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L-1 PCM Principles

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Page 1: PCM and C-DOT Exch

L-1

PCM Principles

Page 2: PCM and C-DOT Exch

PCM PRINCIPLES1.0 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 TECHNIQUES2.1 There are basically two types of multiplexing techniques

i. Frequency Division Multiplexing (FDM)

ii Time Division Multiplexing (TDM)

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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 Multiplexing2.3.1 Basically, time division multiplexing involves nothing more than sharing

a transmission medium by a number of circuits in time domain by establishing

Page 4: PCM and C-DOT Exch

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).

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3.0 PULSE CODE MODULATION SYSTEM3.1 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

Page 6: PCM and C-DOT Exch

• Filtering

• Sampling

• Quantisation

• Encoding

• Line Coding

4.0 FILTERING4.1 Filters are used to limit the speech signal to the frequency band 300-3400

Hz.

5.0 SAMPLING5.1 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.

Page 7: PCM and C-DOT Exch

FIG. 3 : SAMPLING PROCESS

5.3 Sampling Theorem5.3.1 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

Page 8: PCM and C-DOT Exch

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.

Page 9: PCM and C-DOT Exch

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.

5.4The 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

Page 10: PCM and C-DOT Exch

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

Page 11: PCM and C-DOT Exch

6.0 QUANTISATION6.1 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 a level to each intervals.

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

Page 12: PCM and C-DOT Exch

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 systems-

quantizing and encoding are a combined process. For the sake of

understanding, these are treated separately.

6.5 Quantizing Process6.5.1 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.

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

Fig. 6, eight quantizing levels are used for each direction of the

analogue signal. To indicate whether a sample is 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.

Page 14: PCM and C-DOT Exch

6.7 Quantization DistortionPractically 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 Signal

Amplitude Range

Quantizing Interval

(mid value)

Quantizing Level Binary Code

0-10 mv 5 mv 0 1000

10-20mv 15mv 1 1001

20-30 mv 25 mv 2 1010

30-40 mv 35 mv 3 1011

40-50 mv 45 mv 4 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 (V2).

Thus, we see that the error depends upon the size of the step. 12

6.7.3 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

Page 15: PCM and C-DOT Exch

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

Page 16: PCM and C-DOT Exch

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 ENCODING7.1 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 encodingfor + ve 'O' for - ve. in the segmentThe 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.

Page 17: PCM and C-DOT Exch

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).

Page 18: PCM and C-DOT Exch

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.

Page 19: PCM and C-DOT Exch

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.

Page 20: PCM and C-DOT Exch

8.0 CONCEPT OF FRAME8.1 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 transmit and

receive ends, the synchronization data is transmitted through another time slot.

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

Thus the time available per channel would be 3.9 microsecs.

Thus for a 30 chl PCM system,

Frame = 125 microseconds

Time slot per chl = 3.9 microseconds.

8.4 Structure of Frame8.4.1 A frame of 125 microseconds duration has 32 time slots. These slots are

numbered Ts 0 to Ts 31.

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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 SYNCHRONIZATION9.1 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 B1 B2 B3 B4 B5 B6 B7 B8

FAW digit value X 0 0 1 1 0 1 1

9.3 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.

Page 22: PCM and C-DOT Exch

TABLE-2

Frame Remark

Numbers B1 B2 B3 B4 B5 B6 B7 B8

FO X 0 0 1 1 0 1 1 FAW

F1 X 1 Y Y Y 1 1 1 ALARM

F2 X 0 0 1 1 0 1 1 FAW

F3 etc X 1 Y Y Y 1 1 1 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 111110.0 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. For carrying

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synchronization data for all frames, 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).

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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 Structure10.6.1 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

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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 EXCHANGES1.0 Introduction

1.1 To overcome the limitations of manual switching; automatic exchanges, having Electro-mechanical components, were developed. Strowger exchange, the first automatic exchange having direct control feature, appeared in 1892 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 short- comings 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.

Table 1 Chronological Development of Electronic Exchanges.ANALOG

1965 No.1 ESS Local Bell Labs, USA

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1972 D 10 Local and Transit NEC. Japan.1973 Metaconta Local LMT. France1974 No. 1 ESS Centrex Local and Transit Bell Labs. USA1975 Proteo Local & Transit Proteo, Italy1976 AXE Local PTT & LM Ericsson, Sweden1976 No.4 ESS Transit Bell Labs, USA1978 AXE Local LM Eiricsson, Sweden.

Table 2: Development of Electronic ExchangesMODELAnalog

Capacity (in thousands)lines Trunks

TrafficErlangs Call Attempts per second

No. 1 ESS 10-65 - 6,000 30No. 1 ESS 20-128 32 10,000 65NO. 4 A XB ETS - 22.4 6,200 35No. 4 ESS - 107 47,500 150D 10 98 14.3 4,400 30XE 1 - 13 2,500 3.6EWSD 30 - 2,000 11-16EWSP - 13 5,000 -TXE-4 40 - 5,000 50Proteo 30 15 - -AXE 64 - 6,000 35PRX-205 10 - 1,000 10-15Digital Exchange

- - - -

E-10B 30 4 2,400 25Mentaconta 10-60 - 10,000 28-60MT 20 - 64 20,000 83-110E 12 - 65 15,000 86System X 100 60 25,000 800000AXE -10 64 - 26,000 800000FETEX-150L 290 60 24,000 1800000OCB-283 200 60 25,000 800000EWSD 250 60 25,200 1000000No.5ESS - - - -NEAX-61E 100 60 27,000 1000000

TABLE 5- ADVANTAGES OF ELECTRONIC EXCHANGE OVER ELECTROMECHANICAL EXCHANGES

Electromechanical Exchanges - Electronic ExchangesCategory, Analysis, Routing, translation, etc;, done by relays.

Translation, speech path Sub’s Facilities, etc., managed by MAP and other DATA.

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Any changes in facilities require addition of hardware and/or large amount of wiring change. Flexibility limited.Testing is done manually externally and is time consuming. No logic analysis carried out.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.

Changes can be carried out by simple commands. A few changes can be made by Subs himself. Hence, highly flexible.

Testing carried out periodically automatically 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.

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

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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 DevelopmentsMany 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 subscriber’s 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.

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4.2.2 Priority Subscriber LinesPriority 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) RestrictionThe 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 InterceptionIncoming 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 ForwardingThe 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 DisturbThis 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.

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4.2.11 Call RepetitionInstead of camp on busy a call can automatically be repeated. The calling party can replace his hand set after receiving the busy tone. A Periodic check is carried out on the called party’s status. When idle status is ascertained, the connection is set up and ringing current fed to both the parties.

4.2.12 Third party InquiryThis 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 subscriber’s premises the charges of each call made

can be ascertained by him.

4.2.15 Call Charge printout or immediate BillingThe 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) Subscriber’s 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.

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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 CallA 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 callIf 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 AccommodationReduction in switch room accommodation to about 1/6th to 1/4th as compared to Cross-bar system is possible.

4.3.2 Faster installation and Easy ExtensionThe 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 lineRoutine 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.

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4.4 Maintenance Facilities

4.4.1 Fault ProcessingAutomatic fault processing facility is available for checking all hardware components and complete internal working of the exchange. Changeover from a faulty sub-system to stand-by sub-system is automatically affected without any human intervention. Only information is given out so that the maintenance staff is able to attend to the faulty sub-system.

4.4.2 DiagnosticsOnce 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 BlockingIn 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.

5.2 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

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simultaneous paths available through the switching network. However, in electronic exchanges, the prime limitation is the number of simultaneous calls, which can be handled by the control equipment, as it has to execute a number of instructions depending on the type of the call. Therefore the extent of loading of the exchange will be guided solely by the amount of processor loading. Moreover, the facilities to the subscribers will also have to be limited accordingly.

5.3 Power SupplyThe 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 PCB’s therefore, should be done with utmost care and should be minimised care should also be taken to protect the cards from exposure to stray electromagnetic fields.

5.7 PCB RepairThe repair of PCB’s is extremely complicated and sophisticated equipments are required for diagnosing the faults. This results in having costly inventory and a costly repair centre. With the frequent improvement and changes in the cards, proper documentation of cards becomes essential.

5.8 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

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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.

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CHAPTER-2DIGITAL SWITCHING,CONCEPT OF TIME & SPACE SWITCHING1.0 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 divisionmultiplexed (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

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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 Switchingii. 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 time-slot TSx of an I/C Highway can be connected to a different time-slot., TSy, of the outgoing highway. In other words, a time switch is, basically, a time-slot changer.

3.0 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 time-slot.Thus, the crosspoint corresponding to the address, is operated during a particular

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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 rather than serially, through the switching matrix.

3.3.2 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

3.3.3 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

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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

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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 µs)

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

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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 control memory location to halt further transfer of samples.

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

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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

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.

4.3.2 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

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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

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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 two-stage 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

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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 non-folded, having complete isolation of I/C and O/G PCM highways. However, a practical local exchange will have a combination of both types of networks.

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 HWY0Step 2 Space stage (S) Tsx HWY0 to Tsx HWY3Step 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 time-

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slots in each PCM. Accordingly, the speech and control memories in time switches and control memory columns in space switch, will contain 32 locations each.

To establish the connection, the CC searches for free internal time-slots. Let us assume that the first available time-slots are TS10 and TS11, as before. To reduce the complexity of control, the first time stage is designed as output-controlled switch, whereas the second time stage is input-controlled.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.

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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.

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CHAPTER-3

Signaling in Telecommunication subscriber signaling, Trunk signaling (CAS & CCS)

SIGNALLING IN TELECOMMUNICATION1.0. Introduction1.1. 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 informationCall 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.

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ON HOOKOFF-HOOKDIAL TONE(ADDRESS)

ON HOOK

CONNECT

ON HOOK1

2

CALLING SUBSCRIBER ORIGINATING EXCHANGETERMINATING EXCHANGECALLED SUBSCRIBER

LINE TRUNK LINE

TIME

2.1.3 End of selection informationThis 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 informationIt 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 charging may commence.

ii. 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 connectionThe 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

i. 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 start dialling.iii. The called number is transmitted to the exchange, when the calling

subscriber dials the number.iv. if the number is free, the exchange sends ringing current to him.v. Feed-back is provided to the calling subscriber by the exchange by sending.

a. Ring-back tone, if the called subscriber is free(shown in fig.1)b. Busy tone if the called subscriber is busy ( not shown in figure), orc. 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 handsetvii. The exchange recognizing the acceptance terminates the ringing current and the ring-back tone, and establishes a connection between the calling and called subscribers.

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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 SignalingIn automatic exchanges the power is fed over the subscriber’s loop by the centralized battery at the exchange. Normally, it is 48 V. The power is fed irrespective of the state of the subscriber, viz., idle, busy or talking.

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3.2.1.1 Call reportWhen 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 signalAfter 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 DiallingThe 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.

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FIGURE 2. TONE-DIALLING FREQUENCY GROUPS.

3.2.1.3 End of selection signalThe 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) SignalPermanent 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

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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 SignalOn 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 SignalWhen 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 SignalIf 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 SignalWith 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 subscriber’s loop for duration less than the release signal. it may be of 200 to 320 milliseconds duration.

3.3 Inter-exchange Signaling3.3.1 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. PulsedThe 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. ContinuousThe signal consists of transition from one condition to another, a steady state condition does not characterizes any signal.

iii. CompelledIt is similar to the pulsed mode but the transmission is not of fixed duration but condones till acknowledgement of the receiving unit is

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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 the subscriber.ii. Address information in the from of decade pulses.

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. SINGLE FREQUENCY SIGNALING STATES TONE SIGNAL CONDITION

State Forward BackwardIdle (On hook) FORWARDSeizure(off hook)Release (on hook)BACKWARDAnswer(off hook)Clear Back (on hook)Blocking (off hook)

On

offon

offoffon

On

onoff/on

offonoff

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

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((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 Outgoing ExchangeM- lead E-lead

Incoming ExchangeM- lead Elead

Idle(On hook)

Earth Open Earth Open

FORWARDseizure(off hook)

Battery Open Earth Earth

Release(On hook)BACKWARD

Earth Earth/open Battery/Earth Open

Answer(off hook)

battery Earth Battery Earth

Clear Back(On hook)

battery Open Earth earth

Blocking 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.

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outgoing register

incomming register

forwardsignal

time

signal cessation recognition

next forward signal

2-and-2onlysignal recognition

acknowledgement backwardsignal and request for next signal

compelled signal sequence

acknowledgement backward signal

receiving

time

signal cessationrecognition

Sending

Fig.3. Compelled signalling procedureIn 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

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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 signalingThe 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 signalingThe 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 SignallingCCITT 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

Index f 0 f 1 f 2 f 3 f 4

Weight Code 0 1 2 4 7

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TABLE 4-FORWARD SIGNALSSignal Weight Group I Group II

1 0+1 Digit 1 Ordinary subscriber2 0+2 Digit2 Subscriber with

priority Test / Mtce, equipment

3 1+2 Digit3 Spare4 0+4 Digit4 STD Barred5 1+4 Digit5 Spare6 2+4 Digit6 CCB7 0+7 Digit7 Changed Number to

Operator8 1+7 Digit8 Closed Number9 2+7 Digit9 Closed Number10 4+7 Digit0 Spare

TABLE 5 -BACKWARD SIGNALSSignal No. Weight Code Group A Group B1 0+1 Send next digit Called line free with

out metering2 0+2 Restart Changed number 3 1+2 Address complete,

Changeover to reception of group B signals

Called line busy

4 0+4 Calling line identification for malicious calls

Local congestion

5 1+4 send calling subscribers category

Number unobtainable

6 2+4 Set up speech connection

called line fee, with metering

7 0+7 Send last but 1 digit Route congestion8 1+7 Snd last but 2 digit Spare9 2+7 Snd last but 3 digit Route Breakdown10 4+7 Spare Malicious call

blocking

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

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It can be seen from the tables that1. 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 caller’s

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 semi-compelled 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 caller’s category). 3. A5 is acknowledgement by sending any 11-1 to 11-5 by the

originating exchange4. 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 caller’s 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

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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 signallingIn 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 Bit Value Forward backword. af bf ab bbIdle 1 0 1 0

Seizure 0 0 1 0

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Seizure acknowledge

0 0 1 1

Answer 0 0 0 1

Clear Forward 1 0 0/1 1

Clear Back 0 0 1 1

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.

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

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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 signallingIn 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 ProcedureThe 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 SynchronisationSynchronisation 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 and3. 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 protectionTo 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 systemsCurrently 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.

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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 messageFig.5 Typical message formats. The shaded area represent spare bit fields.

4.4.2 CCITT No. 7 SystemIn 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 :-

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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 CCSThe 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 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%

4.5 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.

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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 networksAs 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 Telecommunication1. CDoT :

C-DOT MAX-XL : SYSTEM AND SUBSCRIBER FEATURES1. GENERAL

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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 MODULESC-DOT DSS MAX exchanges can be configured using four basic modules (Fig. 2.1)a. Base Moduleb. Central Modulec. Administrative Moduled. Input Output Module

The Base Module (BM) is the basic growth unit of the system. It interfaces theexternal 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.

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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

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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 UNITRemote 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,

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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 UNITSVarious 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 UnitModule 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 ControllerThis 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 128-

channel link between the terminals (subscribers, trunks, etc.) and the module time

switch.

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4. Message SwitchMessage 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. OPTIMISATIONIn 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. Terminal Interfacing - interfacing lines, analog and digital trunks, CCM, PBX and remote digital lines.

2. Circuit Switching - switching within the Base Module.3. Call Processing - majority of call processing functions.4. 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) Inter-module CommunicationInter-BM and BM-AM communication via the Central Module.

b) Message SwitchingInter-BM and BM-AM control-message communication via the Central Message Switch in the Central Module.

c) Resource AllocationDone by the Administrative Module. THE C-DOT DSS FAMILY

d) Operations and MaintenanceBulk data storage by the Input Output Module and man-machine interface provided by the Administrative Module via the Input Output Module.

e) ServicesAnnouncements 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

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1.1.1. PHILOSOPHY AND GROWTH OF C-DOTThe 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-of-the-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 C-DOT 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

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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 open-ended 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, man- machine 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. Single-chip 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.

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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

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 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 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 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

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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

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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

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mode. In the event of failure of one of the units, other units will share the failed unit’s 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 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.

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

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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 128- channel 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.

╧ ╧ ╧ ╧

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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 0’s and 1’s 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.

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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 bi-directional 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. 0’s and 1’s. 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

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31

31

0

7

TS4 PCM0 TS7 PCM0

100

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

B.M

C.M

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.

0

4

TS4 PCM0

111

B.M

B.M

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

TS7 PCM0

0

7

31

0

4

31

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And as such TS4 PCM0 information will be shifted to TS7 PCM0. Here writing is sequential whereas reading is controlledI/P associated time switch: let us take the same examplei.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.

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0 1 2

CM0 CM1 CM2

0

n-1TS7 PCM2

LOGIC 1

1

2

Pcm 0…………………….

…………………

PCM m-1

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 3.9µ 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

TS4 PCM0

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Cgp trans T.S-------------- cdp receive TSCdp 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 53Cgp trans T.S-------------- cdp receive TS

TS 20 T PCM 53 TS 5 R PCM2Cdp 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.

ISDN Terminal Unit - to support termination of BRI/PRI interfaces and implementation of lower layers of DSS1 signalling protocol.

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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.

2.2.1.1. 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.

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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 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 C-DOT 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.

FIG 2.1

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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).

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F

IG 2

.2

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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

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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

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FIG

2.4

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FIG 2.3

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FIG 2.5

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.

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 Billing/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.

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FIG 2.7

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HARDWARE ARCHITECTURE OF CDOT MBMThe 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. Analog to digital conversion of all signals on analog lines and trunks2. Interface to digital trunks and digital subscribers3. Switching the calls between terminals connected to the same Base Module4. 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

5. 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 3.1The 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.

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1. Analog Subscriber Line CardsTwo 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 32- channel, 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 CardsAnalog 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) CardTerminal 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.

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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.

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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).

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(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

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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. 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 :

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

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

(c) 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 sub-units are realised in the form of the following cards :

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Base Processor Controller (BPC) CardBase 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.

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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 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 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. 3.5 depicts the clock distribution in C-DOT DSS with CM-XL.

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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

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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.

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CAS AND CCS #7 SIGNALLING

7.1.1. OVERVIEW OF SIGNALLINGOne 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 exchangesb) The limitations of CAS systemsc) 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 per-circuit CAS systems, but for funneling all signalling information into a single common-channel, 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.

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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

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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 pre-requisite 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 systemSome 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.

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FIG 7.1

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CCS7 Protocol StackThe 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 1Any 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 2Level 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 3The 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.

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

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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

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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 message-transfer 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

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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.

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