAL SWITCHING,CONCEPT OF TIME & SPACE
SWITCHING
1.0

Introduction

1.1
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.
1.2

In this handout, general principles of time and space switching are

discussed. A practical digital switch, comprising of both time and space stages, is also
explained.

2.0

Time and Space Switching

2.1

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 HWY2, via the digital switch, as shown in Fig1.

FIG 1 DIGITAL SWITCH

2.2
The interconnection of time-slots, i.e., switching of digital signals can be achieved using
two different modes of operation. These modes
are: I. Space Switching
ii. Time switching
Usually, a combination of both the modes is used.
2.2.1 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.
2.2.2 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 timeslot 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.

3.0

Digital Space Switching

3.1

Principle

3.3.1 The Digital Space Switch consists of several input highways, X1, X2,...Xn and several
output highways, Y1, Y2,.............Ym, inter connected by a crosspoint matrix of n rows and m
columns. The individual crosspoint consists of electronic AND gates. The operation of an
appropriate crosspoint 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 2. During other time-slots, the same crosspoint may be used to connect other channels. This
crosspoint matrix works as a normal space divided matrix with full availability between
incoming and outgoing highways during each time-slot.
3.1.2 Each crosspoint column, associated with one O/G highway, is assigned a column of
control memory. The control memory has as many words as there are time-slot per frame in the
PCM signal. In practice, this number could range from 32 to 1024. Each crosspoint in the
column is assigned a binary address, so that only one crosspoint 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.
3.1.3

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 timeslot.Thus, the crosspoint corresponding to the address, is operated during a particular

time-slot. This crosspoint 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 different columns. Depending upon the number of time-slots in one
frame, this time division action increases the utilisation of crosspoint 32 to 1024 times
compared with that of conventional space-devided switch matrix.
3.2

Illustration

3.2.1 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 crosspoint. In other words, the objective is to connect
TS7 of HWY X1 and TS7 of HWY Y3.
3.2.2 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 crosspoint is written in this location, i.e., 1, in binary, is written in location 7, as
shown infig 2.This crosspoint 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.
3.3

Practical Space Switch

3.3.1 In a practical switch, the digital bits are transmitted in parallel

through the switching matrix.
3.3.2
3.3.3

rather than serially,

In a serial 32 time-slot 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.3. 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

3.3.3 Due to reduced bit rate in parallel mode, the crosspoint is required to be operated only for
1/8th of the time required for serial working. It can, thus, be shared by eight times more
channels, i.e.,32 x 8 = 256 channels, in the same frame.
3.3.4

However, since the eight bits of one TS are carried on eight wires, each

cross point have eight switches to interconnect eight input wires to eight output wires.
Each crosspoint (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 s)

Fig 3 Serial parallel converter

3.3.5

For example,to connect 40 PCM I/C highways, a matrix of 40x 40 = 1600

crosspoints 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 crosspoint is sufficient. As eight
switches are required at each crosspoint 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.

4.0

Digital Time Switch

4.1

Principle

4.1.1 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.
4.1.2 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.
4.1.3 The writing/reading operation in the speech memory are controlled by the Control
Memory. It has same number of memory locations as for speech memory, i.e., 32 locations for 32
channelPCM 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.
4.1.4 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.
4.2

Illustration

4.2.1 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.4.
4.2.2

The reading of the sample is controlled by the Control Memory. 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.
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.

4.2.3 For disconnection of the call, the CC erases the contents of the
location to halt further transfer of samples.

control

memory

4.3

Time switch can operate in two modes, viz.,

I.
Output associated control
ii.
Input associated control

4.3.1

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 of Sec.4.2.
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

occurance of the time-slot. The entire process of writing and reading is repeated in every
frame, till the call is disconnected.

FIG 4 OUTPUT ASSOCIATED CONTROL SWITCH

4.3.2

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 fig5.
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 5 INPUT ASSOCIATED CONTROLLED TIMR SWITCH

4.4

Time Delay Switching

4.4.1 The writing and reading, of all time-slots in a frame, has to be completed within one
frame time period (before the start of the next frame). A TS of incoming PCM may, therefore, get
delayed by a time period ranging from 1 TS to 31 TS periods, before being transmitted on
outgoing PCM. For example, consider a case when TS6 of incoming PCM is to be switched to
TS5 in outgoing PCM. In this case switching can be completed in two consecutive frames only,
i.e., 121 microseconds for a 32 channel PCM system. However, this delay is imperceptible to
human beings.
4.5

Non-Blocking feature of a Time Switch

4.5.1 In a Time Switch, there are as many memory locations in the control and speech
memories as there are time-slots in the incoming and outgoing PCM highways, i.e.,
corresponding to each time-slot in incoming highway, there is a definite memory location
available in the speech and control memories. Similarly, corresponding to each time-slot in the
outgoing highway there is a definite memory location available in the control and speech

memories. This way, corresponding to free incoming and outgoing time-slots, there is always a
free path available to interconnect them. In other words, there is no blocking in a time switch.
5.0

Two Dimentional Switching

5.1
Though the electronic crosspoints are not so expensive, the cost of accessing
and selecting them from external pins in a Space Switch, becomes 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 dimentional 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 acceptibility of tolerable limits of blocking, in a practical network. Similarly, a twostage two-dimentional 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
favoured 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., tandom 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 favour TST structure, it is most widely used.
5.2

TST Network

5.2.1

As the name suggests, in a TST network, there are two time stages
seperated 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 ouput 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 crosspoint 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 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 nonfolded, 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.
5.2.2 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 matrix. 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 6.

FIG 6 T S T SWITCH
5.2.3

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 timeslots 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.
For transfer of speech samples from the calling party to the called party of previous
example, CC orders writing of various addresses in 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. 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 crosspoint 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.

FIG 7 T S T SWITCH STRUCTURE

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, avialable to handle next call.

CHAPTER-3

Signaling in Telecommunication subscriber

signaling, Trunk signaling (CAS & CCS)
SIGNALLING IN TELECOMMUNICATION
1.0.
1.1.

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 signalling 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 a
syntax in the form of a complex set of rules. governing the assembly of these signals.

1.2

This handout discusses the growth of signalling and various type of

signalling codes used in Indian Telecommunication.

2.0

Types of signalling information

2.1.

The signaling information can be categorized under four main heads.

2.1.1

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 an appropriate
equipment for receiving address informaion ( 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.

2.1.2

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 signalling.
Address information essentially comprises of full or part of the called subscribers number
and possibly additional service data.

2.1.3

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.

2.1.4

Supervisory information
It specifies the on/off hook condition of a called subscriber after the connection has been
setup
i.
Called subscriber off hook called subscriber has answered and
commence.
ii.

charging may

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.
2.2.

Call connection
The interchange of signaling information can be illustrated with the help of
a typical call connection sequence. The circled number in Fig. 1 correspond to the steps
listed below
CALLING SUBSCRIBER

i.

ORIGINATING EXCHANGE
TERMINATING EXCHANGE
CALLED SUBSCRIBER

A request for originating a call is initiated when the calling subscriber

lifts the handset.
ii.
The exchange sends dial-tone to the calling subscriber to indicate to
him to
LINE
TRUNK
LINE
start dialling.
iii.
The called number is transmitted to the exchange, when the calling
ON
HOOK dials the number.
ON HOOK
ON HOOK
subscriber
TIME
iv. OFF-HOOK
if the number is free, the exchange sends ringing current to him.
TONE is provided to the calling subscriber by the exchange by sending.
v. DIAL
Feed-back
(ADDRESS)
a.
Ring-back tone, if the called subscriber is free(shown in fig.1)
2 called
CONNECT
b.
Busy tone if the
subscriber is busy ( not shown in
figure), or
c.
Recorded message, if provision exists, for non completion of
call
due to some other constraint ( not shown in figure).
vi.
The called subscriber indicates acceptance of the incoming call by lifting
the
handset
3
vii.
The exchange
recognizing the
acceptance terminates the ringing current and the
ADDRESS
10
AUDIBLE
RINGING
ring-back tone, and establishes a connection between
the calling and called
TONE
4 RINGING (20 MHz)
5
subscribers.
6

OFF-HOOK(ANSWER)

AUDIBLE RINGING
TERMINATED

OFF-HOOK

(ANSWER RINGING TERMINATED

SUBSCRIBERS CONNECTED

ON HOOK

OCONVERSATION ENSURES

ON HOOK

DISCONNECT

FIGURE 1. SIGNALLING ON A TYPICAL COMPLETED CALL

viii.

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.
ix
The originating exchange seizes an idle inter exchange trunk,
connected to a digit register at the terminating exchange.
x.
The originating exchange sends the digit. The steps iv to viii
are
then performed to set up the call.
3.0.

Signalling

3.1

Telephony started with the invention of magneto telephone which used a

magneto to generate the ringing current, the only signal, sent ver a dedicated line between
two subscribers. The need for more signals was felt with the advent of manual switching.
Two additional signals were, therefore, introduced to indicate call request and call
release. The range of signals increased further with the invention of electro-mechanical
automatic exchanges and is still growing further at a very fast pace, after the advent of
SPC electronic exchanges.

3.2

Subscriber Line signalling

3.2.1

Calling Subscriber Line Signaling

In automatic exchanges the power is fed over the subscribers 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.

3.2.1.1 Call report

When the subscriber is idle, the line impedance is high. The line impedance falls, as soon
as, the subscriber lifts the hand-set, 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.
3.2.1.2

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
dialling.
1.
Decadic Dialling
The address digits may be transmitted as a sequence of interruption of the DC loop by a
rotary dial or a decadic push-button key pad. The number of interruption (breaks) indicate
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 a 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.
2.
Multifrequency Push-button Dialling
This method overcomes the constraints of the decadic dialling. 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) dialling. 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.2.
By this method, the dialling 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
teminals to a variety of systems, such as computers with voice output. The tones have
been so selected as to minimize harmonic interference and probability of simulation by
human voice.

FIGURE 2. TONE-DIALLING FREQUENCY GROUPS.

3.2.1.3 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.
1.
Ring-back tone to the calling subscriber and ringing current to
the
called subscriber, if the called line is free.
2.
Busy-tone to the calling subscriber, if the called line is busy or
otherwise inaccessible.
3.
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.
3.2.1.4 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 subscriber, e.g., short-circuiting the transmitter of a
CCB, till a proper coin is inserted in the coin-slot.
3.2.1.5 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.
3.2.1.6 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.
3.2.2 Called subscriber line signals.
3.2.2.1 Ring Signal
On receipt of a call to the subscriber whose line is free, the terminating exchange sends
the ringing current to the called telephone. This is typically 25 or 50Hz with suitable
interruptions. Ring-back tone is also fed back to the calling subscriber by the terminating
exchange.
3.2.2.2 Answer Signal
When the called subscriber, lifts the hand-set on receipt of ring, the line impedance goes
low. This is detected by the exchange which cuts off the ringing current and ring-back
tone.
3.2.2.3 Release Signal
If after the speech phase, the called subscriber goes on hook before the calling subscriber,
the state of line impedance going high from a low value, is detected. The exchange sends
a permanent line signal to the calling subscriber and releases the call after a time delay, if
the calling subscriber fails to clear in the meantime.
3.2.3

Register Recall Signal

With the use of DTMF telephones, it is possible to enhance the services, e.g., by dialing
another number while holding on to the call in progress, to set up a call to a third
subscriber. The signal to recall the dialling phase during the talking phase, is called
Register Recall Signal. It consists of interruption of the calling subscribers loop for
duration less than the release signal. it may be of 200 to 320 milliseconds duration.

3.3

Inter-exchange Signaling
3.3.1 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
i. 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
ii. Continuous
The signal consists of transition from one condition to another, a steady
state condition does not characterizes any signal.
iii. 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.
3.3.2

Line signals

3.3.2.4 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.,
i.
Circuit seizure/release corresponding to off/on-hook signal of
subscriber.
ii.
Address information in the from of decade pulses.

the

3.3.2.2 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. The
dialing pulses can also be transmitted by pulsing of the states. 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.
CONDITION

SINGLE FREQUENCY SIGNALING STATES TONE SIGNAL

State

Idle (On hook)

FORWARD
Seizure(off hook)
Release (on hook)
BACKWARD
Answer(off hook)
Clear Back (on hook)
Blocking (off hook)

Forward

Backward

On

On

off
on

on
off/on

off
off
on

off
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 hard-ware costs.
3.3.2.3 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. The signaling states are shown in table2.

TABLE 2. E & M SIGNALING STATES

State
Idle
(On hook)
FORWARD
seizure
(off hook)
Release
(On hook)
BACKWARD
Answer
(off hook)
Clear Back
(On hook)
Blocking

Outgoing Exchange
M- lead
E-lead
Earth
Open

Incoming Exchange
M- lead
Elead
Earth
Open

Battery

Open

Earth

Earth

Earth

Earth/open

Battery/Earth

Open

battery

Earth

Battery

Earth

battery

Open

Earth

earth

Earth

Earth

Battery

Open

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.
3.3.3

Register Signals

3.3.3.1

It was, however felt that the trunk service could not be managed properly
without the trunk register which basically is an address digit receiver, with such
development, the inter-exchange signaling was sub- divided into two categories.
1.
Line signaling in which the signals operate throughout the
duration of call, and
2.

Register signaling during the relatively short phase of setting

up the call, essentially for transmitting the address information.

forward
signal

outgoing register
incomming2-and-2only
register

signal recognition

time
time

signal cessation recognition

signal cessation
recognition

acknowledgement backward
signal and request for next signal
compelled signal sequence

next forward
signal
acknowledgement backward signal

Sending

receiving

Fig.3. Compelled signalling procedure

In other words, register signals are interchanged between registers during a phase
between receipt of trunk seizure signal and the exchange switching to the speech phase.
These signals are proceed-to-send (PTS) signals, address, signals, and signals indicating
the result of the call attempt.
The register signals may be transmitted in band or out of band. however, in the latter
case, the signaling is relatively slow and only limited range of signals may be used. For
example, a single out-of-band frequency may be selected and information sent as pulses.
In-band transmission can be used easily as there can be no possible interference with the
speech signals. To reduce transmission time and to increase reliability, a number of
frequencies are used in groups. Normally 2 out of 6 frequencies are used. To make the
system more reliable compelled sequence is used. Hence, this system is normally called
compelled sequence Multi-frequency (CSMF) signaling as shown in Fig.3. In CCITT

terminology it is termed as R2 system. As the frequencies need be transmitted only for a

short duration to convey the entire information, the post dialling delay is reduced.
3.3.3.2 When more than two exchanges are involved in setting up the
connections
the signaling may be done in either of the two modes
i.
End-to-end signaling
The signaling is always between the ends of the connection, as the call
progresses. Considering a three exchanges, A-B-C, connection, initially
the signaling is between A-B, then between A-C after the B-C connection
is established.
ii.

Link-By-Link signaling
The signaling is always confined to individual links. Hence, initially the
signaling is between A-B, then between B-C after the B-C connection is
established.

Generally supervisory (or line) and subscriber signaling is necessarily on link-by-link

basis. Address component may be signalled either by end-to-end or link-by-link
depending upon the network configuration.
3.3.3.3 R2 Signalling
CCITT standardized the R2 signaling system to be used on national and international
routes. However, the Indian environment requires 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 and the significance of each signal is indicated in Table 4
and 5.
TABLE 3- SIGNAL FREQUENCY INDEX AND WEIGHT CODE
Signal Frequency (Hz)
Forward

1380

1500

1620

1740

1860

Backward

1140

1020

900

780

660

f0

f1

f2

f3

f4

Index
Weight Code

Signal

TABLE 4-FORWARD SIGNALS

Weight
Group I
Digit 1
Digit2

1
2

0+1
0+2

3
4
5
6
7

1+2
0+4
1+4
2+4
0+7

Digit3
Digit4
Digit5
Digit6
Digit7

8
9
10

1+7
2+7
4+7

Digit8
Digit9
Digit0

Signal No.
1

TABLE 5 -BACKWARD SIGNALS

Weight Code
Group A
0+1
Send next digit

2
3

0+2
1+2

0+4

1+4

2+4

7
8
9
10

0+7
1+7
2+7
4+7

Restart
Address complete,
Changeover to
reception of group B
signals
Calling line
identification for
malicious calls
send calling
subscribers category
Set up speech
connection
Send last but 1 digit
Snd last but 2 digit
Snd last but 3 digit
Spare

Note : Signals A2, and A7 to A9 are used in Tandem working only.

Group II
Ordinary subscriber
Subscriber
with
priority Test / Mtce,
equipment
Spare
STD Barred
Spare
CCB
Changed Number to
Operator
Closed Number
Closed Number
Spare

Group B
Called line free with
out metering
Changed number
Called line busy

Local congestion
Number unobtainable
called line fee, with
metering
Route congestion
Spare
Route Breakdown
Malicious call
blocking

It can be seen from the tables that

1.
Forward signals are used for sending the address information of the called
subscriber, and category and address, information of the calling subscriber.
2.
Backward signals are used for demanding address information and callers
category and for sending condition and category of called line.

R2 signaling is fully compelled and the backward signal is transmitted as an

acknowledgement to the forward signal. This speeds up the interchange of information,
reducing the call set up time. However, the satellite circuits are an exception and semicompelled scheme may only be used due to long propagation time.
Register signals may be transmitted on end-to-end basis. It is a self checking system.
Each signal is acknowledgement appropriately at the other end after the receiver checks
the presence of only 2 and only 2 out of 5 proper frequencies.
3.3.3.4 An example of CSMF signaling between two exchanges may be illustrated
by considering a typical case. The various signals interchanged
after seizure of the
circuit using DC signaling are
1.
originating exchange sends first digit
2.
Receipt of the digit is acknowledged by the terminating
exchanges by
sending A5 (demanding the callers category).
3.
A5 is acknowledgement by sending any 11-1 to 11-5 by the
originating exchange
4.
Terminating exchange acknowledges this by A1, demanding
for
next digit.
5.
Originating exchange, acknowledges A1 by sending any of 1-1
to 1-10
sending the digit.
6.
The digits are sent in succession by interchange of
steps v and vi.
7.
On receipt of last digit, the terminating exchange carries out
group
and line selection and then sends A3, indicating
switching over to group B
signals.
8.
This is acknowledgement by the originating exchange by sending
the
callers category again.
9.
The terminating exchange acknowledgements by sending the
called
line condition by sending any of B2 to B6.
10.
In response to B6, the originating exchanges switches through
the
speech path and the registers are released. Alternatively, in response to B2 to B5,
the registers are released and appropriate
tone is fed to the calling subscriber
by the originating
exchange.
4.0

Digital Signalling

4.1

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 signalling can be transmitted over a channel directly associated with the
speech channel, channel-associated signalling (CAS) , or over a dedicated link common
to a number of channels, common channel signalling (CCS). The information transmitted
for setting up and release of calls is same in both the cases. Channel associated signalling
requires the exchanges, to have access to each trunk via the equipment which may be
decentralised, whereas, in common channel signalling, the exchange is connected to only
a limited number of signalling links through a special terminal.

4.2

Channel- Associated signalling

In the PCM systems the signalling information is conveyed on a separate channel which
is rigidly associated with the speech channel. Hence, this method is known as channel
associated signalling (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, repectively, 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 signalling 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 6.
As the dialling 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 signalling for each speech
channel is highly inefficient as it remains idle during the speech phase. Hence, another
form of signalling known as common-channel signalling evolved.

State
Idle
Seizure

Forward
af
1
0

Bit Value
backword.
bf
ab
0
1
0

bb
0
0

Seizure
acknowledge

Answer

Clear Forward

0/1

Clear Back

4.3

Common channel signalling

4.3.1

Common channel signalling (CCS) overcomes the efficiency of the

CAS.
In this method, the signalling channel for a circuit is allotted only for
the duration of
signalling. A separate data-link dedicated to signalling only, is used for the purpose, as
shown in Fig.4.

SIG
OFFICE A

TRUNKS

SIG

SIG

SIG

SIG

SIG

OFFICE B

(a) Per - trunk signalling

OFFICE C

OFFICE D

PROCESSOR

PROCESSOR

CCIS
SIG

CCIS
SIG

(b) Common - channel inter - exchange signalling

CCS SIG Common-Channel Inter-exchange Signalling Equipment SIG per trunk Signalling
Equipment
Figure 4. Inter-exchange Signalling Techniques
In other words, CCS has a pool of signalling channels which are allocated to a speech
circuit, only when the later has any requirements of signalling. Hence, the speech circuits
may have to queue up for a spare signalling circuit. Therefore, the dimensioning of the
pool capacity will depend on the acceptable level of service, and expected signalling
content and frequency per speech circuit.
By using this technique, the signalling equipment can be centralised and made more
compact resulting in advantages of space saving and economy. However, this technique
can be used only by the SPC exchanges for inter processor signalling.
4.3.2

Iner-Processor signalling
In the inter processor signalling, there is a total departure from the conventional
signalling. Instead of exchanging DC signals, tones, frequencies or bit patterns for
hundreds of milliseconds, a single data nessage of 40 to 50 millisecond is sufficient for
conveying the entire information.
The signalling word, also called, signalling unit (SU), is divided into sub words or fields
containing bits to represent.
1.
Actual signal message, i.e., speech circuit number, service indicator
(telephone,data etc.,) and signal information (directory number, etc.)
2.
Transfer control, i.e., informatin for synchronisation, message
numbering and acknowledgement of receipt.
3.
Error protection , i.e., redundant bits for detection of
transmission
error.

4.3.3

Message Transfer Procedure

The contents of the transfer control section depend upon the procedure or protocol
adopted for message transfer which essentially concerns synchronisation and error
correction.

4.3.3.1 Synchronisation
Synchronisation is required at several levels at
1.
data link level to recover bit timing.
2.
message level to detect the start and end of messages and
3.
message sequence level to identify each message in a series of
messages received so that retransmission can be requested if
necessary.
4.3.3.2 Error protection
To detect and correct transmission errors, redundancy must be provided in the transmitted
information, if there is no provision of requesting retransmission of the information.

However, if a return channel is available only error detection, redundancy is necessary

and retransmission of the signal can be requested if the signal is mutilated.
4.4

Practical CCS systems

Currently a signalling system termed as CCITT no.6, recommended by CCITT is in use .
another system termed as CCITT No.7 is being experimented for compatibility with
ISDN. The signalling data is interchanged in digital streams between the two processors
via a special dedicated signalling interface.

4.4.1

CCITT No.6 System

It is designed for use with all types of international circuits, including satellite circuits.
Signalling can be carried over 2400 bits/second over analog links, or at 4 K bits/second
over digital links . The information is transmitted in the form of 28 bit signal units, as
shown in.

Fig5 (a) organised as block of 12 signal units. Error protection is through an error
detecting code and repeat transmission of mutilated message
Fig.5 Typical message formats. The shaded area represent spare bit fields.
4.4.2 CCITT No. 7 System
In view of the introduction of an unprecedented range of new services and facilities for
subscriber, operating companies and telecommunications networks, a new system has been
evolved which will be suitable for the international network (terrestrial and satellite links) and
national network with optional performance in digital network. The signal unit is shown in Fig.
5 (b). The functional breakdown of the system is as under :i.

Level 1 is the signalling data link, comprising of an analog or digital

transmission medium with a bit rate from 2400 bit/s to 64 K bit/s.
ii.
Level 2 is the signalling link function which includes trnsferring the
signalling message over the data link in a signal unit, signal unit delimitation
transmission error detection and correction, and
signalling link failure detection and
recovery.
iii.
Level 3 distributes messages between users and the signalling link.
iv.
Level 4 groups the various user parts. In addition to call processing
the
function of the users may include network administration and
maintenance.
4.5

Advantages of CCS
The other advantages of CCS, in addition to space saving are :i.
Faster call set up by cutting down the post dialler 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 signalling.
ii.
New services can be made available with a better quality. For
example, setting up a call with abbreviated dialling 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.
iii.
More call completion is possibly by re routing the call without
increasing the call set up time to an unacceptable level.
iv.
In MF signalling system it is possible for a clever subscriber to access
the system by generating of generally used signalling tones. By generating tones.
of the correct frequency and at the correct time, such a phone- phreak can make
long distance calls without being charged thus resulting in loss of revenue,
However, phone phreak phree calls are not possible in CCS, as the signalling link
is totally separate from the speech link.
v.
Unified signalling system is possible to provide all existing and
envisage services as required under the integrated services Digital Network
(ISDN).
vi.
Modem 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.
vii.
Traffic engineering becomes more efficient. The speech circuits

4.5

requirements will go down because of substantial reduction of ineffective traffic.

This advantage itself is sufficient to make additional cost of signalling 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

4.51

As in CCS more processing of the signalling is required, the

cost
of hardware and software for the signalling interface will be more. In addition to this,
there would be following constraints of the network.
i.
As a single data link carries signalling information of a large number of
speech circuits, its failure would result in immobilisation of all these speech
circuits.

5.0

Distribution of interface functions

5.1

Signalling functions are distributed between subscriber line units and

junctures according to the nature of the path set up through the switching network.

5.2

Electromechanical switching networks

As ringing current and power feed can pass through the metallic switching network, their
distribution is centralised at the junctor level for all lines. The lines units are employed
only when the lines are idle and during PG conditions. The line units are disconnected
once the lines are connected to a junctor.
Thus the functions of line unit are power feed to idle line, and detection of significant
event, e.g., off hook and end of PG condition. The functions of a junctor are power feed
during speech phase, ringing current connections and tripping, loop supervision,
transmission of tones and recorded announcements and carrying out inter exchange
signaling.

5.3

Electronic Switching Networks.

As the ringing current and power feed current cannot pass through the electronic
crosspoints, they must be connected at the line unit level only. Moreover in view of the
electronic nature of the switching network, lines and trunks must be isolated to prevent
over voltages damaging the exchange equipment.
Hence, the line unit has to provide additional function as viz, power feed to the line
regardless of staus loops supervision, and ringing current connection and tripping. This
results in considerable simplification of design of junctor whose internal functions can be
totally eliminated by using dedicated tone junctors to transmit tones.

6.0

Conclusion

6.1

Looking back over the years, it can be seen that there has been substantial

increase in the services, provided by the telecommunication network. The signalling

system had to grow along with to ensure efficient provision of these services. With the
introduction of computers in the field of telecommunications, new vistas of services have
opened up. The signalling system is also comming abreast to make these services a
reality.

Introduction to Latest Switches in Telecommunication

1. CDoT :
C-DOT MAX-XL : SYSTEM AND SUBSCRIBER FEATURES
1.
GENERAL
C-DOT DSS MAX is a universal digital switch which can be configured for different
applications as local, transit, or integrated local and transit switch. High traffic/load
handling capacity upto 8,00,000 BHCA with termination capacity of 40,000 Lines as
Local Exchange or 15,000 trunks as Trunk Automatic Exchange, the C-DOT DSS family
is ideally placed to meet the different requirements of any integrated digital network.
2.
BASIC GROWTH/BUILDING MODULES
C-DOT DSS MAX exchanges can be configured using four basic modules (Fig. 2.1)
a.
Base Module
b.
Central Module
c.
Administrative Module
d.
Input Output Module
The Base Module (BM) is the basic growth unit of the system. It interfaces the
external world to the switch. The interfaces may be subscriber lines, analogue and digital
trunks, CCM and PBX lines. Each Base Module can interface upto 2024 terminations. The
number of Base Modules directly corresponds to the exchange size. It carries out majority of
call processing functions and, in a small-exchange application, it also carries out operation
and maintenance functions with the help of the Input Output Module.
In Single Base Module (SBM) exchange configuration, the Base Module acts as an
independent switching system and provides connections to 1500 lines and 128 trunks. In such
a configuration, the Base Module directly interfaces with the Input Output Module for bulk
data storage, operations and maintenance functions. Clock and synchronisation is provided
by a source within the Base Module. It is a very useful application for small urban and rural
environments.
With minimum modifications in hardware through only one type of card, a Base. Module
can be remotely located as a Remote Switch Unit (RSU), parented to the main exchange
using PCM links.

Central Module (CM) consists of a message switch and a space switch to provide inter-module
communication and perform voice and data switching between Base Modules. It provides control
message communication between any two Base Modules, and between Base Modules and
Administrative Module for operation and maintenance functions. It also provides clock and
synchronisation on a centralised basis.

Administrative Module (AM) performs system-level resource allocation and processing

function on a centralised basis. It performs all the memory and time intensive call
processing support functions and also administration and maintenance functions. It

communicates with the Base Module via the Central Module. It supports the Input Output
Module for providing man- machine interface. It also supports the Alarm Display Panel
for the audio-visual indication of faults in the system. Input Output Module (IOM) is a
powerful duplex computer system that interfaces various secondary storage devices like
disk drives, cartridge tape drive and floppy drive. It supports printers and upto 8 serial
ports for video display units which are used for man- machine communication interface.
All the bulk data processing and storage is done in this module. Thus, a C-DOT DSS
exchange, depending upon its size and application, consists of Base Modules (maximum
32), Central Module, Administrative Module, Input Output Module and Alarm Display
Panel. The Base Modules can be remotely located or co-located depending on the
requirement.

3.

REMOTE SWITCH UNIT

Remote Switch Unit (RSU) is an integral part of C-DOT DSS architecture. In order to realise a
RSU, the normal BM can be modified for remoting with the host exchange via 2 Mbps digital
links. The number of 2 Mbps links between the Main Exchange and RSU is primarily determined
by the traffic. A maximum 16 PCMs can be provided between a RSU & Main exchange. Analog
and Digital trunk interfaces are also implemented in RSU to support direct parenting of small
exchanges from RSU itself instead of parenting it to the main exchange which will ultimately
save the media required from main exchange. As far as call processing is concerned, RSU is an
autonomous exchange capable of local-call completion. Operation and maintenance functions are
handled by the host exchange. In the event of failure of PCM links, RSU goes into standalone
mode of operation. In case it is not possible to process a call request due to unavailability of links
to the host, the subscriber is connected to appropriate tone or announcement.
During standalone mode of operation, the local and Incoming terminating calls in RSU are
switched and the metering information of all the RSU subscribers is stored in the RSU. It is sent
to the host whenever the PCM links are available again. Only the even numbered BMs can be
configured as RSU i.e. a maximum 16 RSUs are possible in C-DOT DSS MAX-XL and 8 RSUs
in MAX-L.

4.

COMMON CHANNEL SIGNALLING NO. 7 AND ISDN

Common Channel Signalling is pre-requisite to provide any value added service in the network
e.g. Intelligent Network Services, ISDN services. Due to intelligent protocol implementation in
conformance to ITU-T specifications and with the implementation of CCS7 signalling in C-DOT
DSS, it has been made possible to provide the value added services. Also it is possible to
configure the C-DOT DSS as TAX with multiple nodes, connected on CCS7 signalling. ISDN
Services are the most widely used carriers to transport bulk volume of data. With the increasing
use of Internet Access, the use of ISDN interface is likely to go up as it provides the reliable
access to the user at the rate of 64/128 Kbps. In addition to reliable data connection at higher rate,
it integrates computer and Telephone on the single access. In C-DOT DSS, the implementation is
through add-on modules to provide the services in the beginning itself or retrofit as and when
required. This facilitates the network administrator to upgrade the already commissioned
exchanges in future.

5.

REDUNDANCY

To meet the stringent availability requirements, C-DOT DSS employs 'hot standby' technique
for all processor complexes so that in the event of the failure of any one security-block, not
more than 8 subscribers will be affected. Hardware cross-links between processors have been
planned in such a way that even the failure of two dissimilar processors will not affect system
performance. Also, wherever there is no duplication of hardware units, multiple units are
provided to work in a load-sharing mode. In the event of failure of one of the units, other
units will share its load preventing disruption of service. In case of certain service circuits,
n+1 configuration is used for maintaining reliability.
1.
COMMON HARDWARE UNITS
Various hardware units such as controller complexes and message switches have been
standardised for multiple applications. This interchangeability is an important feature of
the system hardware that helps in reducing inventories and increasing system availability.
Some of these standardised units are
2.

Module Control Unit

Module Control Unit is a 16-bit or 32-bit microprocessor complex with associated memory unit.
The same unit can be used as the Base Processor Unit in the Base Module or as the Space Switch
Controller in the Central Module or as the Administrative Processor Unit in the Administrative
Module.

3.

Interface Controller

This is an 8-bit microprocessor based unit with a time-switching network that can be used
to control either terminal interface in the Terminal Unit or service circuit interface in the
Time Switch Unit. In both the cases, its function is to assign time-slots on the 128channel link between the terminals (subscribers, trunks, etc.) and the module time
switch.

4.

Message Switch

Message Switch is implemented as a 32-bit message switch controller which provides upto 38
HDLC/ADLC links for message communication between controllers. In the Base Module, the
message switch can also be implemented as a 16-bit message switch controller and a message
switch device card. In such an implementation, the controller provides upto 22 HDLC/ADLC
links with the help of the device card.

6.
OPTIMISATION
In C-DOT DSS, distribution and centralisation of functions have been optimised. There are
local functions which are entrusted to the growth units, i.e., the Base Modules, for local
switching and interfacing. These functions use resources whose requirement is directly linked
with the number of lines and trunks equipped. These functions are

1.
2.
3.
4.

Terminal Interfacing - interfacing lines, analog and digital trunks, CCM, PBX and
remote digital lines.
Circuit Switching - switching within the Base Module.
Call Processing - majority of call processing functions.
Concentration - for providing upto 2024 subscribers on 512 time-slots.

On the other hand, the functions that are shared globally over the switch are provided by a
central facility which may either be the Central Module or the Administrative Module. These
functions are a)
b)

Inter-module Communication
Inter-BM and BM-AM communication via the Central Module.
Message Switching
Inter-BM and BM-AM control-message communication via the Central Message
Switch in the Central Module.

c)
d)
e)

Resource Allocation
Done by the Administrative Module. THE C-DOT DSS FAMILY
Operations and Maintenance
Bulk data storage by the Input Output Module and man-machine interface
provided by the Administrative Module via the Input Output Module.
Services
Announcements and conference circuits. This approach is also followed while
introducing new services and facilities in order to utilise them most optimally.

INTRODUCTION TO C-DOT
1.1.1. PHILOSOPHY AND GROWTH OF C-DOT
The Centre for Development of Telematics (C-DOT) is the Telecom Technology
development centre of the Government of India. It was established in August 1984 as an
autonomous body. It was vested with full authority and total flexibility to develop state-ofthe-art telecommunication technology to meet the needs of the Indian telecommunication
network. The key objective was to build a centre for excellence in the area of telecom
technology. While the initial mandate of C-DOT in 1984 was to design and develop digital
exchanges and facilitate their large scale manufacture by the Indian Industry, the
development of transmission equipment was also added to its scope of work in 1989.
1.1.2 DEVELOPMENT OF C-DOT FAMILY OF SWITHES
C-DOT DSS MAX is a universal digital switch which can be configured for different
applications as local, transit, or integrated local and transit switch. High traffic/load handling
capacity upto 8,00,000 BHCA with termination capacity of 40,000 Lines as Local Exchange
or 15,000 trunks as Trunk Automatic Exchange, the C-DOT DSS family is ideally placed to
meet the different requirements of any integrated digital network.
The design of C-DOT DSS MAX has envisaged a family concept. The advantages of family
concept are standardised components, commonality in hardware, documentation, training,
installation and field support for all products and minimization of inventory of spares. Infact
this modular design has been consciously achieved by employing appropriate hardware,
software, and equipment practices.
The equipment practices provide modular packaging. Common cards and advanced
components have been used in the system hardware in order to reduce the number and type
of cards. Standard cards, racks, frames, cabinets and distribution frames are used which
facilitate flexible system growth. Interconnection technology has been standardised at all
levels of equipment packaging. All these features, together with ruggedised design, make CDOT DSS MAX easy to maintain and highly reliable.
Another important feature of the design is the provision of both local and centralised
operation and maintenance. Beginning with local operation and maintenance, with the

installation of similar digital switches in the network, centralised operation and maintenance
will provide maintenance and administration services very economically. All these services
are provided through a simple, interactive man-machine interface.

1.1.3. ARCHITECTURE
C-DOT DSS is a modular and flexible digital switching system which provides economical
means of serving metropolitan, urban, and rural environments. It incorporates all important
features and mandatory services, required by the user with option of upgradation to add new
features and services in future. The architecture for the C-DOT DSS is such that it is possible
to upgrade a working C-DOT SBM or MBM Exchange to provide ISDN service by adding
minimum additional hardware modules while retaining existing hardware units. Another
factor of the architecture is to support ISDN subscribers through Remote Switching Unit
(RSU). This remote switching unit is able to provide switching facility locally even in case of
failure of the communication path to the parent exchange. The system employs an openended architecture for flexibility of configuration and growth. The processor architecture is
characterised by distributed control and message-based communication in order to achieve a
loosely-coupled network for flexible system architecture.
Software is written in high level language 'C and distributed over various processors and is
structured as a hierarchy of virtual machines. The software is packaged such that, depending
upon the actual switch configuration, it can be distributed over appropriate controllers. The
software features are implemented by communicating processes. The operating system
provides message communication facilities such that the processes are transparent to their
physical locations.
For inter-processor communication, messages are exchanged over HDLC links that are
implemented either as direct links or switched network paths. This approach hides the
physical details of processes from each other and provides a flexible communication network
between the processors. New modules can be added and existing modules can be modified
without affecting other modules in the system.
Resources are identified as 'global' or 'local' depending upon their distribution in the system.
The resources which depend upon the number of terminals are provided within the basic
growth unit, the Base Module. Base processors are provided for handling call processing
locally. In a small system application, these processors independently support call processing,
exchange operation and maintenance functions.
On the other hand, in order to avoid replication of large data and memory intensive functions,
some features and facilities are provided centrally. Program backup, bulk data storage, manmachine interface, operations and maintenance facilities are therefore provided centrally in
order to provide a means of separating the switch from the operations and maintenance
interface.
1.2.1. TECHNOLOGY

The system employs a T-S-T switching configuration and is based on a 32-channel PCM
structure. It uses a basic rate of 64Kbps and 2Mbps primary multiplexing rate. Control is
distributed over the system by using 32-bit, 16-bit and 8-bit microprocessors. All the critical
control circuitry has built-in redundancy.
System hardware utilises advanced concepts in micro electronics for a compact and optimum
design. Basic memory unit has been implemented as a 16MB dynamic RAM board. Singlechip digital signal processors are used for implementing DTMF and MF receivers. A high
performance, high density VLSI chip detects multiple tones and simultaneously performs
signal filtering on four channels. This approach reduces costs, power dissipation and saves
space on the PCBs.
Analog to digital conversion on the line circuits has been achieved by using a per channel
coder-decoder (CODEC) chip. Customisation based on ASICS/ FPGAs has been used to
optimize space utilisation and reduce the number of components on the line cards.
1.2.2. BASIC GROWTH/BUILDING MODULES
C-DOT DSS MAX exchanges can be configured using four basic modules (Fig. 1.1)
a

Base Module

Central Module

Administrative Module

Input Output Module

The Base Module (BM) is the basic growth unit of the system. It interfaces the external
world to the switch. The interfaces may be subscriber lines including CCM and PBX lines,
analog and digital trunks. Each Base Module can interface upto 2024 terminations. The
number of Base Modules directly corresponds to the exchange size. It carries out majority of
call processing functions and, in a small-exchange application, it also carries out operation
and maintenance functions with the help of the Input Output Module.
In Single Base Module (SBM) exchange configuration, the Base Module acts as an
independent switching system and provides connections to 1500 lines and 128 trunks. In such
a configuration, the Base Module directly interfaces with the Input Output Module for bulk
data storage, operations and maintenance functions. Clock and synchronisation is provided
by a source within the Base Module. It is a very useful application for small urban and rural
environments.
With minimum modifications in hardware through only one type of card, a Base Module can
be remotely located as a Remote Switch Unit (RSU), parented to the main exchange using
PCM links.
Central Module (CM) consists of a message switch and a space switch to provide intermodule communication and perform voice and data switching between Base Modules. It
provides control message communication between any two Base Modules, and between Base
Module and Administrative Module for operation and maintenance functions. It also provides
clock and synchronisation on a centralized basis.

Administrative Module (AM) performs system-level resource allocation and processing

function on a centralised basis. It performs all the memory and time intensive call processing
support functions and also administration and maintenance functions. It communicates with
the Base Module via the Central Module. It supports

FIG 1.1

the Input Output Module for providing man- machine interface. It also supports the Alarm
Display Panel for the audio-visual indication of faults in the system.
Input Output Module (IOM) is a powerful duplex computer system that interfaces various
secondary storage devices like disk drives, cartridge tape drive and floppy drive. It supports
printers and upto 8 serial ports for video display units which are used for man- machine
communication interface. All the bulk data processing and storage is done in this module.
Thus, a C-DOT DSS exchange, depending upon its size and application, consists of Base
Modules (maximum 32), Central Module, Administrative Module, Input Output Module and
Alarm Display Panel. The Base Modules can be remotely located or co-located depending on
the requirement.
Remote Switch Unit (RSU) is an integral part of C-DOT DSS architecture. In order to realise
a RSU, the normal BM can be modified for remoting with the host exchange via 2 Mbps
digital links. The number of 2 Mbps links between the Main Exchange and RSU is primarily
determined by the traffic. A maximum 16 PCMs can be provided between a RSU & Main

exchange. Analog and Digital trunk interfaces are also implemented in RSU to support direct
parenting of small exchanges from RSU itself instead of parenting it to the main exchange
which will ultimately save the media required from main exchange. As far as call processing
is concerned, RSU is an autonomous exchange capable of local-call completion. Operation
and maintenance functions are handled by the host exchange. In the event of failure of PCM
links, RSU goes into stand alone mode of operation. In case it is not possible to process a call
request due to unavailability of links to the host, the subscriber is connected to appropriate
tone or announcement.
During stand alone mode of operation, the local and incoming terminating calls in RSU are
switched and the metering information of all the RSU subscribers is stored in the RSU. It is
sent to the host whenever the PCM links are restored.
Only the even numbered BMs can be configured as RSU i.e. a maximum 16 RSUs are
possible in C-DOT DSS MAX-XL and 8 RSUs in MAX-L.
1.3.1. VALUE ADDED SERVICES:Common Channel Signalling is pre-requisite to provide any value added service in the
network e.g. Intelligent Network Services, ISDN services. Due to intelligent protocol
implementation in conformance to ITU-T specifications and with the implementation of
CCS7 signalling in C-DOT DSS, it has been made possible to provide the value added
services. Also, it is possible to configure the C-DOT DSS as TAX with multiple nodes,
connected on CCS7 signalling.
ISDN Services are the most widely used carriers to transport bulk volume of data. With the
increasing use of Internet Access, the use of ISDN interface is likely to go up as it provides
the reliable access to the user at the rate of 64/128 Kbps. In addition to reliable data
connection at higher rate, it integrates computer and Telephone on the single access.
To support V5.x interface in C-DOT Digital Switching System, a new hardware unit called
VU (V5 Interface unit) is required. All the layer 2 and layer 3 software for V5 interface
resides in this unit. VU works in conjunction with DTU, which in turn extends the 2.048
Mbps digital link (E1) towards Access Network.
In C-DOT DSS, the implementation is through add-on modules to provide the services in the
beginning itself or retrofit as and when required. This facilitates the network administrator to
upgrade the already commissioned exchanges in future.
1.3.2.1. REDUNDANCY
To meet the stringent availability requirements, C-DOT DSS employs 'hot standby' technique
for all processor complexes so that in the event of the failure of any one security-block, not
more than 8 subscribers will be affected.
Hardware cross-links between processors have been planned in such a way that even the
failure of two dissimilar processors will not affect system performance. Also, wherever there
is no duplication of hardware units, multiple units are provided to work in a load-sharing

mode. In the event of failure of one of the units, other units will share the failed units load
preventing disruption of service. In case of certain service circuits, n+1 configuration is used
for maintaining reliability.
1.3.2.2. OPTIMISATION
In C-DOT DSS, distribution and centralisation of functions have been optimised. There are
local functions which are entrusted to the growth units, i.e., the Base Modules, for local
switching and interfacing. These functions use resources whose requirement is directly linked
with the number of lines and trunks equipped.
The local functions are
Terminal Interfacing - interfacing lines, analog and digital trunks, CCM, PBX and remote
digital lines.

Circuit Switching - switching within the Base Module.

Call Processing - majority of call processing functions.

Concentration - for providing upto 2024 subscribers on 512 time-slots.

On the other hand, the functions that are shared globally over the switch are provided by a
central facility which may either be the Central Module or the Administrative Module. These
functions are

Inter-module Communication
Inter-BM and BM-AM communication via the Central Module.

Message Switching
Inter-BM and BM-AM control-message communication via the Central Message
Switch in the Central Module.

Resource Allocation
Done by the Administrative Module.

Operations and Maintenance

Bulk data storage by the Input Output Module and man-machine interface provided
the Administrative Module via the Input Output Module.

by

Services
Announcements and conference circuits.

This approach is also followed while introducing new services and facilities in order to utilise
them most optimally.
1.3.2.3 COMMON HARDWARE UNITS

Various hardware units such as controller complexes and message switches have been
standardised for multiple applications. This interchangeability is an important feature of the
system hardware that helps in reducing inventories and increasing system availability. Some
of these standardised units are

Module Control Unit

Module Control Unit is a 16-bit or 32-bit microprocessor complex with associated
memory unit. The same unit can be used as the Base Processor Unit in the Base Module
or as the Space Switch Controller in the Central Module or as the Administrative
Processor Unit in the Administrative Module.

Interface Controller
This is an 8-bit microprocessor based unit with a time-switching network that can be used
to control either terminal interface in the Terminal Unit or service circuit interface in the
Time Switch Unit. In both the cases, its function is to assign time-slots on the 128channel link between the terminals (subscribers, trunks, etc.) and the module time switch.

Message Switch
Message Switch is implemented as a 32-bit message switch controller which provides
upto 38 HDLC/ADLC links for message communication between controllers. In the Base
Module, the message switch can also be implemented as a 16-bit message switch
controller and a message switch device card. In such an implementation, the controller
provides up to 22 HDLC/ADLC links with the help of the device card.

CHAPTER 2

ARCHITECTURE
2.1.1. PCM PRINCIPLE AND DIGITAL SWITCHING
2.1.1.1 PCM PRINCIPLES
The analog speech coming into the switch- cannot be handled as it is, by the
digital switches like CDOT. It is converted into digital form of 0s and 1s by the
PCM technique. PCM as used in the switch consists of the steps in Coding :
Filtering Sampling Quantising Encoding, by which the analog information
gets converted into digital. By filtering, the speech frequency is restricted to max
4 KHz. As the sampling frequency should be at least twice the cutoff frequency
according to sampling theorem, the analog speech is sampled at 8 KHz or in
other words once in 125 microseconds. Each analog sample (PAM, pulse
amplitude modulation) is quantised & encoded into a digital sample (PCM, of 8
bits). Line coding enables the PCM signal to be sent on the line properly
(example NRZ code converted to HDB3 and sent). Decoding is the reverse
process of converting the digitalised speech back into analog form. CODEC is
the chip which does the combined job of coding and decoding, as part of
BORSCHT functions.
As 3.9 microseconds time is sufficient to handle a channel, 32 channels are
possible in the available time of 125 microseconds. The 125 microseconds
duration is also called time-frame or Frame duration. 3.9 microseconds is the
duration of a time-slot (ts) or channel. This multiplexing done on the basis of
dividing time is termed TDM, Time Division Multiplexing. The PCM in which a
frame has 32 channels is usually called as 30 channel PCM as two time-slots
are reserved, ts0 for Synchronisation i.e. sending/receiving of FAW (Frame
Alignment Word) or Alarm information and ts16 for sending/receiving of
Signalling information. One time-multiframe (MF) has 16 Frames (F0 to F15).
During the 2 milliseconds of a multi-frame, signalling information pertaining to
two speech channels is sent on ts16 of each of its 16 frames except the first.
The ts16 of F0 carries MFAW (Multi Frame Alignment Word). For example, the
digitalised speech belonging to speech channels 15 & 30 are carried in ts15 &
ts31 of each frame; but their signalling information is carried in ts16 of F15 of
every MF. Similarly, ts16 of F4 of each MF carries the signalling information of

speech channels 4 & 19 ; the 8 bits of speech of channel 4 is carried on ts4 of

each frame & 8 bits of speech of channel 19 on ts20 of each frame. The ts0 of
every frame alternately carry FAW and ALM information.
Therefore in one second 500 MF are sent.
Hence the number of bits sent in one second
= (500 MF) X (16 F) X (32 ts) X (8 bits)
= 2048000 bits

( or 2048 Kbps or 2.048 Mbps or simply 2 Mbps.)

This digital stream of 2Mbps is also called basic or first order PCM. One bidirectional PCM link basically caters for 30 speech channels and is further
multiplexed into higher orders such as: 8 Mb (120 channels), 34 Mb (480
channels),140 Mb (1920 channels) and so on.
In Channel Associated Signalling (CAS), ts16 is distinguished as the signalling
time-slot and is rigidly associated with the function of carrying the signalling
information of the 30 speech channels of its own PCM Link only. The 64 Kbps
bit carrying capacity of the CAS-ts i.e ts16 is not fully utilised. The advanced
form of digital signalling namely Common Channel Signalling (CCS) overcomes
the drawbacks of CAS type of digital signalling. Here the CCS-ts @ 64 kbps is
fully utilised and it carries the signalling information of as many PCM links
with the help of different HW & SW. We can have CCS signalling between two
CDOT MBM exchanges as both support it. But we can have only CAS between a
CDOT MBM and a CDOT 256 as CDOT 256 does not support CCS but supports
only CAS.

2.1.1.2. DIGITAL SWITCHING

Switching is a process wherein information is transferred from I/P to O/P. There are two
types of switching:
1. Analog switching: In analog switching the information will be transferred in an analog
form i.e. frequency or earth of battery etc. An input is connected to an o/p by means of a
physical path inside the switching network. This physical path is engaged throughout the
conversation and as such this path cannot be utilized to put through some other call,
resulting in poor utilization.
2. Digital switching: the information is transferred in terms of binary i.e. 0s and 1s.
Digital switching adopts Time Division and PCM techniques. In this case, the same path
is shared by 30 different subscribers in different time domains, resulting in maximum
utilization.

2.1.2. TYPES OF DIGITAL SWITCHING

Digital switching is of two types:
1. Time switching.
2. Space switching
Time switching: the time slot nos will change, while PCM highway no. remains the
same. For ex TS4 PCM0
TS7 PCM0. Further, there are two types of time
switches:
a) Output associated time switch.
b) I/p associated time switch.
Whatever may be the type of switch, two memories are available:
1. Buffer memory.
2. Control or command memory.
Output associated time switch: in this case, as the information of the 32 time slots
pertaining to a PCM highway arrives ,the processor writes it into the corresponding
memory locations.
In our example, TS4 PCM 0 information (8 bit infn) will be written into the
corresponding memory location no. 4 of buffer memory. The command memory receives
the read address cycle of o/g PCM time slots. Here also, there are 32 memory locations.
When the processor comes to memory location no. 7 of command memory, it writes the
address of buffer memory area i.e. 4 in the binary language.
During the next read cycle when the processor comes to memory location no. 7 of CM it
fetches the address of buffer memory as 4 and goes to memory location no. 4 of BM and
reads the 8 bit contents.
B.M

B.M
TS4 PCM0

TS7 PCM0

TS4
PCM0

0
7

31

I/p associated
T.S writing is
controlled.
Reading is
sequential.

-O/p associated T.S writing is sequential.

- is controlled.

31
0

0
100

111
B.M

C.M
31

TS7
PCM0

31

And as such TS4 PCM0 information will be shifted to TS7 PCM0. Here writing is
sequential whereas reading is controlled
I/P associated time switch: let us take the same example
i.e.TS4 PCM0
TS7PCM0. Since the processor knows to which TS no. the
information should be switched, as it receives the TS4 infn in the I/C PCM highway it
writes that information (8 bits) into memory location no 7 of buffer memory.
The control memory receives the read address cycle of 32 time slots pertaining to I/C
PCM highway. When it comes to memory location no 4 the processor writes the buffer
memory address as 7 (i.e. 111) in binary bits.
During the next read cycle when it comes to memory location no 4 of command memory,
it fetches the address of BM area as 7 and as such it goes to location no 7 of BM area and
reads the 8 bit content and thus, TS4 PCM0 TS4 PCM0. Here, writing is controlled
but reading is sequential.
Space switching: in space switching TS no. remains the same,. While the PCM highway
no changes. Let us take as an example
TS4 PCM0

TS4 PCM2

Consider an mXn dimensional space switching network i.e. m I/C PCM highways and n
O/G PCM highways ,connected to the switching network. n - O/G PCM highways are
connected to the switching n/w.

TS4
PCM0

TS7 PCM2

n-1

Pcm 0

LOGIC 1

PCM m-1

CM0

CM1

CM2

The space switch consists of command memories which are equal to no of O/G PCM
highways. In our example, since n - O/G PCM highways are available, there will be n
command memories. In the above example, since O/G PCM highway is no. 2, the
processor picks up command memory no 2. In each command memory 32 memory
locations are available. i.e. 1/TS. Since we are interested in TS4, when the processor
comes to memory location no 4 of command memory 2, it writes the address of I/C PCM
highway no. i.e. 0. As such, logic 1 will be placed on the corresponding AND gate for a
period of 3.9 sec and the 8 bit information of TS4 PCM0 will be shifted to TS4 PCM2.
By employing both time and space switches double connections will be established and
the two subs. can converse with each other.

Double connection means

Cgp trans T.S-------------- cdp receive TS

Cdp trans T.S-------------- cgp receive TS
As an example if cgp and cgp are allotted with time slots TS 5PCM2 and TS 20 PCM 53,
then nature of double connections will be:
TS 5 T PCM2
TS 20 R PCM 53
Cgp trans T.S-------------- cdp receive TS
TS 20 T PCM 53
TS 5 R PCM2
Cdp trans T.S-------------- cgp receive TS
2.2.1. BASE MODULE HARDWARE ARCHITECTURE- TU
Base Module (BM) is the basic building block of C-DOT DSS MAX. It interfaces the
subscribers, trunks and special circuits. The subscribers may be individual or grouped PBX
lines, analog or digital lines. The trunks may be Two Wire Physical, E&M Four Wire, E&M
Two Wire, Digital CAS or CCS. The basic functions of a Base Module are

Analog to digital conversion of all signals on analog lines and trunks

Interface to digital trunks and digital subscribers

Switching the calls between terminals connected to the same Base Module

Communication with the Administrative Module via the Central Module for
administrative and maintenance functions and also for majority of inter-BM switching
(i.e. call processing) functions

Provision of special circuits for call processing support e.g. digital tones, announcements,
MF/DTMF senders/receivers

Provision for local switching and metering in stand alone mode of Remote Switch Unit as
well as in case of Single Base Module Exchange (SBM-RAX)

For these functions, the Base Module hardware is spread over different types of terminal and
other units

Analog Terminal Unit - to interface analog lines/trunks, and providing special circuits as
conference, announcements and terminal tester.

Digital Terminal Unit - for interfacing digital trunks i.e. 2Mbps E-1/PCM links

#7 Signalling Unit Module - to support SS7 protocol handlers and some call processing
functions for CCS7 calls.

1.1.

ISDN Terminal Unit - to support termination of BRI/PRI interfaces and implementation

of lower layers of DSS1 signalling protocol.

V5.x Terminal Unit to support access network through V5.x interface.

Time Switch Unit - for voice and message switching and provision of service circuits.

Base Processor Unit - for control message communication and call processing functions.

Analog Terminal Unit (ATU) Figure 2.1

The Analog terminal unit (ATU) is used for interfacing 128 Analog terminations which may
be lines or trunks. It consists of terminal cards which may be a combination of Line Circuit
Cards (LCC), CCB with Metering (CCM) cards, Two Wire Trunk (TWT) cards, E&M Two
wire (EMT) Trunk cards and E&M Four wire (EMF) trunk cards, depending upon the
module configuration. Also, provision has been made to equip Conference (CNF) card to
support six party conference, Announcement (ANN) to support 15 user friendly
announcement messages, and Terminal Test Controller (TTC) for testing of analog
terminations. Power Supply Unit (PSU-I) provides logical voltages and ringing current in the
ATU.
The Analog Terminal Unit (ATU) is used for interfacing 128 analog terminations
2.2.1.2 Digital Terminal Unit (DTU)
Digital Terminal Unit (DTU) is used exclusively to interface digital trunks. One set of Digital
Trunk Synchronization (DTS) card along with the Digital Trunk Controller performed by the
Terminal Unit. Controller (TUC) in the Digital Terminal Unit.

2.2.1.4 ISDN - Terminal

Unit (ISTU)
One of the four
ATUs/DTUs in a BM
can be replaced by
ISTU
to
provide
BRI/PRI interfaces in
C-DOT DSS. The only
constraint is that ISTU
has to be principal TU
i.e. directly connected
to TSU on 8 Mbps
PCM
link.
The
ATU/DTU cannot be

FIG 2.1

used in concentration with ISTU. By equipping one ISTU in the exchange, a max. of 256 B
channels are available to the administrator which can be configured as BRI, PRI or any mix,
as per site requirement. Depending on the requirement of number of ISDN-Interfaces, one or
more ISTUs can be integrated in C-DOT DSS, either in one BM or distributed across
different BMs.
2.2.1.5 V5.x - Unit (VU)
One of the four ATUs/DTUs in a BM can be replaced by VU to provide V5.x interfaces in CDOT DSS. Hardware architecture of VU (V5 unit) is same as that of SU (SS7 unit). SU
contains software for SS7 signalling whereas VU contains software for V5 interface.

2.2.2. BM HARDWARE ARCHITECTURE- TSU & BPU

2.2.2.1. Time Switch Unit (TSU)
Time Switch Unit (TSU) implements three basic functions as time switching within the Base
Module, routing of control-messages within the Base Module and across Base Modules and
support services like MF/DTMF circuits, answering circuits, tones, etc. These functions are
performed by three different functional units, integrated as time switch unit in a single frame
(refer Fig. 2.2).
Service Unit (SU)
Service Unit is integrated around three different cards as Tone Generator with Answering
Circuit (TGA), Service Circuit Interface Controller (SCIC). and MF/DTMF Controller
(MFC) Card. MF/DTMF circuits (senders/receivers) are implemented by using single-chip,
4-channel Digital Signal Processors (DSPs). Two MFC cards are grouped to form a terminal
group. Upto four MFC Cards can be equipped. The TGA and two groups of MFCs, form
three terminal groups towards the Service Circuits Interface (SCI). Service Circuit Interface
multiplexes these three TGs together with another terminal group from the Base Message
Switch (BMS) to form a 128-channel, 8Mbps link under the control of Service Circuits
Interface Controller (SCIC) and sends it towards the Time Switch.
Base Message Switch (BMS)
Base Message Switch (BMS) routes the control messages within the Base Module, across
different Base Modules, and also Administrative Module via the Central Module. It is
implemented around two different cards as Message Switch Controller (MSC) with six direct
HDLC-links and the Message Switch Device (MSD) Card implementing 16 switched HDLC
links. As a unit, total 22 HDLC channels are implemented for communication with the Base
Processor, Time Switch Controller, Service Circuits Interface Controller, Terminal Interface
Controller within the BM and the four CMS complexes in CM. It acts as a message transfer
point between the Base Processor and these controllers. It receives messages from the Base
Processor and transmits them towards the appropriate controllers.
Time Switch (TS)
The Time Switch complex is implemented using three different functional cards as
multiplexer/demultiplexer (TSM), time switch (TSS) and time switch controller (TSC). The
Time Switch complex receives the following PCM links and performs time- switching on
them for switching within the Base Module :

Four 128-channel multiplexed links from four different Terminal Units which may be
any combination of. ATU, DTU, #7SU and ISTU.

One 128-channel multiplexed BUS from the Service Circuits Interface Controller
(SCIC) in the Time Switch Unit.

Three 128-channel links to support onboard three party conference circuits (3 x 128).

FIG 2.2

It multiplexes above 128-channel links to form a dual 512-channel, 4 Mbps multiplexed bus
towards the Central Module. The individual buses are called Bus0 and Bus1. Besides this, it
also provides network switched path for message communication between Base Modules,
between Base Module and Administrative Module, and between Base Module and Central
Module.
2.2.2.2. Base Processor Unit (BPU)
Base Processor Unit (BPU) is the master controller in the Base Module. It is implemented as
a duplicated controller with memory units. These duplicated sub-units are realised in the
form of the following cards:

Base Processor Controller (BPC) Card

Base Memory Extender (BME) Card

BPC controls time-switching within the Base Module via the Base Message Switch and the
Time Switch Controller. It communicates with the Administrative Processor via Base
Message Switch for operations and maintenance functions. In a SBM configuration, BPC
directly interfaces with the Alarm Display Panel and the Input Output Module.
Figure 2.3 summarises the various units and sub-units of the Base Module.
2.2.3. CENTRAL MODULE (CM) HARDWARE ARCHITECTURE
Central Module (CM) is responsible for space switching of inter-Base Module calls,
communication between Base Modules and the Administrative Module, clock distribution
and network synchronisation. For these functions, Central Module has a Space Switch, Space
Switch Controller and a Central Message Switch.
CM provides connectivity to 16 BMs if it is CM-L and 32 BMs if it is CM-XL. Each BM
interfaces with CM via two 512-channel parallel buses as BUS-0 and BUS-1, each operating
at 4 Mbps. These buses carry voice information of 512 terminations of the Base Module
towards CM. In the reverse direction, after space switching has been done in the Space
Switch under the control of Space Switch Controller (SSC), the same buses carry the
switched voice information for 512 terminations towards BM. Thus, in a 32 Base Module
configuration, there are 64 parallel buses carrying the voice information from Base Modules
to the Central Module, and also the switched voice information in the reverse direction.
2.2.3.1. Space Switch (SS) and Space Switch Controller (SSC)
In order to take care of the large number of interface signals, the switch portion of CM is
divided into three stages viz. MUX stage, Switch stage and DEMUX stage. The MUX and
DEMUX stages are implemented on single card to provide the Base Module to Central
Module interface in each direction. Interfacing and switching are controlled by SSC which
provides control signals for the MUX/DEMUX cards and the Space Switch Switch cards.
Interconnection between MUX/DEMUX cards and the Space Switch is shown in Figure 2.4.
MUX/DEMUX Cards extract the information from time-slots 0 and 1 of Bus0 and Bus1 from
the Base Modules. These time-slots carry control message from each Base Module and these

messages are sent to the Central Message Switch (CMS). The CMS sends these messages to
the Space Switch Controller (SSC) on a 128 kbps link to control space switching based upon
this information.
Four 512-channel buses from four BMs are multiplexed to form a 2048- channel, 16 Mbps
multiplexed BUS which is sent to both copies of the Space Switch Switch Card. Space
switching of these 2048 channels is done based upon the switching information received by
Space Switch Controller (SSC) from CMS.
Clock Distribution
CM provides the central clock for distribution to the Base Modules. The 8MHz clock
may be locally generated at the Central Clock (CCK) card in case of CM-XL and Space
Switch Clock (SCK) card in case of CM-L by using high stability VCXO crystal or may
be derived from an external reference clock using the Network Synchronisation
Controller (NSC) card in case of CM-XL and Network Synchronisation Equipment
(NSE) in case CM-L, under the control of SSC. In the event of failure of external
reference or duplex failure of the NSC cards/NSE, the local clock is fed in the hold-over
mode, synchronised to last reference value. In any arrangement, the local or external
clock is distributed via Central Bus Extender (CBX) cards in case of CM-XL.
The CBX card provides an interface between SSC and SSU. SSC makes any switch card
access through CBX. CBX also handles any power supply errors in SSU and BTU. Each
CCK-CBX-NSC complex form a security block i.e. CBX0 cannot be used with CCK1.
Thus there is a copy 0 complex and a copy 1 complex. The CBX also synchronises all
SSC accesses to SSU with the 16 MHz clock as well as BTU. Fig. 2.5 depicts the clock
distribution in C-DOT DSS with CM-XL.
2.2.3.2. Central Message Switch (CMS)
Central Message Switch (CMS) complex is the central message transfer point of the switch.
It is implemented as four different message switches, working in load-sharing mode. Each
message switch is a high performance message routing block, implemented by using high
speed 32 bit microprocessor MC 68040 in case of CM-XL and 16 bit microprocessor MC
68000 in case of CM-L. This card supports 38 HDLC links in case of CM-XL with flexibility
of programming individual HDLC links up to 750 kbps. All Central Message Switches
(CMS1, 2, 3&4) are used for routing of messages across the Base Modules. On the other
hand only CMS1 and CMS2 interface with the Administrative Module for routing control
message between Base Processors and Administrative Processor. This communication is used
to access office data for routing inter- module calls and administration and maintenance
functions. Fig. 2.6 depicts the Central Message Switch in C-DOT DSS.

FIG 2.4

FIG 2.3

2.2.4. ADMINISTRATIVE MODULE (AM) HARDWARE ARCHITECTURE

Administrative Module (AM) consists of a duplicated 16/32-bit controller called the
Administrative Processor (APC). It communicates with Base Processors via the Central
Message Switch for control messages and with the duplicated Input Output Processors in the
Input Output Module for interfacing peripheral devices Administrative processor is
responsible for global routing, translation, resource allocation and all other functions that are
provided centrally in C-DOT DSS MAX. The implementation of AM is similar to Base
Processor Complex of BM, using the same hardware configuration. As explained earlier,
HPC instead of BPC is used to support 8,00,000 BHCA.
2.2.5. INPUT OUTPUT MODULE (IOM)
Input Output Module (IOM) consists of duplicated Input Output Processor (IOP). The Input
Output Processor (IOP) is a general purpose computer with UNIX Operating System. It is
used as the front end processor in C-DOT DSS. It handles all the input and output functions
in C-DOT DSS. The IOP is connected to AP/BP via HDLC links. During normal operation,
two IOPs interconnected by a HDLC link, operate in a duplex configuration. Working as
front end processor, it provides initial code down load to the subsystems, man machine
interface and data storage for billing and other administrative information. Refer Fig. 2.7 for
IOP connectivity in the system and IOP-VH architecture.
2.2.5.1. IOP-VH Hardware Architecture
The IOP-VH is value engineered high performance IOP, designed using a single card. The
IOP CPU uses MC 68040 (25 MHz) processor on the VHC card. The IOP as a module is
duplicated to provide redundancy for cartridge and disk drives as well as serial
communication terminals and printers.
The system has provision for 7 HDLC channels. Two of these are used to connect the IOP to
both the copies of AP/BP. The third link is for connection with mate IOP when the two are
working in synchronisation i.e. duplex IOP configuration. The rest four links are spare at
present but may be used towards the four CMSs in future. Eight of RS-232C Serial Links
(through ASIO ports) are also implemented for connecting operator terminals and printer to
the IOP in addition to two ports as Console and Host.

FIG 2.6

The monitor based operations are performed only from the Console and the same is true in
case of login to root account. The operations like initial bootup, software link loading etc.
could be performed only from the Console. One X.25 port is implemented for 64Kbps full
duplex link to communicate with Centralise
FIG 2.5Billing/Telecom Management Network Centre.
In addition, one 10 Mbps Ethernet port is also implemented in the IOP-VH which has AUI or
Coaxial interface support at physical level to allow networking of user terminals in future. A
SCSI-2 controller with integrated DMA and SCSI cores is used for interfacing the disk drive
and cartridge tape drive.

FIG 2.7

HARDWARE ARCHITECTURE OF CDOT MBM

The hardware architecture of C-DOT DSS MAX is mapped closely on the system
overview described in the previous chapter. In the following sections, the hardware
architecture of each constituent module is described.
BASE MODULE (BM)
Base Module (BM) is the basic building block of C-DOT DSS MAX. It interfaces the
subscribers, trunks and special circuits. The subscribers may be individual or grouped PBX lines,
analog or digital lines. The trunks may be Two Wire Physical, E&M Four Wire, E&M Two Wire,
Digital CAS or CCS. The basic functions of a Base Module are

1.
2.
3.
4.
5.

Analog to digital conversion of all signals on analog lines and trunks

Interface to digital trunks and digital subscribers
Switching the calls between terminals connected to the same Base Module
Communication with the Administrative Module via the Central Module for
administrative and maintenance functions and also for majority of inter-BM
switching (i.e. call processing) functions
Provision of special circuits for call processing support e.g. digital tones,
announcements, MF/DTMF senders/receivers
6. Provision for local switching and metering in stand alone mode of Remote Switch
Unit as well as in case of Single Base Module Exchange (SBM-RAX) For these
functions, the Base Module hardware is spread over different types of units.

HARDWARE ARCHITECTURE
(a)
Analog Terminal Unit - to interface analog lines/trunks, and providing special
circuits as conference, announcements and terminal tester.
(b)
Digital Terminal Unit - for interfacing digital trunks i.e. 2Mbps E-1/PCM links
(c )
Signalling Unit Module - to support SS7 protocol handlers and some call
processing functions for CCS7 calls.
d)

ISDN Terminal Unit - to support termination of BRI/PRI interfaces and

implementation of lower layers of DSS1 signalling protocol.
e)
Time Switch Unit - for voice and message switching and provision of service
circuits.
f)
Base Processor Unit - for control message communication and call processing
functions.
nalog Terminal Unit (ATU) Figure 3.1
The Analog Terminal Unit (ATU) is used for interfacing 128 analog terminations which may
be lines or trunks. It consists of terminal cards which may be a combination of Line Circuit
Cards (LCC), CCB with Metering (CCM) cards, Two Wire Trunk (TWT) cards, E&M Two
wire (EMT) Trunk cards and E&M Four wire (EMF) trunk cards, depending upon the
module configuration. Also, provision has been made to equip Conference (CNF) card to
support six party conference, Announcement (ANN) to support 15 user friendly
announcement messages, and Terminal Test Controller (TTC) for testing of analog
terminations. Power Supply Unit (PSU-I) provides logical voltages and ringing current in the
ATU.

1.

Analog Subscriber Line Cards

Two variants of subscriber line cards as LCC or CCM with interfaces upto 8
subscribers, provide basic BORSCHT functions for each line. Analog to digital
conversion is done by per-channel CODEC according to A-law of Pulse Code
Modulation. Each CCM card has the provision of battery reversal for all the 8
lines with the last two lines having provision to generate 16 KHz metering pulses
to be sent to subscriber's metering equipment.
The 8-bit digital (voice) output of four LCCs is multiplexed to form a 32channel, 2 Mbps PCM link - also called a terminal group (TG). Since a Terminal
Unit has a maximum of 16 terminal cards, there are four such terminal groups.
The signalling information is separated by a scan/drive logic circuit and is sent to
the signalling processor on four different scan/drive signals. The LCC/CCM also
provides test access relay to isolate the exchange side and line side to test it
separately by using the Terminal Test Controller (TTC).

2.

Analog Trunk Cards

Analog trunk cards interface analog inter-exchange trunks which may be of three
types as TWT, EMT and EMF. These interfaces are similar to Subscriber Line
Card, with only difference that the interfaces are designed to scan/drive events on
the trunks as per predefined signalling requirement.
3. Signalling Processor (SP) Card Signalling Processor (SP) processes the
signalling information received from the terminal cards. This signalling
information consists of scan/drive functions like origination detection, answer
detection, digit reception, reversal detection, etc. The validated events are
reported to Terminal Interface Controller for further processing to relieve itself
from real-time intensive functions. Based on the information received from the
Terminal Interface Controller, it also drives the event on the selected terminal
through scan/drive signals.
4. Terminal Interface Controller (TIC) Card
Terminal Interface Controller (TIC) controls the four terminal groups (TG) of 32
channels, and multiplex them to form a duplicated 128-channel, 8 Mbps link
towards the Time Switch (TS). For signalling information of 128- channels, it
communicates with Signalling Processor (SP) to receive/send the signalling event
on analog terminations. It also uses one of the 64 kbps channel out of 128
channels towards Time Switch, to communicate with Base Processor Unit (BPU).
In concentration mode, three other Terminal Units share this 128-channel link
towards the Time Switch to have 4:1 concentration. Terminal Interface Controller
is built around 8-bit microprocessor with associated memory and interface and it
is duplicated for redundancy.

5.

Special Service Cards

A Terminal Unit has some special service cards such as Conference (CNF) Card
to provide six party conference. Speech samples from five parties are added by
inbuilt logic and sent to the sixth party to achieve conferencing. Terminal Test
Controller (TTC) Card is used to test analog terminal interfaces via the test access
relays on the terminal cards. Announcement Controller (ANN) Card provides 15
announcements on broadcast basis. Only one service card of each type is
equipped in a Base Module with provision of fixed slot for TTC and variable slots
for CNF/ANNC. Announcement and Conference Cards are equipped in Terminal
Unit through S/W MMC command. Two slots are occupied by each card i.e. 16
channels for each card are used out of 128 channels available on a Bus between a
TU & TS.

11.

Digital Terminal Unit (DTU)

Digital Terminal Unit (DTU) is used exclusively to interface digital trunks. One
set of Digital Trunk Synchronization (DTS) card alongwith the Digital Chapter 3.
18 C-DOT DSS MAX Trunk Controller (DTC) card is used to provide one E-1
interface. Each interface occupies one TG of 32 channels and four such interfaces
share 4 TGs in a Digital Terminal Unit. The functions performed by TIC and SP in
Analog Terminal Unit, are collectively performed by the Terminal Unit Controller
(TUC) in the Digital Terminal Unit. The scan functions are - HDB3 to NRZ code
conversion, frame alignment and reconstitution of the received frame. The drive
functions include insertion of frame alignment pattern and alignment information.
Each interface can be configured as CAS or CCS interface.

12.

SS7 Signalling Unit Module (SUM)

Any one of the ATU or DTU in a BM can be replaced by SUM frame to support
CCS7 signalling. Only one such unit is equipped in the exchange irrespective of
its configuration or capacity.

13.
ISDN - Terminal Unit (ISTU)
One of the four ATUs/DTUs in a BM can be replaced by ISTU to provide BRI/PRI interfaces
in C-DOT DSS. The only constraint is that ISTU has to be principal TU i.e. directly
connected to TSU on 8 Mbps PCM link. The ATU/DTU cannot be used in concentration with
ISTU. By equipping one ISTU in the exchange, a max. of 256 B channels are available to the
administrator which can be configured as BRI, PRI or any mix as per site requirement.
Depending on the requirement of number of ISDN-Interfaces, one or more ISTUs can be
integrated in C-DOT DSS, either in one BM or distributed across different BMs.
14.
Time Switch Unit (TSU)
Time Switch Unit (TSU) implements three basic functions as time switching within the Base
Module, routing of control-messages within the Base Module and across Base Modules and
support services like MF/DTMF circuits, answering circuits, tones, etc. These functions are
performed by three different functional units, integrated as time switch unit in a single frame
(refer Fig. 3.2).

(1)

Service Unit (SU)

Service Unit is integrated around three different cards as Tone Generator with
Answering Circuit (TGA), Service Circuit Interface Controller (SCIC). and
MF/DTMF Controller (MFC) Card. MF/DTMF circuits (senders/receivers) are
implemented by using single-chip, 4-channel Digital Signal Processors (DSPs).
Two MFC cards are grouped to form a terminal group. Upto four
MFC Cards can be equipped. The TGA and two groups of MFCs, form three
terminal groups towards the Service Circuits Interface (SCI). Service Circuit
Interface multiplexes these three TGs together with another terminal group from
the Base Message Switch (BMS) to form a 128-channel, 8Mbps link under the

control of Service Circuits Interface Controller (SCIC) and sends it towards the
Time Switch.
(2)

Base Message Switch (BMS)

Base Message Switch (BMS) routes the control messages within the Base
Module, across different Base Modules, and also Administrative Module via the
Central Module. It is implemented around two different cards as Message Switch
Controller (MSC) with six direct HDLC-links and the Message Switch Device
(MSD) Card implementing 16 switched HDLC links. As a unit, total 22 HDLC
channels are implemented for communication with the Base Processor, Time
Switch Controller, Service Circuits Interface Controller, Terminal Interface
Controller within the BM and the four CMS complexes in CM. It acts as a
message transfer point between the Base Processor and these controllers. It
receives messages from the Base Processor and transmits them towards the
appropriate controllers.

Note : To support 8,00,000 BHCA, MSC and MSD cards are replaced by a High
performance Message Switch (HMS) with high speed, 32 bit microprocessor (MC
68040). It implements 38 HDLC links with flexibility of programming individual
link for a speed upto 750 kbps.
3.

(a)
(b)
(c)

Time Switch (TS)

The Time Switch complex is implemented using three different functional cards
as multiplexer/demultiplexer (TSM), time switch (TSS) and time switch controller
(TSC). The Time Switch complex receives the following PCM links and performs
time- switching on them for switching within the Base Module :
Four 128-channel multiplexed links from four different Terminal Units which
may be any combination of. ATU, DTU, #7SU and ISTU.
One 128-channel multiplexed BUS from the Service Circuits Interface Controller
(SCIC) in the Time Switch Unit.
Three 128-channel links to support onboard three party conference circuits (3 x
128).
It multiplexes above 128-channel links to form a dual 512-channel, 4 Mbps
multiplexed bus towards the Central Module. The individual buses are called
Bus0 and Bus1. Besides this, it also provides network switched path for message
communication between Base Modules, between Base Module and
Administrative Module, and between Base Module and Central Module.

15.

Base Processor Unit (BPU)

Base Processor Unit (BPU) is the master controller in the Base Module. It is
implemented as a duplicated controller with memory units. These duplicated subunits are realised in the form of the following cards :

Base Processor Controller (BPC) Card

Base Memory Extender (BME) Card
BPC controls time-switching within the Base Module via the Base Message
Switch and the Time Switch Controller. It communicates with the Administrative
Processor via Base Message Switch for operations and maintenance functions. In
a SBM configuration, BPC directly interfaces with the Alarm Display Panel and
the Input Output Module.
To support 8,00,000 BHCA, the BPC card is replaced by High performance
Processor Card (HPC). It is pin to pin compatible for hardware and also for
software so that they are interchangeable at any site to meet specific traffic
requirement. Figure 3.3 summarises the various units and sub-units of the Base
Module.
16.

CENTRAL MODULE (CM)

Central Module (CM) is responsible for space switching of inter-Base Module
calls, communication between Base Modules and the Administrative Module,
clock distribution and network synchronisation. For these functions, Central
Module has a Space Switch, Space Switch Controller and a Central Message
Switch. CM provides connectivity to 16 BMs if it is CM-L and 32 BMs if it is
CM-XL. Each BM interfaces with CM via two 512-channel parallel buses as
BUS-0 and BUS-1, each operating at 4 Mbps. These buses carry voice
information of 512 terminations of the Base Module towards CM. In the reverse
direction, after space switching has been done in the Space Switch under the
control of Space Switch Controller (SSC), the same buses carry the switched
voice information for 512 terminations towards BM. Thus, in a 32 Base Module
configuration, there are 64 parallel buses carrying the voice information from
Base Modules to the Central Module, and also the switched information in the
reverse direction. Chapter 3.

17.

Space Switch (SS) and Space Switch Controller (SSC)

In order to take care of the large number of interface signals, the switch portion of
CM is divided into three stages viz. MUX stage, Switch stage and DEMUX stage.
The MUX and DEMUX stages are implemented on single card to provide the
Base Module to Central Module interface in each direction. Interfacing and
switching are controlled by SSC which provides control signals for the
MUX/DEMUX cards and the Space Switch Switch cards. Interconnection
between MUX/DEMUX cards and the Space Switch is shown in Figure 3.4.
MUX/DEMUX Cards extract the information from time-slots 0 and 1 of Bus0 and
Bus1 from the Base Modules. These time-slots carry control message from each
Base Module and these messages are sent to the Central Message Switch (CMS).
The CMS sends these messages to the Space Switch Controller (SSC) on a 128
kbps link to control space switching based upon this information.

Four 512-channel buses from four BMs are multiplexed to form a 2048- channel,
16 Mbps multiplexed BUS which is sent to both copies of the Space Switch
Switch Card. Space switching of these 2048 channels is done based upon the
switching information received by Space Switch Controller (SSC) from CMS.
Clock Distribution
CM provides the central clock for distribution to the Base Modules. The 8MHz
clock may be locally generated at the Central Clock (CCK) card in case of CMXL and of Space Switch Clock (SCK) card in case of CM-L by using high
stability VCXO crystal or may be derived from an external reference clock using
the Network Synchronisation Controller (NSC) card in case of CM-XL and
Network Synchronisation Equipment (NSE) in case CM-L under the control of
SSC. In the event of failure of external reference or duplex failure of the NSC
cards/NSE, the local clock is fed in the holdover mode, synchronised to last
reference value. In any arrangement, the local or external clock is distributed via
Central Bus Extender (CBX) cards in case of CM-XL. The CBX card provides an
interface between SSC and SSU. SSC makes any switch card access through
CBX. CBX also handles any power supply errors in SSU and BTU. Each CCKCBX-NSC complex form a security block i.e. CBX0 cannot be used with CCK1.
Thus there is a copy 0 complex and a copy 1 complex. The CBX also
synchronises all SSC accesses to SSU with the 16 MHz clock as well as BTU.
Fig. 3.5 depicts the clock distribution in C-DOT DSS with CM-XL.

18.
Central Message Switch (CMS)
Central Message Switch (CMS) complex is the central message transfer point of the
switch. It is implemented as four different message switches, working in load-sharing
mode. Each message switch is a high performance message routing block, implemented
by using high speed 32 bit microprocessor MC 68040 in case of CM-XL and 16 bit
microprocessor MC 68000 in case of CM L.
This card supports 38 HDLC links in case of CM-XL with flexibility of programming
individual HDLC links upto 750 kbps. All Central Message Switches (CMS1,2,3&4) are
used for routing of messages across the Base Modules. On the other hand only CMS1 and

CMS2 interface with the Administrative Module for routing control message between
Base Processors and Administrative Processor. This communication is used to access
office data for routing inter- module calls and administration and maintenance functions.
Fig. 3.6 depicts the Central Message Switch in C-DOT DSS.

19.
ADMINISTRATIVE MODULE (AM)
Administrative Module (AM) consists of a duplicated 16/32-bit controller called the
Administrative Processor (APC). It communicates with Base Processors via the Central
Message Switch for control messages and with the duplicated Input Output Processors in the
Input Output Module for interfacing peripheral devices Administrative processor is
responsible for global routing, translation, resource allocation and all other functions that are
provided centrally in C-DOT DSS MAX. The implementation of AM is similar to Base
Processor Complex of BM, using the same hardware configuration. As explained earlier,
HPC instead of BPC is used to support 8,00,000 BHCA.
20.

INPUT OUTPUT MODULE (IOM)

Input Output Module (IOM) consists of duplicated Input Output Processor (IOP).
The Input Output Processor (IOP) is a general purpose computer with UNIX
Operating System. It is used as the front end processor in C-DOT DSS. It handles
all the input and output functions in C-DOT DSS. The IOP is connected to AP/BP
via HDLC links. During normal operation, two IOPs interconnected by a HDLC
link, operate in a duplex configuration. Working as front end processor, it
provides initial code down load to the subsystems, man machine interface and
data storage for billing and other administrative information. Refer Fig. 3.7 for
IOP connectivity in the system and IOP-VH architecture.

CAS AND CCS #7 SIGNALLING

7.1.1. OVERVIEW OF SIGNALLING
One of the major factors influencing the development of signalling systems is the
relationship between signalling and the control function of exchanges. Early
telecommunication networks used analogue step-by-step exchanges. In such systems, the
control and switch functions are co-located, and when a call is made, the signalling and
traffic follow the same path within the exchange. This is known as Channel Associated
Signalling (CAS). In such exchange the control mechanism for setting-up and releasing
calls is separated from the switch block.
With Common Channel Signalling (CCS) systems, the philosophy is to separate the
signalling path from the speech path. The separation occurs both within the exchange and
external to the exchange, thus allowing optimisation of the control processes, switch
block and signalling systems.
7.1.2. CAS AND CCS COMPARISON
Common Channel Signalling is being adopted throughout the world in national and
international networks for numerous reasons. The main reasons are:
a) The rapidly changing control techniques of exchanges
b) The limitations of CAS systems
c) The evolutionary potential of CCS systems
One result of the evolutionary process of exchanges described above is to change the
relationship between signalling and call control. In the early exchange systems,
exchanges could communicate, but in a limited and inflexible manner, thus limiting the
flexibility of call control. In a CCS environment, the objective is to allow uninhibited
communication between exchange control functions, or processors, thus tremendously
broadening the scope and flexibility of information transfer.
Further advantages result from the evolutionary process of CCS and call control. The
drive to provide an unrestricted communication capability between exchange processors
eliminates per-circuit signalling termination costs. These costs are inevitable in percircuit CAS systems, but for funneling all signalling information into a single commonchannel, only one signalling termination cost is incurred for each transmission link. There
are cost penalties for CCS systems; e.g. the messages received by an exchange have to be
analysed, resulting in a processing overhead. However, these cost penalties are more than
covered by the advantages of increased scope of inter-processor communication and more
efficient processor activity.
The separation of CCS from traffic circuits, and the direct inter-connection of exchange
processors, are the early steps in establishing a cohesive CCS network to allow
unimpeded signalling transfer between customers and nodes and between nodes in the
network. The concept of a cohesive CCS network opens up the opportunity for the
implementation of a wide range of network management, administrative, operation and
maintenance functions. A major example of such a function is the quasi-associated mode
of operation. This mode of operation provides a great deal of flexibility in network
security, reduces the cost of CCS on small traffic routes and extends the data-transfer
capabilities for non-circuit-related signalling.

CAS systems possess limited information-transfer capability due to:

1. The restricted number of conditions that can be applied (e.g. the limited variations
that can be applied to a D.C. loop or the limited number of frequency
combinations that can be implemented in a voice frequency system)
2. The limited number of opportunities to transfer signals (e.g. it is not possible to
transmit voice-frequency signals during the conversation phase of a call without
inconveniencing the customers or taking special measures).
Neither of these restrictions applies to CCS. The flexible message-based approach allows
a vast range of information to be defined and the information can be sent during any stage
of a call. Hence, the repertoire of CCS is far greater than channel associated versions and
messages can be transferred at any stage of a call without affecting the calling and called
subscribers.
CCS systems transfer signals very quickly, i.e. at 64 Kbps. This speedy signalling also
permits the inclusion of far more information without an increase in post dialling delay.
Techniques used in modern CCS system can further improve the flexibility provided to
customers. User-to-user signalling and end-to-end signalling techniques are used
whereby messages can be transferred from one customer to another without undergoing a
full analysis at each exchange in the network. Whilst forms of end-to end signalling are
possible using CAS systems, the technique can be more efficiently implemented with
CCS systems.
One of the problems that prompted the development of CCS systems was speech
clipping in the international network. In some CAS systems, it is necessary to split the
speech path during call set-up to avoid tones being heard by the calling customer. This
results in a slow return of the answer signal and, if the called customer starts speaking
immediately after answer, then the first part of the statement by the called customer is
lost. As the first statement is usually the identity of the called customer, this causes a
great deal of confusion and inconvenience. CCS systems avoid the problem by
transferring the answer signal quickly. As a result of the processing ability of CCS
systems, a high degree of reliability can be designed into the signalling network. Error
detection and correction techniques can be applied which ensure reliable transfer of
uncorrupted information. In the case of an intermediate exchange failure, re-routing can
take place within the signalling network, enabling signalling transfer to be continued.
While these features introduce extra requirements, the common channel approach to
signalling allows a high degree of reliability to be implemented economically. A major
restriction of CAS is the lack of flexibility, e.g. the ability to add new features is limited.
One factor that led to the development of CCS was the increasing need to add new
features and respond to new network requirements. Responses to new requirements in
CCS can be far more rapid and comprehensive than for channel associated versions. CCS
systems are not just designed to meet current needs. They are designed to be as flexible as
possible in meeting future requirements. One way of achieving the objective is to define
modern CCS systems in a structured way, specifying the signalling system in a number of
tiers. The result is flexible signalling system that reacts quickly to evolving requirements

and future services can be incorporated in a flexible and comprehensive manner. Changes
to existing services can be implemented more quickly and at lower cost than with CAS
systems.
7.1.3. FEATURES OF CCS NO.7 SIGNALLING

Signalling System No.7 (CCS7) is a message based signalling system between Stored
Program Controlled (SPC) switches. Where the intermediate nodes may be used as Signal
Transfer Points (STPs), CCS7 network can be used for transmitting call related messages,
as well as slow speed data packets between ISDN users. The Signalling Connection
Control Part (SCCP) enables it to act like a packet network. Thus it is an important prerequisite to Integrated Service Digital Network (ISDN) and Intelligent Network (IN)
features. Enhanced service for the public telephone network can also be provided using
this message based signalling system
Some of the salient features of CCS7 are:
Fast, reliable and economical
Bit-oriented protocol
Labelled messages
Associated and quasi-associated mode of working
Error correction is supported at link level (level 2) by transmission and sequence
control.
Message routing is supported by signalling message handler at level 3
Redundancy and load sharing is possible on signalling links. Change back on link
restoration is possible
Redundancy and load sharing is possible on signalling routes, along with
diversion on route failure.

FIG 7.1CCS7 Protocol Stack

The CCS7 protocol stack comprises of four layers. With reference to the OSI 7-layer
model, the correspondence between the layers is depicted in Fig.7.1.The functions
defined for each layer or level are briefly described in the following paras.
Level 1
Any node with the capability of handling CCS7 is termed a signalling point. The direct
interconnection of two signalling points with CCS7 uses one or more 'signalling link(s)'.
Level 1 of the 4-level structure (shown in Fig.7.1) defines the physical, electrical and
functional characteristics of the signalling link. Defining such characteristics within level
1 means that the rest of the signalling system (level 2 to 4) can be independent of the
transmission medium adopted. By keeping the interface between levels 1 and 2 constant,
any changes within level 1 do not affect the higher levels. In a digital environment,
usually the physical link is a 64 Kbps channel. This is typically within a digital
transmission system using pulse-code modulation (PCM). However, other types of link
(including analogue) can be used without affecting levels 2 to 4.
Level 2
Level 2 defines the functions that are relevant to an individual signalling link, including
error control and link monitoring. Thus, level 2 is responsible for the reliable transfer of
signalling information between two directly connected signalling points. If errors occur
during transmission of the signalling information, it is the responsibility of level 2 to
invoke procedures to correct the errors. Such characteristics can be optimised without
affecting the rest of the signalling system, provided that the interfaces to levels 1 and 3
remain constant.
Level 3
The functions that are common to more than one signalling link, i.e. signalling network
functions, are defined in level 3 : these include message handling functions and
'signalling network management' functions. When a message is transferred between two
exchanges, there are usually several routes that the message can take including via a
signal-transfer point. The message-handling functions are responsible for the routing of
the messages through the signalling network to the correct exchange. Signalling network
management functions control the configuration of the signalling network. These
functions include network reconfigurations in response to status changes in the network.
For example, if an exchange within the signalling network fails, the level 3 of CCS7 can
re-route messages and avoid the exchange that has failed.
FIG 7.1
Message Transfer Part (MTP)
Levels 1 to 3 constitute a transfer mechanism that is responsible for transferring
information in messages from one signalling point to another. The combination of level 1
to 3 is known as the message transfer part (MTP). The MTP controls a number of
signalling message links and network management functions to ensure correct delivery of
messages. This means that the messages are delivered to the appropriate exchange in an
uncorrupted form and in the sequence that they were sent, even under failure conditions
in the network.
Level 4

Level 4 comprises the user parts. The meaning of the messages transferred by the MTP
and the sequence of actions for a particular application (e.g. telephony) is defined by the
`user parts. A key feature is that many different user parts may use the standardised MTP.
Hence, if new requirements arise, that had not been foreseen previously, the relevant user
part can be enhanced (or a new user part derived) without modifying the transfer
mechanism or affecting other user parts. Three user parts have been defined, the
Telephone User Part (TUP), the ISDN User Part (ISUP) and the Data User Part (DUP).
Along with SCCP, which provides end-to-end signalling capability, MTP constitutes the
Network Services Part (NSP) which provides the Network Layer functionalities of the
OSI model. The user parts of NSP are Operations and Maintenance Application Part
(OMAP) and Mobile Application Part (MAP).
Signalling Connection Control Part (SCCP)
The Signalling Connection Control Part (SCCP) has the functions of the network as well
as the transport layers of the CCS7 protocol stack. Together with the MTP, it provides
true OSI transport layer capabilities. Unlike MTP which provides only datagram service,
SCCP provides connection-oriented and connection-less services as well. Thus, while
MTP is sufficient for circuit switched applications like TUP and ISUP, for non-circuit
related applications, such as database querying, the enhanced addressing capability of
SCCP is required. SCCP has a unique scheme of addressing and routing based on Global
Titles. SCCP utilizes the services of MTP to route its payload from one node to other. In
addition to routing transaction related messages submitted by the Transaction Capabilities
Application Part (TCAP), SCCP also segments and sequences large TCAP messages to fit
into the MTP packet size. At the distant node it is the responsibility of the peer SCCP to
re-assemble the segmented message.
Transaction Capabilities Application Part (TCAP)
TCAP is an application part in the CCS7 stack and is responsible for establishing
dialogue with remote databases. It carries the data of higher layers like INAP and MAP
and invokes remote operations. An operation at remote end requires a series of queries
and responses as part of a TCAP dialogue. Management of a dialogue requires:
Establishing a dialogue
Continuing the dialogue
Terminating the dialogue
Maintaining the integrity of each dialogue in case of multiple dialogue scenario
by assigning unique transaction ids to each dialogue.
Invoking remote operation and managing the operation
TCAP layer is a compound layer in the sense that it is composed of two sublayers,
namely, Transaction Sublayer (TSL) and Component Sublayer (CSL). Transaction sub
layer is responsible for establishing, managing and maintaining the integrity of the
dialogue whereas Component sublayer is responsible for packing the upper layer message
into a component and assigning an invoke ID to the component. When CCS7 is specified
as a signalling system, level 4 specifies a number of call-control functions. Indeed, the
circuit-related mode of CCS7 is so closely associated with controlling the set-up and
release of physical circuits that it is essential that some aspects of call-control are defined
within the user part specification in order to optimise the procedures that are adopted.
Application of the Level Structure

The application of the level structure is illustrated in the above diagram. Exchanges A and
B are directly connected by speech circuits (denoted by the solid lines connecting the
respective switch blocks). A signalling link is also available between Exchanges A and B
(denoted by the dotted line). It is shown that level 4 (the user part) is closely associated
with the control function of the exchange. If the control function of exchange A needs to
communicate with the control function of Exchange B (e.g. to initiate the set-up of a
speech circuit between the exchanges), the control function in Exchange A requests the
level 4 functions to formulate an appropriate message. Level 4 then requests the messagetransfer part (level 1 to 3) to transport the message to exchange B. Level 3 analyses the
request and determines the means of routing the message to exchange B. The message is
then transported via levels 2 and 1. Upon receipt of the message by the MTP of exchange
B, levels 1 and 2 deliver the message to level 3. Level 3 at exchange B recognises that the
message has arrived at the correct exchange and distributes the message to the
appropriate user part at level 4. Level 4 in exchange B then interacts with the control
function to determine the appropriate action and response. If problems arise in the
transmission process between exchanges A and B, causing message corruption, the level
2 functions are responsible for detecting the corruption and retransmitting the
information. If the signalling link between exchanges A and B is not available (e.g. link

has failed), the level 3 functions are responsible for re-routing the information through
the signalling network to exchange B.
7.1.4. IMPLEMETATION OF CCS # 7 SIGNALLING IN CDOT
The ITU-T Signalling System No. 7 (CCS7) capability in C-DOT DSS MAX exchange is
provided in the form of the Signalling Unit Module (SUM). It is a standalone equipment
frame that can be used and retrofitted in any exchange configuration. Only one such unit
is required in an exchange.
The place of SUM in the switch architecture is similar to that of a Terminal Unit (TU).
SUM is equipped in one of the co-located Base Modules in a Terminal Unit frame
position.
Although SUM is a module by itself and contains global resources, it has been
deliberately placed at the front-end in order to make it independent of the switch
configuration. The BM containing the SUM is then called the "home" BM. Figure 7.1.4
depicts the placement of SUM in C-DOT DSS MAX. SUM contains a 68010 or 68040
based generic CPU complex and CCS7 signalling handler terminals. The number of
signalling terminals depends upon the signalling network connectivity and the amount of
signalling traffic to be carried. The CCS7 protocol stack has been implemented according
to ITU-T Recommendations and Indian National Standards. Message Transfer Part
(MTP), ISDN User Part (ISUP), Signalling Connection Controller Part (SCCP) and
Transaction Capabilities Application Part (TCAP) are available for PSTN, ISDN and
Intelligent Network applications. Monitoring and measurements as per ITU-T Rec. Q.752
have been implemented. In future, Mobile Application Part (MAP) and Operations &
Maintenance Application Part (OMAP) will be available.

FIG 7.1.4