Digital Transmission and PCM

7/27/2019 Digital Transmission and PCM

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5Digital Transmission andPulse Code Modulation

Following trials in the196Os, digital telecommunications systems were first idely deployed in the1970s. Since then, the miniaturization and large scale integration of electronic components andthe rapid advance n computer technology have made digital technology the obvioushoice for allnew elecommunications ransmissionandswitchingsystems.Now, n henetworks of mostcountriesnd withwiderworld internat ional satellite andubmarineetworks, digitaltransmission has no rivals. So what exactly is it, and what can we gain by it?

5.1 DIGITAL TRANSMISSIONIn investigating analogue transmission we found a relationship between bandwidth andoverall information carrying capacity, and we described frequency division multiplexing( F D M ) .This was a methodof reducing the number of physical wires needed to carry amultitude of individual channels between two points, and it worked by sharing out theoverall bandwidth of a single set of four-wires (transmit and receive pairs) between allthe channels to be carried. We now discuss digital transmission in detail, how it works,and the equivalents of analogue bandwidth and channel multiplexing. In contrast withanalogue networks, digital networks are deal for the direct carriage of data, because asthe name suggests a digital transmission medium carries information in the form ofindividual digits. Not just any type of digits, but binary digits (bits) in particular.The medium used in digital transmission systems is usually designed so that it is onlyelectrically stable in one ofwo states, equivalent to on (binary value l) or offbinaryvalue 0).Thus a simple form of digital line system might use an electrical current as theconveying medium, and control the current toluctuate between two values, current onand current off. A more recent digital transmission medium using optical fibre (whichconsists of a hair-thin (50 microns in diameter)strand of glass) conveys the digital signalin he form of on and off light signals, usually generated by some sort of semi-conductor electronic device such as a laser or a light e mitting diode ( L E D ) . Any othermedium capable of displaying distinct on/off states could also be used.

55

Networks and Telecommunications: Design and Operation, Second Edition.Martin P. Clark

Copyright 1991, 1997 John Wiley & Sons LtdISBNs: 0-471-97346-7 (Hardback); 0-470-84158-3 (Electronic)


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56 DIGITALRANSMISSION AND PULSEODEODULATIONFor simultaneous two-way (or uplex) digital transmission a four-wire or equivalent)transmission medium is always required. Just as with four-wire analogue transmission(as discussed in Chapter 3) one pair of wires (or its equivalent, for example an opticalfibre) is used for the ransmit (Tx) irection whereas the other pair s used for the receive

(Rx)direction. These allow the digital pulses to pass in both directions simultaneouslywithout nterference. Thus they differ from simple local analogue elephone systemswhich can be made to work adequately in a two-way mode over only two wires (recallFigure 5.6 of Chapter 2).Theprincipaladvantage of digitaloveranalogue ransmission is the mprovedquality of connection. With only twoallowed states on the line (offand on) it is notall that easy to confuse them even when the signal is distorted slightly along the line byelectrical noise or interference or some other cause.Digital line systems are husrelatively immune to interference. As Figure 5.1 demonstrates, the receiving end onlyneeds to detect whether the received signal is above or below a given threshold value.If the pulse shape is not a clean square shape, it does not matter. Allow the sameelectrical dis turbance to nterfere with an analogue signal, and theesult would be a owvolume racklingnoise at the receiving end, which could well make the signalincomprehensible.To make digital transmission still more immune to noise, it is normal practice toregenerate the signals at intervalsalong he ine.Aregenerator reduces the risk ofmisinterpreting he received bitstream at thedistantend of a ong-haul line, bycounteracting the effects of attenuation and distortion, which show up in digital signalsas pulse shape distortions. In thiscorrectivefunctionadigitalregeneratormayberegarded as the equivalent of an analogue repeater.The process of regeneration involves detecting the received signal and recreating anew, clean square wave for onward transmission. The principle is shown in Figure 5.2.The regeneration of digital signals is all that is needed to restore the signal to itsoriginal form; there is no need to amplify, equalise o r process it in any other way. Thefact that the signal can be regenerated exactly is the reason why digital transmissionproduces signals of such high quality.

Pu lses a f f ec t edby noise

on va lue(b inary * l )

1 0 1 0

r \I

Detec t ionthresholdI_ va lue_ _ _ _ _ _ _ - - _ _ - - - - - --l 1 0 1 0 - b i t s t r i n gTransmi t ted-e t e c t e d a l u edespite noises t i l l c o r r e c t ,

Figure 5.1 Digital signal and immunity to noise


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PULSE 57D i s t o r t e de c e i v e di g n a le g e n e r a t e d R e g e n e r a t o r

I O 1 O I O I O 1 0 1 0 1 0 1 0R e c e i v e d bit P a t t e r nr a n s m i t t e di ta t t e r n

Figure 5.2 The principle of regeneration

Errors at the detection stage can be caused by noise, giving the impression of a pulsewhen there is none. Their likelihood can, however, be reducedy stepping up the lectricalpower (which effectively increases the overall pulse size or height), and a probabilityequivalent to one error in several hours or even days of transmission can be obtained(a so-called bit error rute or more correctly bit error ratio ( B E R ) of 1 errored bit in1 million bits is noted as BER = 1 X 1OP6). This is good enough for speech, but ifthe circuit is to be used for data transmission it will not be adequate; the error ratemay need to be reduced still further, and that requires a special technique using anerror checking code. Typical line systems nowadays have BER of 10-9, but with errorchecking techniques this can be improved to l O - I 3 .A digital line system may be designed to run at almost any bit speed, but on a singledigital circuit it is usually 64 kbit/s. This is equivalent to a 4 kHz analogue telephonechannel,as we shall see shortly.The bit speed of adigital line system is roughlyequivalent to the bandwidth of an analogue line system; the more information there isto be carried, the greater the required bit speed. Later in the chapter we also discusshow ndividual 64 kbit/sdigitalchannels can bemultiplexed ogether on a singlephysical circuit, by a method known as ime division multiple x (or T D M ) .TDM has thesame multiplying effect on the circuit carrying capacity of digital line systems as FDMhas for analogue systems.

5.2 PULSECODEMODULATIONYou may well ask, how is a speech signal, a TV signal or any other analogue signal tobe converted into a form that canbe conveyed digitally? The answer lies in a method ofanalogue to digital signal conversion known as pulse code m odulation.Pulse code modulation ( P C M ) unctions by converting analogue signals into a formatcompatible with digital transmission, and it consists of four stages. First, there is thetranslation of analogue electrical signals into digital pulses. Second, these pulses arecoded into a sequence suitable for transmission. Third, they are transmitted over thedigitalmedium. Fourth, they are ranslated back nto heanalogue signal (oranapproximation of it) at the receiving end. PCM was invented as early as 1939, but it wasonly in the 1960s that it began to be widely applied. This was mainly because before theday of solid stateelectronics we did not have he echnology toapply heknownprinciples of PCM effectively.


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58 DIGITALRANSMISSION AND PULSEODEODULATIONSpeech or any other analogue signals are converted into a sequence of binary digitsby sampling the signal waveform at regular intervals. At each sampling instant thewaveformamplitude is determined and,according o tsmagnitude, is assignedanumerical value, which is then coded into its binary form and transmitted over thetransport medium. At theeceiving end, the original electrical signals reconstructed by

translating it back from the incoming digitalized signal. The technique is illustrated inFigure 5.3, which shows a typical speech signal, with amplitude plotted against time.Sampling is pre-determined to occur at intervals of time t (usuallymeasured nmicroseconds). The numerical values of the sampled amplitudes, and their 8-bit binarytranslations, are shown in Table 5.1.Because theuse of decimal points wouldmake thebusiness more complex nd increasethe bandwidth required for transmission, amplitude is represented by integer valuesonly. When the waveform amplitude does not correspond to an exact integer value, asoccurs at time 4 t in Figure 5.3, an approximation s made. Hence at 4t , value -2 is usedinstead of the exact value of -2.4. This reduces the otal number of digits that need to besent. The signal is reconstitutedat the receiving end by generating a stepped waveform,each step of duration t , with amplitude according to the digit value received. The signalof Figure 5.3 is therefore reconstituted as shown in Figure 5.4.In he example, the econstitutedsignalhasasquarewaveform ather han hesmooth continuous form f the original signal. This approximationffects the listenerscomprehension to an extent which depends on the amount f inaccuracy involved. Thesimilarity of the reconstituted signal to the original may be improved by0 increasing the ampling ate i.e. educing he ime eparation of samples) toincrease the number of points on the horizontalaxis of Figure 5.3 at which samplesare taken, and/or

S a m p l e d i g n a lo r i g i n a l )A m p l i t u d e

4430

2 4

4

0-0

-24

Time

-301Figure 5.3 Sampling a waveform


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PULSE 59Table 5.1 Waveform samples from Figure 4.1

Decimal numeric 8-bit binaryTime Amplitude value translation0 0 0 00000000t 2a 2 000000102t U 1 000000013 t 4a 4 00000100

0 increasinghenumberofquantization levels (i.e. wave amplitude levels). Thequantization levels are the points on the vertical scale of Figure 5.3.However, without an infinite sampling rate and an infinite range of quantum values, itis impossible tomatch an originalnalogueignal precisely. Consequently anirrecoverable element of quantization noise is introduced in the course of translatingtheoriginalanalogue signal into ts digitalequivalent.Thesampling ateand henumber of quantization levels need to be carefully chosen to keep this noise down tolevels at which the received signal is comprehensible to the listener. The snag is that thegreater the sampling rate and the greater the number of quantizationevels, the greateris the digital bit rate required to carry the signal. Here again a parallel can be foundwith the bandwidth of an analogue ransmissionmedium,where hegreater is therequired3delity of an analogue signal, the greater is the bandwidth required.The minimum acceptable sampling rate for carrying a given analogue signal usingdigital ransmission is calculatedaccording to a scientific principleknown as heNyquist criterion, (after the man whodiscovered it). The criterion states that the samplerate must be at least double the frequency of the analogue signal being sampled. For a

I R e c o n s t i t u t e d signal

Figure 5.4 Reconstruction of th e waveform of Figure 4.1 from transmitted samples


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60 DIGITALRANSMISSION AND PULSEODEODULATIONstandard speech channel this equals 2 X 4 kHz = 8000 samples per second, the normalbandwidth of a speech channel being 4 kHz.The number of quantization levels found (by subjective tests) to be appropriate forgood speech comprehension is 256. In binary digit (bit) terms this equates ton eight bitnumber, so that the quantum value of each sample is represented by eight bits. Therequired transmission rate of a digital speech channel is therefore 8000 samples persecond, times8 bits, or 64 kbit/s. In other words digital channel of 64 kbit/s capacity isequivalent to an analogue telephone channel bandwidth of 4 kHz. This is the reasonwhy the basic digital channel is designed to run at 64 kbit/s.

5.3 QUANTIZATIONWhen heamplitude level at asamplepoint,does not exactlymatchone of thequantization levels, an approximation is made whichntroduceswhat is calledquantization noise (also quantizingnoise). Now, if the 256 quantization levels wereequally spaced over the amplitude range f the analogue signal, then the low amplitudesignals would ncur far greater percentage quantization errors (and hus distortion)than higher amplitude signals. For this reason, the quantization levels are not linearlyspaced, but instead are moredensely packed around the zero amplitude evel, as shownin Figure 5.5. This gives better signal quality in the low amplitude range and a more

5

Quanti tat ion k v e k h( n o n - l i n e a r )321

- 1

- 2- 3- h

- 5Figure 5.5 Non-linearquantization levels


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QUANTIZATION NOISE 61consistentlyclear signal across hewholeamplitude ange.Twoparticular sets ofquantization levels are in common use for speech signal quantization. They are calledthe A-Law code and he p-Law Mu-L aw) code. Bothhave higher density ofquantization levels around the zero amplitude evel, and bothuse an eight bit (256 level)coding echnique.Theyonly differ in heactualamplitude values chosen as theirrespective quantum levels. The A-law code is theEuropean tandard or speechquantization and the Mu-law code is used in North America. Unfortunately, becausethe odes ave on-corresponding uantization levels, conversionquipment isrequired for nterworkingand hisadds to thequantization noise of aconnectioncomprising both A-law and Mu-law digital transmission plant.Conversion rom A-law to p-law (Mu-law)code or vice versa) amounts o acompromise between the different quantization levels. An 8-bit binary number in one ofthe codes corresponds to a particular quantization value at a particular sample instant.This S-bit number is converted into the S-bit number corresponding to the nearest aluein the other quantization code. The conversion s therefore a relatively simple matter ofmapping (i.e. converting) between one eight bit value and another.

5.4 QUANTIZATIONNOISEMost noise heard by the listener on a digital speech circuit is the noise introducedduring quantization rather than the result of interfering electromagnetic noise addedalong the line, and it is minimized by applying the special A-law and p-law codes asalreadydiscussed. The otal amount of quantizationnoise(quantizing noise) on areceived signal is usually quoted in terms of the number of quantization levels by whichthe signal differs from the original.

This value is quoted as a number of quantization distortion units (or qdus). Typicallythe acceptable maximum number of qdus allowed on a complete end-to-end connectionis less than 10 (taking into account any A-to-p law code conversion or other signalprocessing undertaken on the connection). Another possible type of speech processingis the technique of speech compression, and we shall see in Chapter 38 that the overallbit rate canbe reduced by speech compression, at the cost of some increase in quantiza-tion noise.Quantization noise only occurs in the presence of a signal. Thus, the quiet periodsduring a conversation are indeed quiet. This quietness gives an improved subjectiveview of the quality of digital transmission.

5.5 TIME-DIVISION MULTIPLEXINGAs digital transmission is by discrete pulses and not continuous ignals, it is possible forthe information of more than one64kbit/s channel to be transmitted on the same path,so long as the transmission rate (i.e. bit rate) is high enough to carry the bits from anumber of channels. In practice this is done by interleaving the pulses from the variouschannels in such way that a sequence of eight pulses (called a byte or an c t e t ) from thefirst channel is followed by a sequence of eight from the second channel,and so on. Theprinciple is illustrated in Figure 5.6, in which the TDM equipment could be imagined to


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62 DIGITALRANSMISSION AND PULSE CODE MODULATION

m

c


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TIME-DIVISION MULTIPLEXING 63be a rotating switch, picking up in turn 8 bits (or 1 by te ) from each of the input channelsA, B and C in turn. Thus the output bit stream of the TDM equipment is seen tocomprise, in turn, byte A1 (from channel A), byte B1 (from channel B), byte C1 (fromchannel C), then, cycling again, byte A2, byte B2, byte C2 andso on. Note that higherbit rate is required on the output channel, to ensure that all the incoming data from allthree channels can e transmitted onward. As X 2 = 6 bytes of data are eceived on theincoming side during a time period of 250 microseconds (1 byte on each channel every125 microseconds), all of them have to be transmitted on the outgoing circuit inn equalamount of time. As only a single channel is used for output, this implies a rate of6 X 8 = 48 bits in 250 microseconds, i.e. 192 kbit/s. (Unsurprisingly, the result is equal to3 X 64 kbit/s). Thus, inffect the various channels time-share the outgoing transmissionpath. The technique is known as time-division multiplexing ( T D M ) .TDM can either be carried out by interleaving a byte (i.e. 8 bits) from each tribu-tary channel in turn, or it can be done by single bit interleaving. Figure 5.6 shows themorecommonmethod of byte nterleaving. The use of the TDM technique is socommonon digital line systems that physical circuitscarryingonly64kbit/s areextremely rare, so that digital line terminatingequipmentusually includes amulti-plexing function. Figure 5.7 shows a typical digital ine terminating equipment, used toconvert between a number of individual analogue channels (carried on a number ofindividual physical circuits) and a single digital bit stream carried on a single physicalcircuit. The equipment shown is called a primary multiplexor. A primary multiplexor( P M U X ) contains an analogue to digital conversion facility for individual telephonechannel conversion to 64 kbit/s, and additionally a time division multiplex facility. InFigure 5.7 a PMUX of European origin is illustrated, converting 30 analogue channelsinto A-law encoded 64 kbit/s digital format, and then time division multiplexing all ofthese 64 kbit/schannels into a single 2.048 Mbit/s (El) digital line system. (2.048 Mbit/s

6.4 k b i t l s d a t a- I 0 -- I D- I D30 i n d i v i d u a l - I Da n a l o g u ec i r c u i t s ,(1-15 a n d 17-31) lII

III

30- A I D317 I Dl It

2.0.48 M b i t / s l inew i r es y s tern

A n a l o g u e t o d i g i t a l s i g n a lc o n v er s io n , u s i n g A - l a wp u l s e c o d em o d u l a t i o n ( P C M ]Figure 5.7 Europeanprimary multiplexor


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64 DIGITALRANSMISSION AND PULSE CODEODULATIONis actually equal to 32 X 64 kbit/s, but channels 0 and 16 of the European system aregenerally used for purposes other than carriage of information.)Wecouldequally well have llustrateda NorthAmerican version PMUX.Thedifference would have been the use of p-law encoding and the ultiplexing of 24 channelsinto a 1.544 Mbit/s transmission format also called a T-span, a TI line or a DS 1 . (T =Transmission, DS=digital line system) (1.544 Mbit/s = 24 X 64 kbit/s plus 8 kbit/s).The transmitting equipment of a digital line system has the job of multiplexing thebytes from all theconstituentchannels.Conversely, he receiving equipmentmustdisassemble hesebytes n precisely thecorrectorder.This requires ynchronousoperation of transmitter and receiver, and to this end particular patterns of pulses aretransmitted at set intervals, so that alignment and synchronism can be maintained.These extra pulses are sent in channel of the European 2 Mbit/s digital system, and inthe extra 8 kbit/s of the North American 1.5 Mbit/s system.

5.6 HIGHER BITRATES OF DIGITAL LINE SYSTEMSThe number of channels multiplexed on a carrier depends on the overall rate of bittransmission on the line. Given that each channel mustbe transmitted at 64 kbits/s, theoverall bit speed is usually related to an integer multiple of 64 kbit/s. There are threebasic hierarchies of transmission rates which have been standardized for internationaluse, but these extend to higher bit rates than the 2.048 Mbit/s and 1.544 Mbit/s versionsso far discussed.The ITU-T(formerly CCITT, Consultative Committee for International ele-phonesand Telegraphs), CEPT(EuropeanCommitteeforPostsand Telecommunications)andETSI EuropeanTelecommunicationsStandards nstitute) have tandardized2.048 Mbit/s as the primary igital bandwidth (El line system) and A-Law as thepeechencodingalgorithm.Thishas 32 channels, 30 for speech and two oralignmentsynchronization and signalling, more of which we shall discuss later in the chapter.Higherransmissionates in theEuropean digital ierarchy are ttained byinterleaving a number of 2 Mbit/s systems as illustrated in Figure 5.8. The standardizedrates are:

2.048 Mbit/s, referred to as E l or 2 Mbit/s8.448 Mbit/s, referred to as E2 or 8 Mbit/s (4 X 2 Mbit/s)34.368 Mbit/s, referred to as E3 or 34 Mbit/s (4 X 8Mbit/s)139.264 Mbit/s, referred to as E4 or 140 Mbit/s (4 X 34 Mbit/s)564.992 Mbit/s, referred to as E5 or 565 Mbit/s (4 X 140 Mbit/s)

Multiplexing equipment is available for any of the rate conversions, as Figure 5.8shows.In the second ITU-T standard (which currently predominates in North America), adifferent multiplex hierarchys recommended and is shown below. This is based on a basicblock of 24 X 64 kbit/s channels plus kbit/s forfrarning, giving a bitrate of 1.544 Mbit/s(T1 line system). p-Law encoding is used for pulse code modulation of speech signals.The principles of multiplexing, however, are largely the same, and diagrams similar toFigure 5.8 could have been drawn,


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DIGITAL FRAME FORMATTING AND JUSTIFICATION 65

Fx lLOMbi t /sFigure 5.8 European digitalmultiplexhierarchy

DSO = 64 kbit/s, the basic channelT1 or DS1 = 1.544Mbit/s (this is called the T-span, T1 or DS1 system)T2 or DS2= 6.3 12 Mbit/s (4 X 1.5 Mbit/s)T3 or DS3= 44.736 Mbit/s (7 X 6 Mbit/s), sometimes referred to as 45 Mbit/sDS4 = 139.264 Mbit/s (3 X 45 Mbit/s)278.176 Mbit/s (6 X 45 Mbit/s)In the third system, predominant in Japan, yet another hierarchy is used, thoughthere is some overlap with the North American system. p-Law encoding is applied tospeech pulse code modulation.

DSO= 64 kbit/s, the basic channelJ1 = 1.544Mbit/s (this is called the T-span, T1 or DS1 system)J2= 6.312Mbit/s (4 X 1.5Mbit/s)53= 32.064 Mbit/s (5 X 6 Mbit/s)54= 97.728 Mbit/s (3 X 32 Mbit/s)The various hierarchies are incompatible at ll levels (including the basic speech channellevel, on account of the different quantization code used by the A andp-law PCM algo-rithms). Interworking equipment is therefore required for international links betweenadministrationsemploying he different hierarchies.Ingeneral, his nterworking isundertaken in the country which uses the 1.544 Mbit/s standard.Before we leave the subject of nomenclature for digital stream bitrates, we shouldalso mention the terminology sed particularly for high speed video channels. These areH0 (384 kbit/s, H11 (1536 kbit/s) and H12 1920 kbit/s). These correspond,respectively,to 6 X 64 kbit/s, 24 X 64 kbit/s and 30 X 64 kbit/s. All three bitrates maybe supported byan El line system, only the first two from a T1 or J1 system.

5.7 DIGITAL FRAME FORMATTING AND JUSTIFICATIONAs we noted earlier in the chapter, it is common in a 2.048 Mbit/s system to use onlythirty 64 kbit/s hannels representing only 1.920 Mbit/s) or ctual arriage of


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66 DIGITALRANSMISSION AN D PULSEOD EODULATIONinformation. This leaves an additional 128 kbit/s bit rate available. Similarly, in the1.544 Mbit/s system, the bit rate required to carry twenty-four 64 kbit/s channels is only1.536 Mbit/s, and 8 kbit/s are left over. The burning question: what becomes of thisspare capacity? The answer: it is used for synchronization and signalling functions.Consider a 2.048 Mbit/s bit stream, and in particular the bits carried during a singletime interval of 125 microseconds. During a periodof 125 microseconds a single sampleof 8 bits will have been taken from each of the 30 constituent or tributary channelsmaking up the 2.048 Mbit/s bit stream. These are structured into an imaginary f r a m e ,each rameconsisting of 32 consecutive t imeslots , one timeslot of 8 bits for eachtributary channel. Overall the f r a m e represents a snapshot image, one sample of 8 bitstaken from each of the 30 channels, at a frequency of one frame every 125p . Eachframe is structured in the same way, so that the first timeslot of eight bits holds theeight-bit sample from tributary channel 1, the second timeslot the sample from channel2, and so on. The principle is shown in Figure 5.6. It is very like a single frame of amovie film; the only thing missing s the equivalent of the film perforations which allowamovieprojector to move ach freeze-frameprecisely.This film perforationsfunction is in fact performed by the first timeslot in the frame. It is given the nametimeslot 0. It carries so-called f raming and synchronization information, providing aclear mark o ndicate hestart of each f r a m e and an indication of the exactbittransmission speed. The principles shown in Figure 5.9, which illustrates a ingle frameof 32 timeslots.Timeslot 0 then provides a mark for framing. Timeslots 1-15 and 17-31 are used tocarry the tributary channels. That leaves timeslot 16 which, as we shall see, is used forsignalling.We cannot leave timeslot zero, without briefly discussing its synchronization functionwhichserves to keep the linesystembit raterunning at precisely the ightspeed.Consider a wholly digital network consisting of three digital exchanges A, B and Cinterconnected by 2.048 Mbit/s digital transmission links, as shown in Figure .10, withend users connected to exchanges A and C.Each of the exchanges A, B and C in Figure.10 will be designed to input and eceivedata from thedigital transmission linksA-B and B-C at 2.048 Mbit/s. What happens flink A-B actually runs at 2 048 000bit/s, while link B-C runs at 2 48001bit/s? This,orsomething even worse,couldquite easily happen npractice ifwe did not akesynchronization steps to prevent it. In the circumstances shown, the bit stream receivedby exchange B from exchange A is not fast enough to fill the outgoing timeslots on thelink from B to C correctly, and a slip of 1 wasted bit will occur once per second.Conversely, in the direction from C to A via B, unsent bits will gradually be stored upby exchange B at a rate of 1 extra unsent bit per second, because the exchanges unableto transmit thebits to A as fast as t is receiving them from exchange C. Ultimately bitsare lost when the store in exchange B overflows. Neither slip nor overflow of bits isdesirable, so networks are normally designed to be synchronous at the 2.048 Mbit/slevel, in other words are controlled to run at exactly the same speed. Actually, they runplesiochronously. Some of the bits in timeslot zero of a 2.048 Mbit/s line system areused to try to maintain the synchronization, but systems are not firmly locked in-step.Each system instead runs from its own clock. The synchronization bits adjust the speedof the clock (faster or slower) to keep it in step with other systems, but as there is morethan one clock in the network, there is still a discrepancy in the synchronization of the


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DIGITAL AN D 'JUSTIFICATION 67

mU E


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68 .DIGITAL TRANSMISSION AND PULSEODEODULATION

1 b i t s l i p ' a d d e db y B e a c h s ec o n d

-I 1 b i t S< up Ib y B each second

( r u n s at20L8 000 b i t l s ] 2 0 4 801 b i t / s l( r u n sa t

Figure 5.10 The need for synchronization

various systems, hence the term plesio-chronous. The same plesiochronous functions offraming and synchronization are carried ou t by the surplus 8 kbit/s capacity of he1.544 Mbit/s digital line system.Because the speed of the system is actually slightly greater han he sum of thetributary inputchannels,extra dummy bits(also called justLfication hits or stufing,leading to the term it s tu f ing ) need to be added to the stream. hese can be removed atthe receiving multiplexor. Should one of the tributaries be supplying data (bits)slightlyfaster han tsnominated ate, this can be accommodated by the multiplexor bysubstituting some of the justzjication hits with user data. Similarly, if the rate of theinput channel is slightly too slow, more justification bits (J ) can be added (Figure 5.11).Now let us consider the function of timeslot 16 in a 2.048 Mbit/s digital line system.This timeslot is usually reserved for carrying the signalling information needed to setup the calls on the 30 user channels. The function of signalling information is to conveythe intended destination of a call on a particular channel between one exchange andthe next.From the above, we see that the maximum usable bit rate of a 2.048 Mbit/s system is30 X 64kbit/s or 1.920 Mbit/s.In occasionalcircumstances,however, this can beincreased to 1.984 Mbit/s when the signalling channel (timeslot 16) is not needed.When required on a 1.544 Mbit/s line system, a signalling channel can be made avail-able by stealing a small number of bits (equivalent to 4 kbit/s) from onef the tributary

fast incomingtributary

3JS-p-slow incomingtributary

bitrate adaptor J l J Wlocaloscillator

J I J I J W

Figure 5.11 The process of justification (plesiochronousdigital hierarchy)


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INTERWORKING THE 2MBITjS AND 1.5MBITjSIERARCHIES 69channels thereby educing hecapacity of thatparticular channel to 60 kbit/s).Alternatively, and nowadays more usually, one whole 64 kbit/s channel may be dedi-cated for signalling use. The method of stealing bits to create a 4 kbit/s signallingchannel is known in NorthAmerica as robbed bit signalling. It is only permissible to robthe bits from a voice channel and not froma data channel. Robbing a small number ofbitsfroma voice channel is permissible because thequality ost hereby is almostimperceptible to a human telephone istener. Robbing bits from a channel which iscarrying data, however, will result in quite unacceptable data corruption. As users oftelephone networks have become accustomed to transmitting data signals (e.g. fax),robbed bit signalling has become less acceptable. Where a signalling channel is requiredon a 1.544 Mbit/s digital line system carrying only data ircuits, a whole channel shouldbe dedicated to signalling. Such a dedicated ignalling channel is necessary to create SS7signalling links between computer-controlled telephone exchanges.To return o he two different bit rate hierarchies,observant eadersmayhavenoticed that the higher bit rates of both hierarchies are notexact integer multiplesof thebasic 2.048 Mbit/s and 1.544 Mbit/s tributaries. Instead, some extra fvaming bits havebeen added once again at eachhierarchial level. These are provided for hesameframing reasons as have already been described in connection with the2.048 Mbit/s linesystem, and illustrated nFigure 5.9. However,unlike heir2Mbit/s or1.5Mbit/stributaries,synchronization of higher bit-rate line systems in PDH (plesiochronousdigital hierarchy) is not usually undertaken. Instead, higher order systems are generallyallowed to free-run. The extra bits allow free running, as a slightly higher bit rate isavailable than he ributaries can feed. The higher bit rate ensures that there is nopossibility of bits building up between the tributaries and the igher bit rate line systemitself. Instead there will always be a few bits to spare. The benefit is that the need forsynchronization a t the higher bit rate is avoided, but the penalty is the complicatedframe structure needed a t the higher rates of the hierarchy. The same problem facesusers of the 1.544 Mbit/s hierarchy.

5.8 INTERWORKING THE 2 MBIT/S AND 1.5MBIT/S HIERARCHIESInterworking of digital line systems running in the 2 Mbit/s hierarchy and 1.5 Mbit/shierarchy is relatively straightforward, given the availability of proprietary equipmentfor the conversion. At its simplest, the24 channels of a 1.5Mbit/s system can be carriedwithin a 2 Mbit/s system, effectively wasting the remaining capacity of the 2 Mbit/s sys-tem. Alternatively, a 2 Mbit/s system can be entirely carried on two 1.5Mbit/s systems,wasting 16 channels of the second 1.5 Mbit/s system. More efficiently, however, four1.5 Mbit/s systems fit almost exactly into three 2 Mbit/s systems or vice versa. (Theyappear to fit exactly, but usually some bits are taken for separating the different framesso that the efficiency is reduced slightly).The interworking of one digital hierarchy into the other needs only involve map-ping the individual 8-bit timeslots from one hierarchy into corresponding timeslots inthe other. The technique is called timeslot interchange. The only complication is whenthe 8-bit patterns in the timeslots are not simple data patterns (data patterns shouldbe mapped across unchanged) but whenhey are sample patterns corresponding to A r


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70 DIGITALRANSMISSION AND PULSE CODEODULATION

Timeslotnterchongcequipment1.5Mbi t I s 2Lh .) Z M b i l l s

8ch c1.SMbi t ls 16 eh - 2 M b i t l s- 6ch - c1.5Mbit ls -L ch

c

1.5Mbit ls ZM b i t l s21ch *Mu- low speech l MU to A - l o w P C M lA-lawpeech 1

thosechonnels equiring i t 1Conversion carried outon

Figure 5.12 Timeslot interchange between 1.5Mbit/s and 2Mbit/s

p-law pulse code modulated speech. In this instance, an A- to p-law (Mu-law) speechconversion is also required at the 2 Mbit/s to 1.5 Mbit/s interworking point.Figure 5.12 llustrates a ypical imeslot nterchange between four 1.5 Mbit/s andthree 2Mbit/s digital line systems.Note that the timeslot interchange equipment in Figure 5.12 is also capable of p- toA-law conversion (and vice versa). This has to be available on eachof the channels, butis only employed when he channel is carrying a speech call. When there are consecutivespeech and data calls on the same channel, the p- to A-law conversion equipment willhave to be switched on for the first call and off for the second.Some means is thereforeneeded of indicating to the timeslot interchange equipment whether at any particulartime it is carrying a speech or a data call. Alternatively, particular channels could bepre-assignedeither to speech ordata use. In thiscase he p- to A-law conversionequipment will be permanently on and permanently off, respectively.

5.9 SYNCHRONOUS FRAME FORMATTINGModern linesystems, specifically SD H (synchronousdigitalhierarchy) and S O N E T(synchronous optical network) demand synchronous operation of all the line systemswithin a network (i.e. all must operate using he same clock). In return, t offers asimpler and more regular frame structure of 2Mbit/s and 1.5 Mbit/s tributarieswithinthe higher bitrates (multiples of 155Mbit/s). Aswe will discuss in Chapter 13 this givesmuch greater flexibility in management and administration of the system. A furthersignificant benefit is their support of both 2Mbit /s and 1.5 Mbit/s based hierarchies,creating an easy migration path forworldwide standardization. Table 5.2 lists the basicbitrate hierarchiesof both SDH andSONET.Amore detailedanalysisfollows nChapter 13.


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LINE CODING 71Table 5.2 SDH (synchronousdigitalhierarchy)and SONET (synchronousoptical network)

North American SONET Carried Bitrate Mbit/sDHVT 1.5VT 2.0VT 3.0VT 6.0

-

STS-1 (OC- 1)STS-3 (OC-3)STS-6 (OC-6)STS-9 (OC-9)

STS-12(OC-12)STS- 18 (OC-18)STS-24 (OC-24)STS-36 (OC-36)STS-48 (OC-48)STS-96 (OC-96)STS-192(OC-192)

-

1.5442.0483.1526.3128.44834.36844.736149.7651.84155.52311.04466.56

622.08933.121244.161866.242488.324976.649953.28

VC-l 1VC- 12VC-21VC-22VC-3 1VC-32VC-4STM- 1

-

STM-4-

STM- 16-

STM-64-

5.10 LINE CODINGThe basic information to be transported over any digital line system, irrespective of itshierarchical level, is a sequence of ones and zeros, also referred to asm a r k s and spaces.The sequence is not usually sent directly to line, but is first arranged according to a linecode. Thisaids ntermediate egenerator iminganddistantend receiver timing,maximizing the possible regenerator separation and generally optimizing the operationof the line system. The potential problem is that if either a long string of Os or 1s weresent to line consecutively then the line would appear to be either permanently on orpermanently off. Effectively a direct current condition is transmitted to line. This isnot advisable for two reasons. First the power requirements increased and the attenua-tion is greater for direct as opposed to alternating current. Second, any subsequentdevices in the line cannot distinguish the beginning and end of each individual bit. Theycannot tell if the line is actually still alive. The problem gets worse as the number ofconsecutive Os or 1s increases. Line codes herefore seek to ensure that a minimumfrequency of line state changes is maintained.Figure 5.13 illustrates the most commonly used line codes. Generally they all seek toeliminate long sequences of 1s or Os, and try to be balanced codes, i.e. producing a netzero direct current voltage (thus the three state codes CM1 and HDB3 try to negatepositive pulses with negative ones). This reduces the problems of transmitting poweracross he line. Themore sophisticatedmodern echniquessimultaneously seek to


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72 DIGITALRANSMISSION AND PULSEODEODULATION

NRZ (non return-to-zero)NRZI (non return.to-zero inverted)RZ (return-to-zero)CM1 (coded markinversion)Manchester

diff. Manchester

MillerAMI (alternate markInversion)HDB3 (high densitybipolar, order 3)

1 0 1 0 0 0 0 1

Figure 5.13 Commonly used line codes fo r digital line systems

reduce the frequency of line state changes (thebaud rate)so that higher user bitrates canbe carried.The simplest line code illustrated in Figure 5.13 is a non-return to zero ( NR Z ) ode inwhich 1=on and 0= off. This is perhaps the easiest to understand.In NRZI (non-return-to-zero inverted) it is the presence or absence of a transitionwhich represents a 0 or a 1. This retains the relative simplicity of the code but may beadvantageous where the line spends muchof its time in an idle mode in which a stringof 1s or Os may be sent. Such is the case, for example, betweenan asynchronous terminaland a host computer or cluster controller. NRZI is used widely by the IBM company forsuch connections.A return-to-zero ( R Z )code is like NRZ except that marks return to zero midwaythrough the bit period, and not at the end of the bit. Such coding has the advantage oflower required power and constant mark pulse length in comparison with basic NRZ.The length of the pulserelative to he otal bit period is known as he dutycycle.Synchronization and timing adjustment can thus e achieved without affecting themurkpulse duration.A variationof he NRZ and RZ codes is the C M I (codedmark nversion) coderecommended by ITU-T. In CMI, a 0 is represented by the two signal amplitudes A1and A 2 which are transmitted consecutively, each for half the bit duration. 1s are sentas fullbit duration pulses of one of the two inesignalamplitudes, heamplitudealternating between A1 and A2 between consecutive marks.


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LINE CODING 73In the Manchester code, a higher pulse density helps to maintain synchronizationbet,ween the two communicating devices. Here the transition from high-to-low repre-sents a 1 and the reverse transition (from low-to-high) a. The Manchester code is usedin e th ern e t L A N s (Chapter 19).In he differentialManchestercode avoltage ransition at hebitstartpoint is

generated whenever a binary 0 is transmitted but remains the same for binary 1. TheIEEE 802.5 specification of the token ring LAN (Chapter 19) demands differentialManchester oding. nbothhe Manchester and dlfferentialManchester codingschemes, two extra coding violation symbols exist, J and K. These allow for bi t s tu f ingas previously discussed.In the Miller code, a transition either low-to-high or high-to-low represents a 1. Notransition means a 0.The A M I (a lt ern a te ma rk in versio n)nd HDB3 (high density bipolar) odes defined byITU-T (recommendation G.703) are both three-state, rather than simple two-state (on/off) codes. In these codes, as canbe seen in Figure 5.13, the two extreme states aresedto representm a r k s , and the mid state is used to represent paces. The three states couldbe positive and negative values, with a mid value of 0. In the case of optical fibres,where light is used, the three states could be off, low intensity and high intensity.In both AMI and HDB3ine codes, alternativema rks are sent as positive and negativepulses. Alternating he polarity of the pulseshelps to prevent direct current being

Figure 5.14 Digital signal pattern. These oscilloscope patterns result from esting bf circuitsusing a standard line format for the Bell Systems digital network. (Courtesy of ATBrT)


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74 DIGITALRANSMISSION AND PULSEODEODULATIONtransmitted to line. In a two-state code, a string of marks would have the effect ofsending a steady on value to line.The HDB3 code used widely in Europe and n international transmission systems) san extended form of AMI in which the numberof consecutive zeros that maybe sent toline is limited to three. Limiting the number of consecutive zeros brings two benefits:first a null signal is avoided, and second a minimum mark density can be maintained(even during idle conditions such as pauses in speech). A high mark density aids theregenerator timing and synchronization.In HDB3, the fourth ero in a stringof four is marked (i.e. forcibly set o 1) but this isdone in such a way that the zero value of the original signal may be recovered at thereceiving end. The recovery is achieved by marking fourth zeros in violation, that is tosay, in the same polarity as the previous mark, rather than in opposite polarity mark(opposite polarity of consecutive marks being the normal procedure).

5.11 OTHER LINE CODES AND THEIR LIMITATIONSOne of the inecodesused n hepast n North America nassociationwith he1.5Mbit/s line system is called zero code suppression. The technique seeks to elimi-nate patterns of 8 or more consecutive zeros, but it does so in an irreversible mannerby forcibly changing the value of the eight consecutive bit of value 0, so that insteadof transmitting 00000000, 00000001 is transmitted. Unfortunately, as it is only a two-state code, the receiving end device, unlike an HDB3 receiver, is unable to tell that theeighthbitvaluehas been altered.Anerrorresults.Theerror is not perceptible tospeech users, but would cause unacceptable corruption of data carried on a 64 kbit/schannel.Once eighth bit encoding using the zero code suppression technique had begun, itbecame acceptable to rob the eighth bit for other internal network uses. A robbed bitsignalling channel,equivalent to heEuropeans timeslot 16 signallingchannel,wascreated, as already discussed.Both of the above uses of eighth bit encoding reduced the usable portion of the64 kbit/s channel. For his reason, it s common for data erminals in North America tocommonly use only seven of the eight available bits in each byte. This has the effect ofreducing the usable bit rate to 56 kbit/s (8000 samples of 7 bits per second) even though64 kbit/s is carriedon he ine. North Americanreaders may befamiliar with the56 kbit/s user rate.

In connections from Europe to North America where the 56 kbit/s user data rate isemployed, it is necessary to employ rate daptor theEuropean ndoaccommodate the lower rate. In essence the rate adaptor is programmed to waste theeighthbit of eachbyte, giving a 56 kbit/s user rate even at theEuropeanend.Alternatively a rate adaptor may be used at both ends to employ an even lower bit rate,such as the ITU-T standard bit rate of 48 kbit/s. In this case two bits of each byte areignored. Stimulated by worldwide customer pressure for 64 kbit/s services (includingISDN, see Chapter lO ) , the restriction to 56 kbit/s channel capacity in North Americalooks set to disappear with the adoption f various new line codes. These includeB8ZS(bipolar 8-zero substitution) and ZBTSI (zero byte time slot interchange). Like HDB3,


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OTHER LINE CODES AND THEIR LIMITATIONS 75both eliminate long strings of zeros, but in a recoverable way. The BSZS code, forexample changes a string of eight zeros to the pattern OOOVBOVB.Some US readersmay dditionallyhave ome crosshe 64 kbitls restrictedbandwidth. As its name suggests, this provides a bit rate close to 64 kbit/s. It derivesfrom the use zero code suppressed channel prior to the availability of either BSZS orZBTSI. Any bit pattern can be sent by the user a t the normal 64 kbit/s rate so long asthe OOOOOOOO pattern is never used.

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