Mạng và viễn thông P33

594 PRACTICAL NETWORK TRANSMISSION PLANNING ITU-T’s recommendations on transmission planning include design guidance on the overall signal volume at all points through the network the c

be of sufficient strength (or volume) as to be clearly distinguishable by the receiver The reliability

of the service is also important Meeting these objectives requires careful network transmission planning This chapter covers two aspects of planning, first describing the electrical engineering design guidelines usually laid out in a formal network ‘transmission plan’, and then going on to discuss the general administrative and operational practicalities of ‘lining-up’ and operating a transmission network Resource management is crucial to ensure that lineplant and circuits are available when needed, that radio bandwidth is available without interference, that cables have been laid and satellites put in orbit

33.1 NETWORK TRANSMISSION PLAN

A set of guidelines, laying down simple ‘rules’ for planning and commissioning new line

systems or new circuits, is usually set out in a formal transmissionplan These guidelines

are intended to ensure that the electrical principles of telecommunications theory are adhered to The transmission plan should include, for example, stringent rules on the

use, positioning, and strength of amplifiers or regenerators to correct the effects of attenuation The plan needs to give guidelines on minimization of noise and interference,

on the use of echo control devices to eliminate echo, and on the use of equalizers to rectify frequency attenuation distortion A good transmission plan is a guideline for network design, ensuring that circuits are electrically stable and perform to a standard acceptable to the majority of their users A typical target is ‘to ensure that 90% of people consider the performance to be “fair”, or “better”’ Acceptable values of attenuation and distortion can be established by subjective tests, and then the network can be designed to accord with them

593

Networks and Telecommunications; Design and Operations (Second Edition)

Martin P Clark Copyright © 1991, 1997 John Wiley & Sons Ltd ISBNs: 0-471-97346-7 (Hardback); 0-470-84158-3 (Electronic)

594 PRACTICAL NETWORK TRANSMISSION PLANNING

ITU-T’s recommendations on transmission planning include design guidance on the overall signal volume at all points through the network

the control of signal loss and the circuit’s electrical stability

the limits on acceptable signal propagation times (excessive propagation times mani- fest themselves as a delay in transmission, a potential cause of slow data throughput and response times, or of unacceptable gaps and pauses in conversation)

the limits on acceptable noise disturbance

the control of sidetone and signal echo (both these effects manifest themselves to

speakers, who hear their own words echoed back to them)

the minimization of signal distortion, crosstalk and interference

(for digital circuits) maximum allowed bit error rate (or bit error ratio, BER), jitter and quantisation distortion

maximum line length (e.g approximately 15 metres for V.24 interface, 100 metres for X.21, etc.)

The recommendations lay down a standard reference system, against which national and international transmission networks may be designed or calibrated These facilitate the correct placing of circuit conditioning equipment, and the establishment of cor- rect signal volumes and controlled levels of signal distortion at all points along a connection

When designing telephone or other voice transmission networks, the strength of the electrical signal induced by the microphone at the sending end and the final sound volume produced by the receiver is all-important It varies according to the proximity of the speaker to the microphone and the listener to the ear piece, the total loss being composed of three basic parts; the loss incurred when inducing the electrical signal in the microphone, the electrical signal loss across the transmission network itself, and

finally the sound induction loss in the receiver The relationship is straightforward to

understand, as Figure 33.1 illustrates, but the problem for the network designer lies in

deciding just how much signal loss is acceptable in the main part of the network itself With no overall loss, the listener will hear too loud a signal, equivalent to a talker at normal volume speaking directly into his ear On the other hand too much loss will result in an inaudible signal

T o ease the design dilemma, the system of reference equivalents was developed by

CCITT (the forerunner to ITU-T) The send reference equivalent ( S R E ) of a telephone

microphone is the signal loss expressed in dB (decibels) when the electrical signal volume is compared with the original speech volume Similarly, the receive reference equivalent ( R R E ) of the earpiece is the signal loss when the final (heard) sound volume

is compared with the electrical signal volume input to the earpiece Both SRE and RRE

SEND AND RECEIVE REFERENCE EQUIVALENTS 595

Network

I

Sending loss - - - Network loss L Receiving loss

Figure 33.1 Send and receive reference equivalents

are shown on Figure 33.1 To measure the sending reference equivalent (SRE) of a

handset, a special NOSFER equipment such as that shown in Figure 33.2 is used This

compares the equipment with a standard device, named after this system of calibration,

fundamental system for determining reference equivalents)

The NOSFER equipment in Figure 33.2 consists of a high quality microphone with a

device which maintains it at a fixed distance from the speaker's mouth This equipment

is connected to the listener's earpiece via a large attenuator, of strength A dB To per- form the calibration, a person talks alternately into microphone a and microphone b, while the listener adjusts the variable attenuation B to a value at which the volumes

appear to be the same The difference between values B and A will then give a measure

of the relative microphone efficiencies The more that attenuation B must be reduced

(i.e the lower the value of B ) the less efficient the microphone and the larger the

numerical value of A - B, which is termed the send reference equivalent ( S R E ) In other

Standard equipment Fixed (large) attenuation

596 PRACTICAL NETWORK TRANSMISSION PLANNING

Fixed (large) attenuation

Standard equipment

L

Earpiece Microphone

Talker

Listener under test between

Figure 33.3 Measuring receive reference equivalent (RRE)

words, the higher the value of SRE, then the lower the efficiency of the microphone and the greater the loss of signal strength in it Thus SRE is a measure of the signal loss in the sending device

Receive reference equivalents ( R R E s ) similarly are a measure of the signal loss in the receiving device, and they may be measured by an equivalent NOSFER equipment; but

in this case the comparison must be made against a standard high quality receiver, as Figure 33.3 shows

REFERENCE EQUIVALENT

For the design of the network itself, all telephone handsets are assumed to have typical nominal SRE and RRE values The values chosen are usually quoted for the handset

loss relative to an imaginary reference point on the customer’s line side of the telephone

exchange, and they take account not only of typical handset SREs and RREs, but also

of the loss encountered on the local line Typical values are 13 dB SRE and 2-3 dB

RRE In practice, customer lines have SRE and RRE of varying values, depending

on the exact handset in use and the length of the line However, choosing a nominal design value is important as a guide for handset design and also as a benchmark for the required performance of the network itself

In Figure 33.4 two reference points (a) and (b), have been marked, one at each end of the connection, corresponding to the points relative to which SRE and RRE values apply These points are marked on the diagram as 0 dBr The nomenclature dBr stands

for decibels-relative, and it is used to indicate the received signal loudness at any point,

measured relative to the signal strength at the reference point Hence, at the reference point itself, where the relative strength is equal to the signal strength at the reference point, the value will be zero dBr

If the network introduces a loss of 3 dB in the transmission direction from A to B,

and 3.5 dB in the transmission direction from B to A, then the received signal strengths

at points (b) and (a), respectively, will be -3 dBr and -3.5 dBr (i.e signals 3 dB and

CONNECTION REFERENCE POINTS AND OVERALL REFERENCE EQUIVALENT 597

(a1 Reference point

Figure 33.4 Connection ‘reference point’

3.5dB weaker than those sent from the opposite reference points) These values have been marked on the diagram in Figure 33.5

Now we can meaningfully discuss the end-to-end loss, called the overall reference

equivalent (or ORE) Surprisingly, the ORE may not be precisely equal to the sum of the SRE, the RRE and the cross-network loss This is because both the SRE and RRE are only subjective measurements made in isolation on the network’s general speech carrying performance, and the ORE is a more stringent measure of the actual end-to-

end loss, usually measured using a single frequency tone of 800 Hz Thus the formula

ORE = SRE + network loss + RRE does not apply

Initially, network planners thought the discrepancy was small enough to be ignored, thereby greatly easing the task of network design, and CCITT used to recommend that the network loss be adjusted to conform with an overall reference equivalent not exceeding

598 PRACTICAL NETWORK TRANSMISSION PLANNING

40dB However, a number of network operators found difficulty with the system, reporting discrepancies of up to 5 dB, so in 1976 CCITT developed a new measure of

performance called the loudness rating Still using the NOSFER equipment as a funda-

mental reference, and based on similar principles and reference points, the loudness rating method of network design was developed in such a way that loudness ratings (LRs) added to give a reliable and accurate algebraic sum Send loudness ratings ( S L R s ) , receive ( R L R s ) and overall loudness ratings ( O L R ) were defined in a similar manner to

the equivalent RES, but now the formula holds true

OLR = SLR + network loss + RLR The method achieves greater reliability because it uses standard equipment, which although similar in principle to a NOSFER equipment, more closely responds to the

frequencies of speech The equipment is called an intermediate reference system (ZRS)

First the IRS is calibrated against NOSFER, then the system under test is calibrated The difference in the two attenuation values (needed for each system individually to

give performance equal to NOSFER) is the loudness rating ( L R ) Values of loudness rating differ from the reference equivalent by varying amounts, up to 3 dB Loudness

ology for determining LR is given in recommendation P.65

Following on from the successful introduction of its loudness rating method, CCITT

in 1980 upgraded its original method of reference equivalents, applying a correction,fac- tor to them to enable the values to be added algebraically A simple mathematical form-

ula converts reference equivalents (RES) into corrected reference equivalents (CREs)

Corrected reference equivalents are today’s standard method for designing networks against transmission loss The following relationship applies

OCRE = SCRE + network loss + RCRE

3dB) from corresponding reference equivalents ( R E S ) , but are usually a fixed amount

(5 dB) greater than corresponding loudness ratings ( L R s )

ITU-T nowadays recommends that the maximum overall loudness rating ( O L R )

should not exceed 29 dB and has set an objective optimum value of 10 dB Such targets should be easily achievable with the excellent performance of modern digital networks

In practice the OLR or OCRE may differ slightly even from the planned value, but this will not matter if sufficient safety margin is allowed during planning, and provided that the differential losses in the two directions of transmission are not too dissimilar In Figure 33.5, for example, the ORES in the two directions are not quite the same ITU-T recommends that the difference be limited to 8dB

33.4 MEASURING NETWORK LOSS

That the loss in decibels incurred by a signal traversing a network will depend on the frequency of that signal is a fact that we learned in Chapter 3 With that in mind, how

CORRECTING SIGNAL STRENGTH 599

can we meaningfully quote a value for network loss? The answer is that it is defined as the signal loss incurred by a standard tone of 1020 Hz frequency (formerly 800 Hz and l000Hz tones were also used) It is measured using a standard 1020Hz tone source which is set to generate a known absolute signal power at a refence point known as the

measured at any other required points in the same transmit path, and compared with the nominal values laid down by the formal network transmission plan

33.5 CORRECTING SIGNAL STRENGTH

The transmission plan (ITU-T recommendations G.lO1, G.102, G.103, G l l l and G.121) lays out a rigid framework for the expected signal strength at all points along

a connection Any necessary adjustments in strength are achieved by the use of

and are boosted by inserting amplifiers into the connection The position of the ampli- fiers is critical If insufficient amplification is used the signal strength becomes too weak, becomes inaudible, and is affected by noise interference Subsequent amplifica- tion (if attempted) does not correct the situation, because it amplifies the noise as much

as the original signal (in other words a given minimum signal-to-noise ( S j N ) ratio needs

to be maintained)

If over-amplification is used it is likely to cause the electrical circuit to saturate (i.e overload), resulting in signal distortion and electrical instability (manifested as a

whistling and loud feedback signal) Furthermore, over-amplification can lead to inter-

ference in adjacent circuits Unduly strong signals are easily corrected either by reducing the amplification or by the introduction of attenuators

There are standard positions in the circuit, where amplifiers or attenuators may be used The strength necessary is determined by comparing the actual signal strength with the pre-determined value appropriate for that particular reference point As well as there being a reference point in the customer’s local exchange, reference points can also

be defined at other points in the network, at least at each exchange This allows each

individual section of an overall transmission link to be designed and lined-up accord-

ingly The sub-sections may then be joined to form a high quality, end-to-end connection If instead the circuit is lined up once (as a single long section) there could be

a counteracting effect between the performance of the various sub-sections which would mask some of the impairment that exists We have, for example, already noted that it is not acceptable for us to allow the signal to fade to a strength equal to that of the inter- fering noise before we start amplifying it Hence the need for a segmented approach to transmission planning Figure 33.6 illustrates the transmission plan for the North American telephone network while it was analogue (The illustrated plan has been superseded by the digital transmission plan, but it is still useful in illustrating the use of such a plan) Each link is marked with its maximum permitted loss

Any new circuit between any two exchanges in the network must be lined up to have

an overall loss within the ranges shown on the plan Thus the maximum end-to-end network loss is assured to be 19.5 dB The Americans call this end-to-end loss between

end offices the via net loss ( V N L )

600 PRACTICAL NETWORK TRANSMISSION PLANNING

Figure 33.6 The North American telephone network transmission plan

It is best for longer links to be lined-up with loss values towards the upper end of the

permitted range This ensures greater electrical stability Another important feature of the plan is that greater losses are permitted on the links at the bottom of the network hierarchy (i.e between Class 4 and Class 5 exchanges) This is because the vast majority

of lines are in this category, and considerable savings are possible if amplification can be avoided

Analogue and digital networks both need transmission plans, but they take different forms In digital networks, the signal strength in decibels is only an important consider- ation in mixed networks at the points of analogue-to-digital signal conversion, because the regeneration of digital signals by amplification is quite unnecessary in pure digital networks The digital transmission plan concerns itself instead with such factors as

maximum permissible bit error ratios (BERs) and the extent of the quantization limit,

which we shall describe more fully later in the chapter In the meantime, Figure 33.7 shows the signal loss transmission plans for the UK analogue and new UK digital networks

Figure 33.7(a) shows the connection of two normal telephones via two-wire analogue lines to their respective local exchanges, and then over four-wire digital transmission across both exchanges and the interconnecting link At the originating exchange the speech signal is converted from a two-wire analogue form into a four-wire digital form

by an analogue/digital conversion device, adjusted to produce a net loss of 1 dB The signal is then switched through both exchanges in a digital form without further loss

CORRECTING SIGNAL STRENGTH 601

2 to L wire and analogue to digital converter

X Exchange switch motrix

2-wire line loss

X Exchange switch matrix

Figure 33.7 (a) The UK digital network transmission plan (b) The UK analogue network

transmission plan

until it reaches the digital-to-analogue converter at the other end Unlike the first converter, this one adds a further 6 d B loss, giving an overall network performance of

7 d B loss between reference points The ORE can then be calculated by adding the

relevant SRE and RRE values This part of the transmission plan is very similar to the

analogue plan (Figure 33.7(b)) which it replaces, except that although the total four-

wire section loss is 7dB the recommended signal strengths at the various intermediate

reference points in this part of the connection are different Another difference is that the analogue plan allows a small number of two-wire links to be used in the connection

602 PRACTICAL NETWORK TRANSMISSION PLANNING

Sidetone is the name given to an effect on two-wire systems (e.g basic analogue telephones) where the speaker hears his own voice in his own earphone while he is speaking (we first introduced it in Chapter 2) Too little sidetone can make speakers think their telephone is dead, but too much leads them to lower their voices ITU-T recommends sidetone reference equivalents of at least 17 dB

By ensuring that the ITU-T recommended loss of 6 to 7 dB at least is encountered between the two reference points at either end of the four-wire part of a connection (be it digital or analogue), we not only safeguard the circuit against the effects of instability; we also make

it unlikely that any signal echo is generated by the two-to-four-wire converter (the so- called hybridconverter) These echoes are caused by reflection of the speaker’s voice back from the distant receiving end, and Figure 33.8 shows a hybrid converter causing a signal

echo of this kind The problem of echo arises whenever a hybrid converter is used, and it results from the non-ideal electrical performance (i.e the mismatch) of the device No matter whether the four-wire line is digital or analogue, the result is an echo

To understand the cause of the echo we have first to consider the composition of the hybriditself It consists of a bridge of four pairs of wires, one pair corresponds to the two- wire circuit, two more pairs correspond to the receiver and transmit pairs of the four-wire circuit, and the final pair is a balance circuit, the function of which is explained below The wires are connected to the bridge in such a way as to create a separation of the receive and transmit signals on the two-wire circuit from or onto the corresponding receive and transmit pairs of the four-wire circuit It is easiest to understand the type of hybrid which uses a pair of cross-coupled transformers as a bridge This is shown in Figure 33.9

Any signal generated at the telephone in Figure 33.9 appears on the transmit pair

(but not on the receive pair), and any incoming signal on the receive pair appears on the telephone (but not on the transmit pair) It works as follows

Electrical signal output from the telephone produces equal magnetic fields around windings W1 and W2 Now the resistance in the balance circuit is set up to be equal to

that of the telephone to induce equal fields around windings W3 and W4, but the cross-

coupling gives them opposite polarity The fields of windings W1 and W3 tend to act

together and to induce an output in winding W.5 Conversely, the fields of windings W2

ECHO CONTROL AND CIRCUIT INSTABILITY 603

Transformer I

Transformer 2

Figure 33.9 A hybrid transformer

and W4 cancel one another (due to the cross-coupling of W4), with the result that no output is induced in winding W6 This gives an output on the transmit pair as desired, but not on the receive pair

In the receive direction, the field around winding W6 induces fields in W2 and W4 This produces cancelling fields in windings W1 and W3 because of the cross-coupling of windings W3 and W4 An output signal is induced in the two-wire telephone circuit but not in the transmit pair, winding W5

Unfortunately, the balance resistance of practical networks is rarely matched to the resistance of the telephone For one thing, this is because the tolerance of workmanship

in real networks is much greater In addition, the use of exchanges in the two-wire part

of the circuit (if relevant) means that it is impossible to match the balance resistance to the resistance of all the individual telephones to which the hybrid may be connected The fields in the windings therefore do not always cancel out entirely as intended So,

for example, when receiving a signal via the receive pair and winding W6, the fields pro- duced in windings W1 and W3 may not quite cancel, and a small electric current may be induced in winding W5 This manifests itself to the speaker as an echo The strength of the echo is usually denoted in terms of its decibel rating relative to the incoming signal This is a value called the balance return loss, or sometimes, the echo return loss The more

efficient the hybrid, the greater the balance return loss (the isolation between receive and transmit circuits)

33.8 ECHO CONTROL AND CIRCUIT INSTABILITY

A variety of problems can be caused by echo, the three most important of which are

0 electrical circuit instability (and possible feedback)

0 talker distraction

data corruption

604 PRACTICAL NETWORK TRANSMISSION PLANNING

If the returned echo is nearly equal in volume to that of the original signal, and if a

rebounding echo effect is taking place at both ends of the connection, then the volume

of the signal can increase with each successive echo, leading to distortion and circuit

overload This is circuit instability, and as we already know the chance of it occurring is considerably reduced by adjusting the four-wire circuit to include more signal attenua- tion The total round-loop loss in the UK digital network shown in Figure 33.7 is at least

14 dB, probably inflicting at least 30 dB attenuation on echoes even if the hybrid has only

a modest isolating performance Talker distraction (or data corruption) is another effect

of echo, but if the time delay of the echo is not too long then distraction is unlikely, because all talkers hear their own voices anyway while they are talking

The echo delay time is equal to the time taken for propagation over the transmission link and back again, and is thus related to the length of the line itself The longer the line, the greater the delay Should the one-way signal propagation time exceed around

8 ms, giving an echo delay of 15 ms or more, then corrective action is necessary to eliminate the echo which most telephone users find obstrusive A one-way propagation time of 8 ms is inevitable on all long lines over 2500 km, so that undersea cables of this length and all satellite circuits usually require echo suppression Further propagation delay can also be caused by certain types of switching and transmission equipment Indeed a significant problem encountered with digital transmission media is that the time required for intermediate signal regeneration (detection and waveform reshaping) means that the overall speed of propagation is actually reduced to only 0.6 of the speed

of light This means that even quite short digital lines require echo suppression Two

methods of controlling echoes on long distance transmission links are common These

are termed echo suppression and echo cancellation Echo cancellation is nowadays most

common

An echo suppressor is a device inserted into the transmit path of a circuit It acts to

suppress retransmission of incoming receive path signals by inserting a very large attenuation into the transmit path whenever a signal is detected in the receive path Figure 33.10 illustrates the principle

Actually, the device in Figure 33.10 is called a half echo suppressor as it acts to suppress only the transmit path A full echo suppressor would suppress echoes in both transmit and receive paths

It is normal for a long connection to be equipped with two half-echo suppressors, one

at each end, the actual position being specified by the formal transmission plan Ideally half echo suppressors should be located as near to the source of the echo as possible (i.e as near to the two-to-four-wire conversion point as possible), and works best when near the ends of the four-wire part of the connection In practice it may not be economic

to provide echo suppressors at all exchanges in the lower levels of the hierarchy, and so they are most commonly provided on the long lines which terminate at regional (class 1

of Figure 33.6) and international exchanges

Sophisticated inter-exchange signalling is used to control the use of half echo sup- pressors Such signalling ensures that on tandem connections of long-haul links inter- mediate echo suppressors are ‘turned off in the manner illustrated by Figure 33.1 1 This ensures that a maximum of two half echo suppressors (one at each end of the four-wire part of the connection) are active at any one time

The amount of suppression (i.e attenuation) required to reduce the subjective disturb- ance of echo depends on the number of echo paths available, the echo path propagation

ECHO CONTROL AND CIRCUIT INSTABILITY 605

National network B (no echo suppressors 1

t- -f- Transmit and receive circuit

ES Half -echo suppressor

Figure 33.11 Controlling intermediate half-echo suppressors

606 PRACTICAL NETWORK TRANSMISSION PLANNING

time, and on the tolerance of the telephone users (or data terminal devices) ITU-T recommends that echo suppression should exceed (1 5 + n ) dB where n is the number of links in the connection

Unfortunately echo suppressors cannot be used on circuits carrying data, because the switching time between attennuation-on and attennuation-off states is too slow and can itself cause loss or corruption of data Most data modems designed for use on telephone circuits are therefore programmed to send an initiating 2100 Hz tone over the circuit, to

disable the echo suppressors

Another form of echo control device, called an echo canceller, can be used on either

voice or data circuits, and is the most common form of echo control device used in conjunction with digital circuits Like an echo suppressor, an echo canceller has a signal detector unit in the receive path However, instead of using it to switch on a large attenuator, it predicts the likely echo signal (digitally) and literally subtracts this prediction from the returning ‘transmit’ signal, thereby largely ‘cancelling’ out the real echo signal Other signals in the transmit path should be unaffected Listeners rate the performance of echo cancellers to be better than that of echo suppressors This, coupled with the fact that they do not corrupt data signals, has made them standard equipment

An important consideration of the network transmission plan is the overall signal delay

or propagation time Excessive delay brings with it not only the risk of echo, but also a

number of other impairments In conversation, for example, long propagation times between talker and listener can lead to confusion In the course of a conversation, when

we have said what we want to say, we expect a fairly prompt response If we are met with a silent pause, caused by a propagation delay, then we may well be tempted to speak again, to check that we have been heard (‘Are you still there?’) Sure as fate, as soon as we do that, the other party appears to start saying its piece, and everyone is talking at once

On video the effect of signal delay is even more revealing For example, on live satellite television broadcasts whoever is at the far end always gives the impression of pausing unduly before answering any question

Nothing can be done to reduce the delay incurred on a physical cable or satellite transmission link Thus intercontinental telephone conversations via satellite are bound

to experience a one-way propagation delay of about 1/4 second, giving a minimum pause between talking and response of l / 2 second Furthermore, the extremely rapid bit speeds and response times that computer and data circuitry are capable of, can be affected by line lengths of only a few centimetres or metres Line lengths should therefore be mini- mized and circuitous routings avoided as far as possible It is common for maximum physical line lengths to be quoted for data networks Similarly, in telephone networks, rigid guidelines demand that double or treble satellite hops or other excessive delay paths (i.e those of 400 ms one-way propagation time or longer) are avoided whenever possible Excessive delays can be kept in check by appropriate network routing algorithms, as we saw in Chapter 28

NOISE AND CROSSTALK 607

With the emergence of ATM and other ‘fast packet switching’ technology in some voice telephone networks (e.g corporate telephone networks) the problem of speech propagation delay has been exacerbated by the time required to fill an ATM cell of

48 bytes (48j8000 = 0.6 ms) and by the delay caused by the exit buffer (sometimes up to 30ms) The exit buffer is intended to eliminate quality problems caused by cell delay

which is the obligation to use echo cancellation devices to control echo

The use of speech compression techniques also leads to increased signal propagation delays (primarily due to the processing time needed to compress the signal, but maybe additionally due to the increased time now needed to fill a frame or ATM cell (at 32 kbit/s

an ATM cell takes 12 ms to fill)) Multiple occurrences of compression and decompression are therefore to be avoided This requires careful design of the network topology and planning of relevant echo cancellation

adjacent power lines, electrostatic interference, or other telecommunication lines The only reliable way of controlling them is by careful initial planning and design of the trans- mission system and the route One source of noise results from the induction of signals onto telecommunications cables which pass too close to high power lines Another source of noise is poorly soldered connections or component failures Both of these are easily avoided However, by far the most serious source of noise in telecommunications networks and the type most difficult to contain is that caused by electromechanical exchanges themselves This type of noise results from the electrical noise associated with the electrical pulses needed to activate the exchange, and also from the mechanical

‘chatter’

Noise is minimized by ensuring that the signal strength is never allowed to fade to a

volume level comparable with that of the surrounding noise Thus a minimum signal-

ensures that the real signal is still perceptible amongst all the background If the signal becomes too weak in comparison with the noise; unfortunately amplification is then of little value because it boosts signal and noise strength equally It is difficult to remove noise without affecting the signal itself, but there is some scope for removing noise which lies outside the frequency spectrum of the signal itself by simple filtering This can

slightly improve the signal-to-noise ratio ( S j N )

The noise caused by atmospheric interference and magnetic or electrostatic induction has a random nature and is heard as low level hum, hiss or crackle When this type of noise is measured using a special type of filter, weighted to reflect the human audible range, then the noise strength can be quoted in psophometric units ITU-T recommends that the strength of psophometric noise should not exceed an electromotive force (e.m.f )

at the receiving end of 1 millivolt (1 mV) Depending on the length of the connection, the transmission network designer may decide the maximum allowable noise disturbance

per kilometre of line Typically this value is around 2-3 picowatts (a very small unit of power) per kilometre This might be written 2 pWOp/km, where pW stands for picowatts,

608 PRACTICAL NETWORK TRANSMISSION PLANNING

0 indicates that the value is measured relative to the 0 dBr reference point, and p denotes

psophometric noise

adjacent transmission lines (see Chapter 3) The presence of crosstalk is usually an indication that the circuit has a fault which is causing it to perform outside its design range Perhaps an amplifier is turned up too much or some other fault has caused an abnormally high volume signal in the adjacent transmission line, or perhaps there is a short circuit or an insulation breakdown caused by damp The transmission plan normally seeks to minimize crosstalk by ensuring that the transmit and receive signal levels at any point along the connection never differ by more than 20dB

In Chapter 3 we discussed the use of equalizers to counteract the attenuation distortion

of signals which is caused by differential attenuation of the component frequencies Equalizers are generally located in transmission centres, alongside amplifiers The

equalizers which correct frequency distortion (also called attenuation distortion) make

sure that the attenuation of component frequencies is less than a specified maximum ITU-T recommendations for speech channels suggest a maximum differential attenua- tion of 9 dB, using appropriate equalization to ensure that the frequency response of the circuit is within the musk illustrated in Figure 33.12

However, frequency distortion equalizers are not the only type of equalizer Another type is called a group delay equalizer This acts by removing the distorting effects of

Shaded region is the unallowed region, or ‘mask’

The frequency response line must lie wlthin the mask

Figure 33.12 Frequency distortion: CCITT’s G132 ‘Mask’

TRANSMISSION PLAN FOR DIGITAL AND ‘DATA’ NETWORKS 609

slightly different propagation times of the component frequencies within the signal Group delay is particularly harmful to data signals passing over FDM (frequency divi-

those frequency components which are received first, to even up the delay incurred by all the frequency components

Both normal frequency attenuation equalizers and group delay equalizers are cali- brated when the circuit is initially lined up A range of different frequencies of known phase and signal strength is sent along the transmission line, and the characteristics of the received signals are carefully measured The equalizers are then adjusted accord- ingly and placed in the circuit A quick re-test should reveal perfect circuit equalization

‘DATA’ NETWORKS

Transmission planning of a digital network, as compared with its analogue equivalent,

is relatively straightforward primarily because the technique of regeneration practically eliminates the problems of crosstalk, noise, attenuation and interference However, a

number of new factors need to be taken into account in a digital transmission plan; they are

e the digital line bit error rate (or bit error ratio, B E R )

e the synchronization of the network

e the quantization distortion

e (during the period of network transition) the number of analogue-to-digital and digital-to-analogue conversions

Let us take them in order

so frequently along the length of the line to ensure that marks (1 S) and spaces (Os) are not

confused by the receiver In addition, although less prone than analogue lines, it is still prudent to protect a digital line as far as possible from noise interference and crosstalk to prevent spurious bit errors The frequency of such error is measured in terms of the error

rate (the bit error rate or bit error ratio, B E R ) Typical acceptance values of BER range

from around 1 bit error in 105 bits (BER = 10-5) to one error in 109 bits, depending on the application Some types of modern transmission system (including fibre transmission systems and modern digital radio systems) operate a t very low bit error rates (BER = 10-l2) The error rate is usually checked when the digital line system is first established All channels in the same line system (e.g each 64 kbit/s channel of a 2 Mbit/s

line system) will experience the same bit error rate ( B E R )

The synchronization of digital networks ensures that there is no build-up or loss of information in the line system Without synchronization, if a transmitter sent data faster than the receiver was ready to receive it, the result would be a build-up, and ultimately loss

of information Conversely, if the receiver was expecting data to arrive at a rate faster than the transmitter could send them, then imaginary ‘fill-in’ data would need to be created for

610 PRACTICAL NETWORK TRANSMISSION PLANNING

the missing bits Synchronization of digital networks is usually carried out in a hier- archical manner, with one exchange designated to house the muster clock, providing syn- chronization for other exchanges Figure 33.13 illustrates a typical three-tier synchroniza-

tion plan At the top of the hierarchy, a single exchange has a master clock In the second tier a number of main exchanges receive synchronization (i.e a data transmitting and

receiving rate) from the master clock exchange; additionally they are locked together by two-way synchronization links, keeping them all rigidly in step Finally, at the bottom of the hierarchy the smaller exchanges merely receive synchronization sources from the

higher level exchanges, but do not have two-way synchronization links Any digital clocks which are located any of the exchanges of the network can be adjusted to run faster or slower to keep track with the synchronization clock In case the main clock fails it is usual for one of the second tier exchanges to act as a standby master clock, ready to take over should the exchange with the master clock go off-air

TRANSMISSION PLAN FOR DIGITAL AND ‘DATA’ NETWORKS 611

The third important element of a digital transmission plan is the control of

quantization distortion arises because the quantum amplitude values which are available to represent the signal always differ slightly from the actual value The relationship between the representative digital signal and the original is therefore less than perfect In turn, there will be differences between the final signal and the original, manifested as slight (maybe even imperceptible) distortion This type of quantization distortion occurs during the initial conversion of any analogue (e.g speech) signal into its digital equivalent and cannot be recovered on re-conversion Further quantization distortion can occur should certain other transmission equipments be installed on the digital transmission path The use of such equipments may be unavoidable, but the transmission plan should clearly set out how much extra distortion is permissible Either the equipments need to be designed to conform with these limits, or the quality

of transmission will suffer Examples are echo cancellers (discussed earlier in this chapter) and circuit multiplication devices (discussed in Chapter 38)

Quantization distortion is measured in quantization distortion units (q.d.u.s) One q.d.u is equivalent to a difference in amplitude between the digital signal and the original analogue signal equal to one digital quantum level (for explanation of quantum levels, return to Chapter 5) Thus if the final signal is reproduced with amplitudes varying from that of the original signal by a whole quantum amplitude step, then

1 q.d.u of distortion has been encountered A digital transmission plan needs to lay out

strict limits on the maximum quantization distortion that can be allowed in any link or part of the network CCITT 1984 recommendations (still valid today) suggested a maximum end-to-end quantisation distortion of 14 q.d.u.s, allocating 5 q.d.u.s for each national network and a 4q.d.u limit for the international connection in between Typical values of quantization distortion are given in the table of Table 33.1 They

reflect the planning values given in ITU-T recommendation G 1 13

From the table of Table 33.1 it is clear that quantization distortion occurs as a result of

any form of signal processing, and is particularly sensitive to analogue/digital con- version processes and speech compression using either 32 kbit/s ADPCM or 16 kbit/s

CELP algortihms (as we discuss more in Chapter 38) For this reason, connections of interleaved analogue and digital sections, as well as multiple speech compression/ decompression stages should be avoided as far as possible ITU-T states this require- ment by recommending the limitation of the number of unintegrated PCM digital sections

to 3 or 4 and no more than 7 Of course, as the network evolves and digital transmission becomes more widespread this problem will disappear

Table 33.1 Typical quantization distortion values

Quantization Digital process distortion units A/D conversion by 8-bit PCM 1

A/D conversion by 7-bit PCM 3

A-law to Mu-law conversion 0.5

ADPCM (see Chapter 20) 3.5