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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 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. 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.


There are basically two types of multiplexing techniques i. ii Frequency Division Multiplexing (FDM) Time Division Multiplexing (TDM)


Frequency Division Multiplexing Techniques (FDM)

The FDM techniques is the process of translating individual speech circuits (3003400 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.

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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 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.

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 are connected to individual gates which are opened one by one in a fixed sequence. At the PCM Principles, Multiplexing & Signaling Page 2 of 26

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


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 following processing steps are required Filtering Sampling Quantisation Encoding Line Coding

3.1 3.2

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

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

Fig. 3: Sampling Process

3.2.1 Sampling Theorem 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. 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 Let us say our voice signals are band limited to 4 KHz and let sampling frequency be 8 KHz. Time period of sampling Ts = 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 1 sec 8000

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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.

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 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. 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. The signals on the common medium (also called the common highway) of a TDM system will consist of a series of pulses, the amplitudes of which are proportional to the amplitudes of the individual channels at their respective sampling instants. This is illustrated in Fig. 5

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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 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. 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.

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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. 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 10-20mV, 3040mV 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. 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 the quantizing intervals are of the same size. This is called Linear Quantizing.


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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 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 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'.


3.1.1 Relation between Binary Codes and Number of levels. 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. 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 Signal Quantizing Interval Amplitude Range (mid value) 0-10 mv 10-20mv 20-30 mv 30-40 mv 40-50 mv 5 mv 15mv 25 mv 35 mv 45 mv Quantizing Level Binary Code

0 1 2 3 4

1000 1001 1010 1011 1100

If a sample has an amplitude of say 23 mv or 28 mv, in either case it will be assigned \he \eve\ "2". This Is represented in binary code 1010. When this is decoded at the receiving end, the decoder circuit on receiving a 1010 code will convert this into an analogue signal of amplitude 25 mv only. Thus the process' of quantization leads to an approximation of the input signal with the detected signal having some deviations in amplitude from the actual values. This deviation between the amplitude of samples at the transmitter and receiving ends (i.e. the difference between the actual value & the reconstructed value) gives rise to quantization distortion. 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. 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. To reduce error, we, therefore, need to reduce step size or in other words, increase th,e number of steps in the given amplitude range. This would however, increase the transmission bandwidth because bandwidth B = fm log L. where L is the number of quantum steps and fm is the highest signal frequency. But as we knows from speech statistics that the probability of occurrence of a small amplitude is much greater than large one, it seems appropriate to provide more quantum levels (V = low value) in the small amplitude region and only a few (V = high value) in the region of higher amplitudes. In this case, provided the total number of specified levels remains unchanged, no increase in transmission bandwidth will be required. This will also try to bring about uniformity in signal to noise ratio at all levels of input signal. This type of quantization is called non-uniform quantization. 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 a sample value it is necessary to know the following: 1. 2. 3. The sign of the sample (positive or negative excursion) The segment number 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 16 segments will be taken into account.



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

Polarity bit 1 encoding for + ve 'O' for - ve.

Segment Code

Linear in the segment

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

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

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 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 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, 1 lying between levels + vm/64 -vm/64 being colinear are taken as one segment. The voltage to be encoded corresponding to 2 ends of successive segments are in the ratio of 2. That is vm, vm/2, vm/4, vm/8, vm/16, vm/32, vm/64, vm/128 (vm being the maximum voltage). There are 128 quantification levels in the positive part of the curve and 128 in the negative part of the curve. 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 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.

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 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.



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 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 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. 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.


Structure of Frame

A frame of 125 microseconds duration has 32 time slots. These slots are numbered Ts 0 to Ts 31. Information for providing synchronization between trans and receive ends is passed through a separate time slot. Usually the slot Ts 0 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 are utilized for voltage signal of channels 16 to 30 respectively.



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. The digits or bits of FAW occupy seven out of eight bits of Ts 0 in the following pattern. Bit position of Ts 0 FAW digit value 1 B1 X B2 0 B3 0 B4 B5 1 B6 1 B7 0 B8 1

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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. T AB L E -2 Frame Numbers FO F1 F2 F3 etc B1 X X X X B2 0 1 0 1 B3 0 Y 0 Y B4 1 Y 1 Y B5 1 Y 1 Y B6 0 1 0 1 B7 1 1 1 1 B8 1 1 1 1 FAW ALARM FAW ALARM Remark

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 B3 B4 B5 are set 0. An urgent alarm is indicated by transmitting "all ones". The code word for an urgent alarm would be of the form. X 111 1111



In a telephone network,-the signaling 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 "signaling" in PCM systems. 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 a slow 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 is allocated for carrying signaling information.

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The time slot 16 of each frame carries the signaling 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 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 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 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 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 of frames 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 carrying signaling 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.


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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 worldwide hierarchy for the FDM system (although the term hierarchy is not used 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) (b) (c) (d) (e) the bit rate of the principal signal sources. traffic demand, network topology, operational features, flexibility of the network. time division and multiplexing plant requirements. compatibility with analog equipment. 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 24channel PCM primary multiplex at the various levels of the hierarchy : Level in hierarchy First level Second level Third level Fourth level Fifth level Bit rate 1544 kbit/s 6312 kbit/s 46304 kbit/s 280000 kbit/s 568000 kbit/s T1 T2 L5 (Jumbo Grp) WT4 (Wave guide) T5 Trans. line

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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 3.1 For 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. 3.2 It is also an appropriate bit rate for inter-connection to radio-relay links planned for use at various frequencies. 3.3 At 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) 3.4 In 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. 3.5 The 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. 3.6 Moreover, 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. 3.7 The 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 4.2

The critical point in this hierarchy is whether or not the third level at 34368 kbit/s should exist. The C.C.I.T.T. has agreed after long discussions on the following (Recommendation G.751) that there should be a 4th order bit rate of 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 4th order 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.


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.


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.


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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 8448 kbit/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 PCM Principles, Multiplexing & Signaling Page 22 of 26

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. xxxx SIGNALLING IN TELECOMMUNICATIONS The term signaling, when used in telephony, refers to the exchange of control information 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 the case 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 to significant 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 a form 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 in the 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 control lines 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 a communications 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 is sufficient 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 traffic providing 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 two signaling methods are very similar since they share a common heritage and in some cases, 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 what CCS and CAS are to the channel. Signaling System Number #7 SS7 is a set of telephony signaling protocols which are used to set up most of the world's public 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 7 and CCIS7.

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There is only one international SS7 protocol defined by ITU-T in its Q.700-series recommendations. 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 the originating 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 made up 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 SS7 network. 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, SS7 links 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 to interconnect 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 called High 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-T Recommendation 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 Links utilize 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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