3. PCM, Multiplexing & Signalling

8/13/2019 3. PCM, Multiplexing & Signalling

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PCM Principles, Multiplexing & Signaling Page 1 of 26

PCM PRINCIPLE

1.0 INTRODUCTION

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 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 forsome 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. Similarlyincreasing the number of open wire pairs that can be installed on one alignment due toheadway consideration and maintenance problems. Similarly increasing the number of

pairs to the underground cable is uneconomical and leads to maintenance

problems.

It, therefore, became imperative to think of new technical innovations whichcould exploit the available bandwidth of transmission media such as open wire lines or

underground cables to provide more number of circuits on one pair. The technique used

to provide a number of circuits using a single transmission link is called Multiplexing.

2.0 MULTIPLEXING TECHNIQUES

There are basically two types of multiplexing techniques

i. Frequency Division Multiplexing (FDM)

ii Time Division Multiplexing (TDM)

2.1 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 transmissionmedium. 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 varyingsignals.


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Fig. 1 FDM Principle

2.2 Time Division Multiplexing

Basically, time division multiplexing involves nothing more than sharing

a transmission medium by a number of circuits in time domain by establishing asequence of time slots during which individual channels (circuits) can be transmitted. Thus

the entire bandwidth is periodically available to each channel. Normally all time slots1are

equal in length. Each channel is assigned a time slot with a specific common repetitionperiod called a frame interval. This is illustrated in Fig. 2.

Fig. 2 Time Division Multiplexing

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 areconnected to individual gates which are opened one by one in a fixed sequence. At the


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receiving end also similar gates are opened in unision with the gates at the transmittingend.

The signal received at the receiving end will be in the form of discretesamples 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 sequentialsampling 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 DivisionMultiplexing (TDM).

3.0 Pulse Code Modulation

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.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.

Basic Requirements for PCM System

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

Filtering

Sampling

Quantisation

Encoding

Line Coding

3.1 FILTERING

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

3.2 SAMPLING

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 theamplitude 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


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samples are to be sent for any band limited signal to get a good approximation of theoriginal analogue signal and the same is defined by the sampling Theorem.

Fig. 3: Sampling Process

3.2.1 Sampling Theorem

A complex signal such as human speech has a wide range of frequencycomponents 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 componentsoccupy 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 highestfrequency components of the signal.

Sampling Theorem States

"If a band limited signal is sampled at regular intervals of time and at a rate equal toor 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

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


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channels one by one at the rate specified by the sampling theorem, then the time availablefor sampling each channel would be equal to Ts/N microseconds.

FIG. 4: Sampling and combining Channels

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 bya 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 channelis 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 ofchannels to be combined and the clock pulse frequency i.e. the sampling frequency.

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 samplingi.e..the interval between 2 consecutive samples of a channel is 125 microseconds. Thisduration i.e. 125 microseconds is called Time Frame.

The signals on the common medium (also called the common highway)

of a TDM system will consist of a series of pulses, the amplitudes of which are

proportional to the amplitudes of the individual channels at their respective samplinginstants. This is illustrated in Fig. 5


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i

Fig 5 : PAM Output Signals

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

3.3 Quantization

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 instantvalues. 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 signaltransmission. Therefore, in PCM systems, PAM signals are converted into digital form

by using Quantization Principles. The discrete level of each sampled signal is quantifiedwith reference to a certain specified level on an amplitude scale.

The process of measuring the numerical values of the samples and giving thema 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.


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Quantising, in other words, can be defined as a process of breaking down acontinuous amplitude range into a finite number of amplitude values or steps.

A sampled signal exists only at discrete times but its amplitude is drawn from acontinuous 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 toget 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 10-20mV, 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 thesake of understanding, these are treated separately.

Quantizing Process

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 thequantizing intervals are of the same size. This is called Linear Quantizing.

FIG. 7: QUANTIZING-POSITIVE SIGNAL


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Assigning an interval of 5 for sample 1, 7 for 2 etc. is the quantizingprocess. Giving, the assigned levels of samples, the binary code are

called coding of the quantized samples.

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

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

3.1.1 Relation between Binary Codes and Number of levels.

Because the quantized samples are coded in binary form, the quantization intervalswill 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.

3.1.2 Quantization Distortion

Practically in quantization we assign lower value of each interval to a sample

falling in any particular interval and this value is given as:


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Table-1: Illustration of Quantization Distortion

Analogue Signa

Amplitude Range

uantizing 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 atthe 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 inamplitude 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.

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

In linear quantization, equal step means equal degree of error for all inputamplitudes. In other words, the signal to noise ratio for weaker signals will be poorer.

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 ofquantum steps and fm is the highest signal frequency. But as we knows from speech

statistics that the probability of occurrence of a small amplitude is much greater than

large one, it seems appropriate to provide more quantum levels (V = low value) in the

small amplitude region and only a few (V = high value) in the region of higheramplitudes. 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 ofquantization is called non-uniform quantization.

In practice, non-uniform quantization is achieved using segmented quantization

(also called companding). This is shown in Fig. 9 (a). In fact, there is equal number of


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segments for both positive and negative excursions. In order to specify the location of asample 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

Fig. 9 (a) Segmented coding curve

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 16segments will be taken into account.

3.4 Encoding

Conversion of quantised analogue levels to binary signal is called encoding. To

represent 256 steps, 8 level code is required. The eight bit code is also called an eight bit

"word".

The 8 bit word appears in the form

P ABC WXYZ

Polarity bit 1 Segment Code Linear

encoding

for + ve 'O' for - ve. in the segment


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The first bit gives the sign of the voltage to be coded. Next 3 bits gives the segmentnumber. There are 8 segments for the positive voltages and 8 for negative voltages.

Last 4 bits give the position in the segment. Each segment contains 16 positions.

Referring to Fig. 9(b), voltage Vc will be encoded as 1 1 1 1 0101.

FIG. 9 (b) : Encoding Curve with Compression 8 Bit Code

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

converts PAM signals (i.e. after sampling) into an 8 bit binary signal. The coding isdone 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 beencoded.

It is logarithmic 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, 1lying 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 inthe ratio of 2. That is vm, vm/2, vm/4, vm/8, vm/16, vm/32, vm/64, vm/128 (vm being the

maximum voltage).

There are 128 quantification levels in the positive part of the curve and 128 in the

negative part of the curve. In a PCM system the channels are sampled one by one by


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applying the sampling pulses to the sampling gates. Refer Fig. 10. The gates open onlywhen 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 quantizedand encoded. The encoded samples of all the channels and signals etc are combined in the

digital combiner and transmitted.

Fig. 10

The reverse process is carried out at the receiving end to retrieve 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 individualframes. 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.4.0 CONCEPT OF FRAME

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 forthe duration "St". This duration, which is the width of the sampling pulse, is called the

"time slot" for a given channel.


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Since Ts is much larger as compared to St. a number of channels can be sampledeach 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 tothe 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 theduration 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.

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.

5.0 Structure of Frame

A frame of 125 microseconds duration has 32 time slots. These slots arenumbered Ts 0 to Ts 31. Information for providing synchronization between trans andreceive ends is passed through a separate time slot. Usually the slot Ts 0 carries the

synchronization signals. This slot is also called Frame alignment word (FAW).

The signaling information 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 areutilized for voltage signal of channels 16 to 30 respectively.

6.0 SYNCHRONIZATION

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 anddiscriminate 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 patterncalled 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.The digits or bits of FAW occupy seven out of eight bits of Ts 0 in the followingpattern.

Bit position of Ts 0 B1 B2 B3 B4 B5 B6 B7 B8

FAW digit value

X 0 0 1 1 0 1

1


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The bit position B1 can be either ' 1 ' or '0'. However, when the PCM system is to belinked 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 theformer are fixed at T. The FAW and alarm signals are transmitted alternatively as

shown in Table - 2.

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 synchronization alarm. If Y = 1 in

B4, it indicates high error density alarm. When there is no alarm condition, bits B3B4 B5 are set 0. An urgent alarm is indicated by transmitting "all ones". The code

word for an urgent alarm would be of the form.

X 111 1111

7.0 SIGNALLING IN PCM SYSTEMS

In a telephone network,-the signaling information is used for proper routingof 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 "signaling" in PCMsystems. The signaling information can be transmitted in the form of DC pulses (as

in step by step exchange) or multi-frequency pulses (as in cross bar systems) etc.

The signaling pulses retain their amplitude for a much longer period than the

pulses carrying speech information. It means that the signaling information is aslow varying signal in time compared to the speech signal which is fast changing in

the time domain. Therefore, a signaling channel can be digitized with less number of

bits than a voice channel. In a 30 chl PCM system, time slot Ts 16 in each frame isallocated for carrying signaling information.


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The time slot 16 of each frame carries the signaling datacorresponding to two VF channels only. Therefore, to cater for 30 channels, we

must transmit 15 frames, each having 125 microseconds duration. For carrying

synchronization data for all frames, one additional frame is used. Thus agroup of 16 frames (each of 125 microseconds) is formed to make a "multi-

frame". The duration of a multi-frame is 2 milliseconds. The multi-frame 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 andsynchronization 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 multi-frame is

illustrated in Fig. 11 (a) & 11 (b).

Fig. 11 (a) Multi-frame Formation


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FIG. 11 (b) 2.048 Mb/s PCM Multi-frame

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 is 256 x 8000 = 2048 K/bits.

Thus, a 30 channel PCM system has 2048 K bits/sec.

8.0 Multi-frame Structure

In the time slot 16 of FO, the first four bits (positions 1 to 4) contain the multi-frame

alignment signal which enables the receiver to identify a multi-frame. The other four bits

(no. 5 to 8) are spare. These may be used for carrying alarm signals. Time slots 16 offrames F1 to FT5 are used for carrying the signaling information. Each frame carries

signaling, data for two VF channels. For instance, time slot Ts 16 of frame F1 carries the

signal data for VF channel 1 in the first four bits. The next four bits are used for carryingsignaling information for channel 16. Similarly, time slot Ts16 of F2 carries signalling

data of chls 2 and 17. Thus in multi-frame structure, four signaling bits are provided for

each VF channels. As each multi-frame includes 16 frames, so the signaling of each

channel will occur at a rate of 500 per sec.

xxxx


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DEFINITION AND DESCRIPTION OF DIGITAL HIERARCHIES

1.0 INTRODUCTION AND DEFINITION

The term digital hierarchy has been created when developing digital

transmission systems. It was laid down when by multiplexing a certain number of PCM

primary multiplexers were combined to form digital multiplexers of higher order (e.g.

second-order multiplex equipments).

Consequently, a digital hierarchy comprises a number of levels. Each level is

assigned a specific bit rate which is formed by multiplexing digital signals, each having

the bit rate of the next lower level. In CCITT Rec. G.702, the term digital multiplex

hierarchy is defined as follows :

A series of digital multiplexes graded according to capability so that

multiplexing at one level combines a defined number of digital signals, each having the

digit rate prescribed for the next lower order, into a digital signal having a prescribed

digit rate which is then available for further combination with other digital signals of the

same rate in a digital multiplex of the next higher order.

2.0 WHY HIERARCHIES ?

2.1 Before considering in detail the digital hierarchies under discussion we are

going to recapitulate in brief, why there are several digital hierarchies

instead of one only. It has always been pointed out that as far as the

analogue FDM technique is concerned, the C.C.I.T.T. recommends the

world wide use of the 12-channel group (secondary group). Relevant

C.C.I.T.T. Recommendation exists also for channel assemblies with more

than 60 channels so that with certain exceptions there is only one world-

wide hierarchy for the FDM system (although the term hierarchy is notused in the FDM technique).

2.2 In the digital transmission technique it was unfortunately not possible to

draw up a world-wide digital hierarchy. In practice, equipment as

specified in C.C.I.T.T. Recommendation G.732 and 733, they do not only

differ completely in their bit rates, but also in the frame structures, in


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signalling, frame alignment, etc. Needless to say that, as a consequence,

the higher order digital multiplexers derived from the two different PCM

primary multiplexers and thus the digital hierarchies differ as well.

2.3 Since these two PCM primary multiplexers are available, two digital

hierarchies only would have to be expected. In reality, however, two

digital hierarchies with several variants are under discussion because the

choice of the fundamental parameters of a digital hierarchy depends not

only on the PCM primary multiplex, which forms the basic arrangement in

that hierarchy, but on many other factors such as :

(a) the bit rate of the principal signal sources.

(b) traffic demand, network topology, operational features, flexibility

of the network.

(c) time division and multiplexing plant requirements.

(d) compatibility with analog equipment.

(e) characteristics of the transmission media to be used at the bit rates

for the various levels of the hierarchies.

Since today these factors which are essential for forming digital

hierarchies vary from country to country, it is no wonder that we now have

to consider more than two proposals for digital hierarchies.

3.0 DIGITAL HIERARCHIES BASED ON THE 1544 KBIT/S PCM PRIMARY

MULTIPLEX EQUIPMENT

It was around 1968 that Bell labs. proposed a digital hierarchy based on the 24-

channel PCM primary multiplex at the various levels of the hierarchy :

Level in hierarchy Bit rate Trans. line

First level 1544 kbit/s T1

Second level 6312 kbit/s T2

Third level 46304 kbit/s L5 (Jumbo Grp)

Fourth level 280000 kbit/s WT4 (Wave guide)

Fifth level 568000 kbit/s T5


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This proposal was modified during the following years. At the end of the study

period 1968/72, the following digital network hierarchy was finally proposed as given in

Fig.1.

Fig. 1

Encoded FDM (Master Group) USA & Canada

3.1For the various bit rates at the higher levels of the two proposals, different reasons

have been indicated. The bit rate of 44736 kbit/s was selected to provide a

flexibility point for circuit interconnection and because it was a suitable coding

level for the 600 channel FDM mastergroup.

3.2It is also an appropriate bit rate for inter-connection to radio-relay links planned

for use at various frequencies.

3.3At the same time, N.T.T. published its PCM hierarchy are concerned (1554 and

6112 kbit/s, respectively), these two proposals are identical. They differ, however,


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in the higher levels as shown in Fig.2.

Fig. 2

Encoded TDM (Japanese)

3.4In the N.T.T. proposal the bit rate of 32064 kbit/s at the third level of the

proposed hierarchy might be considered a suitable bit rate to be used on

international satellite links perhaps for administrations operating different PCM

primary multiplex equipments. It is also a convenient bit rate for encoding the

standardized 300-channel FDM mastergroup. Delta modulation and differential

PCM for 4 MHz visual telephone are also suitable for this bit rate. Transmission

of 32064 kbit/s via a special symmetrical cable of new design is also possible.

3.5The above fact shows that the differing bit rates of the third level indicated in the

two hierarchy proposals can, therefore, be justified by technical arguments. As far

as the differing bit rates of the fourth level are concerned, only a few technical

reasons are included in the two proposal. In both cases coaxial cables are used as

a transmission medium so that the medium does not call for different bit rates.

3.6Moreover, it seems that at present the specifications of the fourth level (and

higher ones) in the two proposed hierarchies is not yet considered so urgent. For

the time being the third level seems to be more important.

3.7The C.C.I.T.T. faced with this situation has reached finally the solution which is

covered by CCITT recommendation G.752 as one can see from this

recommendation, two different hierarchical levels are existing in the third level of

this hierarchy, namely 32064 kbits/s and 44736 kbit/s respectively. Higher level

have not been specified so far.

4.0 DIGITAL HIERARCHY BASED ON THE 2048 KBIT/S PCM PRIMARY

MULTIPLEX EQUIPMENT

For this digital hierarchy, two specifications have at present been laid down only

for the first level at 2048 kbit/s and for the second level at 8448 kbit/s.

As for the higher levels, the situation is just contrary to that existing in the case of

digital hierarchies derived from 1544 kbit/s primary multiplex, i.e. general

agreement has more or less been reached on the fourth level having a bit rate of

139264 kbit/s. 5th order system where bit rate of 565 Mb/s have also been

planned now.


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4.1 The critical point in this hierarchy is whether or not the third level at 34368 kbit/s

should exist.

4.2 The C.C.I.T.T. has agreed after long discussions on the following

(Recommendation G.751) that there should be a 4th

order bit rate of 139264

kbit/s in the digital hierarchy which is based on the 2nd

order bit rate of 8448

kbit/s.

There should be two methods of achieving the 4thorder bit rate :

Method 1 by using a 3rd

order bit rate of 34368 kbit/s in the digital hierarchy.

Method 2 by directly multiplexing sixteen digital signals at 8448 kbit/s. The

digital signals at the bit rate of 139264 kbit/s obtained by these two methods

should be identical.

The existence of the above two methods implies that the use of the bit rate of

34368 kbit/s should not be imposed on an Administration that does not wish to

realize the corresponding equipment.

4.3 In accordance with the above two methods the following realizations of digital

multiplex equipments using positive justification are recommended :

Method 1: Realization by separate digital multiplex equipments : one type which

operates at 34368 kbit/s and multiplexes four digital signals at 8448 kbit/s; the

other type which operates at 139264 kbit/s and multiplexes four digital signals at

34368 kbit/s.

Method 2 : Realization by a single digital multiplex equipment which operates at

139264 kbit/s and multiplexes sixteen digital signals at 8448 kbit/s.

Method 1 has been put into practice.

4.4 Where the fifth level is concerned, some preliminary proposals (e.g. 565148

kbit/s) have been submitted which were not discussed in detail.

Therefore, the present structure of this digital hierarchy is as given in Fig.3.

139.264


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Fig. 3

Encoded TDM (European)

5.0 Most of the administrations favour the specification of a third level at 34368

kbit/s, mainly as a suitable flexibility point for the operation of the network and as

an adequate bit rate for digital line systems which are to be set up either on new

cables (screened symmetrical or micro-coaxial cables) or an radio-relay links.

Other administrations do not consider the specification of a third level to be

advantageous for their networks. On the contrary they regard it to be more

economical to go directly from the second level at 8448 kbit/s so the fourth level

at 139264 kbit/s, is also achieved by multiplexing four digital signals at 34368

kbit/s, each of which is obtained by multiplexing first four digital signals at 8448kbit/s. However, this is a matter of internal multiplexing only, i.e. digital

multiplex equipment of this type has no external input or output at 34368 kbit/s.

All administrations interested in the third level at 34368 kbit/s would thus be

offered the possibility of using this level. Their digital multiplex equipment which

multiplexes in the same way each of the four digital signals at 8448 kbit/s has to

provide external outputs for the resulting signal at 34368 kbit/s. The digital

multiplex equipment which multiplexes each of the four digital signals at 34368

kbit/s has to provide four inputs for these bit rates and one output for the resulting

bit rate of 139264 kbit/s.

5.1 Outlook

The above context indicates that at the moment the discussion of digital

hierarchies is still underway and is mainly concentrated on the third and fourth

levels. Although certain trends are evident the specification of these and higher

levels will take some time. In the interest of a comprehensive specification of the

digital hierarchies to be drawn up as soon as possible, it is to be hoped that all

parties concerned perform their studies with high priority.

All digital multiplexes and hierarchies proposed till date are operating in an

asynchronous mode (positive justification, positive stuffing, bit-interleaved). It

is likely that in the future, synchronous digital multiplex equipment has to be

considered when setting up digital hierarchies. For various digital line systems

being developed in many countries non-hierarchical bit rates have provisionally

been adopted with due regard to the characteristics of the transmission media


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used. These non-hierarchical bit rates for digital line systems have also to be born

in mind when defining the digital hierarchies and may affect the hierarchical bit

rates.

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SIGNALLING IN TELECOMMUNICATIONS

The term signaling, when used in telephony, refers to the exchange of controlinformation associated with the establishment of a telephone call on a

telecommunications circuit. An example of this control information is the digits dialed by

the caller, the caller's billing number, and other call-related information.

When the signaling is performed on the same circuit that will ultimately carry the

conversation of the call, it is termed Channel Associated Signaling (CAS). This is thecase for earlier analogue trunks, MF and R2 digital trunks, and DSS1/DASS PBX trunks.

In contrast, SS7 signaling is termed Common Channel Signaling (CCS) in that the

path and facility used by the signaling is separate and distinct from the

telecommunications channels that will ultimately carry the telephone conversation. With

CCS, it becomes possible to exchange signaling without first seizing a facility, leading tosignificant savings and performance increases in both signaling and facility usage.

Channel Associated Signaling

Channel Associated Signaling (CAS), also known as per-trunk signaling (PTS), is aform of digital communication signaling. As with most telecommunication signaling

methods, it uses routing information to direct the payload of voice or data to its

destination. With CAS signaling, this routing information is encoded and transmitted inthe same channel as the payload itself. This information can be transmitted in the same

band (in-band signaling) or a separate band (out-of-band signaling) to the payload.

CAS potentially results in lower available bandwidth for the payload. For example, in the

PSTN the use of out-of-band signalling within a fixed bandwidth reduces a 64 kbit/s DS0

to 56 kbit/s. Because of this, and the inherent security benefits of separating the controllines from the payload, most current telephone systems rely more on Common Channel

Signaling (CCS).

Common Channel Signaling

In telephony, Common Channel Signaling (CCS) is the transmission of signaling

information (control information) on a separate channel from the data, and, more

specifically, where that signaling channel controls multiple data channels.


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For example, in the public switched telephone network (PSTN) one channel of acommunications link is typically used for the sole purpose of carrying signaling for

establishment and Tear down of telephone calls. The remaining channels are used

entirely for the transmission of voice data. In most cases, a single 64kbit/s channel issufficient to handle the call setup and call clear-down traffic for numerous voice and data

channels.

The logical alternative to CCS is Channel Associated Signaling (CAS), in which each

bearer channel has a signaling channel dedicated to it.

CCS offers the following advantages over CAS, in the context of the PSTN:

Faster call setup.

No falsing interference between signaling tones by network and speech

frequencies. Greater trunking efficiency due to the quicker set up and clear down, thereby

reducing traffic on the network.

No security issues related to the use of in-band signaling with CAS.

CCS allows the transfer of additional information along with the signaling trafficproviding features such as caller ID.

The most common CCS signaling methods in use today are Integrated Services Digital

Network (ISDN) and Signaling System 7 (SS7).

ISDN signaling is used primarily on trunks connecting end-user private branch exchange

(PBX) systems to a central office. SS7 is primarily used within the PSTN. The twosignaling methods are very similar since they share a common heritage and in somecases, the same signaling messages are transmitted in both ISDN and SS7.

CCS is distinct from in-band or out-of-band signaling, which are to the data band whatCCS and CAS are to the channel.

Signaling System Number #7

SS7is a set of telephony signaling protocols which are used to set up most of the world'spublic switched telephone network telephone calls. The main purpose is to set up and tear

down telephone calls. Other uses include number translation, prepaid billing mechanisms,short message service (SMS), and a variety of other mass market services.

It is usually abbreviated as Signaling System No. 7, Signaling System #7, or just SS7. In

North America it is often referred to as CCSS7, an acronym for Common Channel

Signaling System 7. In some European countries, specifically the United Kingdom, it is

sometimes called C7(CCITT number 7) and is also known as number 7and CCIS7.


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There is only one international SS7 protocol defined by ITU-T in its Q.700-seriesrecommendations. There are however, many national variants of the SS7 protocols. Most

national variants are based on two widely deployed national variants as standardized by

ANSI and ETSI, which are in turn based on the international protocol defined by ITU-T.Each national variant has its own unique characteristics. Some national variants with

rather striking characteristics are the China (PRC) and Japan (TTC) national variants.

SS7 is designed to operate in two modes: Associated Mode and Quasi-Associated Mode.

When operating in the Associated Mode, SS7 signaling progresses from switch to

switch through the PSTN following the same path as the associated facilities that carry

the telephone call. This mode is more economical for small networks. The Associated

Mode of signaling is not the predominant choice of modes in North America.

When operating in the Quasi-Associated Mode, SS7 signaling progresses from theoriginating switch to the terminating switch, following a path through a separate SS7

signaling network composed of STPs. This mode is more economical for large networks

with lightly loaded signaling links. The Quasi-Associated Mode of signaling is the

predominant choice of modes in North America.

SS7 clearly splits the signaling planes and voice circuits. An SS7 network has to be madeup of SS7-capable equipment from end to end in order to provide its full functionality.

The network is made up of several link types (A, B, C, D, E, and F) and three signaling

nodes - Service switching point (SSPs), signal transfer point (STPs), and Service Control

Point (SCPs). Each node is identified on the network by a number, a point code.

Extended services are provided by a database interface at the SCP level using the SS7network.

The links between nodes are full-duplex 56, 64, 1,536, or 1,984 kbit/s graded

communications channels. In Europe they are usually one (64 kbit/s) or all (1,984 kbit/s)timeslots (DS0s) within an E1 facility; in North America one (56 or 64 kbit/s) or all

(1,536 kbit/s) timeslots (DS0As or DS0s) within a T1 facility. One or more signaling

links can be connected to the same two endpoints that together form a signaling link set.

Signaling links are added to link sets to increase the signaling capacity of the link set.

In Europe, SS7 links normally are directly connected between switching exchanges using

F-links. This direct connection is called associated signaling. In North America, SS7links are normally indirectly connected between switching exchanges using an

intervening network of STPs. This indirect connection is called quasi-associated

signaling. Quasi-associated signaling reduces the number of SS7 links necessary tointerconnect all switching exchanges and SCPs in an SS7 signaling network.

SS7 links at higher signaling capacity (1.536 and 1.984 Mbit/s, simply referred to as the

1.5 Mbit/s and 2.0 Mbit/s rates) are calledHigh Speed Links (HSL)in contrast to the low


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speed (56 and 64 kbit/s) links. High Speed Links (HSL) are specified in ITU-TRecommendation Q.703 for the 1.5 Mbit/s and 2.0 Mbit/s rates, and ANSI Standard

T1.111.3 for the 1.536 Mbit/s rate. There are differences between the specifications for

the 1.5 Mbit/s rate.High Speed Linksutilize the entire bandwidth of a T1 (1.536 Mbit/s)or E1 (1.984 Mbit/s) transmission facility for the transport of SS7 signaling messages.

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Documents

3. PCM, Multiplexing & Signalling