Pulse-code Modulation Technique For Digital Telephone
. 1911NEAR EAST TECHNICAL UNIVERSI1YELECTRIC AND ELEC1RONIC ENGINEERING DEPARTMENTPULSE-CODE MODULATION TECHNIQUEFOR DIGITAL TELEPHONESUPERVISORProf.Dr. FahretttinSADIKOGLU MAMEDOV
ACKNOWLEDGEMENTSThe authors would like to thank Prof. Dr. Fahrettin M. Sadikoglufor his supervision and constructive criticism in the development of thisthesis.The authors also wishes to express his deep gratitude to his parents for their support throughout his life.
\,)/TABLE OF CONTENTS1INTRODUCTION.CHAPTER I : THE FUNCTION BLOCK OF THE PCM SYSTEM. 41.1. Sampling. 61.2. Quantizing;. 81.3. Coding & Companding. 91.4. Electrical Representationof PCM Signals1.5. Demodulation1618CHAPTER II: PCM TRANSMISSION SYSTEM2.1. First Order PCM Systems21212.2. First Order PCM System Recommended by CCITIFrame Structures232.3. Second Order PCM systems282.4. tdm Hierarchy29CHAPTER III: THE TELEPHONE NEIWORK363.1. The Public Switched Telephone Network (PSTN)363.1.1. Switching Hierarchy of North America383.2. Technical ConsiderationsIn The Planning ofDigital Network For Multiexchange Areas3.2.1. Numbering40413.2.2. TransmissionSystem423.2.3. TransmissionPlanning433.2.4. Signalling443.3. Network Structure In The Rural Area463.4. A Completely Digital Network47CONCLUSIONREFERENCES50
. INTRODUCTIONPulse - code modulation (PCM)was developed in 1937 at the ParisLaboratories of American Telephone and Telegraph Company (AT&T) byAled Reeves. Reeves conducted several succesful transmission experiments across the English channel using the modulation techniqe, including pulse-widthmodulation(PWM). pulse-amplitudemodulation(PAM) and pulse-position modulation (PPM).At the time circuits involvedwas enormously complex & expensive. Although the significance of Reeve'sexperiments was acknowledged by Bell laboratories, it was no until thesemiconductor industry evolved in the 1960s that PCM because moreprevalent. Currently in the world PCM is the prefered method of transmission within the Public Swiched Telephone Network (PSTN).PCM is a methode of digital transmission of an analog signal. Themotives for using PCM in the telephone network can be summarized in thefollowing7 points:0 Transmission quality almost independent of distance. A char-acteristic of a digital signal is its immunity to interference. Digital signalscan be regenerated at intermediate points along a transmission line without loss of quality. This is not the case with analogue signals where notonly the signal but also the noise is amplified at the intermediate amplification points. In the digital case the repeaters have to make the simpledecision as to whether an incoming pulse is a "one" or a "zero". After thedecision has been taken a fresh pulse is transmitted. It is certain that anumber of incoming pulses will be so distorted that they cannot be recognized correctly, but this failure rate can be made as low as required. Itshould however be observed, that the PCM-systems used in practice and1
specified by CCITT are not, from a transmissionquality point of view bet-ter than the FDM-system specified by CCITT.0 Time-division-multiplex.The TDM principle allows an increase incapacity on cable pairs originally used for single telephone channels. Itmay be feasible to introduce PCM transmission on these pairs instead oflaying new cables.0 Economy for certain links. In certain applications, especially inthe junction network, PCM transmission has shown itself to be competitive with any other method of transmission. The length of the transmission links should be in the inttermediate region where normal voicefrequency (VF) links tend to be too long and FDM links too short.This optimal distance interval is very dependent on local factorssuch as subscriber density, topology of the country etc . and therefore varies widely. However,the figures in the graph can be taken as being typicalfor the first order PCMmultiplex system.O Economy in combination with digital switching. A high proportionof the cost for PCM-systems lies in the terminal cost. The introduction ofdigital switching will lsubstanttally reduce this cost since switching is performed directly on the digital bit stream and no costly analogue/ digitalconversion is needed. A combination of digital switching and transmissionwill therefore tend to lower the overall cost.0IC-technology. Developments in integrated circuit technologyseem to point to favourable cost levels and a high degree of reliabilty.0Integrationof services. As a digital medium a PCM-link cantransmit not only speech but also data, telex, encoded visual informationetc. A PCM-channel has a capacity of 64,000 bit/s which makes it a verypowerful data channel.2
New transmission media. Future wide-band transmission media,such as waveguides and optical fibres are more suitable for digital thananalogue transmission.The project consists of 3 chapters:Chapter 1 presents five major principles: sampling, quantizing,electrical representation of PCM signals, Coding and Companding,de-modulation for PCM systems. This theoretical material is used it be representation first order PCM system. The functional and timing diagram issystem are presented.Chapter 2 starts with description 30 and 24 channels standardPCM systems rdecommended by CCIT.The more important accept of PCMmodulation technique-digital multiplexing hierarchy is analysed. The evaluation of Transmission rate for different level hierarchy are clearly calculated and explained.Chapter 3 includes an application PCM system in the publicswitched telephone network. Different types of network is considered. Newmethod of switching is given to interconnect ceutial office through the useof interoffice trunks and tudem trunus. Outside the local area told trunusand intertool trunus are considered.The text is arrauged so that it can be used by students to studysame part of the subject "Telecommunication". It can be useful for engineers too.3
CHAPTER ITHE FUNCTION BLOCK OF THE PCM SYSTEMThe statement above gives some idea about the basic processes inpulse code modulation. Here we shall give these processes their rightnames.The process of choosing measuring points on the analoguee speechcurve is called sampling. The measurement values are called samples.When samlpling, we take the first step towards a digital representation ofthe speech signal as the chosen sampling instants give us the time coordinates of the measuring points.The amplitudes of the samples can assume each value in the amplitude range of the speech signal. When measuring the sample amplitudes we have to round off for practical reasons. In the rounding-offprocess, or the quantizing process, all sample amplitudes between twomarks on the scale will be given the same quantized value. The number ofquantized samples is discrete as we have only a discrete number of markson our scale.Each quantized sample is then represented by the number of thescale mark, i.e. we know now the coordinates on the amplitude axis of thesamples.The processes of sampling and quantizing yield a digital representation of the original speech signal, but not in a form best suited totransmission over a line or radio path. Translation to a different form ofsignal is required. This process is known as encoding. Most often the sample values are encoded to binary form, so that each sample value is repre4
.sented by a group of binary elements. Typically, a quantized sample canassume one of 256 values. In binary form the sample will be representedby a group of 8 elements. This group is in the followingcalled a PCMword.For transmission purposes the binary values O and I can be taken as corresponding to the absence and presence of an electrical pulse.On the transmission line the pulses in the PCM words will becomegradually more distorted. However,as long as it is possible to distinguishbetween the absence and the presence of a pulse, no information loss hasoccurred. If the pulse train is regenerated, i.e. badly distorted pulses arereplaced by fresh pulses at suitable intervals, the information can betransmitted long distances with practically no distortion at all. This is oneof the advantages of digital transmission over analogue transmission; theinformation is contained in the existence or not of a pulse rather than inthe form of the pulse.In our picture of the graph and the table this is analogous to thefact that the information in the table is not affected if the digits are badlywritten as long as they are legible. But if the graph is badly drawn, loss ofinformation is inevitable.On the receiving side the PCM words are decoded. t.e. they aretranslated back to quantized samples. The analogue speech signal is thenreconstructed by interpolation between the quantized samples. There is asmall difference between the analogue speech signal on the receiving sideand the corresponding signal on the transmitting side due to the roundingoff of the speech samples. This difference is known as quantizing distortion.The function blocks in the pulse code modulation process areshown in figure 1. 1.5
AnaloguesignalReceiverTransmlsslon linePCM signalFigure 1 - 1. Pulse code modulation Function blocks1.1. SamplingIn the practical electrical meaning, to sample is to take instaneousvalues of the analogue signal at equal time intervals. See figure 1-2.fAOrlqlnal' ".'Sampler1 -- :;/,Sampleds ---- o,m,Amplitude /JI-'SamplesII I"'limeSampling Intervalt,f J Sampling rate't,Figure 1 - 2. The sampling processThe sampled signal is a train of pulses, whose envelope is the original signal.Now, what should be thesampling rate, i.e. the number of sampllesper second? The answer to this question is given by the Sampling Theo6
rem, which also illustrates the fundamental fact that the information contained in the signal is not affected by sampling:The sampled signal contains within it all information about theoriginal signal if:0 the original signal is band limited, i.e. it has no frequencycomponents in its spectrum beyond some frequency B0 the sampling rate is equal to or greater than twice B,i.e. fs 2B.The sampling theorem is illustrated in figure I-3. Obviously, thespectrum of the sampled signal contains the spectrum of the original signal, i.e. no information loss has occurred.c JCC .frequencya) band limited signal: b'"nh ch lIBIsb) sampled signal, Is 2 · B21sfrequencyFigure 1 - 3 . Spectrum of a) band-limited signalb) sampled signalIn telephony, the part of the speech spectrum between 300 and3400 Hz is used. The human speech spectrum extends from a lowest fre-quency of some 100 Hz up to very high audio frequencies. The telephoneset reduces this frequency range, but not enough at high frequencies so inorder to come below this band limit at 3400 Hz, the speech signal must below-pass-filtered before sampling.7
. plitude. This means that a loud talker and quiet talker let a listener hearthe same quantizing distortion. Relative to the speech levels, the quiettalker generates much more distortion than the loud talker. Furthermore,a statistical analysis shows that for an individual talker, small amplitudesare much more probable than large ones.amplitude ,sampled signal(PAM signal)quantizerquanllzedsignalI quantizing)------Intervalquantizing levelFigure I - 4 . The quantizing process1.3. Coding & CompandingPractical PC systems use seven-and eight-levelbinary codes, or27 28 256 quantum steps128 quantum stepsTwo methods are used to reduce the quantum steps to 128 or 256without sacrificing fidelity. These are nonuniform quantizing steps andcompanding before quantizing, followed by uniform quantizing. unlikedata transmission, in speech transmission there is a much greater likelihood of encountering signals of small amplitudes than those of largeramplitudes.A secondary but equally important aspect is that coded signals aredesigned to convey maximum information, considering that all quantiumsteps (meanings or characters) will have an equally probable occurrence9
A sampling rate of 8000 Hz is used for PCM systems in telephony.This rate is somewhathigher than twice the highestfrequencyin theband, 3400 Hz, due to difficulties in making low-pass filters steep enough.The sampled signal is often said to be pulse amplitude modulatedas it consists of a train of pulses, whose amplitudes have been modulatedby the original signal. Pulse Amplitude Modulation (PAM) is an analoguepulse modulation method as the amplitudes of the pulses may vary continously in accordance with the original signal variations.The relative simplicity of PAM systems makes them attractive forsome telephony applications. However,PAMis unsuitable for transmissionover long distances owing to the difficulty of pulse regeneration with sufficient accuracy, which is important as the PAM pulses contain the information, in the pulse form.1. 2. QuantizingThe continuous pulse amplitude range is broken down to a finitenumber of amplitude values in the quantizing process. The aplitude valuesin the quantizing process. The amplitude range is divided into intervals,and all samples whose amplitudes fall into one specific quantizing intervalare given the same output amplitude. See figure I-4. The rounding off ofthe samples causes an irretrievable error, quanting distortion, in the signal.This voluntary sacrifice, which can be brought down to suitablelow limits by making the number of permitted amplitude levels largeenough, is accepted because it makes error-free transmission possible byonly having a discrete number of amplitudes.In figure I-4 the quantizing distortion is independent of sample am8
.(i.e., the signal-level amplitude is assumed to follow a uniform probabutydistribution between O and the maximum voltage of the channel). To circumvent the problem of nonequiprobabilityof signal level for voice signals,specifically, that lower - level signals are more probable than higher-levelsignals, larger quantum steps are used for the larger-amplitudeportion ofthe signal, and finer steps are used for the signals with low amplitudes.The two methods of reducing the total number of quantumsteps can nowbe more precisely labeled:* Nonuiform quantizing performed in the coding process.*Companding(compression) before the signals enter the coder,which now performs uniform quanttizingon the resulting .stgnal beforecoding. At the receive end, expansion is carriedout after decoding.Most practical PCM systems use complandingularity(more stepss)stantaneousto thesmalleramplitudeto give finer gran-signals.Thisis in-companding, as compared to the syllabic companding used inanalog carrier telephony.Compressionplitude signals. The compressionimpartsmore gain to lower am-and later expansionfunctionsare log-arithmic and follow one of two laws, the A law or the "mu" (µ)law. Thecurve for the A law may be plotted from the formulaAlxl ) lxl l ln(A) x1 Inlxl ) 1( 1 ln(A) A(1s lxJ A1where A 87.6. The curve for theµ law may be plotted from the formula:y1 µJxlln (1 µ) ln (where x is signal imput amplitudeand µ 100 for the original NorthAmerican Tl system (now out dated) and 255 for later North American(DSI) systems and the CCITI 24-channel system.10
A common expression used in dealing with the "quality" of aPCMsignal is the signal-to-distortion ratio (expressed in decibels). Parameters Aandµ determine the range over which the signal-to-distortion ratio is comparatively constant. This is the dynamic range. Using aµ of 100 can provide a dynamic range of 40 dB of relative linearity in the signal-todistortion ratio.In actual PCM systems the companding circuitry does not providean exact replica of the logarithmic curves shown. The circuitry producesapproximate equivalents using a segmented curve, and each segment islinear. The more segments the curve has, the more it approaches the truelogarithmic curve desired. Such a segmented curve is shown in Figure 1-5.If µ law were implemented using a seven (hetghtj-segment linear approximate eqivalent, it would appear as shown in Figure 1-5. Thus oncoding, the first three coded digits would indicate the segment number(e.g. 23 8). Of the seven-digit code, the remaining four digits would divide each segment into 16 equal parts to identify further the exact quantum step (e.g., 24 16). For small signals, the companding improvement1.0 "'ciE0u'-::---'-- 1.l -f-1.0Inputj J1.0Figure I - 5. Seven-segmentlinear approximate of the logarithmiccurvefor µlaw(µ 100)11
is approximatelyA law: 24 dBµlaw: 30 dBusing a seven-level code. These values derive from the equation of companding improvement orGc!B Uniform (linear) scaleCompanded scaJeCoding in PCM systems utilizes straighforward binary codes. Examples ofsuch coding are shown in Figure l-5a, which is expanded in Figure 9.7,and in Figure 9.8, which is expanded in Figure 1.6.b showing a number ofexample code levels.The coding process is closely related to quantizing. In practical systems, whether the A law or the µ law is used, quantizing employes segmented equivalents of the companding curve (Figures I-6 andl-8), as discussed earlier. Such segmenting is a handy aid to coding. Consider theEuropean 30 2 PCM system, which uses a 13-segment approximation ofthe A-law curve (Figure 1-6). The first code element indicates whether thequantum step is in the negative or positive half of the curve. For example,if the first code element were a 1, it would indicate a positive value (e.g.,the quantum step is located above the origin). The followingthree-code elements (bits) identify the segment, as there are seven segments above andseven segments below the origin (horizontal axis).The first four elements of the fourth segment are 1101. The first1 indicates it is above the horizontal axis (e.g., it is positive). The next12
three elements indicate the fourth step or0-1000 and 10011-10102-1011 3-11004 -11015-1110 etc.CorleSegmentI1 1 1 1 X XX X0 ,,2 --·OG80364248321 1 1 0 X X X XIII- I1101xxxx,ooxxxxIII241011xxxx101oxxxx324J. (VI641001xxxx0IL1v1JI14tVIIL j1 oooxxxx0 4 01ooxxxx0101xxxx011oxxxx0111xxxxFigure I - 6. Quantization and coding used in the CEPT 30 2 PCMsystem.13
. Figure 1. 7 shows a "blowup" of the uniform quantizing and subsequent straightforward binary coding of step 4. This is the final segmentcoding, the last four bits of a PCM code word for this system. Note the 16steps in the segment, which are uniform in size.110101()() ----Figure I - 7. The CEPT30 2 PCM system, coding of segment (4 positive).EncodingCode96804832161 35768 Segments16 Steps eachEncoder inputFigure I - 8. Positive portion of segmented approximation of u lawquantizing curve used in North American (AIT) DS1 PCM channelizing equipment. Courtesy of IIT Telecommunications,Raleigh, N.C.14
The North AmericanproximationDSI PCM systemusesa 15-segmentof the logarithmic µ law. Again, there ase actuallyap-16 seg-ments. The segments cutting the origin are colinear and counted as one.The quantizationin the DSI system is shown in Figure 1-8 for the positiveportion of the curve. Segment 5, representingquantizing steps 64 through80, is shown blown up in Figure 1-8, Figure 1-9 shows the DSI coding. Ascan be seen in the figure, again the first code element, whether a 1 or a 0,indicates whether the quantumstep is above or below the horizontal axis.The next three elements identify the segment, and the last four elements(bits) identify the actual quantumlevel inside the segment. Of course, wesee that the DSI is a basic 24-channel system using eight-level coding withµ-law quantizationcharacteristicwhereµ 255.l igi1 N 11111herI2'.I4: de I .evvl(l't'ak posili\'c25523!122320711!11)',17: ): !I14'.112712fi1119: ;( 79le\'l'I)(( .euu-r levels)( Nominal tt'ro)I()()()()(l't'ak lll'gati\'t' h-vr-l)0()III*I()()*One digit is ad,ll-d to eusurc that rl«: 1i111ing conte111 of the uansruiued paltt'rn ismainraiucr],·Figure I - 9. Eight-level coding of North American (ATT) DS1 PCMsystem. Note that there are actually only 255 quantizing steps becausesteps O and 1 use the same bit sequence, thus avoiding a code sequencewith no transitions ii.e., O's only).15
As we know, pulses with two levels, i.e. binary pulses,tractive for transmissionas they are easy to regeneratemission line. It is not dfficult to build regeneratorare at-on the trans-circuitsable to de-termine whether a pulse is present or not.Present-day practical system use binary encoding of the quantizedspeech samples. See figure 10. As telephony uses 256 quantizinglevels,each sample will be encoded to a code group. or PCM word. consisting of 8binary pulses (8 bits).amplllude. .,. "' r111 3110 2quantized000001.o,I I I I I010-21100-0-11,I,. IIIEncoderI011 -3 IIIpulse codemodulatedsignal. . . .,, , . , . . , , ., . '--.--' .,100110111101001010000101 O 2 3 1-1-2-0 1Figure 1- 10. Encoding of quantized samples with 8 quantizing levels (3 binary digits/codeword).As the sampling rate used is 8000 samples/second, one pulse codemodulated speech signal will generate a 64 kbit/s digital signal.1.4. Electrical Representation of PCM signalsDigital signals within the terminal are usually transmitted in the formof a unipolar pulse train in the nonreturn-to-zero (NRZ) mode, see figure 11.This signal form is not appropriate for transmission over long distances.16
. IIOIII 0II](rFigure 1- 11. Binary information represented in:I a unipolar nonreturn-to-zero (NRZ)pulse train.II · a bipolar return-to-zero (RZ)pulse train.A better form is a bipolar return-to-zero (RZ) signal. The advanteges of this signal are :* it has no power in the lower parts of its spectrum, i.e. it has nodirect current component; this is due to the alternating polarities of thepulses*the intersymbol interference is reduced by the return-to-zerofeature.Of course, even this signal will be attenuated and distorted duringtransmission, and noise will be added to It.At some point on the transmission line the signal must be restored.This is done by inserting a device on the line that first examines the distorted pulse train to see whether the likely binary value is 1 or 0, and thengenerates and transmits to the line new pulses according to the result of theexamination. Such a device is called a regenerative repeater. See figure 12.At the same time as the pulses are reshaped, the noise added duringtransmission is eliminated at the least if the noise signal amplitude is notlarge enough to bring the received code signal to the wrong side of a regenerator decision level. Normally,the regenerated code signal is identical tothe transmitted original code signal. Even after a large number of re17
generative repeaters,the code signal is practically identical to the originalsignal. This is the reason for the high transmissionable with PCM transmissionsystem.PCMI". 0 ·- ---ir1,ansm111.,. l ".1'.!' · 11 · JRegenor llverepeaterquality that is obtain-timing- - - - - - "": rogeneraledcode. n· · ·· · ,:s;. : \.T/ \ \.? ::Z ,l I t I t I fI I """,-,.,.,,0 1d 11ocsonlevels Figure 12. Pulse forms on a transmission line.1.5. DemodulationThe processes in the receiver that convert the incoming PCM signalto an analogue speech signal again are regeneration, decoding and reconstruction.The regeneration process has the same aim and is performed in thesame way as on the transmission line, i.e. the distorted pulses are replaced bynew square pulses, see figure 12. Beforeentering the decoder the bipolar signalis reconverted to unipolar. In the decoding process the code words generateamplitude pulses, whose hetghtts are the same as the heights of the quantizedsamples, which generated the code words. So after passing through the decoder the train of quantized samples is retrieved. See figure 13.18
. 0 2. 3 .t1-2 O 1, .,. , ,. . ,--.,,. .---.,.- ,,--. .--,,,. pulse codemodulated slgnal10011011110100101000010\amplitude111 Jdecoderquantizedsamples110 2101100000001 010-2011 -31 O11-1timeFigure 13. Decoding of encoded amplitude levelsThe analogue signal is reconstructed in a low pass filter. figure14a. This can be seen from figure 14b. The spectrum of a sampled signalcontains the spectrum of the original signal as has been shown in figure I3. A low pass filter with a cut-off frequency at B Hz takes away all frequency components in the spectrum above B Hz and the spectrum of thedesired analogue signal is left.quantizedsampleslow passJIiterreconstructed o s! 9 11amplltuda IIIII,II.,time! ' 7 llmoFigure I - 14.a Reconstruction of the analogue19
,. TQ"'"dc\ oh l:' T''" ·rrnm '"'L-,LJB"'"'I-,---1Is21aarnplad ''""" """'"" /////fl@/////ft//Pu// /////4,Power /lrequencylow pass tiller characlerlsllcfrequencyanaloguesignal------ 1sr----------spectrum of reconstructed analoguesignaltrequency Figure I - 14b. Reconstruction of the analogue signal shown byspectrum diagram.20
CHAPTER IIPCM TRANSMISSION SYSTEMWe have now dealt with the fundamental principles of PCM. In thispart of the chapter we shall describe how these principles are used tobuild up practical PCM transmission systems. However,we shall start byexplaining the time-division multiplex principle as this makes PCM transmission systems for telephony economic?-11Yattractive.2.1. First Order PCM SystemsSeveral signals inpulse from can use a common transmissionpath if the signals have different phases. Figure 2-1 shows how threePAM-signalsare time-division multiplexed on the same transmission line.The pulses of the three signals are interleaved by opening the samplinggates one by one cyclically.During one cycle the transmission line receivesone PAMpulse from each of the participating signals. Such a set of pulsesis called one frame. The time interval that each of these pulses occupies iscalled a time slot. In this example each frame has three time slots.On the receiving side the pulses are distributed again. This is doneby opening the sampling gates cyclically in the same manner as on thetransmitting side. Due regard must of course be paid to the transmissiondelay. This delay has been omitted in figure 2-1 for clarity.In the case of PCM signals the time-division multiplexing is mostofter carried out before the samples are pulse coded, i.e., the samples fromthe participating analogue signals are combined on a common PAMtransmission line. See figure 2-2.21
. ifb-II({o- iVF ({o- '1.:::IVF v I.P!' TD"r:-@--8-low pass filter1IIII II1trame11 time I IIpJP . Gb---\I: 11III:IIIIII.IIII1III IplTDMp I--, P f IIIIII-------- 7 .VFI: - :::::::::J. Gn---f .lp.n p.n pst: Cro---IJ1IP T DMpoecoder]4-rIIII1sampler·h i,.· . ."- .Figure 2-1. A PAM transmission system using time-division multiplex(TDM)@low pass lllterHsampler'transmission lineIl-,IIIFigure 2-2. A PCM-TDMtransmission system. Attenuation and delay on the PAM and PCM transmission lines are not shown.22
.In this way the coding equipment can be used in time-division multiplex. We see from the figure that the PCMpulses are not nterleaved pulseby pulse, but PCMword by PCMword. This is often called time slot interleaving. PCM systems used in telephony are most often TDM systems, so\when we read or hear the term "PCMsystem" it is almost always referringto a PCM-TDMsystem. Howeverit must not be forgotten that PCMin itselfcan be used, and is used in some cases, on a one-channel basis.2.2. First Order PCMSystem Recommended by CCITTFrameStructuresAs mentioned in chapter I, CCITTrecommends two ftrst order, orprimary, PCM systems for use in telephony: the 30-channel system, proposed by CEPT, and the 24-channel system proposed by AT & T. The firstorder systems will form the basis for hierarchies of digital transmissionsystems.We have to distinguish between the PCM multiplex equipment, orthe PCM terminal, and the/PCM transmission line. The multiplex equipment converts a number of analogue signals (30 or 24) to a digital signalon the transmitting side and carries out the inverse functions on the receiving side. The transmission line conveys the digital signalsbetweentwo multiplex equipment units. See figure 2-3.In the following,the 30-channel multiplex equipment will be treated in some detail, as this multiplex forms the basis for the subsequentpresentation of digital telephony in this book. A summary of the most important data on the 24-channel multiplex is also given. The presentationof first order PCM systems will end with a brief description of the transmission lines.23
-H--Transmitter.,.Receiver30 or 24VF channels ' PCM1 J -·-4- - -1I\ Mul iplexI equipmentI (PCMR \PCM transmission lineR: Regenerative repeatersIIter inal)\ -t., tr 3 -\. . lI11,,,,--Re-c-ei-ve-r II\Figure 2-3. Two multiplex equipment units.Thirty analogue speech channels together with. associated signalling are converted to one digital signal by the 30-channel system.The structure of this digital signal is shown in figure 2-4.JTimeiPrimary multiplexuooo Ir arnes/s204a kblt/s·O1iFrame !256 bits'Time slot125µs \8. bitsiSync. ch: synchronization channelSi;in. ch: signalling channel3.9 µs10011101Figure 2-4. Frame structure of the 30-channel primary multiplex.24i)\\\',Transmitter1------130or 24I PCMI VF channelsI multiplex II equipment II (PCMI terminal)" -j}'"
The digital signal is divided into frames, with a repetition rate of8000 frames/sec.This is of course because8000 Hz and the fact that the frame containsthe samplingfrequency isone binary coded samplefrom each of the analogue signals. Each frame consists of 32 eight bit timeslots. Of these, 30 time slots are used for PCM channelsmaining two for synchronizationThe PCM channelsand the re-and signalling.carry analogue
1.5. Demodulation 18 CHAPTER II: PCM TRANSMISSION SYSTEM 21 2.1. First Order PCM Systems 21 2.2. First Order PCM System Recommended by CCITI Frame Structures 23 2.3. Second Order PCM systems 28 2.4. tdm Hierarchy 29 CHAPTER III: THE TELEPHONE NEIWORK 36 3.1. The Public Switched Telephone Network (PSTN) 36 3.1.1.
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