M01- Communication Basics

TC31009

Certified Wireless Maintenance Professional

MODULE - 01

COMMUNICATION BASICS

H.O- First Floor, Rajput House, Gokhale Road, Thane (West)-Pin : 400602

Phone: 044-2537 6408 / 5592 1830

Email : [email protected] Website : www.teleman.in

Communication Basics Module - 01

TC31009D10 2

Module - 01

Communication Basics

INDEX

Chapter Topic Pages

Transmission

01 Transmission Units and Impairments 4 - 18 02 Modulation and Multiplexing 19 – 47 03 Transmission Media 48 – 60 04 Optical Fiber Communication 61 – 79 05 Optical Fiber Cable Splicing 80 – 97 06 OFC Test and Measurements 98 – 104 07 Optical Connectors and Couplers 105 – 115 08 Coding Theory 116 – 125 09 Basics of Transmission Systems 126 - 147 Switching

10 Introduction to Electronic Exchanges 148 – 174 11 Switching Concepts 175 – 180 12 Digital Switching 181 – 192 13 Signaling in Telecom Systems 193 – 219 14 ISDN 220 – 225 15 Telephone Traffic Engineering 226 – 247 16 Long Distance Switching Plans 248 – 291 17 Network Management 292 - 303


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Course Name

Teleman Certified Wireless Maintenance Professional

(TCWMP)

Course Code TC31009 Document Code TC31009D10

Module No.

Topic

No. of Chapters

01 / 05


17

Issue No.

Revision No.

Issue Date

Effective Date

1

0

15-12-2008

01-01-2009

Teleman (A Division of Plug-n-Work Directions) January 2009 Copyright © P-n-W 2009, All rights reserved No part of this publication may be reproduced, stored in retrieval system or transmitted in any form or by any means, electronic, mechanical, photocopying, recording or otherwise, without the prior permission of the publisher. Trademark acknowledgements: All products are registered trademarks of their respective organisation.


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Module - 01


Chapter – 01

Transmission Units and Impairments

Session Objectives: On completion of this session, you will be able to understand the concepts and able to explain:

• Transmission Units • Transmission Impairments • Decibel and Neper • Basic derived decibel units • Signal-to-Noise Ratio • Digital Transmission - Performance Criteria • Cross-Talk in Transmission Media


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1. Historical overview

Telecommunication systems started with the transmission of digital signals. In fact, non–electric signaling systems date back over 2000 years. The Greek General Polybius is known to have used a scheme based on an array of 10 torches in 300 B.C. and Roman armies made extensive use of a form of semaphore signaling. Claude Chappe, Sommering, Whetstone and Cook were all experimenting with different kinds of Telegraphy till it was perfected by Mores. In all this, only written message was transmitted and message was converted to a coded signal to match the characteristics of a transmission line. Gary, Baudot and others developed other codes, which were mainly used in Telegraph network. Thus, we can say, by 1972 most of the basic techniques of digital transmission had been discovered. In 1876, Alexander Graham Bell invented the Telephone and as means of communication, the telephone was fast, personal and convenient. It needed no training in the use of codes and so made electrical communications directly accessible to the general public. Thus, telephone began to dominate the development of communications. Telephony involves the transmission of analog signals and when a practical amplifying service appeared in the form of the thermionic valve, this also proved suitable for dealing with analog signals. Hence, after 1880, the developing Telecom networks were basically designed to handle analog transmissions and to an increasing extent, the digital transmission in the form of telegraphy had to be adopted to fit in with the characteristics of these networks. By 1950s, the world's communications systems were based entirely on analog transmission. However, interest in the digital transmission received an impetus after the publications of classic papers of Nyquist and Shannon. With the invention of pulse code modulation by Reeves in 1938, the basic principles for digitising analog speech signals were established. However, the technical means for transmitting digitized speech signals were not available at that time. It was not until the transistor came into use that indications of the economic advantages of digital techniques as compared to analog methods became apparent. LSI and VLSI techniques that are now available have made digital communications far more economical as compared to analog methods became apparent. LSI and VLSI techniques that are now available have made digital communications far more economical as compared to analog systems.


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Digital transmission systems are gaining more acceptances in view of:

• Introduction of digital switching systems • The need to transmit non-voice signals which are increasingly

becoming important instead of the plain old Telephone service. • The introduction of new media like optical fibers, wave-guide

that are more suitable for digital transmission systems, will be introduced in the network and by the turn of the century, most of the countries would have gone completely digital.

2. Transmission Units

The study of transmission units has a unique importance for communication engineer who has to maintain and install telecommunication equipments achieving the standards set up by international consultation committees. In order to control the quality of wanted signal in the presence of many undesired signals, we should be able to specify the amount of wanted and unwanted signals at a point in the telecommunications network. The components used in the telecommunication circuit either give loss or gain to the signals they handle. There are certain specific operating conditions to be satisfied for various components without which the optimum performance cannot be obtained from these components. For this, it is essential to define conditions that control those operating conditions. This can be done only if the conditions are specified in terms of certain units of the quantity the components are to handle.

3. Transmission Impairments

With analog transmission systems using copper cable there are three major categories of impairments. They are attenuation, noise, and distortion.

• Attenuation: There are two commonly used processes to compensate (overcome) for attenuation or loss:

(a) Repeaters are the most commonly used devices to compensate for "Loss." However, repeaters amplify the noise along with the signal resulting in a poor signal to noise ratio.

(b) Signal to Noise Ratio: The ratio of the average signal power (strength) to the average noise power (strength) at any point in a transmission path.


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• Noise: Any random disturbance or unwanted signal on a transmission facility that obscures the original signal. The environment in which the system is operating generally causes noise.

• Distortion: Inaccurate reproduction of a signal caused by changes in the signal's waveform, either amplitude or frequency, to compensate for distortion equalizers may be used. One type of equalizer used in the analog environment is the load coil. Load coils are used to flatten the frequency response.

The higher voice frequencies attenuate at a higher rate than the lower voice frequencies.

4. Decibel and Neper Historically speaking ‘attenuation’ was first of all defined in terms of the attenuation produced by a standard reference cable known as “mile of standard cable”. It consists of 88 ohms series impedance and 0.54 µF as shunt impedance. The fundamental objection to this unit was the fact that the attenuation of the standard cable varied with frequency. With the introduction of systems operating over different frequency ranges, it became necessary to define a unit which was independent of frequency .The unit which represents the useful and convenient concepts in connection with the transmission of signals over telephone lines has been named and defined as “Bel” (which comes from the name Alexander Graham Bell -the inventor of Telephone). In practice, however, a smaller and more convenient unit called decibel (abbreviated as dB) is used.

DECIBEL (dB) One tenth of the common logarithm of the ratio of relative powers, equal to 0.1 B (bel). The decibel is the conventional relative power ratio, rather than the bel, for expressing relative powers because the decibel is smaller and therefore more convenient than the bel. The ratio in dB is given by

X = log P2/P1 B i.e. = 10 log P2/P1 dB

Where P 1 and P 2 are the actual powers. Power ratios may be expressed in terms of voltage and impedance, E and Z, or current and impedance, I and Z. Thus dB is also given by;

X = 20 log V2/ V1 dB. (When Z 1 = Z 2 )


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Note: The dB is used rather than arithmetic ratios or percentages because when circuits are connected in tandem, expressions of power level, in dB, may be arithmetically added and subtracted. For example, in an optical link if a known amount of optical power, in dBm, is launched into a fiber, and the losses, in dB, of each component (e.g., connectors, splices, and lengths of fiber) are known, the overall link loss may be quickly calculated with simple addition and subtraction.

Example 1 Let us look at the following network: The input is 1W and its output 2W, therefore,

Gain = 10 log (output)/ (input) dB. = 10 log 2/1 dB= 10 (0.3010) dB=3.010 dB = 3dB approximately

Example 2 Let us look at another network:

Loss = 10 log Input/Output =10 log 1000/1 dB =10 log 103

dB =30 log 10 dB = 30 dB

Thus a network with an input of 5 W and output of 10 W is said to have Gain = 10 log 10/5 dB

= 10 log 2 dB =3.103 dB = 3 dB.

Let us remember that doubling the power means a 3 dB gain; likewise halving the power means a 3dB loss. Example 3 Consider a network with a 13 dB gain:

0.1W ?

Gain = 10 log P2/P1 dB = 10 log P2/0.1 dB =13db i.e., log P2/0.1 = 1.3 or P2/0.1 = antilog 1.3 or P2 = 0.1 antilog 1.3 P2 = 2W

Network 13 db gain

Net Work 1 W 2 W

Net Work 1000 W 1 W


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Example 4 Consider the following network

1W

What is the power output of this network? To do this without pencil and paper, we would proceed as follows:

• Suppose the network attenuated the signal by 30 dB. Then the output would be 1/1000 of the input or 1mW.

• Now 27 dB loss is 3dB less than 30 dB. • Thus the output would be twice 1m W i.e., 2mW. • (Because the loss is less by 3 dB, the corresponding output will

be more i.e. double but not half) • It is quite simple. Thus, if we have multiples of 10 or 3 up or 3

down from these multiples, we can work it out in our mind without pencil and paper.

NEPER The natural logarithm of the ratio of two voltages (or currents) expresses the loss or gain in Nepers, N

i.e. X= loge V1/V2 (N)

When the loss (gain) is X Neper, V1 and V2 are voltages, then

ex = V1/V2

Example The loss of a transmission system is 1N when 2.72 V input voltage produces 1 V output voltage.

Comparing powers, X= 1/2 loge P1/P2 (N) or e2x = P1/P2 Other transmission units 1 deciNeper (dN) = 0.1 N

1 Centi Neper (cN) = 0.01N 1 MilliNeper (mN) = 0.001 N

5. Basic derived decibel units DBm Till now decibel has referred to ratios or relative units. We cannot say that the output of an amplifier is 33 dB. We can say that an amplifier has a gain of 33 dB or that a certain attenuator has a 6 dB loss. These figures or units don't give any idea whatsoever of absolute level. Whereas, several derived decibels units do. Perhaps the dBm is the most

Network 27 dB loss ?W


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common of these. By definition dBm is a power level related to 1 mw. The most important relationship to remember is:

0 dBm = 1mW.

The dBm formula may then be written as:

Power (in dBm) = 10 log Power (mW)/(1mW)

Example An amplifier has an output of 20 W; what is its output in dBm?

Power (dBm) = 10 log 20 W/1 mW = 10 log 20x103 mW/1mW = +43 dBm. (The plus sign indicates that the quantity is above the level of reference, 0 dBm.) dBmO Decibel referred to 1 mw at zero (0) Transmission level point. dBmO is a measure of power with reference to Zero dBm at the Reference Transmission Level Point (RTLP). The RTLP is also known as Zero Transmission Level Point (0TLP). Powers measured at any transmission level point can be expressed in dBmO, by correcting the power measured for the difference in level between the point of measurement and the RTLP. For example, a level of +25 dBm measured at a +17 dB transmission level point is equivalent to 8 dBmO. Conversely a level of +8 dBmO is also equivalent to +3 dBm measured at a -5 dB transmission level point. A level expressed in dBmO is, therefore, only a relative level.

Conversion from Neper to decibel and Vice Versa

We know that decibel is fundamentally a unit of power ratio but it can be used to express current ratios when the resistive components of the impedance, through which the current flows, are equal.

The Neper, on the other hand, is fundamentally a unit of current ratio but it can also be used to express power ratios when the resistive components, of the impedance, through which the current flows, are equal. Because of its derivation from the exponential e, the Neper is the most convenient unit for expressing attenuation in theoretical works. The decibel, on the other hand, being defined in terms of logarithms to base 10, is a more convenient unit in practical calculations using the decimal system.


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The conditions under which the two units may be used can be summarised in the following equations, the notation of which is indicated in Fig below.

Where Z1 and Z2 are characteristic impedances R1 and R2 are pure resistances G1 and G2 are leakances β1 and β 2 are phase angles X1 and X2 are reactance.

Atténuation in decibels = 10 log10 |PS/PR |.................. (i) Attenuation in Nepers = log e| IS/IR|.......................................... (ii)

Equation (i) can be expressed as current ratio and also as voltage ratio as follows

Attenuation in dB = 20 log10 IS/IR if R1= R2 (iii)

= 20 log10 ES/ER if G1 = G2 (iv) Attenuation in Nepers = log e ES/ER if Z1 =Z2

(v) =1/2 log e PS/PR (if R1=R2 (vi)

If the resistive components of the impedance at the input and output of the network are equal, then the attenuation may be readily converted from one notation to another, for:

Attenuation in dB = 20 log10 IS/IR = 20 loge IS/IR log10 e (log10 e= .4343) = 8.686 loge IS/IR = 8.686 x (attenuation in Nepers)

Thus, Attenuation in dB = 8.686 x attenuation in Nepers

(vii) (provided that R1 = R2)

Attenuation in Nepers = 0.1151 x attenuation in dB (viii) (provided that R1 =R2)


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Other Units

In Analogue Transmission system, the quality of communication is mainly assessed by the value of Signal to noise ratio. 6. Signal-to-Noise Ratio

It is popularly known as SNR. SNR is the ratio of signal power to the noise power at any point in a circuit. This ratio is usually expressed in Decibels (dB). For satisfactory operation of a channel the value of SNR should be sufficiently high i.e., the signal power should be sufficiently higher than the noise power. SNR at any point in a circuit is given as: SNR = S/N = Signal Power / Noise Power Both powers are expressed in watts. Expressing dBs: SNR = 10 log10 (S/N) dB. Example: Signal voltage Vs = 0.923 µV; Noise voltage Vn = 0.267 µV, then calculate the signal-to-noise ratio. S/N = Vs2 / Vn2 = 0.923/0.267)2 = 11.95 In decibels, S/N = 10 log10 (11.95) = 10.77 dB.

In Digital Transmission system, the quality of communication is mainly assessed by two factors.

1. BER (Bit Error Ratio) 2. Jitter

These two factors can be taken as Quality Factors as they are used for judging the quality of Digital Transmission. Bit Errors

In the digital transmission, the bits transmitted at the transmitting end (1 or 0 ) are not always detected as 1 or 0 at the receiving end. When the transmitted bit 1 or 0 is not identified as 1 or 0 at the receiver, the bit is counted as an error bit.

For assessing the real error performance, the bit error ratio (BER) is to be calculated instead of actual error bits. Bit Error Rate (BER)

The BER is the measure or error bits with respect to the total number of bits transmitted in a given time. The total number of bits transmitted can be known from the bit rate of the digital signal. The bit rate is the number of bits transmitted in one second and is specified for each transmission system. Hence, the total number of bits transmitted in a given time can be counted. In the measurement of BER, generally the


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measuring instrument measures the number of bits transmitted in a given time.

The time setting can be from a few seconds to a few hours, depending on the feasibility. The standards are set by ITU (International Telecommunication Union). The time set for the measurement of BER, is called gating time. Larger the gating time better is the assessment of BER. But for the measurement of BER, the Digital Equipment has to be taken off-line.

Digital communication can just run with one error bit in one thousand bits received. For more than one error bit, in one thousand bits received, communication gets affected. For good quality communication, the requirement is not more than one error bit in one million bits. Jitter

Abrupt and unwanted variations of one or more signal characteristics, such as the interval between successive pulses, the amplitude of successive cycles, or the frequency or phase of successive cycles. Jitter must be specified in qualitative terms (e.g., amplitude, phase, and pulse width or pulse position) and in quantitative terms (e.g., average, RMS, or peak-to-peak). The low-frequency cut-off for jitter is usually specified at 1 Hz. Contrast with drift, wander.

Short-term variations of the significant instances of a digital signal from their reference position in time. (Short term frequency equal to or greater than 10 Hz.). Long term variations of significant instances of a digital signal from their ideal positions in time, are called wander. (Long-term variations – frequency less than 10 Hz).

Drift

A comparatively long-term change in an attribute or value of a system or equipment operational parameter. The drift should be characterized, such as "diurnal frequency drift" and "output level drift." Drift is usually undesirable and unidirectional, but may be bi-directional, cyclic, or of such long-term duration and low excursion rate as to be negligible. Wander

Relative to Jitter and swim, long-term random variations of the significant instants of a digital signal from their ideal positions. Wander variations are those that occur over a period greater than 1 s (second). Jitter, swim, wander, and drift have increasing periods of variation in that order.

Digital Transmission Analyser (DTA) is used for the measurement of both BER and Jitter.


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7. Digital Transmission - Performance Criteria (General) 1 in 106 (1.OE – 6) : Better 1 in 105 (1.OE – 5) : Good

1 in 104 (1.OE – 4) : Reasonably good 1 in 103 (1.OE – 3) : Just Acceptable More than 1 in 103 : Unacceptable Bit errors greatly affect data service. For data channels 1 in 109 (1.OE – 9) is normally realizable.

Quality Parameters

To pin point the exact number of seconds or minutes, in which the bit errors take place and up to what extent, the quality parameters are defined.

The quality parameters are:

1. Error Seconds (ES) 2. Severely Error Seconds (SES) 3. Non Severely Error Seconds (NSES) 4. Degraded Minutes (DM).

Error Seconds (ES): Number of one-second intervals with one or more errors. Severely Error Seconds (SES): Number of one-second intervals with an error rate, worse than 1.OE-3

Non-Severely Error Seconds (NSES): Number of one-second intervals with an error rate, better than or equal to 1.OE-3. Degraded Minutes (DM): Number of one-second intervals with a bit error rates worse than 1.OE-6. Available and non-available time

A period of available time begins with a period of ten consecutive seconds each of which has a BER better than 1.0E-3. These 10 seconds are considered to be available time.


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A period of unavailable time begins when the bit error rate in each second is worse than 1.0E-3 for a period of 10 consecutive seconds. These 10 consecutive seconds are considered to be unavailable time. 8. Cross-Talk in Transmission Media

Any disturbing signal produced by transfer of unwanted power

from one transmission path (called disturbing circuit) to another transmission path (called disturbed circuit) is known as cross talk. Cross talk may be produced by:

• Galvanic, capacitive or inductive couplings between transmission media (Linear cross-talk) e.g. between pairs of a VF (voice frequency) cable system.

• Poor control of frequency response i.e. defective filters or poor filter design is the cause.

• Non-linear performance in analogue (FDM) multiplex systems. A signal transmitted on one circuit or channel of a transmission system (multi-channel) creates an undesired effect in another circuit or channel (non-linear cross talk)

Types of cross talk Broadly speaking, cross talk is of six types.

1. Near-end cross- talk (NEXT). 2. Far-end cross talk (FEXT). 3. Intelligible cross-talk 4. Unintelligible cross-talk 5. Interaction cross-talk 6. Reflected cross-talk

Near-end cross talk (NEXT)

Near-end cross talk occurs if the cross talk power in the disturbed channel propagates in the direction opposite to the propagation of useful power in the disturbing channel. Refer to Fig 1.1 for illustration of near-end cross talk.

The terminals of the disturbed channel, at which the near-end cross talk is present, and the energized terminal of the disturbing channel, are usually near each other. The near-end cross talk is much stronger than far-end cross talk because the magnetic (or galvanic) and electrostatic inductions are additive in the case of near-end cross talk and the inducing current in the disturbing circuit is much stronger.


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Fig. 1.1 Near End Cross Talk Far-end cross-talk (FEXT)

It occurs if the cross talk power in the disturbed channel propagates in the direction of the propagation of the useful power in the disturbing channel. Refer to Fig.1.2 for illustration of far-end cross talk. The terminals of the disturbed channel, at which the far-end cross talk is present, and the energized terminals of the disturbing channel, are usually remote from each other. Far-end cross talk is less effective in impairment of the original signal in the disturbed circuit because the magnetic and electrostatic inductions are subtractive. Also the inducing current in the disturbing circuit gets very much attenuated after it has traveled to the far end.

Fig. 1.2 Far End Cross Talk

Intelligible cross talk The cross talk is intelligible when the whole or an important part,

of the speech on the disturbing circuit is intelligible on the disturbed circuit. Between circuits transmitting the same frequency band or working without frequency translation (audio-frequency) only intelligible cross talk can arise. As the secrecy of the conversation is affected by


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intelligible cross talk, steps should be taken to see that intelligibility of sentence articulation of the cross talk should be less than 10%. Unintelligible cross talk (also called noise)

The cross talk is unintelligible when the disturbing circuit gives rise only to noise in the disturbed circuit. It decreases the intelligibility but does not endanger the secrecy of conversation. Unintelligible cross talk occurs

• Between carrier channels having different frequency allocations. • Between carrier channels having virtual carrier frequencies

essentially differing from each other and • In consequence of non-linear distortion.

Interaction cross talk (Indirect cross talk) Interaction cross talk conveyed by a third circuit from the

disturbing circuit to the disturbed circuit, where it causes far end cross talk (fig.3). This type of cross talk is also called double near-end cross talk. It occurs mainly in two-wire carrier systems fitted with intermediate repeaters. Reflected cross-talk

Indirect cross talk caused by reflection due to mismatch of the circuit is called reflected cross talk. Causes of cross-talk

Cross talk is mainly caused by two types of induction viz., Magnetic and Electrostatic. Magnetic induction

It is well known that a change in magnetic lines of forces is associated with the flow of electric currents. The magnetic lines of forces due to currents flowing through circuit A will also embrace the wires of circuit B. As the current in circuit A alternates, the magnetic field also alternates, and according to Faraday' law it induces e.m.fs in the wires of circuit B Electrostatic induction

Electrostatic induction occurs due to the capacitance between four wires of the two circuits that are built side by side.

Practically it is noted that the current due to magnetic induction flows in one direction in the entire circuit, whereas that due to the electric induction flows through the two sections in opposite directions


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Methods for reducing cross talk

There are a number of methods for eliminating or at least substantially reducing cross talk in open-wire lines. Since the cross talk reduction in open wire lines depends upon three factors viz. Wire configuration, transposition and resistance unbalance. One possible method is to arrange the wires in such a configuration that the effect of the electric and magnetic fields of one pair will be the same on both wires of the disturbed pair, thus leaving no residual difference to cause currents in the disturbed circuit.

Another method to reduce the cross-talk is to reduce the separation between the wires of either or both disturbing and the disturbed pairs and, if possible, to increase the separation between the pairs themselves.

The most commonly used method is the use of “transpositions". Transposition means interchanging the position of the two wires forming the pair at regular intervals on the pole route, right through the length of the pair. The transposition is by far the most effective and practical method of reducing cross talk.


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Module - 01


Chapter – 02

Modulation and Multiplexing


• Need for Modulation • Analog-to-Analog Modulation • Analog-To-Digital Conversion • Pulse Code Modulation • Digital-To-Analog Conversion • Multiplexing Techniques • Digital Hierarchies • Multiple Access Methods


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1. Introduction The basics of modulation is to take a message bearing signal like an audio signal and superimpose it upon a carrier signal for transmission. For ease of transmission such carrier signals use generally high frequencies:

� For easy propagation as electromagnetic waves with low loss and low dispersion

� Simultaneous transmission without interference from other signals

� Enables the construction of small antennas (a fraction, usually a quarter of the wavelength)

Enables the multiplexing (combining) multiple signals for transmission at the same time over the same carrier Different modulation schemes are possible. Well known examples of high frequency carrier signals are:

• AM radio is 550-1600 KHz • FM radio is 88 MHz-108 MHz • TV is 52-88 MHz (channels 1-6), 174-216 MHz (channels 7-12)

and 470-900 MHz (UHF) • Microwave and satellite signals are of the order of several GHz • Infra red fiber optic signals are of the order of 200-300 THz.

2. Analog-to-Analog Modulation Analog-to-analog modulation is the representation of analog information by an analog signal. Radio, that familiar utility, is an example of an analog-to-analog communication. The following Fig 2.1 shows the relationship between the analog information, the analog-to-analog modulation hardware, and the resultant analog signal.

Fig 2. 1 Analog-to-analog modulation Analog-to-analog modulation can be accomplished in three ways

• Amplitude modulation (AM) • Frequency modulation (FM) • Phase modulation (PM)


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2.1 Amplitude modulation (AM) In AM transmission, the carrier signal is modulated so that its amplitude varies with the changing amplitudes of the modulating signal. The frequency and phase of the carrier remain the same; only the amplitude changes to follow variations in the information. The following fig show this concept works. The modulating signal becomes an envelope to the carrier.

Fig 2.2 Amplitude modulation 2.2 Frequency Modulation (FM)

In FM transmission, the frequency of the carrier signal is modulated to follow the changing voltage level (amplitude) of the modulating signal. The peak amplitude and phase of the carrier signal remain constant, but as the amplitude of the information signal changes, the frequency of the carrier changes correspondingly. The following Fig 2.3 shows the relationships of the modulating signal, the carrier signal, and the resultant FM signal.


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Fig 2.3 Frequency modulation 2.3 Phase Modulation Due to simpler hardware requirements, phase modulation (PM) is used in some systems as an alternative to frequency modulation. In PM transmission, the phase of the carrier signal is modulated to follow the changing voltage level (amplitude) of the modulating signal. The peak amplitude and frequency of the carrier signal remain constant, but as the amplitude of the information signal changes, the phase of the carrier changes correspondingly. The analysis and the final result (modulated signal) are similar to those of frequency modulation. 3. Analog-To-Digital Conversion We sometimes need to digitize an analog signal. For example, to send human voice over a long distance, we need to digitize it since digital signals are less prone to noise. This is called an analog-to-digital conversion or digitizing an analog signal. The following Fig 2.4 shows the analog-to-digital converter, called a codec (coder-decoder).

Fig 2.4 Analog-to-digital conversion


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In analog-to-digital conversion, we are representing the information contained in a continuous wave form as a series of digital pulses (1s or 0s). 3.1 Pulse Amplitude Modulation (PAM) The first step in analog-to-digital conversion is called pulse amplitude modulation (PAM). This technique takes an analog signal, samples it, and generates a series pulses based on the results of the sampling. The term sampling means measuring the amplitude of the signal at equal intervals. The method of sampling used in PAM is more useful to other areas of engineering than it is to data communication. However, PAM is the foundation of an important analog-to-digital conversion method called pulse code modulation (PCM). Fig 2.5 PAM In PAM, the original signals sampled at equal intervals as shown above. PAM uses a technique called sample and hold. At a given moment, the signal level is read, and then held briefly. The sampled value occurs only instantaneously in the actual waveform, but is generalized over a still short but measurable period in the PAM result. The reason PAM is not useful to data communications is that, although translates the original wave form to a series of pulses, these pulses are still of any amplitude (still an analog signal, not digital). To make them digital, we must modify them using pulse code modulation (PCM). 4. Pulse Code Modulation (PCM) PCM modifies the pulses created by PAM to create a completely digital signal. To do so, PCM first quantizes the PAM pulses. Quantization is a method of assigning integral values in a specific range to sampled instances. The result of quantization is presented in the Fig. 2.6.


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Fig 2.6 Quantized PAM signal The binary signals are then transformed into a digital signal using one of the digital-to-digital encoding techniques. The following Fig 2.7 shows the result of the pulse code modulation of the original signal encoded finally into a unipolar signal. Only the first three sampled values are shown.

Fig 2.7 4.1 From analog signal to PCM digital code PCM is actually made up of four separate processes

• PAM • Quantization • Binary encoding • Digital-to-Digital encoding


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The following Fig shows the entire process in graphic form.

Fig 2.8 (a) From Analog to PCM

Fig 2.8 (b) PCM Frame structure


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4.2 Sampling Rate The accuracy of any digital reproduction of an analog signal depends on the number of samples taken. Using PAM and PCM, we can reproduce the waveform exactly by taking infinite samples, or we can reproduce the barest generalization of its direction of change by taking three samples. According to the Nyquist theorem, to ensure the accurate reproduction of an original analog signal using PAM, the sampling rate must be at least twice the highest frequency of the original signal. So if we want to sample telephone voice with maximum frequency 4000 Hz, we need a sampling rate of 8000 samples per second. 5. Digital-To-Analog Conversion Digital-to-analog conversion or digital-to-analog modulation is the process of changing one of the characteristics of an analog signal based on the information in a digital signal (0s and 1s). The following Fig 2.9 shows the relationship between the digital information, the digital-to-analog modulating hardware, and the resultant analog signal.

Fig 2.9 Digital-to-analog modulation The following are the techniques adapted for digital-to-analog modulation

Fig 2.10 Types of digital-to-analog encoding


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5.1 Aspects of Digital-to-Analog conversion Bit Rate and Baud Rate Two terms used frequently in data communication are bit rate and baud rate. Bit rate is the number of bits transmitted during one second. Baud rate refers to the number of signal unit per second that are required to represent those bits. The baud rate determines the bandwidth required to send the signal. Bit rate equals the baud rate times the number of bits represented by each signal unit. The baud rate equals the bit rate divided by the number of bit represented by each signal shift. Bit rate is always greater than or equal to the baud rate. Carrier signal In analog transmission, the sending device produces a high-frequency signal that acts as a basis for the information signal. This base signal is called the carrier signal or carrier frequency. The receiving device is tuned to the frequency of the carrier signal that it expects from the sender. Digital information is then modulated on the carrier signal by modifying one or more of its characteristics (amplitude, frequency, and phase). This kind of modification is called modulation (or shift keying) and the information signal is called a modulating signal. 5.2 Amplitude Shift Keying (ASK) In amplitude shift keying (ASK), the strength of the carrier signal is varied to represent binary 1 or 0. Both frequency and phase remain constant while the amplitude changes. Which voltage represents 1 and which 0 is left to the system designers. A bit duration is the period of time that defines one bit. The speeds of transmission using ASK is limited by the physical characteristics of the transmission medium. The following fig gives a conceptual view of ASK.


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Fig 2.11 ASK

ASK transmission is highly susceptible to noise interference. The term noise refers to unintentional voltages introduced onto a line by various phenomena such as heat or electromagnetic induction created by other sources. Noise usually affects the amplitude; therefore, ASK is the modulating method most affected by noise. 5.3 Frequency Shift Keying (FSK) In frequency Shift Keying (FSK), the frequency of the carrier signal is varied to represent binary 1 or 0. The frequency of the signal during each bit duration is constant and its value depends on the bit (0 or1); both peak amplitude and phase remain constant. The following Fig gives the conceptual view of FSK.

Fig 2.12 FSK


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5.4 Phase Shift Keying (PSK) In phase shift keying (PSK), the phase of the carrier is varied to represent binary 1 or 0. Both peak amplitude and frequency remain constant as the phase changes. For example, if we start with a phase of 0 degrees to represent binary 0, then we can change the phase to 180 degrees to send binary 1. The phase of the signal during each bit duration is constant and its value depends on the bit (0 or1). The following Fig gives a conceptual view of PSK.

Fig 2.13 PSK The above method is often called 2-PSK, or binary PSK, because two different phases (0 and 180 degrees) are used. The following figure makes this point clearer by showing the relationship of phase to bit value. A second diagram, called a constellation or phase-state diagram, shows the same relationship by illustrating only the phases.

Fig 2.14 PSK constellation PSK is not susceptible to the noise degradation that affects ASK, or to the bandwidth limitations of FSK. This means that smaller variations in the signal can be detected reliably by the receiver. Therefore, instead


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of utilizing only two variations of a signal, each representing one bit, we can use four variations and let each phase shift represent two bits.

Fig 2.15 4-PSK The constellation diagram for the signal shown above, a phase of 0 degrees now represents 00; 90 degrees represents 01; 180 degrees represents 10; and 270 degrees represents 11. This technique is called 4-PSK or Q-PSK. The pair of bits represented by each phase is called a dibit. We can transmit data two times as fast using 4-PSK as we can use 2-PSK.

Fig 2.16 4-PSK characteristics We can extend this idea to 8-PSK. Instead of 90 degrees, we now vary the signal by shifts of 45 degrees. With eight different phases, each shift can represent three bits (one tribit) at a time. The following figure shows the relationships between the phase shifts and tribits each one represents: 8-PSK is three times faster than 2-PSK.


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Fig 2.17 8-PSK characteristics 5.5 Quadrature Amplitude Modulation (QAM) PSK is limited by the ability of the equipment to distinguish small differences in phase. This factor limits its potential bit rate. So far, we have been altering only one of the three characteristics of a sine wave at a time, but what if we alter two? Bandwidth limitations make combinations of FSK with other changes practically useless. But why not combine ASK and PSK? Then we could have x variations in phase and y variations in amplitude, giving us x times y possible variations and the corresponding number of bits per variation. Quadrature amplitude modulation (QAM) does just that. Possible variations of QAM are numerous. Theoretically, any measurable number of changes in amplitude can be combined with any measurable number of changes in phase. The following figure shows two possible configurations, 4-QAM and 8-QAM. In both cases, the number of amplitude shifts is fewer than the number of phase shifts. Because amplitude changes are susceptible to noise and require greater shift differences than do phase changes, the number of phase shifts used by a QAM system is always larger than the number of amplitude shifts.


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Fig 2.18 4-QAM and 8-QAM constellations The time-domain plot corresponding to the 8-QAM signal is shown below

Fig 2.19 Time domain for an 8-QAM signal 5.6 Bit/Baud Comparison Assuming that an FSK signal over voice-grade phone lines can send 1200 bits per second, the bit rate is 1200 bps. Each frequency shift represents a single bit; so it requires 1200 signal elements to send 1200 bits. Its baud rate, therefore, is also 1200 bps. Each signal variation in an 8-QAM system, however, represents three bits. So a bit rate of 1200 bps, using 8-QAM, has a baud rate of only 400. The following table gives a comparative bit and baud rates for the various methods of digital-to-analog modulation.


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Modulation Units Bits/Baud Baud Rate Bit

Rate ASK, FSK, 2-PSK Bit 1 N N 4-PSK, 4-QAM Dbit 2 N 2N 8-PSK, 8-QAM Tribit 3 N 3N 16-QAM Quadbit 4 N 4N 32-QAM Pentabit 5 N 5N 64-QAM Hexabit 6 N 6N 128-QAM Septabit 7 N 7N 256-QAM Octabit 8 N 8N

5.7 PDM (Pulse Duration Modulation) PDM is a method of pulse modulation in which the duration of the pulse train is used to transfer the binary signal information. 5.8 PPM (Pulse Position Modulation) The amplitude and width of the pulse is kept constant in the system. The position of each pulse, in relation to the position of a recurrent reference pulse, is varied by each instantaneous sampled value of the modulating wave. PPM has the advantage of requiring constant transmitter power since the pulses are of constant amplitude and duration. It is widely used but has the big disadvantage that it needs a synchronization between transmitter and receiver. 5.9 PWM (Pulse Width Modulation) Pulse Width Modulation refers to a method of carrying information on a train of pulses, the information being encoded in the width of the pulses. In applications to motion control, it is not exactly information we are encoding, but a method of controlling power in motors without (significant) loss. In battery systems PWM is the most effective way to achieve a constant voltage for battery charging by switching the system controller's power devices on and off. 6. Multiplexing Techniques Suppose a company with a link between two cities wished to maximize the traffic between them. First, the data must be sent faster. Then, more links must be acquired. This is the basis of multiplexing, transmission technology deals with. The transport network has been defined as a set of links between telecommunication sites. Before multiplexing was discovered, each


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telephone call needed its own link to be transmitted. Many telephone calls needed many links, which was expensive. A way to put more than one telephone call on each link must be found to save money. The best way to put more than one telephone call on each link is to multiplex the calls. This makes best use of the links. The easiest way to understand multiplexing is to remember the transmission game one played as a child: two tin cans connected by a piece of string (see Figure 2.20). In essence, that was a private link. Multiplexing, however, enables several telephone calls to be sent on the same line. The end users have the illusion of being on their own private link. In effect, multiplexing creates a virtual telephone link for all of the users, which is an early telephone version of virtual reality. Transmission systems use a different type of multiplexing.

Figure 2.20 Multiplexing Transmission systems that are designed according to European rules work with groups of 30 telephone calls. A group of 30 telephone calls is multiplexed into a 2–Mbps digital signal, and throughout most transmission documents and presentations, constant references to 2–Mbps channels may be found. These 2–Mbps streams are the basic building blocks for multiplexing. There are basically three types of multiplexing techniques.

• Frequency Division Multiplexing (FDM) • Time division Multiplexing (TDM) • Code Division Multiplexing (CDM)


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6.1 Frequency Division Multiplexing (FDM) Frequency Division Multiplexing (FDM) technique is the process of translating the frequencies of individual channels, having the bandwidth 300-3400 Hz, into pre-assigned frequency slots within the bandwidth of the transmission medium. The frequency translation is done by Amplitude Modulation (AM) of an appropriate carrier frequency by the Audio Frequency. At the output of the modulator, a filter network is connected to select either the lower or upper of the side band. Since the intelligence is carried in either of the two side bands, Single Side Band Suppressed Carrier (SSBSC) mode of AM is used. This has advantages of substantial saving of bandwidth and also permits the use of low power amplifiers. In this way number of channels can be combined by using different carrier frequencies for different channels, and transmitting all of them on a single medium. Application of FDM Techniques FDM techniques usually find application in the following analogue transmission systems i.e., in a system used for transmitting continuously varying signals.

• Carrier Systems • Coaxial Systems • Microwave systems • Satellite systems

Limitations of FDM Analogue Systems

• Interference (Near-end and Far-end cross-talk) • Noise (increases with the length of the system) • Distortion (increases with the length of the system)

Because of the various limitations and transmission impairments in analogue transmission system, need for digital systems using Time Division Multiplexing arises. 6.2 Time Division Multiplexing (TDM) Basically, TDM involves sharing a transmission medium by a number of circuits, in time domain. This can be achieved by establishing a sequence of time periods during which individual channel (circuit) is transmitted. Thus the entire bandwidth is periodically available to each channel. Each channel is sampled at a specified rate and transmitted for a fixed duration. All channels are sampled, one by one and transmitted. Normally, all the time durations are equal. Each channel is assigned time duration with a specific common repetition period. The channels are connected to individual gates, which are opened one by one, in a fixed sequence. At the receiving end also, similar gates are opened in unison with the gates at the transmitting end.


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The signal received at the receiving end will be in the form of discrete samples, which are combined to reproduce the original signal. Thus, at a given instant of time, only one channel is transmitted through the medium, and by sequential sampling, a number of channels can be staggered in time, instead of transmitting all the channels at the same time as in FDM systems. TDM Technique is used in Digital Transmission Systems and Digital Switching Systems, where the discrete values of the codified signal are employed. 6.3 Code Division Multiplexing (CDM) Code Division Multiplexing is a technique in which each channel transmits its bits as a coded channel-specific sequence of pulses. This coded transmission typically is accomplished by transmitting a unique time-dependent series of short pulses, which are placed within chip times within the larger bit time. All channels, each with a different code, can be transmitted on the same media and asynchronously demultiplexed. Code Division Multiple Access (CDMA), a method for transmitting simultaneous signals over a shared portion of the spectrum. The foremost application of CDMA is the digital cellular phone technology from QUALCOMM. Unlike the other digital systems that divide the spectrum into different time slots, CDMA's spread spectrum technique overlaps every transmission on the same carrier frequency by assigning a unique code to each conversation. The often-used analogy for this is your ability to detect your own language in a room full of people speaking other languages. 7. Digital Hierarchies The term “digital hierarchy” has been created when developing digital transmission systems. It was laid down when by multiplexing a certain number of PCM primary multiplexers were combined to form digital multiplexers of higher order (e.g. second-order multiplex equipments). Consequently, a digital hierarchy comprises a number of levels. Each level is assigned a specific bit rate which is formed by multiplexing digital signals, each having the bit rate of the next lower level. 7.1 Definition

In CCITT Rec. G.702, the term “Digital Multiplex Hierarchy” is defined as follows:


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“A series of digital multiplexes graded according to capability so that multiplexing at one level combines a defined number of digital signals, each having the digit rate prescribed for the next lower order, into a digital signal having a prescribed digit rate which is then available for further combination with other digital signals of the same rate in a digital multiplex of the next higher order”.

7.2 Why Hierarchies?

FDM System

Before considering in detail the digital hierarchies under discussion we are going to recapitulate in brief, why there are several digital hierarchies instead of one only. It has always been pointed out that as far as the analogue FDM technique is concerned, the C.C.I.T.T. recommends the world wide use of the 12-channel group (secondary group). Relevant C.C.I.T.T. Recommendation exists also for channel assemblies with more than 60 channels so that with certain exceptions – there is only one world-wide hierarchy for the FDM system (although the term “hierarchy” is not used in the FDM technique).

Digital Scenario

In the digital transmission technique it was unfortunately not possible to draw up a world-wide digital hierarchy. In practice, equipment as specified in C.C.I.T.T. Recommendation G.732 and 733, they not only differ completely in their bit rates, but also in the frame structures, in signaling, frame alignment, etc. Needless to say that, as a consequence, the higher order digital multiplexers derived from the two different PCM primary multiplexers and thus the digital hierarchies differ as well.

Digital Hierarchy based on the 2048 Kbps PCM

For this digital hierarchy, two specifications have at present been laid down only for the first level at 2048 Kbps and for the second level at 8448 Kbps.

As for the higher levels, the situation is just contrary to that existing in the case of digital hierarchies derived from 1544 Kbps primary multiplex. General agreement has more or less been reached on the fourth level having a bit rate of 139264 Kbps. 5th order system where bit rate of 565 Mb/s have also been planned now.

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


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139264 Kbps in the digital hierarchy which is based on the 2nd order bit rate of 8448 Kbps”.

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

Method 1 by using a 3rd order bit rate of 34368 Kbps in the digital hierarchy.

Method 2 by directly multiplexing sixteen digital signals at 8448 Kbps. The digital signals at the bit rate of 139264 Kbps obtained by these two methods should be identical.

The existence of the above two methods implies that the use of the bit rate of 34368 Kbps should not be imposed on an Administration that does not wish to realize the corresponding equipment.

Encoded TDM (European)

In accordance with the above two methods the following realizations of digital multiplex equipments using positive justification are recommended:

Method 1: Realization by separate digital multiplex equipments: one type which operates at 34368 Kbps and multiplexes four digital signals at 8448 Kbps; the other type which operates at 139264 Kbps and multiplexes four digital signals at 34368 Kbps.

Method 2: Realization by single digital multiplex equipment which operates at 139264 Kbps and multiplexes sixteen digital signals at 8448 Kbps.

Method 1 has been put into practice.

Where the fifth level is concerned, some preliminary proposals (e.g. 565148 Kbps) have been submitted which were not discussed in detail.

Therefore, the present structure of this digital hierarchy is as given in Figure below.

Fig. 2.21 Encoded TDM (European)


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Most of the administrations favor the specification of a third level at 34368 Kbps. The same is mainly as a suitable flexibility point for the operation of the network and as an adequate bit rate for digital line systems which are to be set up either on new cables (screened symmetrical or micro-coaxial cables) or a radio-relay links. Other administrations do not consider the specification of a third level to be advantageous for their networks. On the contrary they regard it to be more economical to go directly from the second level at 8448 Kbps so the fourth level at 139264 Kbps, is also achieved by multiplexing four digital signals at 34368 Kbps, each of which is obtained by multiplexing first four digital signals at 8448 Kbps. However, this is a matter of internal multiplexing only, i.e. digital multiplex equipment of this type has no external input or output at 34368 Kbps.

All administrations interested in the third level at 34368 Kbps would thus be offered the possibility of using this level. Their digital multiplex equipment which multiplexes in the same way each of the four digital signals at 8448 Kbps has to provide external outputs for the resulting signal at 34368 Kbps. The digital multiplex equipment which multiplexes each of the four digital signals at 34368 Kbps has to provide four inputs for these bit rates and one output for the resulting bit rate of 139264 Kbps. 7.3 Higher order Multiplexing Second order Multiplexing The CCITT Recommendation G.742 deals with Second order digital multiplex equipment operating at 8448 Kbps and using positive justification. This 2nd order digital multiplex equipment using positive justification is intended for use on digital paths using 2048 Kbps primary multiplex equipments. Bit rates

The nominal bit rate should be 8448 Kbps. The tolerance on this rate should be +30 PPM.

Third order digital multiplexing

Third order digital multiplexing equipment operates at 34368

Kbps. The nominal bit rate should be 34368 Kbps. The tolerance on the

rate should be + 20 PPM

Fourth order digital multiplexing

The nominal bit rate should be 139264 kb/s. The tolerance on the

rate should be +15 PPM


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8. Multiple Access Methods The radio frequency spectrum, a finite natural resource, has

greater demands placed on it every day. In an effort to make the most efficient use of this resource, various technologies have been developed so that multiple, simultaneous users can be supported in a finite amount of spectrum. This concept is called "multiple access." The three most commonly used access methods are frequency division multiple access (FDMA), time division multiple access (TDMA), and code division multiple access (CDMA).

FDMA and TDMA are currently being used to support conventional and trunked radio systems, as well as commercial cellular systems. CDMA is being used primarily in cellular systems at this time. 8.1 FDMA, TDMA, and CDMA - a non technical example The best way to describe the differences between FDMA, TDMA, and CDMA technologies is with an example of how they work. The following example is one of the best. Picture a large room with a group of people divided up into pairs. Each pair would like to hold their own conversation with no interest in what is being said by the other pairs. For these conversations to take place without interruption from other conversations, it is necessary to define an isolated environment for each conversation. In this example, the room should be considered as a slice of the radio spectrum specifically allocated to be used by this group of people. Imagine each pair communicating through cellular telephones or radios. Applying an FDMA system to this analogy, the single large room (slice of spectrum) would be partitioned with many dividing walls and creating a large number of smaller rooms. A single pair of people would enter each small room and hold their conversation. Each room is like a single frequency/channel. No one else could use the room (or frequency) until the conversation was complete, whether or not the parties were actually talking. When the conversation is completed, the first pair of people would leave and another pair would then be able to enter that small room. In a TDMA environment, each of the small rooms would be able to accommodate multiple conversations “simultaneously.” For example, with a three-slot TDMA system, each “room” would contain up to three pairs of people, with the different pairs taking turns talking. According to this system, each pair can speak for 20 seconds during each minute. Pair A would use 0:01 second through 0:20 second, pair B would use 0:21 second through 0:40 second, and pair C would use 0:41 second through 0:60 second. However, even if there were fewer than three pairs in the small room, each pair would still be limited to 20 seconds per minute. Using the CDMA technology, all the little rooms would be eliminated. All pairs of people would enter the single large room (our spectrum space). Each pair would be holding their conversations in a


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different language and therefore they could use the air in the whole room to carry their voices while experiencing little interference from the other pairs. The air in the room is analogous to a wideband “carrier” and the languages represent the “codes” assigned by the CDMA system. In addition, language “filters” would be incorporated so that, for example, people speaking French would hear virtually nothing from those speaking another language. Additional pairs could be added, each speaking a unique language (as defined by the unique code) until the overall “background noise” (interference from other users) made it too difficult to hold a clear conversation. By controlling the voice volume (signal strength) of all users to a minimum, the number of conversations that could take place in the room could be maximized (i.e., maximize the number of users per carrier). Additional pairs can be easily added to the room without much interference to the other pairs. 8.2 FDMA - Frequency Division Multiple Access Frequency division is the original multiple access technique. Currently, most legacy public safety wireless networks use FDMA to improve spectrum efficiency. FDMA is used throughout the commercial wireless industry. Legacy commercial telecommunication networks (analog networks based on Advanced Mobile Phone Service [AMPS] and Total Access Communications System [TACS] standards) are built on a backbone of cellular base stations, using the FDMA technology. However, due to increased spectrum efficiency of CDMA and TDMA systems, very few, if any, new cellular systems are using FDMA. How it Works: FDMA systems separate a client's large frequency band into several smaller individual bands/channels. Each channel has the ability to support a user. Guard bands are used to separate channels to prevent interference. They are used to isolate channels from adjacent-channel interference.

Figure 2.22 FDMA

FDMA permits only one user per channel because it allows the user to use the channel 100 percent of the time. Therefore, only the frequency “dimension” is used to define channels. Each block represents a different user


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When the FDMA technique is employed, each user is assigned a discrete slice of the radio frequency (RF) spectrum, a “channel” of spectrum space that will vary in size depending on the type of signal being transmitted. In a given amount of spectrum space, the user is granted access to a small sliver of the overall allocation. As long as the user is engaged in “conversation,” no other user can access the same spectrum space. An example of this type of access is use of the spectrum by commercial radio broadcasters. In the commercial radio broadcast bands, 535–1705 kHz for amplitude modulation (AM) and 88–108 megahertz (MHz) for frequency modulation (FM), each local broadcast station (user) is assigned a specific slice of spectrum within the frequency band allocated for that purpose. As long as the station broadcasts, no other radio station in the same area can use that radio frequency bandwidth to send a signal. Another broadcast station can use that same bandwidth only when the distance between the stations is sufficient to reduce the risk of interference. In a conventional two-frequency public safety radio system, one frequency is used to transmit and the other is used to receive. Each channel has its own center frequency and each channel has a bandwidth that is a fraction of the original allotted bandwidth. In this type of system, if an FDMA channel is in use, other users cannot use it until the “conversation” is complete. This is one of the inefficiencies of FDMA systems. Figure 2 graphically displays a two-frequency conventional system. The mobile and portable radio users transmit on frequency F1 to the repeater; the repeater then retransmits back to the users on frequency F2. In Figure 2, the F1 lightning symbol is an uplink to the repeater while the F2 lightning symbol is a downlink.

Project 25's (P25) Phase I standard requires upgrades from standard analog technology with a 25 kHz bandwidth to digital technology with a narrower bandwidth of 12.5 kHz. Implementation of an FDMA system would give each user access to two separate frequency allotments, each with a 12.5 kHz bandwidth. Under P25, this newer equipment is also required to be “backward compatible” to the legacy 25 kHz analog equipment to allow a smooth transition.

Because adjacent channel interference is an important factor in channel quality, frequency planning is a key consideration when selecting fixed or base station locations. Frequency planning is complicated and difficult. Available frequency bands must be researched and analyzed. Transceiver transmission strength affects fixed station range while antenna design affects its coverage patterns. These are also important factors in frequency planning. Figure 3 is a sample base station coverage scheme for a cellular system.


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Advantages

• Simple to implement, from a hardware standpoint • Fairly efficient with a small base population and when traffic is

constant • Backward compatible to analog radio equipment

Disadvantages

• Network and spectrum planning are intensive • Poor spectrum efficiency, because channels are allocated for one

user. • Frequency planning is time-consuming

8.3 TDMA - Time Division Multiple Access As the frequency spectrum experiences more traffic, spectrum efficiency becomes increasingly important. TDMA systems were developed as FDMA system spectrum efficiency became insufficient. Not only do TDMA systems split users into an available pair of channels, but they also assign each user an available time-slot/cell within that channel. TDMA systems have the capability to split users into time slots because they transfer digital data, instead of analog data commonly used in legacy FDMA systems. Each of the users takes turns transmitting and receiving in a round-robin fashion. Frequency division is still employed, but these frequencies are now further subdivided into a defined number of time slots per frequency. In reality, only one user is (actually) using the channel at any given moment. Each user is transmitting and receiving in short “bursts.” Because TDMA systems do not transmit all of the time, their mobile phones have an extended battery life and talk time. How it works: Similar to an FDMA trunked system, when a user depresses the Push-To-Talk (PTT) switch in a TDMA system, a control channel registers the radio to the closest base station. During registration, the base station assigns the user an available pair of channels, one to transmit and the other to receive. But, unlike an FDMA system registration, a TDMA system registration also assigns an available time-slot within the channel. The user can only send or receive information at that time, regardless of the availability of other time-slots. Information flow is not continuous for any user, but rather is sent and received in bursts. The bursts are re-assembled at the receiving end and appear to provide continuous sound because the process is very fast. In Figure 2.23, each row of blocks represents a single channel divided into three time-slots. Calls in a TDMA system start in analog format and are sampled, transforming the call into a digital format. After the call is converted into digital format, the TDMA system places the call into an assigned time slot.


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Figure 2.23 TDMA TDMA increases the number of users who have access to particular channel by dividing that channel into time-slots. Figure 6 is also a graphical display of the efficiency of a TDMA system. The improved efficiency of TDMA over FDMA can be realized through a quick glance at Figures 2.22 and 2.23. In Figure 2.22, the FDMA system supports 4 users while in Figure 2.23, the TDMA system supports 12 users within the same bandwidth as the FDMA system. There are systems in place today that allow an increase of up to six times the capacity of FDMA alone. Because TDMA systems also split an allotted portion of the frequency spectrum into smaller slots (channels), they require the same level of frequency planning as FDMA systems. The same careful steps in frequency planning must be taken in both FDMA and TDMA systems. Advantages

• Extended battery life and talk time • More efficient use of spectrum, compared to FDMA • Will accommodate more users in the same spectrum space than an

FDMA system which improves capacity in high traffic areas, such as large metropolitan areas

• Efficient utilization of hierarchical cell structures – pico, micro, and macro cells

• Can handle video and audio data efficiently Disadvantages

• Network and spectrum planning are intensive • Multipath interference affects call quality • Dropped calls are possible when users switch in and out of

different cells • Frequency planning is time consuming • Frequency guard bands add to spectrum inefficiency • Too few users result in idle channels (rural versus urban

environment) • Higher costs due to greater equipment sophistication


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8.4 CDMA - Code Division Multiple Access

CDMA is a spread spectrum technique used to increase spectrum efficiency over current FDMA and TDMA systems. Although spread spectrum’s application to cellular telephony is relatively new, it is not a new technology. Spread spectrum has been used in many military applications, such as anti-jamming (because of the spread signal, it is difficult to interfere with or jam), ranging (measuring the distance of the transmission to determine when it will be received), and secure communications (the spread spectrum signal is very hard to detect).

How it works: With CDMA, unique digital codes (Walsh Codes), rather than separate radio frequencies/ channels, are used to differentiate users. The Walsh codes are shared by the mobile phone and the base station, and are called “pseudo-Random Code Sequences.” All users access the entire spectrum allocation all of the time. That is, every user uses the entire block of allocated spectrum space to carry his/her message. A user's unique Walsh Code separates the call from all other calls. Figure 2.24 graphically shows each user simultaneously accessing the fully allotted frequency spectrum.

Figure 2.24 CDMA CDMA allows all users access to their entire allocated spectrum. CDMA, being a “spread-spectrum” technology, spreads the information contained in a signal over the entire available bandwidth and not simply through one frequency. Due to the wide bandwidth of a spread-spectrum signal, it is very difficult to cause jamming, difficult to interfere with, and difficult to identify. It appears as nothing more than a slight rise in the “noise floor” or interference level, unlike other technologies where the power of the signal is concentrated in a narrower band making it easier to detect. Therefore CDMA systems provide more privacy than FDMA or TDMA systems. These are great advantages over technologies using a narrower bandwidth.

CDMA channels can handle an unspecified number of users. There is not a fixed number. The capacity of the system depends on the quality of current calls. As more users are added, noise is added to the wideband frequency and therefore decreases the quality of current calls. Each


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user's transmission power increases the level of the frequency spectrum's "noise floor" and therefore decreases the overall call quality for all users. To help eliminate the "noise floor," CDMA mobile phones and base stations use the minimum amount of power required to communicate with each other. They use precise power control to decrease users' transmission power. By decreasing a user's transmission power, the mobile phone has added battery life, increased talk time, and smaller batteries. Because CDMA is a spread spectrum technology, it requires less frequency planning. The full original spectrum is not divided into separate blocks/channels, like it is in FDMA and TDMA systems. Therefore, there is no need to plan for multiple frequency guard bands. Because all users have access to the entire spectrum at all times, frequency planning only needs to consider one frequency/channel. Advantages

• Greatest spectrum efficiency: capacity increases of 8 to 10 times that of an analog system and 4 to 5 times that of other digital systems which makes it most useful in high traffic areas with a large number of users and limited spectrum

• CDMA improves call quality by filtering out background noise, cross-talk, and interference

• ”Soft handoffs”— Because of the multiple diversities in use, handoffs between cells are undetected by the user

• Simplified frequency planning - all users on a CDMA system use the same radio frequency spectrum

• Detailed frequency plans are not necessary. • Frequency re-tunes for expansion are eliminated. • Fewer cells are required for quality coverage • Random Walsh codes enhance user privacy; a spread-spectrum

advantage • Precise power control increases talk time and battery size for

mobile phones Disadvantages

• Backwards compatibility techniques are costly • Currently, base station equipment is expensive • Difficult to optimize to maximize performance • Low traffic areas lead to inefficient use of spectrum and equipment

resources


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The following table shows the differences between each of the technologies. Each technology is rated based on their performance with respect to the ideal performance level.

Performance FDMA TDMA CDMA Capacity (Spectrum Efficiency)

Poor Medium Good

Security Poor Poor Good Ease of Network Planning Poor Poor Medium Ease of Implementation Very Good Very Good Medium Cost of Implementation Good Good Medium Backwards Compatibility Very Good Good Poor


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Module - 01


Chapter – 03

Transmission Media


• Various types of Transmission Media • Overhead Lines • Underground Copper Cables • Coaxial Cables • Optical Fiber Cables • Fiber Geometry • Construction of the OFC • Optical Fiber Cable Splicing


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1. Transmission Media Basically there are two ways in which information of any type can be transmitted over telecommunication media – analog or digital. Analog means that the amplitude of the transmitted amplitude signal varies over a continuous range. Digital transmission means that streams of on/off pulses are sent on the transmission media. The pulses are referred to as bits. Examples of analog signals are human voice, hi–fi music, temperature reading, etc. while those of digital are data, telegraphy signals. There are four types of media that can be used in transmitting information in the telecommunications world:

• Overhead Lines • Underground Copper Cables • Coaxial cable (actually an adaptation of copper wire) • Fiber • Wireless

In days of old, copper wire was the only means of transmitting information. Technically known as unshielded twisted pair (UTP), this consisted of a large number of pairs of copper wire of varying size in a cable. The cable did not have a shield and therefore the signal—primarily the high-frequency part of the signal—was able to leak out. Also, the twisting on the copper pair was very casual, designed as much to identify which wires belonged to a pair as to handle transmission problems. However, this is the way it was done, and for voice communications it was quite satisfactory. Consequently, there are millions of miles of copper in the PSTN—miles that must be used. 2. Overhead Lines All are familiar with kilometers of pole lines on city streets, rural highways and along railway tracks without knowing what the conductors of these lines are meant for. These conductors are the transmission lines consists of a pair of conductors across which voltage is applied for transmission to some distant point. Since transmission of voltage takes place from one end to another end over these conductors, it is called a transmission line. In telecommunications, the open wire transmission lines may transmit telegraph or telephone signals at milliwatt levels from direct current to alternating currents of up to some kilo Hertz.

The open wire lines may be used as: • Telephone subscriber lines • Telegraph lines • Long distance, trunks or carrier lines.


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3. Underground Copper Cables Cables are bunch of high conductivity annealed copper conductors of small diameter, well insulated from each other and compactly packed inside a sheath. The sheath is mechanically protected by providing steel tape, etc. The cables nomenclature identifies the size in number of pairs, gauge of the conductor, type of insulation, core filling material, armored or unarmoured, etc.

The Advantages of Underground cables over overhead lines are: 1. Compactness in construction. 2. More number of subscribers can be provided using small space. 3. Better and efficient performance. 4. Less constructional difficulties. 5. Less maintenance cost 6. Less fault liability

Junction Network The junction network comprises of the linkages between local

exchanges and between tandem exchange and local exchange. Provision of junctions, routing of traffic and transmission media for junction networks has to be carefully planned. It is very important to select proper transmission media and junction network.

In the majority of networks the most commonly used medium for junctions is underground cables. These suffer from several drawbacks like poor transmission quality due to high attenuation and poor reliability. With the advent of digital transmission systems PCM systems (2 Mb) are also in use in our networks. The transmission rate cannot be increased beyond 2 Mb due to poor X-Talk performance of cables. The data over voice technology enables combined voice and data operation on the existing unloaded subscriber cable loops. The data and speech are integrated at customer premises equipment and separated and segregated at exchange. The limitation is that the data calls through PSTN are limited to voice band modem speeds in two wire mode and require external modems. Subscriber Network The subscriber network consists of the circuits between the local exchange and subscribers up to customer premises include cables, cross connection points, and wires. The Local Exchange network diagram is given in Figure 3.1. As the over head line is not reliable cable is extended to the nearest locations in the area for serving the customer needs. It consists of Primary cable, Secondary cable and Distribution cables. These cables are terminated at cross connection points like cabinets, pillars and DPs of various capacities to suit different cable network systems.


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Fig. 3.1 External Plant Structure in a Local Telephone Exchange System

The subscriber network consists of the following characteristics. • One pair exclusively for each subscriber • Very low traffic occupancy and subs network difficult to replace. • Transmission quality is poor and accounts for large part of overall

transmission loss which directly shows the efficiency of the system. • Substantial part of overall investment goes in to the subscriber

network.

The factors driving towards replacement of present predominantly analog network by digital subs network are mainly : 1. Technological evolution, with Optical fibers, VLSI 2. Reduced costs due to OFC 3. Need for new services with a long term objective of ISDN lines. The services which can be provided with existing subscriber network based on copper cable pairs are Telephony, Telefax, and Teletext. With phased digitalisation the services that can be made available on copper wire are Telephony (7KHz), Audio conferencing (64Kb/s), Videotext, Image transmission, Computer communications and ISDN Classification of Underground Cables with regard to design features are: 1. Place where it is used - Underground / overhead / submarine 2. Insulation material used - Paper / polythene cables 3. The filling compound - Dry core / jelly filled cables 4. Mechanical protection - Armored / un-armored cables 5. Place of utilization - Primary / Distribution / Junction cable. 6. System for which used - Co-axial / PCM 7. Type of conductor - Copper cable / Optical fiber cable 8. Gauge of the conductor - 0.40 mm / 0.50 mm / 0.63 mm/0.90mm 9. Pressurization of core - Pressurized / un-pressurised cables


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Limitations copper cable in local loop 1. The local loop is the suspected link for most of the local faults. The old and worn-out cables have poor insulation which results in cross talk. 2. Water seepage during rainy season is the major cause of faults in copper cables. Repeated digging and opening of joints further increase the possibility of occurrence of faults. 3. The local loop is also vulnerable to electromagnetic interception and hence misuse. 4. The limitations on loop length and unpredictable growth of demand restrict the quick expansion capability and ease in network flexibility. 5. Large size copper cable calls for more space for leading in cables in MDF and congestion in duct. 6. Copper pairs are band width limited and are not able to support

new broad band services. 4. Coaxial Cables Coaxial cable consists of a single strand of copper running down the axis of the cable. This strand is separated from the outer shielding by an insulator made of foam or other dielectrics. A conductive shield covers the cable. Usually an outer insulating cover is applied to the overall cable—this has nothing to do with the carrying capacity of the cable. Because of the construction of the cable, obviously coaxial in nature, very high frequencies can be carried without leaking out. In fact, dozens of TV channels, each 6 MHz wide, can be carried on a single cable. The coaxial cable consists of an inner solid cylindrical conductor placed along the axis of an outer hollow cylindrical conductor. A coaxial cable may consist of two or more cores layered up with suitable lay with proper insulation along with quads laid in the interstices between them all enclosed in a lead sheath. The cable is recognized with:

• Number of cores i.e. either 2 core or 4 core • Size of the inner diameter of tube - r i.e. 0.375 types (large

tube – 0.375”) or 0.174 types (small tube – 0.174”). The interstice Quads or pairs having diameter of 0.9mm Fig: 3.2 Two Core 375 coaxial cable


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5. Optical Fiber Cables

Fiber-optic communications is based on the principle that light in a glass medium can carry more information over longer distances than electrical signals can carry in a copper or coaxial medium. The purity of today's glass fiber, combined with improved system electronics, enables fiber to transmit digitized light signals well beyond 100 km without amplification. With few transmission losses, low interference, and high bandwidth potential, optical fiber is an almost ideal transmission medium.

Fiber is the third transmission media, and it is unquestionably the transmission medium of choice. Whereas transmission over copper utilizes frequencies in the megahertz range, transmission over fiber utilizes frequencies a million times higher. This is another way of saying that the predominant difference between electromagnetic waves and light waves is the frequency. This difference, in turn, permits transmission speeds of immense magnitudes. Transmission speeds of as high as 9.9 Gbps have become commonplace in the industry today. At this speed, the entire fifteen-volume set of Encyclopedia Britannica can be transmitted in well under one second.

In order to over come the limitations of copper cable and to be able to support value added broad band service like data cable video HDTV and increased use of computer which requires band width on demand, optical fiber is introduced in the local loop . In the light of infinite bandwidth and the reliability, optical fiber is the automatic choice in the local loop Laying fiber, on a per-mile basis, still costs somewhat more than laying copper. However, on a per-circuit basis there is no contest; fiber wins hands down. However, if a local loop is being laid to a residence, there is little justification to installing fiber—there will never be a need for more than one or two or three circuits. This realization has led to a transition in our thinking. Shortly after the commercialization of fiber, we talked about fiber-to-the-home (FTTH). It was then realized that there was little need to install fiber for a final several hundred yards, so the industry shied away from fiber-to-the-curb (FTTC). In such a system, fiber would carry a plurality of channels to the "curb," whereupon they would be broken down and applied to the copper drop leading to the home. In many cases even this was overkill, and fiber-to-the-neighborhood (FTTN) is now being used. The message is clear: apply fiber when it is economical to do so and otherwise rely on copper.


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6. Fiber Geometry An Optical fiber consists of a core of optically transparent material usually silica or borosilicate glass surrounded by a cladding of the same material but a slightly lower refractive index. Fiber themselves have exceedingly small diameters. Figure shows cross section of the core and cladding diameters of commonly used fibers. The diameters of the core and cladding are as follows.

Core (µµµµm) Cladding (µµµµm)

8 125 50 125

62.5 125 100 140

Fig. 3.3 Typical Core and Cladding Diameter Fiber sizes are usually expressed by first giving the core size followed by the cladding size. Thus, 50/125 means a core diameter of 50 µm and a cladding diameter of 125 µm. Fiber Types The refractive index profile describes the relation between the indices of the core and cladding. Two main relationships exist: Step Index Graded Index The step index fiber has a core with uniform index throughout. The profile shows a sharp step at the junction of the core and cladding. In contrast, the graded index has a non–uniform core. The index is highest at the center and gradually decreases until it matches with that of the cladding. There is no sharp break in indices between the core and the cladding. By this classification, there are three types of fibers:

• Multimode Step Index Fiber (Step Index Fiber). • Multimode Graded Index Fiber (Graded Index Fiber). • Single–mode Step Index Fiber (Single Mode Fiber).


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Figure 3.4(a) OF Cable construction

Figure 3.4 (b) Single-Mode and Multimode Fibers

Figure 3.4 (c) Single-Mode and Multimode Fibers – Cross section


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Step Index Multimode Fiber This fiber is called "Step Index" because the refractive index changes abruptly from cladding to core. The cladding has a refractive index somewhat lower than the refractive index of the core glass. As a result, all rays within a certain angle will be totally reflected at the core–cladding boundary. Rays striking the boundary at angles greater than the critical angle will be partially reflected and partially transmitted out through the boundary. After many such bounces the energy in these rays will be lost from the fiber. The paths along which the rays (modes) of this step index fiber travel differ, depending on their angles relative to the axis. As a result, the different modes in a pulse will arrive at the far end of the fiber at different times, resulting in pulse spreading which limits the bit–rate of a digital signal which can be transmitted. This type of fiber results in considerable model dispersion, which affects the fiber's bandwidth.

Fig. 3.5 Step Index fiber

Graded Index Multi–mode Fiber This fiber is called graded index because there are many changes in the refractive index with larger values towards the center. As light travels faster in a lower index of refraction, so, the farther the light is from the center axis, the greater is its speed. Each layer of the core refracts the light. Instead of being sharply reflected as it is in a step index fiber, the light is now bent or continuously refracted in an almost sinusoidal pattern. Those rays that follow the longest path by traveling near the outside of the core have a faster average velocity. The light traveling near the center of the core has the slowest average velocity. As a result all rays tend to reach the end of the fiber at the same time. That causes the end travel time of different rays to be nearly equal, even though they travel different paths.


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The graded index reduces model dispersing to 1 ns/km or less. Graded Index fibers have core diameter of 50, 62.5 or 85 µm and a cladding diameter of 125 µm. The fiber is used in applications requiring a wide bandwidth and low model dispersion. The number of modes in the fiber is about half that of step index fiber having the same diameter and numerical aperture.

Fig. 3.6 Graded Index fiber Singe Mode Fiber Another way to reduce model dispersion is to reduce the core's diameter, until the fiber only propagates one mode efficiently. The single mode fiber has an exceedingly small core diameter of only 5 to 10 µm. Standard cladding diameter is 125 µm. Since this fiber carries only one mode, model dispersion does not exists. Single mode fibers easily have a potential bandwidth of 50 to 100 GHz–km. The core diameter is so small that the splicing technique and measuring techniques are more difficult. Light source must have a very narrow spectral width and they must be very small and bright in order to permit efficient coupling into the very small core diameter of these fibers. One advantage of single mode fiber is that once they are installed, the system's capacity can be increased as newer, higher capacity transmission system becomes available. This capability saves the high cost of installing a new transmission medium to obtain increased performance and allows cost effective increases from low capacity system to higher capacity system.


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Fig . 3.7 Optical Fibers – Principle and Types As the wavelength is increased, the fiber carries fewer and fewer modes until only one remains. Single mode operation begins when the wavelength approaches the core diameter. At 1300 nm, the fiber permits only one mode; it becomes a single mode fiber. As optical energy in a single mode fiber travels in the cladding as well as in the core, therefore, the cladding must be a more efficient carrier of energy. In a multimode fiber cladding modes are not desirable; a cladding with inefficient transmission characteristic can be tolerated. The diameter of the light appearing at the end of the single mode fiber is larger than the core diameter, because some of the optical energy of the mode travels in the cladding. Mode field diameter is the term used to define this diameter of optical energy. Optical Fiber Parameters Optical fiber systems have the following parameters:

• Wavelength • Frequency • Window • Attenuation • Dispersion • Bandwidth

Fiber Optic – Cable being used in India The cable being used for long distance route has been supplied by M/s SIECOR, U.S.A., a joint venture of Siemens and Corning glass works. The cable is non metallic with 12 fibers single mode, to be operated at 1300 nm. In fact, DOT has standardised single mode fibers at 1300 nm for all long distance routes and for junction working.


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7. Construction of the OFC An optical fiber consists of a core of optically transparent material usually doped silica or borosilicate glass surrounded by a cladding of the same material but of a slightly lower refractive index. Nominal diameter of core is 9 µm and of cladding is 125 µm in Siecor optical fiber cable single mode. Material used for primary coating is Dzg resin and for secondary coating is UV curable resin. OF Cable Construction Cabling is an outer protective structure surrounding one or more fibers. Cabling protect fibers environmentally and mechanically from being damaged or degraded in performance. SIECOR Fiber Optic Cables have the following parts:

• Optical Fiber • Buffer tube • Strength member • Jacket

The cable buffer tube is one of two types, namely Loose buffer or Tight buffer. The loose buffer uses a hard plastic tube having an inside diameter several times that of the fiber. One or more fibers lie within the buffer tube. As the cable expands and shrinks with temperature changes, it does not affect the fiber as much; the fiber in the tubes is slightly longer than the tube itself. SIECOR cable is loose buffer tube cable.

Fig. 3.8 Cross Section of Optical Fiber (Single Mode)


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Fiber Optic Construction Specifications

Primary coating – 250 + 15 micron of UV cured acryl ate.

Secondary coating – Nominal 2.4 mm loose dual layered buffer tube filled with paraffin based gel.

Central member – Solid GRP non–metallic in the center of the cable core : Outer dia. = 2.1 + 0.15 mm.

Core – Loose buffer tubes and fillers stranded around the central member.

Buffer tube filling compound

– Paraffin gel inside tube

Interstitial filling compound

– Petroleum gel filled areas between the loose buffer tubes.

Filler – Nominal 2.4 mm natural coloured solid P.E. filler

Core Wrap – Continuous layer of non hygroscopic dielectric material applied longitudinally

Inner P.E. Sheath – Not less than 2.0 mm thick P.E. inner sheath (Black colour).

Outer Nylon sheath – Not less than 0.7 mm thick Nylon sheath (Orange colour).


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Module - 01


Chapter – 04

Optical Fiber Communication


• Introduction • Fiber Optics Characteristics • Fiber types • Splicing • Bend Radius & Tensile Loading • Fiber optic Communication System


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1. Introduction An Optical Fiber Transmission System uses light waves as carrier of the information signals. These light waves are transmitted from one place to another place through an optical fiber using the principle of Total Internal Reflection. Over an Optical Fiber System, large bulk of information can be sent with very low distortion. 2. Fiber-Optic Characteristics Optical-fiber systems have many advantages over metallic-based communication systems. These advantages include interference, attenuation, and bandwidth characteristics. Furthermore, the relatively smaller cross section of fiber-optic cables allows room for substantial growth of the capacity in existing conduits. Fiber-optic characteristics can be classified as linear and nonlinear. Nonlinear characteristics are influenced by parameters, such as bit rates, channel spacing, and power levels. Wavelength It is a characteristic of light that is emitted from the light source and is measured in nanometers (nm). In the visible spectrum, wavelength can be described as the colour of the light. For example, Red light has a longer wavelength than Blue light. Typical wavelengths for fiber optic are 850nm, 1300nm and 1550nm, all of which are invisible. Frequency It is the number of pulse per second emitted from a light source. Frequency is measured in units of hertz (Hz). It terms of optical pulse 1 Hz = 1 pulse/sec. Window A narrow window is defined as the range of wavelengths at which a fiber best operates. Typical windows are given below:

Window Operational Wavelength

800 nm–900 nm 850 nm 1250 nm – 1305 nm 1300 nm 1500 nm – 1600 nm 1550 nm

Interference Light signals traveling via a fiber-optic cable are immune from electromagnetic interference (EMI) and radio-frequency interference (RFI). Lightning and high-voltage interference is also eliminated. A fiber network is best for conditions in which EMI or RFI interference is heavy


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or safe operation free from sparks and static is a must. This desirable property of fiber-optic cable makes it the medium of choice in industrial and biomedical networks. It is also possible to place fiber cable into natural-gas pipelines and use the pipelines as the conduit. Linear Characteristics Linear characteristics include attenuation, chromatic dispersion (CD), polarization mode dispersion (PMD), and optical signal-to-noise ratio (OSNR). Attenuation Several factors can cause attenuation, but it is generally categorized as either intrinsic or extrinsic. Intrinsic attenuation is caused by substances inherently present in the fiber, whereas extrinsic attenuation is caused by external forces such as bending. The attenuation coefficient α is expressed in decibels per kilometer and represents the loss in decibels per kilometer of fiber. Intrinsic Attenuation Intrinsic attenuation results from materials inherent to the fiber. It is caused by impurities in the glass during the manufacturing process. As precise as manufacturing is, there is no way to eliminate all impurities. When a light signal hits an impurity in the fiber, one of two things occurs: It scatters or it is absorbed. Intrinsic loss can be further characterized by two components:

• Material absorption • Rayleigh scattering

Material Absorption Material absorption occurs as a result of the imperfection and impurities in the fiber. The most common impurity is the hydroxyl (OH-) molecule, which remains as a residue despite stringent manufacturing techniques. Figure shows the variation of attenuation with wavelength measured over a group of fiber-optic cable material types. The three principal windows of operation include the 850-nm, 1310-nm, and 1550-nm wavelength bands. These correspond to wavelength regions in which attenuation is low and matched to the capability of a transmitter to generate light efficiently and a receiver to carry out detection.


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Fig 4.1 : Attenuation versus Wavelength The OH- symbols indicate that at the 950-nm, 1380-nm, and 2730-nm wavelengths, the presence of hydroxyl radicals in the cable material causes an increase in attenuation. These radicals result from the presence of water remnants that enter the fiber-optic cable material through either a chemical reaction in the manufacturing process or as humidity in the environment. The variation of attenuation with wavelength due to the water peak for standard, single-mode fiber optic cable occurs mainly around 1380 nm. Recent advances in manufacturing have overcome the 1380-nm water peak and have resulted in zero-water-peak fiber (ZWPF). Examples of these fibers include SMF-28e from Corning and the Furukawa-Lucent OFS AllWave. Absorption accounts for three percent to five percent of fiber attenuation. This phenomenon causes a light signal to be absorbed by natural impurities in the glass and converted to vibration energy or some other form of energy such as heat. Unlike scattering, absorption can be limited by controlling the amount of impurities during the manufacturing process. Because most fiber is extremely pure, the fiber does not heat up because of absorption. Rayleigh scattering As light travels in the core; it interacts with the silica molecules in the core. Rayleigh scattering is the result of these elastic collisions between the light wave and the silica molecules in the fiber. Rayleigh scattering accounts for about 96 percent of attenuation in


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optical fiber. If the scattered light maintains an angle that supports forward travel within the core, no attenuation occurs. If the light is scattered at an angle that does not support continued forward travel, however, the light is diverted out of the core and attenuation occurs. Depending on the incident angle, some portion of the light propagates forward and the other part deviates out of the propagation path and escapes from the fiber core. Some scattered light is reflected back toward the light source. This is a property that is used in an optical time domain reflectometer (OTDR) to test fibers. The same principle applies to analyzing loss associated with localized events in the fiber, such as splices. Short wavelengths are scattered more than longer wavelengths. Any wavelength that is below 800 nm is unusable for optical communication because attenuation due to Rayleigh scattering is high. At the same time, propagation above 1700 nm is not possible due to high losses resulting from infrared absorption. Extrinsic Attenuation Extrinsic attenuation can be caused by two external mechanisms: macro bending or micro bending. Both cause a reduction of optical power. If a bend is imposed on an optical fiber, strain is placed on the fiber along the region that is bent. The bending strain affects the refractive index and the critical angle of the light ray in that specific area. As a result, light traveling in the core can refract out, and loss occurs. A macro bend is a large-scale bend that is visible, and the loss is generally reversible after bends are corrected. To prevent macro bends, all optical fiber has a minimum bend radius specification that should not be exceeded. This is a restriction on how much bend a fiber can withstand before experiencing problems in optical performance or mechanical reliability. The second extrinsic cause of attenuation is a micro bend. Micro bending is caused by imperfections in the cylindrical geometry of fiber during the manufacturing process. Micro bending might be related to temperature, tensile stress, or crushing force. Like macro bending, micro bending causes a reduction of optical power in the glass. Micro bending is much localized, and the bend might not be clearly visible on inspection. With bare fiber, micro bending can be reversible. Chromatic Dispersion Chromatic dispersion is the spreading of a light pulse as it travels down a fiber. Light has a dual nature and can be considered from an electromagnetic wave as well as quantum perspective. This enables us to quantify it as waves as well as quantum particles. During the propagation of light, all of its spectral components propagate accordingly. These spectral components travel at different group velocities that lead to dispersion called group velocity dispersion (GVD). Dispersion resulting


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from GVD is termed chromatic dispersion due to its wavelength dependence. The effect of chromatic dispersion is pulse spread. Polarization Mode Dispersion Polarization mode dispersion (PMD) is caused by asymmetric distortions to the fiber from a perfect cylindrical geometry. The fiber is not truly a cylindrical waveguide, but it can be best described as an imperfect cylinder with physical dimensions that are not perfectly constant. The mechanical stress exerted upon the fiber due to extrinsically induced bends and stresses caused during cabling, deployment, and splicing as well as the imperfections resulting from the manufacturing process are the reasons for the variations in the cylindrical geometry. PMD is not an issue at low bit rates but becomes an issue at bit rates in excess of 5 Gbps.

Fig 4.2: Polarization Mode Dispersion Polarization Dependent Loss Polarization dependent loss (PDL) refers to the difference in the maximum and minimum variation in transmission or insertion loss of an optical device over all states of polarization (SOP) and is expressed in decibels. A typical PDL for a simple optical connector is less than .05 dB and varies from component to component. Optical Signal-to-Noise Ratio The optical signal-to-noise ratio (OSNR) specifies the ratio of the net signal power to the net noise power and thus identifies the quality of the signal. Attenuation can be compensated for by amplifying the optical signal. However, optical amplifiers amplify the signal as well as the noise. Over time and distance, the receivers cannot distinguish the signal from the noise, and the signal is completely lost. Regeneration helps mitigate these undesirable effects before they can render the system unusable and ensures that the signal can be detected at the receiver. Optical amplifiers add a certain amount of noise to the channel. Active devices, such as lasers, also add noise. Passive devices, such as taps and the fiber, can also add noise components. In the calculation of


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system design, however, optical amplifier noise is considered the predominant source for OSNR penalty and degradation. OSNR is measured in decibels. The higher the bit rate, the higher the OSNR ratio required. 3. Fiber Types The MMF and SMF types currently used for premise, metro, aerial, submarine, and long-haul applications. The International Telecommunication Union (ITU-T), which is a global standardization body for telecommunication systems and vendors, has standardized various fiber types. These include the 50/125-µm graded index fiber (G.651), Non dispersion-shifted fiber (G.652), dispersion-shifted fiber (G.653), 1550-nm loss-minimized fiber (G.654), and NZDSF (G.655). Multimode Fiber with a 50-Micron Core (ITU-T G.651) The ITU-T G.651 is an MMF with a 50-µm nominal core diameter and a 125-µm nominal cladding diameter with a graded refractive index. The attenuation parameter for G.651 fiber is typically 0.8 dB/km at 1310 nm. The main application for ITU-T G.651 fiber is for short-reach optical transmission systems. This fiber is optimized for use in the 1300-nm band. It can also operate in the 850-nm band. Non dispersion-Shifted Fiber (ITU-T G.652) The ITU-T G.652 fiber is also known as standard SMF and is the most commonly deployed fiber. This fiber has a simple step-index structure and is optimized for operation in the 1310-nm band. It has a zero-dispersion wavelength at 1310 nm and can also operate in the 1550-nm band, but it is not optimized for this region. The typical chromatic dispersion at 1550 nm is high at 17 ps/nm-km. Dispersion compensation must be employed for high-bit-rate applications. The attenuation parameter for G.652 fiber is typically 0.2 dB/km at 1550 nm, and the PMD parameter is less than 0.1 ps/√ km. An example of this type of fiber is Corning SMF-28. Low Water Peak Non dispersion-Shifted Fiber (ITU-T G.652.C) The legacy ITU-T G.652 standard SMFs are not optimized for WDM applications due to the high attenuation around the water peak region. ITU G.652.C-compliant fibers offer extremely low attenuation around the OH peaks. The G.652.C fiber is optimized for networks where transmission occurs across a broad range of wavelengths from 1285 nm to 1625 nm. Although G.652.C-compliant fibers offer excellent capabilities for shorter, unamplified metro and access networks, they do not fully address the needs for 1550-nm transmission. The attenuation parameter for G.652 fiber is typically 0.2 dB/km at 1550 nm, and the PMD parameter is less than 0.1 ps/√ km. An example of this type of fiber is Corning SMF-28e.


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Dispersion-Shifter Fiber (ITU-T G.653) Conventional SMF has a zero-dispersion wavelength that falls near the 1310-nm window band. SMF shows high dispersion values over the range between 1500 nm and 1600 nm (third window band). The trend of shifting the operating transmission wavelength from 1310 nm to 1550 nm initiated the development of a fiber type called dispersion-shifted fiber (DSF). DSF exhibits a zero-dispersion value around the 1550-nm wavelength where the attenuation is minimum. The DSFs are optimized for operating in the region between 1500 to 1600 nm. With the introduction of WDM systems, however, channels allocated near 1550 nm in DSF are seriously affected by noise induced as a result of nonlinear effects caused by FWM. This initiated the development of NZDSF. G.53 fiber is rarely deployed any more and has been superseded by G.655. 4. Splicing Fiber-optic cables might have to be spliced together for a number of reasons—for example, to realize a link of a particular length. Another reason might involve backhoe fade, in which case a fiber-optic cable might have been ripped apart due to trenching work. The network installer might have in his inventory several fiber-optic cables, but none long enough to satisfy the required link length. Situations such as this often arise because cable manufacturers offer cables in limited lengths—usually 1 to 6 km. A link of 10 km can be installed by splicing several fiber optic cables together. The installer can then satisfy the distance requirement and avoid buying a new fiber-optic cable. Splices might be required at building entrances, wiring closets, couplers, and literally any intermediate point between a transmitter and receiver. Connecting two fiber-optic cables requires precise alignment of the mated fiber cores or spots in a single-mode fiber-optic cable. This is required so that nearly all the light is coupled from one fiber-optic cable across a junction to the other fiber-optic cable. Actual contact between the fiber-optic cables is not even mandatory. The splices offer sophisticated, computer-controlled alignment of fiber-optic cables to achieve losses as low as 0.02 dB. Typical fusion-splice losses can be estimated at 0.02 dB for loss-budget calculation purposes. Mechanical splices are easily implemented in the field, require little or no tooling, and offer losses of about 0.5 to 0.75 dB. 5. Bend Radius and Tensile Loading An important consideration in fiber-optic cable installation is the cable’s minimum bend radius. Bending the cable farther than its minimum bend radius might result in increased attenuation or even broken fibers. Cable manufacturers specify the minimum bend radius for


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cables under tension and long-term installation. The ANSI TIA/EIA-568B.3 standard specifies a bend radius of 1.0 inch under no pull load and 2.0 inches when subject to tensile loading up to the rated limit. Cable tensile load ratings, also called cable pulling tensions or pulling forces, are specified under short-term and long-term conditions. The short-term condition represents a cable during installation and it is not recommended that this tension be exceeded. The long-term condition represents an installed cable subjected to a permanent load for the life of the cable. Typical loose-tube cable designs have a short-term (during installation) tensile rating of 600 pounds (2700 N) and a long-term (post installation) tensile rating of 200 pounds (890 N). 6. Fiber-Optic Communications System As shown Figure below, information (voice, data, and video) from the source is encoded into electrical signals that can drive the transmitter. The fiber acts as an optical waveguide for the photons as they travel down the optical path toward the receiver. At the detector, the signals undergo an optical-to-electrical (OE) conversion, are decoded, and are sent to their destination.

Fig 4.3 : Fiber Optic Communication System


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Transmitter The transmitter component of Figure 4.3 serves two functions. First, it must be a source of the light launched into the fiber-optic cable. Second, it must modulate this light to represent the binary data that it receives from the source. A transmitter’s physical dimensions must be compatible with the size of the fiber-optic cable being used. This means that the transmitter must emit light in a cone with a cross-sectional diameter of 8 to 100 microns; otherwise, it cannot be coupled into the fiber-optic cable. The optical source must be able to generate enough optical power so that the desired BER can be met over the optical path. There should be high efficiency in coupling the light generated by the optical source into the fiber-optic cable, and the optical source should have sufficient linearity to prevent the generation of harmonics and inter-modulation distortion. If such interference is generated, it is extremely difficult to remove. This would cancel the interference resistance benefits of the fiber-optic cable. The optical source must be easily modulated with an electrical signal and must be capable of high-speed modulation; otherwise, the bandwidth benefits of the fiber-optic cable are lost. Finally, there are the usual requirements of small size, low weight, low cost, and high reliability. The transmitter is typically pulsed at the incoming frequency and performs a transducer electrical-to-optical (EO) conversion. Light-emitting diodes (LEDs) or vertical cavity surface emitting lasers (VCSELs) are used to drive MMF systems, whereas laser diodes are used to drive SMF systems. Two types of light-emitting junction diodes can be used as the optical source of the transmitter. These are the LED and the laser diode (LD). LEDs are simpler and generate incoherent, lower-power light. LEDs are used to drive MMF. LDs generate coherent, higher-power light and are used to drive SMF. Figure 4.4 below shows the optical power output, P, from each of these devices as a function of the electrical current input, I, from the modulation circuitry. As the figure indicates, the LED has a relatively linear P-I characteristic, whereas the LD has a strong nonlinearity or threshold effect. The LD can also be prone to kinks when the power actually decreases with increasing input current. LDs have advantages over LEDs in the sense that they can be modulated at very high speeds, produce greater optical power, and produce an output beam with much less spatial width than an LED. This gives LDs higher coupling efficiency to the fiber-optic cable. LED advantages include a higher reliability, better linearity, and lower cost.


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Fig 4.4 : LED and LD- PI Characteristics A key difference between the optical output of an LED and a LD is the wavelength spread over which the optical power is distributed. The spectral width, σ, is the 3-dB optical power width (measured in nanometers or microns). The spectral width impacts the effective transmitted signal bandwidth. A larger spectral width takes up a larger portion of the fiber-optic cable link bandwidth. Figure 4.5 shows the spectral width of the two devices. The optical power generated by each device is the area under the curve. The spectral width is the half-power spread. An LD always has a smaller spectral width than an LED. The specific value of the spectral width depends on the details of the diode structure and the semiconductor material. However, typical values for an LED are around 40 nm for operation at 850 nm and 80 nm at 1310 nm. Typical values for an LD are 1 nm for operation at 850 nm and 3 nm at 1310 nm.


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Fig 4.5: LED and LD Spectral Width Other transmitter parameters include packaging, environmental sensitivity of device characteristics, heat sinking, and reliability. With either an LED or LD, the transmitter package must have a transparent window to transmit light into the fiber-optic cable. It can be packaged with either a fiber-optic cable pigtail or with a transparent plastic or glass window. Some vendors supply the transmitter with a package having a small hemispherical lens to help focus the light into the fiber-optic cable. Packaging must also address the thermal coupling for the LED or LD. A complete transmitter module can consume more than 1 watt, which could result in significant heat generation. Plastic packages can be used for lower-speed and lower reliability applications. However, high-speed and high-reliability transmitters need metal packaging with built-in fins for heat sinking. There are several different schemes for carrying out the modulation function. These include intensity modulation (IM), frequency shift keying (FSK), phase shift keying (PSK), and polarization modulation (PM). Within the context of a premise fiber-optic data link, the only one really used is IM. IM is used universally for premise fiber-optic data links because it is well matched to the operation of both LEDs and LDs. The carrier that each of these sources produces is easy to modulate with this technique. Passing current through them operates both of these devices. The amount of power that they radiate (sometimes referred to as the radiance) is proportional to this current. In this way, the optical power takes the shape of the input current. If the input current is the waveform m (t) representing the binary information stream, the resulting optical signal looks like bursts of optical signal when m (t) represents a 1 and


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the absence of optical signal when m (t) represents a 0. This is also known as direct modulation of the LED or LD. Receiver Figure 4.6 below shows a schematic of an optical receiver. The receiver serves two functions: It must sense or detect the light coupled out of the fiber-optic cable and convert the light into an electrical signal, and it must demodulate this light to determine the identity of the binary data that it represents. The receiver performs the OE transducer function. Fig 4.6 : Schematic of an Optical Receiver A receiver is generally designed with a transmitter. Both are modules within the same package. The light detection is carried out by a photodiode, which senses light and converts it into an electrical current. However, the optical signal from the fiber-optic cable and the resulting electrical current will have small amplitude. Consequently, the photodiode circuitry must be followed by one or more amplification stages. There might even be filters and equalizers to shape and improve the information-bearing electrical signal. The receiver schematic in Figure shows a photodiode, bias resistor circuit, and a low-noise pre-amp. The output of the pre-amp is an electrical waveform version of the original information from the source. To the right of this pre-amp are an additional amplification, filters, and equalizers. All of these components can be on a single integrated circuit, a hybrid, or discretely mounted on a printed circuit board. The receiver can incorporate a number of other functions, such as clock recovery for synchronous signaling, decoding circuitry, and error detection and recovery. The receiver must have high sensitivity so that it can detect low-level optical signals coming out of the fiber-optic cable. The higher the sensitivity, the more attenuated signals it can detect. It


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must have high bandwidth or a fast rise time so that it can respond fast enough and demodulate high-speed digital data. It must have low noise so that it does not significantly impact the BER of the link and counter the interference resistance of the fiber-optic cable transmission medium. There are two types of photodiode structures: positive intrinsic negative (PIN) and the avalanche photodiode (APD). In most premise applications, the PIN is the preferred element in the receiver. This is mainly due to fact that it can be operated from a standard power supply, typically between 5 and 15V. APD devices have much better sensitivity. In fact, APD devices have 5 to 10 dB more sensitivity. They also have twice the bandwidth. However, they cannot be used on a 5V printed circuit board. They also require a stable power supply, which increases their cost. APD devices are usually found in long-haul communication links and can increasingly be found in metro-regional networks (because APDs have decreased in cost). The demodulation performance of the receiver is characterized by the BER that it delivers to the user. The sensitivity curve indicates the minimum optical power that the receiver can detect compared to the data rate, to achieve a particular BER. The sensitivity curve varies from receiver to receiver. The sensitivity curve considers within it the SNR parameter that generally drives all communications-link performance. The sensitivity depends on the type of photodiode used and the wavelength of operation. 1. Optical Sources The Figure 4.7 shown below is a basic block diagram of an optical transmitter. It contains an electronic processing circuit and a light source. The input to the processing circuit is the signal from the carrier multiplexing equipment (for example 140 CMI signal). The output from the electronic processing circuit is the current required to operate the source.

Fig. 4.7 Basic Transmitter Block Diagram

The circuit of transmitter must accept standard analog or digital signal levels. Digital systems use either Transistor–transistor logic (TTL) or emitter–coupled logic (ECL) or complementary metal oxide semiconductor (CMOS) logic levels. The circuitry has to perform the following functions in a digital transmitter.


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• Separation of clock and data. • Retiming and reshaping of pulses. • Line coding–conversion to optical line code. • Scrambling or randomizing base band input signal. • Combining order wire, supervisory, protection switching signals

with the main signal. The two types of sources employed in fiber optics communication systems as carriers of information are the light emitting diode (LED) and laser diode (LD). Analogous to electronic oscillators in radio systems, these are the optic oscillators in fiber system. The source provides suitable power at the required wavelengths for long distance applications. The sources found suitable in communication applications are the semiconductor sources (LED and LD). 2. Optical Sources Requirements To meet the communication needs in optical fiber technology, the requirements have been summarized in Table 1 below:

Property Requirement Target

Primary Characteristic Wavelength

It must operate at a wavelength which gives low loss and low dispersion in fibers

850, 1300 and 1550 nm.

Reliability

Long life, good stability of operation and good reproducibility of output characteristics are necessary.

Life = 106 hrs.

Output power System demands must be met > 1 mW

Power efficiency It must operate with a low power and low voltage and heat generated must be small.

> 10%

Desirable Properties Spectral width (temporal coherence)

It must have a spectral width which enables the maximum bandwidth of the optical fiber to be realized

Target must be determined by the scale of the system.

Focusing effect (spatial coherence)

It should be possible to focus the output onto the fiber and to obtain high coupling efficiency.

Modulation Direct modulation must be possible or coupling to an external modulation made easy.

Size and weight It must be small and light in weight.

Cost Mass production and low cost.


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Like in many other cases, the practical sources fall short of the above requirements particularly in respect of stability and emission of single frequency. The efforts required to achieve stability are not as difficult as the generation of a single frequency output. Continuous development work is in progress to achieve the latter requirement. Light emitting diode optical sources find application in short haul and medium haul communication systems where the power requirement is small and bit rates are low. For long distance, high bit rate applications, Laser diode is the answer. The two types of sources are discussed below. The optic beams generated by these two light sources carry the information through the process of intensity modulation. The operating principles, transfer characteristics, modulation and stabilization of output pertaining to the two types (LD and LED) of semiconductor optical sources are considered. 3. Light Emitting Diodes Operating Principle: An LED is a semiconductor p–n junction diode and is forward biased. It emits light under this condition. When the p and n types of semiconductor materials are brought together, the resulting energy barrier under zero external applied voltage prevents the movement of the n and p charge carriers. When the barrier energy level is lowered by the application of an external voltage (in forward direction), the movement of electrons and holes have taken place and some of the charge carriers recombine in the transition region. The energy lost in the transition is converted to optical energy giving rise to photon. The diode is modulated by a current source that turns the LED on or off as shown below. Unlike analog modulators, digital drivers need not provide d–c bias current. In the off state, the emission of the LED should be low.

Fig. 4.8 Series–switched and parallel–switched digital modulators


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Long Term Stability Lifetime of the LED is about 106 hours and large temperature variations are tolerated but at the cost of output power. The decline in output power is approximately 1% per degree centigrade. Different types of packages are available. The pigtail construction and micro lens type are a few. 4. Laser Diode A laser has an optical cavity formed by two parallel mirrors at the end facets of the semiconductor crystal as in Fig. 4.9. For most applications, it is sufficient to use the natural cleavage facets of the crystal as mirrors. To increase the reflectivity, the mirrors can be coated with a metallic film.

Fig. 4.9 Resonant Cavity

Operating Principles A few characteristics of lasers are: 1) Pumping threshold: The input power must be above a threshold before emission of light takes place. This is not required in LED. 2) Radiation pattern: The angle of radiation depends on the size of the emitting area and on the modes of oscillation within the laser. 3) Output spectrum: The output power is spread over a range of frequencies and power does not vary smoothly over this range but has peaks and valleys. The number of such peaks depends on spectral width, refractive index of material and dimension of cavity. Operating Characteristics The output power–forward input current characteristics are shown in Fig. 4.10. The threshold current is approximately 75 mA and the


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voltage at threshold is 1 to 2 volts. Operating current is generally about 20 to 40 mA above the threshold current. A short kink is observed near the threshold. Digital modulation of laser diode differs from the modulation used in LED, in that a d–c bias current is added.

Fig. 4.10 Operating Characteristics Temperature stability Laser diode is more temperature sensitive than LEDs. At high temperatures, threshold current increases. The threshold increases 1.5% per degree increase in temperature. If the current remains constant, the output of the laser diode decreases. Output Spectrum When the drive current is near threshold, lasers produce multimode spectra. As the current increases, total line width decreases and number of longitudinal modes decreases. At sufficiently high currents, the spectrum contains just one mode. The light from laser beam is confined to a narrow angular region.


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5. Comparison of LED and LASER Diode (Table 2)

Property LED Laser Diode

Single Mode

Laser Diode

Spectral Width (nm) 20–100 1–5 <0.2

Rise Time (ns) 2–250 0.1–1 0.05–1

Modulation Bandwidth (MHz)

< 300 < 2000 ~ 2000

Coupling Frequency Very low Moderate High

Compatible Fiber MM (SI&GI)

MMGI, SM Single Mode

Temperature Sensitivity Low High High

Circuit Complexity Simple Complex Complex

Lifetime (Hours) 105 104–105 104–105

Costs Low High Highest

Path Length Moderate Long Very Long

Data Rate Moderate High Very High


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Module - 01


Chapter – 05

Optical Fiber Cable Splicing


• Definition • Splicing Methods • Example of Mechanical Splicing • Preparations of Splicing Closure • Fusion Splicer (X 76) • Fusion Splicing methods


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1. Definition Splices are permanent connection between two fibers. The splicing involves cutting of the edges of the two fibers to be spliced. There are several reasons for splicing a fiber cable, these include:

• To join two fibers due to a breakage. • To connect some of the cores straight through a patch cabinet. • To extend a cable run. • To reduce losses, a fusion splice has much lower losses than

two connectorized cables joined through a coupler. • To attach a pre-terminated pigtail.

A Pigtail is a short length of fiber with a factory fitted and polished connector. In the past these were used in preference to field terminations because of the complexities at the time of manually terminating optical fibers. These days pigtails are mainly used where the environment isn't suitable for manual terminations or where speed is a factor. 2. Splicing Methods Single–Fibre Mechanical Splicing

• Single Fibre Capillary • Aligns two fiber ends to a common centerline, thereby aligning

cores. • Clean, cleaved fibers are butted together and index matched. • Permanently secured with epoxy or adhesive.

Examples: Siecor, See Splice GTE Elastomeric Splice.

Fig. 5.1 Mechanical Splice

The following three types are widely used:

• Adhesive bonding or Glue splicing. • Mechanical splicing. • Fusion splicing.

Splice

Uncos

ted Cos

ted


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Adhesive Bonding or Glue Splicing This is the oldest splicing technique used in fiber splicing. After fiber end preparation, it is axially aligned in a precision V–groove. Cylindrical rods or another kind of reference surfaces are used for alignment. During the alignment of fiber end, a small amount of adhesive or glue of same refractive index as the core material is set between and around the fiber ends. A two component epoxy or an UV curable adhesive is used as the bonding agent. The splice loss of this type of joint is same or less than fusion splices. But fusion splicing technique is more reliable, so at present this technique is very rarely used. Mechanical Splicing This technique is mainly used for temporary splicing in case of emergency repairing. This method is also convenient to connect measuring instruments to bare fibers for taking various measurements. The mechanical splices consist of 4 basic components: (i) An alignment surface for mating fiber ends. (ii) A retainer (iii) An index matching material. (iv) A protective housing A very good mechanical splice for M.M. fibers can have an optical performance as good as fusion spliced fiber or glue spliced. But in case of single mode fiber, this type of splice cannot have stability of loss. Fusion Splicing The fusion splicing technique is the most popular technique used for achieving very low splice losses. The fusion can be achieved either through electrical arc or through gas flame. The process involves cutting of the fibers and fixing them in micro–positioners on the fusion splicing machine. The fibers are then aligned either manually or automatically core aligning (in case of S.M. fiber) process. Afterwards the operation that takes place involve withdrawal of the fibers to a specified distance, preheating of the fiber ends through electric arc and bringing together of the fiber ends in a position and splicing through high temperature fusion. If proper care taken and splicing is done strictly as per schedule, then the splicing loss can be minimized as low as 0.01 dB/joint. After fusion splicing, the splicing joint should be provided with a proper protector to have following protections:

• Mechanical protection • Protection from moisture.

Sometimes the two types of protection are combined. Coating with Epoxy resins protects against moisture and also provides mechanical strength at the joint.


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Now–a–days, the heat shrinkable tubes are most widely used, which are fixed on the joints by the fusion tools. The fusion splicing technique is the most popular technique used for achieving very low splice losses. The introduction of single mode optical fiber for use in long haul network brought with it fiber construction and cable design different from those of multimode fibers. The splicing machine begins to the core profile alignment system, the main functions of which are:

• Auto active alignment of the core. • Auto arc fusion. • Video display of the entire process. • Indication of the estimated splice loss.

The two fiber ends to be spliced are cleaved and then clamped in accurately machined vee–grooves. When the optimum alignment is achieved, the fibers are fused under the microprocessor control; the machine then measures the radial and angular off–sets of the fibers and uses these figures to calculate a splice loss. The operation of the machine observes the alignment and fusion processes on a video screen showing horizontal and vertical projection of the fibers and then decides the quality of the splice. The splice loss indicated by the splicing machine should not be taken as a final value as it is only an estimated loss and so after every splicing is over, the splice loss measurement is to be taken by an OTDR (Optical Time Domain Reflectometer). The manual part of the splicing is cleaning and cleaving the fibers. For cleaning the fibers, Dichlorine Methyl or Acetone or Alcohol is used to remove primary coating. With the special fiber cleaver or cutter, the cleaned fiber is cut. The cut has to be so precise that it produces an end angle of less than 0.5 degree on a prepared fiber. If the cut is bad, the splicing loss will increase or machine will not accept for splicing. The shape of the cut can be monitored on the video screen, some of the defect noted while cleaving are listed below: (i) Broken ends. (ii) Ripped ends. (iii) Slanting cuts. (iv) Unclean ends. It is desirable to limit the average splice loss to be less than 0.1 dB. 3. Example of Mechanical splicing The 3M brand Optical Fiber Splicing System provides permanent mechanical splices for single or multi mode fiber having 125 µm diameter cladding. Three color coded versions of the Fibrlok Splice are available for splicing 250 µm and 900 µm diameter plastic coated fiber.


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Fig. 5.2 Mechanical Splice (3M brand)

Fig. 5.3 Fiber assembly tool (3M brand)


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Splicing Set-Up

• The splicing area should be clean, dry and well-lighted. A clean, well organized splicing area will improve splicing efficiency and minimize the risk of contamination of fibers or splices.

• Open the buffer tubes, expose and clean the fibers as specified in practice.

• Load the splice into the assembly tool by pressing firmly at the ends of the splice.

• Remove the minimum length of fiber required to prepare and splice the fibers.

• Strip approximately 1 to 2 inches (2.5 to 5 cm) of plastic coating from the fiber using a mechanical stripper.

• Clean the bare glass by pulling the fiber through an alcohol soaked lint-free wipe. This will remove any fragments or dirt remaining on the fiber.

• Push the fiber down into the fiber retention pad on the proper side of the splice. Fiber should be inserted into the splice immediately following cleaning and placing in retention pad to minimize exposure to the atmosphere and reduce the risk of contamination.

• Prepare second fiber (strip, clean and cleave) as described for fiber 1.

• Lay fiber into foam retention pad and begin to insert the fiber end into the splice.


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• Pivot the handle of the Assembly Tool down until it contacts the cap of the Splice. Squeeze the handle of the assembly tool as shown in order to close

• cap and actuate the splice. When possible, secure the tool to a work surface for added support. A snap sound will be heard when the splice is actuated.

• Remove the Splice from the Assembly Tool by first removing the fibers from the foam retention pads and then lifting the splice from the splice holding cradle.

4. Preparation of Splicing Closure There are two types of splice closures – one to be installed in a pit (SC–4) and the other to be installed on wall (Wall Splicing Closure - WSC). The general steps related to splice closure are given below:

• Locate the splice pit. • Mark the splice number and route name on the joint indicator. • Open the splice pit. • Record the meter marks of the cable. • Coil the spare cable. • Prepare end caps including fixing of cable in end cap. • Assemble the closure.

The details of WSC and SC4 are given below: Wall Splice Closure (WSC) The front view of the WSC with its components is shown in Fig. Pigtail cables–tight–buffered fibers (with connectors on one end) that are spliced to the outside cable fibers.


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• Anchoring screw – used to secure the strength member to the strain relief bracket.

• Strain relief brackets – used as strain relief for multifibre cable. • Fastener locations – used with fasteners to secure the front

panel to the unit. • Cable ties – used to secure cables.

Fig. 5.4 Components of WSC

• Cable contains fibers from the outside environment which are spliced to pigtail cables or other multifibre cables.

• Knockouts – when removed provide location for cable routing. • Cable retaining guide – used to hold the cable in position. • Wire stacker – used to store splice trays. • Screw holes – locations used to mount strain relief brackets. • Splice tray – used to store splices. • Identification label – used to make an installation plan as well

as a record of splicing information; located on the front panel. All vital information about the installation should be recorded on this label.

SC–4 Type Closure The application of SC–4 closure is shown in Fig 5.5. The diagram shows the placement of SC4 closures and the cable racking arrangement. There is a long list of parts in SC4 due to its underground applications. Underground application needs the water proofing of the closure.


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Fig. 5.5 Application of SC-4 Type closure Loading of Splice Trays in SC4: The following steps are followed for field splicing:

• Remove 150 cms of buffer tubes. • Clean the fibers. • Load the buffer tubes in the tray • Place the fibers in the tray • Close the trays.

5. Fusion Splicer (X 76) As with all fiber termination methods, safety is very important so first some safety tips.

• Always work in a clean and tidy area. • Fiber off cuts are hard to see and can easily penetrate the skin

especially if they get into your clothes, so care must be taken to ensure the safe disposal of all off cuts. Dispose of fiber scraps immediately using a suitable container and do not throw into a waste paper bin. * Because of the dangers of ingesting a fiber, do not eat or drink in the termination area.

• Fusion splicers use an electric arc to fuse the fibers together so they should never be used in an environment where flammable gases or liquids are present.* Never look into the end of a live fiber connector. Holding some multimode fibers up to a piece of paper may prove the presence of light and therefore prove that it is live, but it doesn't prove that it isn't live! Some laser powered equipment use light which is outside of the visible spectrum, so err on the side of caution.

A fusion splice is a way of joining two fiber cores by melting the ends together using an electric arc. A splicing machine is used because


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an extremely high degree of accuracy is needed; the machine first has to align the cores and then apply the exact amount of heat to melt the ends before pressing them together. There are four basic steps to fusion splicing 1 - Strip back all coatings down to the bare fibers and clean using isopropyl alcohol. 2 - Cleave the fibers using a precision cleaving tool and put the heat shrink tube on to one of the ends. 3 - Fuse the fibers together in the fusion splicer. 4 - Put the heat shrink protector on the fiber joint. 6. Fusion Splicing Method Stripping Strip back the external sheathing of the cable using a rotary stripping tool. Cut back the aramid strength member using ceramic or kevlar scissors. Strip the primary buffer from the fiber using fiber strippers not ordinary wire strippers. Do this a small section at a time to prevent the fiber breaking, about 10mm (3/8 in) on each cut is fine until you get used to it. Strip back about 35mm (1.5 in). Clean the bare fiber with a lint free wipe and isopropyl alcohol, it will "squeak" when it is clean. Cleaving The cleaver first scores the fiber and then pulls the fiber apart to make a clean break. It is important that the ends are smooth and perpendicular to get a good joint; this is why a hand held cleaver will not do. Cleavers vary from manufacturer to manufacturer and you should read the instructions for the one you are using. Basically the operation consists of putting the fiber into the groove and clamping, then close the lid and press the lever. Fusion Process Once the fiber ends are prepared they are placed in the fusion splicer. Press the button and the machine takes care of the rest of the fusion process automatically. First the two fibers are aligned; you can see this on the photo where a much magnified image shows the two fiber ends. The display also shows how well the cleaver does its job of producing a perfect 90 degree cut.


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If you watch very carefully in the video you can see the X and Y alignment that takes place. The splicer aligns the fibers on one axis and then from another camera angle set at 90 degrees, it aligns the other axis. This high precision alignment is critical for a low loss joint; any mismatch of the fiber cores will significantly reduce the propagation of light through the joint. Bearing in mind that we are dealing with two very small glass rods of only 125 microns in diameter, it brings it home as to how extremely accurate these machines are. Once the fibers are aligned the splicer fires an electric arc between the two ends which melts them immediately and pushes them together, or fuses them into one piece of fiber. The fusion splicer then tests for dB loss and tensile strength before giving the "OK" beeps for you to remove the splice from the machine Protection The splicer in the video has a built in heat shrink oven, so when the fiber is taken out of the machine the protective tube is slid into place and the whole assembly is put into the oven to shrink the tube on to the splice. The protective tube gives physical protection to the splice and further protection is provided by placing the splice into a splice tray. Once all of the fibers have been joined the whole tray is then fixed into a splice box which protects the cable joint as a whole and the cable clamps are then tightened to prevent any external forces from pulling on the splices.


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Fusion Splicing Procedure


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Preparing the Fiber Ends Note: Minimum splice loss can only be achieved if the fiber ends have been prepared carefully. Preparation of the Fiber includes the following steps • Stripping the coating • Cleaning the fibers • Cutting the fibers with cleaver • Assessing quality of the fibers and faces (monitor) Follow the instructions for the fiber cleaver Remove the coating

Fig. 5.6 Stripping the coating and cleaning the Fiber Cleaning the fibers Clean the ends of the fibers over the length of about 10cm (coated and uncoated fibers) With a lint free paper cloth dipped in alcohol or with alcohol in ultrasonic bath Danger : there is a risk of fire if the solvent is spilled


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Cutting the fiber Using the cleaver S46999 M9-A8

• Select the correct fiber guides for the coating diameter (250 µm, 500 µm, or 900 µm

• Open the flap of the cleaver and place the fiber in the cleaver so that the end of the coating is 10mm mark

• Close the flap of the cleaver and then press down carefully • Open the flap before you cut fiber in order to avoid the damage

to the end face of the fiber.

Fig. 5.7 Cutting the Fiber with RXS cleaver Note: If you press down the flap too quickly the fiber and the face quality will be poor and you will damage the diamond. Inserting the fibers Insertion in the fiber holder

• Open the electrode flap and the clamps flap for the tensile test equipment and the fiber holder flaps

• Place the fibers in the V Groove of the slide so that the ends of the fibers are between the electrodes.


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Fig. 5.8 Inserting the Fiber in the Fiber Holder

• Close the fiber holder flap prepare the second fiber and insert in the same way on the other side.

• Close the Electrode flap. If you wish to perform a tensile test on the splice after fusing, close the flaps of the tensile test device as well.

Note: Ensure that the fiber ends are visible on the monitor after insertion. Otherwise the resulting splice may be poor or the travel of the positioner may not be adequate. Checking the fiber end faces

• After inserting the fibers in the in the splicers select menu items (“searching fibers”?) or Automatic or fully automatic.

• The fiber ends are displayed in two views from the direction of the X axis ( upper half of the monitor) and from the direction of the Y axis ( lower half of the monitor)

Note: the fibers must be

• Clean • Free of protrusions and indentation • Flat and square to the fiber axis • The quality of the splice is always determined by the quality of

the fiber end faces.


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Fig. 5.9 Assessing the quality of the Fiber end faces

• If the quality of the fiber end is poor the end face must be prepared again.

• If you are in “fully automatic” or “automatic mode” and notice that a fiber end is of poor quality you can terminate the proceeding process by pressing key.

Splicing fibers Splicing the fibers includes the following steps • Cleaning the fibers • Adjusting the fibers (Z axis gap X/Y offset) • Fusing the fibers • Assessing the splice quality (with fully automatic and automatic) Cleaning Clean the fibers, repeat the process if the fiber ends are still visibly contaminated. If the contamination is still not removed completely after this the fiber must be prepared again. Adjusting and fusing Adjustment of the fibers in X-Y-Z –axis and initiation of the fusion process can be implemented fully automatically, automatically or manually depending on which splicing process you have selected. Manual position is only possible for the Z-axis. Procedure for each splicing process is already described.


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Assessment of the Splice and Refusing Visual assessment If a good splice has been made, the cladding has a smooth surface. There should be no visible faults or shadows inside the fiber image as shown in the next page. The bright lines in the center of the fiber are result of the lens effect.

Fig. 5.10 Visual Splice assessment Splice loss indication In fully automatic and automatic mode the splice loss is indicated in dB to an accuracy of +/- 0.1 dB. Causes of Poor Splice quality

• Quality of the fiber end faces unsatisfactory • Fusion parameters are not set correctly and not optimized.

Fit the splice protector as follows:

Connect the heater to one of the terminals on the splicer. Make sure that the splicer is connected to a battery or AC power supply. Close the cover of the heater and check that the red LED lights. Open the cover again. Push the heat shrink splice protector over one of the fiber ends before splicing. Splice the fibers as described and then push the splice protector centrally over the splice.


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Note: The splice protector must be pushed over one of the fiber ends before fusion since this is no longer possible once the fiber ends have been fused. Place the fiber with the splice protector in the heater. Only apply light tension to the splice in order to avoid damage. Make sure that the fiber remains in position in the splice protector. Cleaning: Remove any dirt and pieces of fiber. Flaps closed? Close all flaps on the splicer. Case lid: If you are working with a splicer integrated in the case, clip in the lid of the case and close the case. Warning: Do not use force to close the case. Make sure that all parts of the splicer are in their transport position.


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Module - 01


Chapter – 06

OFC Test and Measurements


• Introduction • Calibrated Light Source • The Light Power Meter • The Optical Attenuator • Optical Time Domain Reflectometer (OTDR) • Fiber Optic Power Meter (FOT-12A-50) • Optical Talk Set (TSH-01 Ver.2 1550nm) • Continuity Test • Cable Loss


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1. Introduction Low loss, wide bandwidth, non–inductive transmission without crosstalk and highly insulated, thin and lightweight optical fiber cables have revolutionised the field of Telecommunication. Testing a Fiber–Optic based system requires special instrumentation if the link itself must be checked. The basic structure of communication system using Optical Fiber is shown in Fig. 6.1 below:

Fig. 6.1 System Composition

Measurement Requirements The most commonly used tests in field are Optical Power and Cable loss. The instruments needed for these tests include a calibrated light source, a light power meter, an optical attenuator, and optical Time Domain Reflectometer. Optical Devices 2. Calibrated Light Source The calibrated light source is the equivalent of a signal generator. It must generate light energy signals of known power levels. These light signals come out through an LED or LASER. The other requirement may be the need to vary the wave–length (frequency) of the light wave, just as a signal generator allows the user to vary the frequency. The reason for this is that optical fibers perform best at specific light wavelengths only (depending on the exact type of glass used) and the light supplied to the optical fiber must have the matching wave-length for that fiber type. 3. The Light Power Meter The light power meter measures the power of any light signal, which typically ranges from 1 nanowatt (nW) to 2 milliwatts (mW). This is similar to measuring the voltage or power of an electrical signal. The power meter must also be calibrated carefully, since its readings form the


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basis for determining the amount of light energy that is present. The power meter uses a precise light–to–electrical–energy transducer and then measures its electrical output. 4. The Optical Attenuator The optical attenuator is similar to a simple potentiometer or circuit used to reduce a signal level. The attenuator is used whenever performance tests must be run. For example, to see how the bit error rate is affected by varying the signal level in the link. The optical equivalent is much more complex than the one used in electronic circuitry. It can be implemented in two ways. One way is to have a precise mechanical setup in which the optical signal passes through a glass plate with differing amounts of darkness and then back to the optical fiber, as shown in Fig. 6.2. The glass plate has gray density ranging from 0 percent at one end to 100 percent at the other end. As the plate is moved across the gap, more or less light energy is allowed to pass. This type of attenuator is very precise, and can handle any light wavelength (since the plate attenuates any light energy by the same amount, regardless of wavelength), but it is mechanically expensive.

Fig. 6.2 An attenuator for fiber–optic

5. Optical Time Domain Reflectometer (OTDR) The OTDR is an important tool to characterize a fiber’s attenuation, uniformity, splice loss, breaks and length. Its main advantage is one–port operation at the fiber input with no need to access the fiber output. These are measured by launching an optical pulse into one end of the optical fiber and detecting reflected pulses (back–scattered light) returning to the instrument. The measuring conditions, items and measured results are displayed on CRT.


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An optical pulse is launched into one end of fiber. This optical pulse, on encountering microscopic non–uniformities of route, will cause light scattering in all directions. If the size of non–uniformity is much less then, the scattering phenomenon is governed by Rayleigh scattering principle (Scattering coefficient = 1/λ4, i.e. light scattering decreases as the wavelength increases) out of this scattered light a portion of it, i.e. back scattered light, may reach back to the fiber input. Infect, due to limited numerical aperture of the fiber, only a portion of back-scattered light reaches the fiber input. When the optical pulse launched into the fiber encounters a transition in media, for example, at a fiber 'break (Fig. 6.3), connectors, splicing points and free fiber ends, reflection occurs in addition to refraction. Depending upon the reflection coefficient at that point, the reflected light is received back at the sending end of the fiber. This reflected light is typically 4% of incident light. Thus, the reflected light received at the sending has a much higher power level as compared to back–scattered light, which is typically 50 dB lower than the incident optical power. A typical OTDR arrangement is shown in the Fig. 6.4.

Fig. 6.3 Break in Fiber

Fig. 6.4 OTDR Instrument Principle


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A pulse generator drives a laser diode which the launches high power optical pulses (100 nW up to several mW) with pulse widths of 100 ns to 4 µs and repetition rate of few kHz into fiber using a polarising beam splitter. Refractive index of fiber core, under test, up to four decimal points can be entered in the instrument for exact display of fiber length on x–axis. An APD is used as a detector. Its signal is fed to amplifier. The diagram in Fig. 6.5 shows a typical signal display on OTDR screen. The reflection from the fiber front is the largest signal in an OTDR. The attenuation of cable section can be obtained from equation: α = 1/L x 0.5 x 10 log (p1/p2) dB/km Back scattered light from two non–uniformities reach the input at the same time if they are spaced apart by half of pulse length in fiber as shown below.

Fig. 6.5 OTDR Signal Display

Fig. 6.6 Explanation of the ∆z/2 Uncertainty of the OTDR Signal


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It may be noted that at the receiving end, it cannot be distinguished whether the received back–scattered light pertained to discontinuities of z or z' points. It may be concluded that z–z', i.e. dz/2 is the unresolvable length of OTDR signal. For example: A pulse width W of 10 ns and a refractive index n = 1.5 results in Group velocity = (3 x 108)/1.5. Pulse length in fiber = Pulse width W x Group velocity Pulse length in fiber = (3 x 108)/(1.5 x 10–8) = 2 m. Uncertainty(minimum resolvable length in fiber)=2m/2 =1m or +0.5 m Improvement in received power is possible by increasing pulse width at the expense of length resolution. Back-scattered signals are normally buried in noise because of their tiny amplitudes. Special techniques are used to improve S/N of received signal in OTDR. Concept of Dead zone The dead zone is the distance corresponding to the initial length of fiber under test whose trace is missed in OTDR (as shown in Fig. 6.6) due to the saturation of the receiver circuit of OTDR by strong signal like Fresnel reflection and due to imperative property of this circuit that a certain time is lost in transforming the time into distance. The back-scattered signals received from the initial distance (dead zone) take too less a time to be transformed into corresponding trace by the receiver circuit. The dead zone is the sum of pulse–width and the distance corresponding to recovery time of the receiver circuit. The smaller the pulse–width, the smaller the dead zone.

7. Optical Talk Set This is a dedicated Talk set for fiber optic engineers involved in telecommunications plant installations and maintenance. This set allows Full-Duplex conversation between up to four operators with headset or voice gated speakerphone over a single optical fiber. Designed for long distance single mode applications. The features of the set include volume control (both in headset and speakerphone modes) for the best listening comfort and a Call function for alerting the companion user. 8. Continuity Test The simplest is the continuity test, to find out whether there is any break in the fiber. It requires only a light source and optical power meter. Some light energy is coupled into the cable, and some should come out the other end. The test is a good/bad test, which doesn't indicate anything about the cable loss. The light source and power meter do not have to be calibrated or accurate for this simple test as in figure below.


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Fig. 6.7 Continuity Test Sometimes, a simple flashlight is used as the light source, and the light coming out can be seen by eye. This is not a recommended practice because in a multifibre bundle, the fiber the technician looks into, may not be the one with the flashlight at the other end. Instead, it may be an active fiber that has a laser LED as its source, and eye damage can result. The flashlight and eye should only be used for a single strand of fiber, when the entire length of fiber is visible, and there can be no mistakes.

Fig. 6.8 Visual Inspection 9. Cable Loss To check cables loss, a measured amount of light energy at a specific wavelength is coupled into the cable, and the amount of output light power is measured. The cable loss is specified in decibels. Since the amount of energy coupled into the cable must be known, the light source must be calibrated, or the power meter must first be used to measure the output of the source directly and then the output through the cable using the same source.


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Module - 01


Chapter – 07

Optical Connectors and Couplers


• Connectors • Connector Requirement • Connector Composition • SC Connectors • ST Connectors • LC Connectors • FC/PC Connector • MT-RJ Connectors • MTP/MPO Connectors • Fibre Connecting Techniques • Optical Connector types and insertion loss • Couplers


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Fiber-Optic Cable Termination 1. Connectors The connectors are rematable interconnect devices which provide flexibility required in a Fiber Optical Transmission system. The basic function required of connectors is to allow transfer of optical power from one fiber component to another with minimum loss and possibility of disconnection and remitting number of times with minimum insertion loss. 2. Connector Requirement

• The attenuation in optical fiber connectors should be less than 1 dB.

• The connector must provide consistent performance on each remitting.

• The connector must provide protection to the fiber so that it does not break while being handled.

• The connectorisation technique should be simple. • The connector size should not be very much bigger than the

fiber size and it should not be too small. • Connector must be cost effective.

3. Connector Composition Connector fundamentally consists of two parts, a plug and an adapter. For fiber to fiber connections, the fibers are terminated in individual plugs and mated in the adapter. For fiber to device connection, the devices may be housed in the adapter part and the fiber in the plug part. The fixing of the fiber in the plug may be achieved directly or by using sleeves commonly known as ferrules. The proper centering in these ferrules could be achieved by using precision drilled holes, jewels or rods depending on the arrangement. The adapter provides the alignment mechanism. The performance of the connectors depends on the accuracy of the alignment of the optical elements to be connectorised. The basic elements in the connectors are fiber fixing mechanism and the alignment mechanism. The alignment accuracy required is of very high to avoid losses and are consequently quite costly. There are many types of optical connectors. The one you use depends on the equipment you are using it with and the application you are using it on. The connector is a mechanical device mounted on the end of a fiber-optic cable, light source, receiver, or housing. The connector allows the fiber-optic cable, light source, receiver, or housing to be mated to a similar device. The connector must direct light and collect light and must be easily attached and detached from equipment.


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A connector marks a place in the premises fiber-optic data link where signal power can be lost and the BER can be affected by a mechanical connection. Of the many different connector types, those for glass fiber-optic cable and plastic fiber-optic cable are discussed in this chapter. Other considerations for terminations are repeatability of connection and vibration resistance. Physical termination density is another consideration. Commonly used fiber-optic connectors are shown in Figure. 7.1 Fig 7.1 : Fibre Optic Connectors 4. SC Connectors SC connectors are used with single-mode and multimode fiber-optic cables. They offer low cost, simplicity, and durability. SC connectors provide for accurate alignment via their ceramic ferrules. An SC connector is a push-on, pull-off connector with a locking tab. Typical matched SC connectors are rated for 1000 mating cycles and have an insertion loss of 0.25 dB. From a design perspective, it is recommended to use a loss margin of 0.5 dB or the vendor recommendation for SC connectors.


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5. ST Connectors The ST connector is a keyed bayonet connector and is used for both multimode and single mode fiber-optic cables. It can be inserted into and removed from a fiber-optic cable both quickly and easily. Method of location is also easy. ST connectors come in two versions: ST and ST-II. These are keyed and spring-loaded. They are push-in and twist types. ST connectors are constructed with a metal housing and are nickel-plated. They have ceramic ferrules and are rated for 500 mating cycles. The typical insertion loss for matched ST connectors is 0.25 dB. From a design perspective, it is recommended to use a loss margin of 0.5 dB or the vendor recommendation for ST connectors. 6. LC Connectors LC connectors are used with single-mode and multimode fiber-optic cables. The LC connectors are constructed with a plastic housing and provide for accurate alignment via their ceramic ferrules. LC connectors have a locking tab. LC connectors are rated for 500 mating cycles. The typical insertion loss for matched LC connectors is 0.25 dB. From a design perspective, it is recommended to use a loss margin of 0.5 dB or the vendor recommendation for LC connectors. 7. FC/PC Connector FC/PC has been one of the most popular single-mode connectors for many years. It screws on firmly, but make sure you have the key aligned in the slot properly before tightening. SCs and LCs are replacing it. 8. MT-RJ Connectors MT-RJ connectors are used with single-mode and multimode fiber-optic cables. The MT-RJ connectors are constructed with a plastic housing and provide for accurate alignment via their metal guide pins and plastic ferrules. MT-RJ connectors are rated for 1000 mating cycles.


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The typical insertion loss for matched MT-RJ connectors is 0.25 dB for SMF and 0.35 dB for MMF. From a design perspective, it is recommended to use a loss margin of 0.5 dB or the vendor recommendation for MT-RJ connectors. 9. MTP/MPO Connectors MTP/MPO connectors are used with single-mode and multimode fiber-optic cables. The MTP/MPO is a connector manufactured specifically for a multifiber ribbon cable. The MTP/MPO single-mode connectors have an angled ferrule allowing for minimal back reflection, whereas the multimode connector ferrule is commonly flat. The ribbon cable is flat and appropriately named due to its flat ribbon-like structure, which houses fibers side by side in a jacket. The typical insertion loss for matched MTP/MPO connectors is 0.25 dB. From a design perspective, it is recommended to use a loss margin of 0.5 dB or the vendor recommendation for MTP/MPO connectors.


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• Connectors are rematable devices • Provide flexibility for interconnection of different section of fiber in

Fiber Optical Transmission System Requirements • Attenuation should be low <1db • Should be good in performance • Should protect the fiber while handling • Should be made with simple method - No special tools should be

required for making the connection • Small in size • Cost effective Connectors & Couplers composition • Connectors are plug in type • Couplers are socket or adapter type • Fibers terminated in plug • Adapters have two sides. Each side plug is screwed 10. Fibre Connecting Techniques • Core Centered Connectors – Core is centered in the plug. Pig tails are

this type. Made in factory only • Core Aligned Connectors – Core is aligned in the plug by a special

machine. Made in factory only • Precision Mould Connectors – Made from plastic materials and

molded in machine • Optical Tech. – Optically collimating, Focussing connection, Fibre

taper connection


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• Optically collimating and focusing tech. Use lenses to collimate or focus the light

• Fibre taper provide large area of contact and reduce loss Types of Connectors

• FC, D3, D4, Dorran type connectors are used • These connectors have pin and notches arrangement to provide good

performance and less loss using repeatedly • FC type connectors are used widely now Fiber-to-fiber interconnection can consist of a splice, a permanent connection or a connector, which differs from the splice in its ability to be disconnected and reconnected. Ferrule

• The fiber is mounted in a long, thin cylinder, the ferrule, which acts as a fiber alignment mechanism.

• The ferrule is bored through the center at a diameter that is slightly larger than the diameter of the fiber cladding.

• The end of the fiber is located at the end of the ferrule. • Ferrules are typically made of metal or plastic. Connector Body

• The connector body holds the ferrule. • It is usually constructed of metal or plastic and includes one or more

assembled pieces which hold the fiber in place. • The Connector body assemblies vary among connectors. • The ferrule extends past the connector body to slip into the coupling

device.


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Connector and Coupling Device The cable is attached to the connector body. It acts as the point of entry for the fiber. Typically, a strain-relief boot is added over the junction between the cable and the connector body, providing extra strength to the junction. The Coupling Device

• Do not use the male-female configuration. • Instead, a coupling device such as an alignment sleeve is used to

mate the connectors. • These devices are also known as feed-through bulkhead adapters. 11. Optical Connector types and insertion loss

C o n n e c to r In s e rtio n L o s s R e p e a ta b ility F ib e r T yp e

F C

0 .5 0 -1 .0 0 d B 0 .2 0 d B S M , M M

F D D I

0 .2 0 -0 .7 0 d B 0 .2 0 d B S M , M M

L C

0 .1 5 d b (S M ) 0 .1 0 d B (M M )

0 .2 d B S M , M M

M T A rra y

0 .3 0 -1 .0 0 d B 0 .2 5 d B S M , M M

S C

0 .2 0 -0 .4 5 d B 0 .1 0 d B S M , M M

S C D u p le x

0 .2 0 -0 .4 5 d B 0 .1 0 d B S M , M M

S T

T yp . 0 .4 0 d B (S M ) T yp . 0 .5 0 d B (M M )

T yp . 0 .4 0 d B (S M ) T yp . 0 .2 0 d B (M M )

S M , M M


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Various Optical Connectors E2000 Connector: E2000/LX-5 is like a LC but with a shutter over the end of the fiber FDDI Connector Developed by ANSI for use in FDDI network, this is a duplex connector using two 2.5mm ferrules.


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12. Couplers FC Coupling SC Coupling ST Coupling LC Coupling Damaged Connector

• A number of events can damage fiber optic connectors. • Unprotected connector ends can experience damage by impact,

airborne dust particles, or excess humidity or moisture. Effects on Fibre Optics Connectors

• Never clean an optical connector attached to a fiber that is carrying light.

• The micro-explosions at the tip of the connector can leave pits in the end of the connector and crack the connector’s surface, destroying its ability to carry light with low loss.

Cleaning

• The fiber end face and ferrule must be absolutely clean before it is inserted into a transmitter or receiver.


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• Dust, lint, oil (from touching the fiber end face), or other foreign particles obscure the end face, compromising the integrity of the optical signal being sent over the fiber.

• Larger dust particles (9 µm or larger) can completely obscure the core of a single-mode fiber.

• Fiber optic connectors need to be cleaned every time they are mated and unmated.

• Cover a fiber optic connector when it is not in use. • Unprotected connector ends are most often damaged by impact, such

as hitting the floor. • Most connector manufacturers provide some sort of protection boot. • The protectors cover the entire connector end, but they are generally

simple closed-end plastic tubes that fit snugly over the ferrule only. • These boots will protect the connector's polished ferrule end from

impact damage that might crack or chip the polished surface. Cleaning Technique

• Use only industrial grade 99% pure isopropyl alcohol and lint-free tissue required.

• Fold the tissue twice so it is four layers thick. • Saturate the tissue with alcohol. First clean the sides of the connector

ferrule. • Place the connector ferrule in the tissue, and apply pressure to

the sides of the ferrule. • Rotate the ferrule several times to remove all contamination from the

ferrule sides. • Now take a clean tissue and saturated with alcohol and that it is still

four layers thick. • Put the tissue against the end of the connector ferrule. • Put your fingernail against the tissue so that it is directly over the

ferrule. • Now scrape the end of the connector until it squeaks. It will sound

like a crystal glass that has been rubbed when it is wet. • Mate the connector immediately! Don’t let the connector lie around

and collect dust before mating. • Air can be used to remove lint or loose dust from the port of a

transmitter or receiver to be mated with the connector. Never insert any liquid into the ports.

Handling

• Never touch the fiber end face of the connector. • Connectors that are not in use should be covered over the ferrule by a

plastic dust cap.


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Module - 01


Chapter – 08

Coding Theory

Session Objectives: On completion of this session, you will be able to understand the concepts and explain the features of transmission systems such as:

• Digital-To-Digital encoding • Unipolar • Polar • Bipolar (Biphase) • Line Codes • AMI Code • HDB3 Code • CMI Code (Coded Mark Inversion Code) • 5B6B Code


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1. Digital-To-Digital encoding Digital-to-digital encoding is the representation of digital information by a digital signal. For example, when we transmit data from the computer to a printer, both the original data and the transmitted data are digital. In this type of encoding, the binary 1s and 0s generated by a computer are translated into a sequence of voltage pulses that can be propagated over a wire.

Fig 8.1 Digital to digital encoding There are three types of digital encoding :

• Unipolar • Polar • Bipolar

2. Unipolar Unipolar encoding is very simple and very primitive. Although it is almost obsolete today, its simplicity provides an easy introduction to the concepts developed with the more complex encoding systems and allows us to examine the kinds of problems that any digital transmission system must overcome. Unipolar encoding is so named because it uses only one polarity. Therefore, only one of the two binary states is encoded, usually the 1. The other state, usually the 0, is represented by zero voltage, or an idle line.

Fig 8.2 Unipolar encoding


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3. Polar Polar encoding uses two voltage levels; one positive and one negative. By using both levels, in most polar encoding methods, the average voltage level on the line is reduced. Types of polar encoding 1. NRZ

• NRZ-L • NRZ-I

2. RZ 3. BIPHASE

• Manchester • Differential Manchester

Non-Return to Zero (NRZ) In NRZ encoding, the level of the signal is always either positive or negative. Unlike in unipolar encoding, where a 0 bit is represented by an idle line, in NRZ systems if the line is idle, it means no transmission is occurring at all. The two most popular methods of NRZ transmission are discussed below. NRZ-L In NRZ-L encoding, the level of the signal depends on the type of bit is represents. A positive voltage means the bit is a 1, and negative voltage means the bit is a 0; thus, the level of the signal is dependent upon the state of the bit.

Fig 8.3 NRZ-I and NRZ-L Encoding


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NRZ-I In NRZ-I, an inversion of the voltage level represents a 1 bit. It is the transition between a positive and negative voltage, not the voltages themselves that represents a 1 bit. A 0 bit is represented by no change. An advantage of NRZ-I over NRZ-L is that because the signal changes every time a 1 bit is encountered, it provides some synchronization. A series of seven 1s will cause seven inversions. Each of those inversions allows the receiver to resynchronize its timer to the actual arrival of the transmission. Statistically, strings of 1s occur more frequently in transmissions than do strings of 0s. Return to Zero (RZ) As we can see, anytime the original data contain strings of consecutive 1s or 0s, the receiver can lose its place. As we mentioned in our discussion of unipolar encoding, one way to assure synchronization is to send a separate timing signal on a separate channel. However, this solution is both expensive and prone to errors of its own. A better solution is to somehow include synchronization in the encoded signal, something like the solution provide by NRZ-I, but one capable of handling strings of 0s as well as 1s. To assure synchronization, there must be a signal change for each bit. The receiver can use these changes to build up, update, and synchronize its clock. As seen above, NRZ-I accomplishes this for sequences of 1s. But to change with every bit, we need more than just two values. One solution is return to zero (RZ) encoding, which uses three values: positive, negative, and zero. In RZ, the signal changes not between bits but during each bit. Like NRZ-L, a positive voltage means 1 and a negative voltage means 0. But, unlike NRZ-L, halfway through each bit interval, the signal returns to zero. A 1 bit is actually represented by positive-to-zero, and a 0 bit by negative-to-zero, rather than by positive and negative alone. The following figure illustrates the concept.

Fig 8.4 RZ Encoding


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The main disadvantage of RZ encoding is that it requires two signal changes to encode one bit and therefore occupies more bandwidth. But of the three alternatives we have examined so far, it is the most effective. 4. Bipolar (Biphase) Bipolar encoding, like RZ, uses three voltage levels: Positive, negative and zero. Unlike RZ, however, the zero level in bipolar encoding is used to represent binary 0. Positive and negative voltages represent alternating 1s. If the first 1 bit is represented by the positive amplitude, the second will be represented by the negative amplitude, the third by the positive amplitude, and so on. This alternation occurs even when the 1 bits are not consecutive. Probably the best existing solution to the problem of synchronization is bipolar encoding. In this method, the signal changes at the middle of the bit interval but does not return to zero. Instead, it continues to the opposite pole. As in RZ, these mid interval transitions allow for synchronization. As mentioned earlier, there are two types of bipolar encoding in use on networks today:

• Manchester • Differential Manchester.

Manchester Manchester encoding uses the inversion at the middle of each bit interval for both synchronization and bit representation. A negative-to-positive transition represents binary 1 and positive-to-negative transition represents binary 0. By using a single transition for a dual purpose, Manchester encoding achieves the same level of synchronization as RZ but with only two levels of amplitude. Differential Manchester In Differential Manchester, the inversion at the middle of the bit interval is used for synchronization, but the presence or absence of an additional transition at the beginning of the interval is used to identify the bit. A transition means binary – and no transition means binary 1. Differential Manchester requires two signal changes to represent binary 0 but only one to represent binary1.


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Fig 8.5 Manchester and Differential Manchester Encoding Three types of bipolar encoding are in popular use by the data communications industry, such as AMI, B8ZS and HDB3. 5. Line Codes If encoder output of PCM equipment is transmitted over the transmission medium, viz. VF cable pair, the signal is likely to under go high frequency attenuation distortion and cross–talk. Moreover, the signal has strong DC content and thus prevents the use of AC coupled circuits. For distortion free transmission, the encoder output should be converted into a suitable code which will match the characteristics of the medium. This code is called the "Line Code" and the signal converted to the line code is called the line signal. Characteristics The line code needs to have the following characteristics:

• Restricted bandwidth • Low energy in the upper part of the signal spectrum to

reduce attenuation distortion. • Low energy in the lower part of the spectrum to reduce

cross–talk. • No DC component so that transformers can be used for

coupling purposes. • Contain adequate timing information. • Have an in–built error monitoring capability.


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Following codes are available as Line codes.

• NRZ Binary • RZ Binary • Bipolar (AMI) • HDB3 • CMI • 5B6B

Because of the strong DC component and low frequency content, NRZ and RZ codes are not suitable for transmission. 6. AMI Code AMI stands for "Alternate Mark Inversion". This code solves the DC content problem. Here, a logic '0' is represented by 0 volt and logic '1' is alternately encoded with positive and negative voltages. Therefore, the average voltage is maintained very close to zero and hence there is no DC component. Under steady state conditions, a low DC of the order of 0.4 to 0.8 volts only remains. By inverting on each occurrence of a 1, bipolar AMI accomplishes two things:

• The DC component is zero, • A long sequence of 1s stays synchronized.

Three is no mechanism to ensure the synchronization of a long string of 0s.

Fig. 8.6 AMI Code Signal Waveform Two variations of bipolar AMI have been developed to solve the problem of synchronizing sequential 0s. The first, used in North America, is called bipolar 8-zero substitution (B8ZS). The second, used in Europe and Japan, is called high-density bipolar3 (HDB3). Both are


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adaptations of bipolar AMI that modify the original pattern only n the case of multiple consecutive 0s. 7. HDB3 Code To overcome the timing difficulties in the AMI code, another code called the HDB3 code has been devised. The abbreviation HDB stands for HIGH DENSITY BIPOLAR code. The HDB3 code is actually a code from a family of codes derived from what is called Binary N Zero Substitution or BNZS method. In this method, the PCM signal is usually transmitted according to the AMI code, but when a string of N zeros is encountered, the N zeros are replaced by a special code which will deliberately introduce a bipolar deviation or violation. Normally in the AMI code, if there are N zeros, they will be transmitted as such. But in the BNZS method, a '1' pulse is introduced deliberately. The polarity of this '1' depends upon the polarity of the previous mark encountered. This additional '1' pulse introduced in place of a '0' is called a 'violation'. When the substitution of a zero by a violation pulse is done for 4 zeros, i.e. N = 4, the BNZS code is called the B4ZS code. Since this code precludes strings of zeros greater than three, it is also referred to as a HDB3 code. Here, when the number of zeros is more than 3, the fourth bit position is filled with a violation pulse. Consecutive violations are made to be of opposite polarity so that these violations themselves do not produce any DC component. The violation pulse is always placed in the last bit position. Suppose there are 4 zeros coming in a row. Then the HDB3 code for this would be ‘ B00V ’ in general where V is the violation pulse. The polarity of this depends on the polarity of the last '1' and the number of '1's encountered prior to the four zeros.

a) NRZ Code b) RZ Code c) HDB3 Code Fig. 8.7 Example of HDB3 Code Conversion


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Fig 8.8 HDB3 Encoding The first bit of the code was shown as B in above. B is set to '0' if the number of '1's encountered prior to the violation is ODD. If it is EVEN or ZERO, then the "B" bit is filled with a '1' whose polarity is in accordance with the AMI. Examples of HDB3 Code Conversion Notice that up to pulse Z, the HDB3 code follows the AMI code. After pulse Z, we have four consecutive zeros. This calls for a violation. Prior to the arrival of these zeros, three '1's were encountered, i.e. number of '1's preceding the violation is ODD. This means that the HDB3 substitution for the zeros will be of the form 000V. Also, the polarity of the last '1' before the arrival of the zeros is positive. Therefore, code, i.e. if the previous '1' was positive, then B is '1' with negative polarity and vice versa. 8. CMI Code (Coded Mark Inversion Code) This is a 2 level NRZ code in which a binary '0' is coded as '1' and binary '1's are coded alternatively as a logic '0' or '1'. In case of a binary '0', the two CMI bits '0' and '1' are for half clock duration whereas for binary '1's the '0' and '1's are for full clock duration. This is illustrated in Fig. 8.9. This is basically a binary code and the bit rate of the code is twice the Bipolar AMI code. For this reason CMI code is grouped with IB2B family of line codes. The CMI code has high clock content and for this


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reason the CMI code is recommended by CCITT for 140 Mb/s multiplex equipment.

Fig. 8.9 CMI Code – An Example 9. 5B6B Code The objective of this coding is to modify the input bit pattern such that the continuity of ones or zeros is reduced and a mark ratio of nearly ½ is maintained in the resultant output bit pattern.


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Module - 01


Chapter – 09

Basics of Transmission Systems Session Objectives: On completion of this session, you will be able to understand the concepts and explain the features of:

• Plesiochronous Digital Hierarchy • Synchronous Digital Hierarchy • Microwave Systems • MW Applications • MW System Capacity • Communication through Satellite • Features of Satellite Communications • Advantages of Satellite Communications • Frequency Bands • VSAT System Components and Specifications • HVNET – DOT VSAT Network


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1. Plesiochronous Digital Hierarchy (PDH) The PDH is a technology used in telecommunications networks to transport large quantities of data over digital transport equipment such as fiber optics and microwave systems. The term Plesiochronous is derived from Greek plesio, meaning near, and chronos, time, and refers to the fact that PDH networks run in a state where different parts of the network are almost, but not quite perfectly, synchronised. PDH is now being replaced by Synchronous Digital Hierarchy (SDH) equipment in most telecommunications networks. PDH allows transmission of data streams that are nominally running at the same rate, but allowing some variation on the speed around a nominal rate. By analogy, any two watches are nominally running at the same rate, clocking up 60 seconds every minute. However, there is no link between watches to guarantee they run at exactly the same rate, and it is highly likely that one is running slightly faster than the other. The European and American versions of the PDH system differ slightly in the detail of their working, but the principles are the same. In European system, basic data transfer rate is a data stream of 2.048 Mbps (megabits/second). For speech transmission, this is broken down into 30 x 64 kbit/s (kilobits/second) channels plus 2 x 64 kbit/s channels used for signaling and synchronisation. Alternatively, the whole 2 Mbit/s may be used for non speech purposes, for example, data transmission. The exact data rate of the 2 Mbit/s data stream is controlled by a clock in the equipment generating the data. The exact rate is allowed to vary some percentage either side of an exact 2.048 Mbit/s. This means that different 2 Mbit/s data streams can be (probably are) running at slightly different rates to one another. In order to move multiple 2 Mbit/s data streams from one place to another, they are combined together, or "multiplexed" in groups of four. This is done by taking 1 bit from stream #1, followed by 1 bit from stream #2, then #3, then #4. The transmitting multiplexer also adds additional bits in order to allow the far end receiving multiplexer to decode which bits belong to which 2 meg data stream and so correctly reconstitute the original data streams. These additional bits are called "justification" or "stuffing" bits. Because each of the four 2 Mbit/s data streams is not necessarily running at the same rate, some compensation has to be made. The transmitting multiplexer combines the four data streams assuming that they are running at their maximum allowed rate. This means that occasionally, (unless the 2 Mbit/s really is running at the maximum rate) the multiplexer will look for the next bit but it will not have arrived. In this case, the multiplexer signals to the receiving multiplexer that a bit is "missing". This allows the receiving multiplexer to correctly reconstruct


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the original data for each of the four 2 Mbit/s data streams, and at the correct, different Plesiochronous, rates. The resulting data stream from the above process runs at 8.448 Mbit/s (about 8 Mbit/s). Similar techniques are used to combine four x 8 Mbit/s together, giving 34 Mbit/s. Four x 34 Mbit/s, gives 140. Four x 140 gives 565. 565 Mbit/s is the rate typically used to transmit data over a fiber optic system for long distance transport. Recently, telecommunications companies have been replacing their PDH equipment with SDH equipment capable of much higher transmission rates. 2. Synchronous Digital Hierarchy (SDH) Synchronous Digital Hierarchy (SDH) signals the beginning of a new phase in evolution of the world’s telecommunication network. SDH will bring a revolution in telecommunication services, which will have far-reaching effects for end users, service providers and equipment manufacturers alike. With introduction of SDH, the transmission network will enter a new era, which can be compared in scale to that occurred following the introduction of PCM and Optical fiber. As end users (particularly business users) become more dependent on effective communications, pressure builds up for a reliable and flexible network with unlimited bandwidth. The complexity of the current network, based on Plesiochronous transmission systems, means that network operators are unable to meet this demand. The current Plesiochronous Digital Hierarchy (PDH) evolved in response to the demand for plain voice telephony. (Sometimes called POTS- Plain old Telephone Service) is not ideally suited to the efficient delivery of and management of high bandwidth connections. Synchronous transmission systems address the shortcomings of the PDH. Using essentially the same fiber, the synchronous network is able to significantly increase the available bandwidth while reducing the amount of equipment in the network. In addition the provision within the SDH for sophisticated network management includes significantly more flexibility in the network. Deployment of synchronous transmission systems will be straightforward due to their ability to inter work with existing Plesiochronous systems. The SDH defines a structure which enables Plesiochronous signals to be combined together and encapsulated within a standard SDH signal. This protects network operators’ investment in Plesiochronous Equipments and enables them to deploy synchronous equipment in a manner suited to the particular needs of their network. As Synchronous Equipments become established within the network, the full benefits it brings will become apparent. The network operator will experience significant cost savings associated with reduced amount of hardware in the network and increased efficiency and


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reliability of the network will lead to savings resulting from a reduction in operation and maintenance costs. The sophisticated network management capabilities of a synchronous network will give a vast improvement in the control of transmission networks. Improved network restoration and reconfiguration capabilities will result in better availability and faster provisioning of services. The SDH offers network operators a future proof network solution. It has been designed to support future services such as Metropolitan Area Network (MANs), Broadband ISDN etc. 2.1 SDH Evolution PDH has reached a point where it is no longer sufficiently flexible or efficient to meet the demands being placed on it. As a result, synchronous transmission was thought to overcome the problems associated with Plesiochronous transmission. In particular, the inability of PDH to extract individual circuits from high capacity systems without having to demultiplex the whole system. Attempts to formulate a set of standards covering optical transmission of synchronous signals began in the U.S. at beginning of 1984. The aim was to have a synchronous standard to allow the interconnection of equipment from more than one vendor. In order to move away from proprietary interfaces and achieve true interconnectivity between vendors, subcommittee T1X1of the American National standard institute (ANSI) began work in 1985 on developing a Standard Optical NETwork (SONET) based on a proposal by Bell core. In 1986, the CCITT became interested in the work being carried out on SONET and after much debate on how to incorporate both US and European transmission hierarchies, final agreement was reached in February 1988 and CCITT working group XVIII brought out the recommendations on SDH, published in CCITT Blue Book 1989. Since then, an ongoing standards effort has continued to develop and refine the SDH standards. S.D.H. evolution is possible because of the following factors:

• Fiber Optic Bandwidth: The bandwidth in Optical Fibre can be increased and there is no limit for it. This gives a great advantage for using SDH.

• Technical Sophistication: Although, SDH circuitry is highly complicated, it is possible to have such circuitry because of VLSI technique which is also very cost effective.

• Intelligence: The availability of cheaper memory opens new possibilities.


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• Customer Service Needs: The requirement of the customer with respect to different bandwidth requirements could be easily met without much additional equipment.

The different services it supports are: • Low/High speed data. • Voice • Interconnection of LAN • Computer links • Feature services like H.D.T.V. • Broadband ISDN transport (ATM transport)

2.2 Advantages of SDH The advantages of the SDH system are listed below:

• First world standard in digital format. • First optical Interfaces. • Transversal compatibility reduces networking cost. Multi-

vendor environment drives price down • Flexible synchronous multiplexing structure. • Easy and cost-efficient traffic add-and-drop and cross connect

capability. • Reduced number of back-to-back interfaces improves network

reliability and serviceability. • Powerful management capability. • New network architecture. Highly flexible and survivable self

healing rings available. • Backward and forward compatibility: Backward compatibility

to existing PDH • Forward compatibility to future B-ISDN, etc.

2.3 When do we use SDH?

• When networks need to increase capacity, SDH simply acts as a means of increasing transmission capacity.

• When networks need to improve flexibility, to provide services quickly or to respond to new change more rapidly.

• When networks need to improve survivability for important user services.

• When networks need to reduce operation costs , which are becoming a heavy burden


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3. Microwave Systems With the advent of mass scale industrialisation in our country, the demand for more communication facilities came up. Several new telephone exchanges have been installed throughout the country for local communication and more and more carrier channels have been provided for carrying the trunk traffic. With the planned introduction of Subscriber Trunk Dialing throughout the country, the number of carrier chls required to interconnect different cities became too high to be accomplished by overhead lines. Thus, U/G Cables Carrier Systems were introduced, the first of them being the symmetrical pair Cable Carrier System between Calcutta and Asansol with an ultimate capacity of 480 channels. Then came the Co–axial Cable Carrier System linking all major cities in the country. With the development of Microwave technique, which can provide large block of circuits at comparative cost, the problem of long distance communication circuits appear virtually solved. A brief description of the Microwave technique is attempted in the following paragraphs. Electromagnetic waves can be broadly classified in terms of frequencies as follows Range Name Wavelength Uses

0–30 KHz V.L.F. Up to 10 km. Used for long communication. Has limited information. Bandwidth requires very high power.

30–300 KHz L.F. 10 km to 1 km 0.3–3 MHz M.F. 1 km to 100 m Radio Broadcast, Marine Power

in KW, ground wave propagation, i.e. follows the curvature of the Earth.

3–30 MHz H.F. 100 m to 10 m Long haul point to point communication. Propagation is by one or more reflections from ionosphere layers and so subject to variations.

30–300 MHz V.H.F. 10 m to 1 m Line of sight, Troposcatter communication.

0.3–3 GHz U.H.F. 1 m to 10 cm. –––––– do –––––– 3–30 GHz S.H.F. 10 cm to 1 cm. Line of sight, terrestrial M/W

and Satellite communication. 30–300 GHz E.H.F. 1 cm to 1 mm. Experimental.


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The term SHF corresponds to "MICROWAVE" Centrimetric waves. As a convention frequencies, above 1 GHz and up to 40 GHz are termed as Microwave. However, most of the m/w systems available are in the range of 1 to 18 GHz. 3.1 MW Applications M/W frequency bands are used for the following services:

• Fixed Radio Communication Services. • Fixed Satellite Services. • Mobile Services. • Broadcasting Services. • Radio Navigation Services. • Meteorological Services. • Radio Astronomy Services.

To meet the requirements of all above mentioned services, co–ordination among the users of M/W spectrum is necessary. In this regard (in the national context) the wireless planning and co–ordination wing (WPC) of the ministry of communication has allotted m/w frequencies spectrum, on the basis of various wireless users classified as general users and major users. Wireless users who are permitted to plan their services and take action for the development of the required equipments are major users. DOT has been nominated as a major wireless user by the WPC in 1981 in the following sub base band of the m/w spectrum for fixed radio communication. Microwave Spectrum Available for DOT (Table 1)

Band Bandwidth Available Spectrum Space

2 GHz 300 MHz 2000–2300 MHz 4 GHz 900 MHz 3300–4200 MHz 6 GHz 1185 MHz 5925–7110 MHz 7 GHz 300 MHz 7425–7725 MHz 11 GHz 1000 MHz 10,700–11,700 HHz 13 GHz 500 MHz 12,750–13,250 MHz

In India the first M/w System was completed in December, 1965 between Calcutta and Asansol with a system capacity of 1200 channels. At present thousands of kilometers of M/W systems are scattered throughout the country and further expansion is taking place at a very large rate.


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3.2 Frequency Characteristics Microwaves are very short frequency radio waves that have many of the characteristics of light wave in that they travel in line–of–sight paths and can be reflected, boomed and focused. By focusing these ultra high radio waves into a narrow beam, their energies are concentrated and relatively low transmitting power is required for reliable transmission over long distance. 3.3 MW System Capacity Microwave communication systems are used to carry telephony, television and data signals. Majority of the systems, however, carry multi–channel telephone signals. The spectrum of the multichannel telephone signal is shown in Table 1. This signal is also called base band. Individual telephone channels, 4 KHz wide (300 to 3400 Hz for speech and the remaining for signaling and guard band) are multiplexed together in a multiplex equipment to get the base band. The base band frequency occupied by some typical channel capacity system is given in Table below:

Channel Capacity Base band frequency(in KHz)

60 channels 12–252 60 channels 60–300 120 channels 60–555 300 channels 60–1300 600 channels 60–2540 960 channels 60–4028 1800 channels 312–8120/316–8204 2700 channels 312–12336/316–12388

The system capacity of line of sight systems ranges from 60 telephone channels to 2700 channels over a Radio bearer with a few systems of lower capacities varying from 60 to 600 channels. On the same m/w route one can use more than one radio channels, thus getting still larger capacity. As an example one can accommodate 8 go and 8 return RF channels each with a capacity of 1800 telephone channels in a 500 MHz bandwidth. Of course, in such cases usually one or two RF channels are kept as a standby which is switched over automatically on fading or equipment failure. Usually the system with capacities up to 300 channels is called narrow band system and the systems providing more than 300 channels are called wide band system. Microwave systems used to provide communication on major trunk routes with high traffic density and serving long distances are classified as long haul m/w systems. 2, 4, and 6 GHz systems are long haul systems. Systems used to provide communication over short distances for trunk routes with


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light traffic density are classified as short haul system. 7 and 11 GHz systems are short haul systems.

Fig. 9.1 Typical Microwave Radio System

Fig. 9.2 Schematic of a Microwave Terminal


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4. Communication through Satellite Long distance communication using conventional techniques like coaxial cable or microwave radio relay links involves a large number of repeaters. For radio relay links of repeater spacing is limited by line of sight and is of the order of tens of kms. As the number of repeaters increase system performance and reliability are degraded. Tropo scatter propagation can cover several hundred kms. but the channel capacity is limited and costs are high due to necessity of large antennas and high transmit power. HF communication is subject to fading due to ionosphere disturbances and channel capacity is severely restricted due to limited bandwidth available. Large areas could be covered if the height of microwave repeater could be increased by putting it on board an artificial earth satellite. Science Fiction writer Arthur C. Clarke in an article in Wireless World in 1945 proposed that worldwide coverage could be obtained by using three microwave repeaters placed in a geostationary orbit at the height of about 36000 kms. with a period of 24 hours (Fig. 9.3).

Fig. 9.3 Modes of Communication


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Fig. 9.4 Global Coverage with Geo stationary Satellite Satellite communication provides a practical and economical means of long haul communication traffic in a country with a large geographical area. It also enables communication service to those areas which are virtually INACCESSIBLE by other conventional forms of communication system due to natural physical barriers.


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4.1 Principles and Features of Satellite Communications Principles In satellite communications, a satellite with microwave radio repeater equipment receives and amplifies radio waves sent from earth stations and returns them to the earth. A geo-stationary satellite is launched above the equator 36,000 km high above the earth. Its period round the earth coincides with that of the earth rotation. Therefore, the satellite looks as if it is stationary from the earth. If three (3) communication satellites are launched equidistantly above the equator (See Fig. 9.4), it can serve almost all communication networks round the world. Therefore, to facilitate public international telecommunications, INTELSATS IV and V have been launched above the Atlantic, Pacific, and Indian Oceans. These networks cover almost all countries around the world. Features For international communication, a submarine cable along the Atlantic Ocean was installed in 1857. Also, short–wave radio communication (invented by Marconi in 1886) has been in use. However, short wave radio communication has disadvantages of:

• Small transmission capacity; only small telephone channels can be used to transmit.

• Fading in wave propagation; interferes with stability of transmission. Although over–the–horizon propagation is used for short distance international communications, it is impossible to apply it to transoceanic long distance communications.

Unlike other system, geo-stationary satellite communication systems have the following features

• Stable and large capacity communication. • Costs of establishment and maintenance do not depend on

communication distance. The costs of submarine and over–the–horizon systems are proportional to the length, but those of the satellite system do not affect the communication distance. Therefore, the satellite system is ideal for long distance communications.

• Multiple accesses are possible. Signals sent from an earth station can be received at several earth stations simultaneously. Therefore, it can transmit signals to many stations simultaneously, such as TV. Actually, increasing of submarine cable's capacity and distance between repeaters, can make submarine cables competitive to satellite communication specially when very large capacity is required but for small traffic size countries, satellite


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communication is unavailable for the independent communication services.

4.2 Advantages of Satellite Communications Large coverage: Almost one–third of the earth with exception of polar regions is visible from geostationary orbit. It is, thus, possible to cover about 10,000 kms. distance irrespective of intervening terrain with a single satellite. High quality: Satellite links can be designed for high quality performance. The link performance is highly stable since it is free from ionospheric disturbances, multipath effects or fading. High reliability: Reliability is high since there is only one repeater in the link. High capacity: With microwave frequencies, wide bandwidths are available and large communication capacity can be obtained. Flexibility: In a terrestrial system, communication is tied down to the links installed. On the other hand, satellite communication is well suited for changing traffic requirements, locations and channel capacities. Speed of installation: Installation of earth terminals can be achieved in a short time as compared to laying of cables or radio relay links. Mobile, short–term or emergency communications: With airliftable or road transportable terminals, short–term or emergency communications can be quickly provided. Reliable long distance land mobile, maritime mobile and aeronautical mobile services are feasible only by means of satellite. Satellite communication is ideally suited for point to multipoint transmission on broadcasting over large areas. Application of satellites for TV broadcasting, audio and video distribution and teleconferencing, facsimile, data and news dissemination is, therefore, increasing rapidly. All types of common services are possible. 4.3 Satellite Communication Network Satellite Communication Network could be defined as an ensemble of earth stations of pre–determined size spread over a pre–defined coverage area, interconnected through a suitably designed satellite, placed at a pre–determined location in properly chosen orbit around the earth. Thus, two important elements of a satellite communication network are:

• Space Segment • Ground Segment


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Uplinks and Down Link Uplink is the radio path from Ground segment, i.e. earth station to the Space segment, i.e. satellite, whereas Downlink is the radio path from space segment, i.e. satellite to the ground segment, i.e. earth station. 4.4 Frequency Bands Choice of Frequency band for space communication depends upon

• Band–width required. • Noise consideration • Propagation factors • Technological developments with regard to component and

device. As the signal levels from the satellite are expected to be very low, any natural phenomenon to aid the reception of the incoming signals must be exploited. Between the frequencies of 2 GHz to 10 GHz, the level of the sky–noise reduces and this band of frequencies is known as the 'microwave window'. The most of the communication satellites as on today are using a frequency of 6 GHz for "Up link" and 4 GHz for "Down link" transmission. These frequencies are preferred because of

• Less atmospheric absorption than higher frequency. • Less noise both galactic and manmade. • Less space loss compared to higher frequency. • A well developed technology available at these frequencies. • 6 GHz/4 GHz bands are shared with terrestrial services,

creating interference problem. • As equatorial orbit is filling with geostationary satellites, RF

interference is increasing from one satellite system to another is increasing.

• 14/11 and 30/20 GHz systems for telecommunication and broadcasting satellite services are slowly coming.


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4.5 Frequency bands in use for satellite communication are:

"L" BAND 1830–2700 MHz

"S" BAND 2500–2700 MHz Insat is Using

"C" BAND 5925–6425 MHz UP 3700–4200 MHz DOWN

Insat is Using

"X" BAND 7900–8400 UP 7250–7750 DOWN

"KU" BAND

14.000–14.500 Hz. UP 10950–11200 GHz/DN. 11450–11700 GHz/DN.

"K" BAND 27.5–30 GHz UP 17.7–21.2 GHz DOWN

Extended C BAND 6725–7025 UP 4500–4800 DOWN

Insat is Using

V BAND 40–51 GHz UP 40–41 GHz DOWN

V Band Inter-satellite 59–64 GHz 54–58 GHz

4.6 Time Delay

The total earth–satellite–earth path length may be as much as 74,000 km thus giving a one–way propagation delay of 250 ms. The effect of this delay on telephone conversations, where a 500 ms gap can arise between one person asking a question and hearing the other person reply, has been widely investigated, and was found to be less of a problem than had been anticipated. With geostationary satellites, two–hop operation sometimes unavoidable and gives rise to a delay of over one second.


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4.7 Geographical Advantage A station which is located closer to the sub–satellite point, as demonstrated in Fig. 9.5 will have an advantage in received signal level with respect to one at the edge of the service area of the satellite. For a global coverage satellite, this can be as much as 4.3 dB.

Fig. 9.5 Example of Geographical Advantage 4.8 Kinds and Systems of Communication Satellite Kinds of Communication Satellites – depends on type of orbit and frequency band used. During the early experimental stage of communication satellites, a passive satellite was used without any amplifiers and it only reflected radio waves sent from the earth station. But, later on active satellite with amplifiers was developed and put into practical use. Communication Satellite can be classified by the orbit used and also by frequency band used. Before discussing satellite orbits in a more generalized manner, however, it is necessary to be aware of the natural laws that control the movement of satellites. These are based on Kepler's laws and basically stated are:

• The orbit plane of any earth satellite must bisect the Earth centrally.

• The Earth must be at the centre of any orbit. The choice of orbit is restricted to three basic types, namely: polar, equatorial and inclined as illustrated in Fig. 9.6. The actual shape of the orbit is limited to circular and elliptical. Any combination of type and shape is possible but observations are made only of the circular polar, elliptically inclined and the circular equatorial.


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Fig. 9.6 Three Basic Orbits

Circular polar orbit This is the only orbit that can provide full global coverage by one satellite, but requires a number of orbits to do so. In a communications sense where instantaneous transfer of information is required, full global coverage could be achieved with a series of satellites, where each satellite is separated in time and angle of its orbit. However, this produces economic, technical and operational disadvantages and is thus not used for telecommunications though it is favored for some navigation, meteorological and land resource satellite system. Elliptically inclined orbit An orbit of this type has unique properties that have been successfully used for some communications satellite system, notably the Russian domestic system. For this system, the elliptical orbit has an angle of inclination of 63 degrees and a 12–hour orbit period. By design, the satellite is made to be visible for eight of its 12–hour orbit period to minimize the handover problem while providing substantial coverage of the temperate and polar regions. By using three satellites, suitably phased, continuous coverage of particular temperate region can be provided that would not be covered by other orbits. The elliptically inclined orbit is used exclusively by the Russians for their Orbital and Molniya systems, but since coverage is limited to particular areas (higher latitudes), it is, therefore, not suitable for a global network. Circular Equatorial Orbit Circular orbits in the equatorial plane permit fewer satellites and ground stations to be used, and satellites with long orbital periods (at high altitudes) have greater mutual visibility. A satellite in a circular orbit at 35,800 km has a period of 24 hours and consequently appears


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stationary over a fixed point on the earth's surface. The satellite is visible from one third of the earth's surface, up to the Arctic circle, and this orbit is almost universally preferred for satellite communications system. Stabilization of the satellite is necessary since the earth is not truly spherical, and the moon, sun and the earth's tidal motion have gravitational effects on the satellite, tending to make it drift from its correct position. Inclination to the equatorial plane produces a sinusoidal variation in longitude, seen from earth as motion around an ellipse once every 24 hours, with peak deviation equal to the inclination angle. Incorrect velocity results incorrect altitude, and a drift to the east or to the west. When a non re–usable launcher is utilized, injection of the satellite into geostationary orbit requires two rocket burns: the first to get the vehicle into a parking orbit, and the second via an elliptical transfer orbit to geostationary altitude. The spacecraft's own apogee motor then increases its velocity to about 10,000 fps to maintain the geostationary orbit. When launched from the Space Transportation System (Shuttle), a booster rocket is attached to the satellite to boost it to the geostationary orbit. The satellite must then be correctly positioned and held in position for its required lifetime (typically 7 to 10 years). This is done by using hydrazine (liquid nitrogen plus ammonia) and cold gas jets. About 40 lbs. of hydrazine are required for corrections to maintain geostationary position within q 0.1x for five years, but since hydrazine is also used for initial positioning, the quantity available depends on the accuracy of the launch. To extend the life of the satellites, less frequent corrections may be made allowing the satellite to drift. F = Noise Figure of Receiver. The antenna noise is expressed in degrees kelvin and is called noise temperature of antenna. It can be converted to familiar units of power, watts by multiplying it with Boltzmann's constant K = 1.38 x 10–29 joule/kelvin and the bandwidth. Noise temperature of an antenna is of the order of 20–50oK. Geostationary Satellite This satellite revolves above the equator round the earth at a height of 35,790 km. Its period of revolving round the earth is same as that of the earth rotation on its own axis. Therefore, it looks as if it is stationary. This system was contributed to the "WIRELESS WORLD" by Mr. A.C.Clark, Dr. Rosen (an American) and others. It launched a Syncom communication satellite in 1963. Syncom No. 1 failed to launch in February, 1963. But, Syncon No. 2 finally succeeded in July 1963. This satellite centered the equator and moved like a figure eight (8). This was not a complete geostationary satellite, but it came into practical use (24 hours) as synchronous satellite.


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This satellite is advantageous because:

• Its large antenna at an earth station is easy to track. • Twenty–four (24) hours communication can be made with

even only one satellite. • The satellite looks at the earth as if it was stationary, and it

radiates highly effective wave power. • Visibility from one (1) satellite is very wide, and global

communication can be made using only three (3) satellites. Its drawback, however, is its delay caused in long distance transmission. But, the system is economical and accordingly, it is widely used for both international and regional domestic communications 5. VSAT System Components and Specifications The C–200 Series Micro Earth Station is a part of two way data communication system. The network consists of:

• Master Earth Station • Micro Earth Station • A Geosynchronous Satellite

It operates at C–band Microwave frequency which travels in straight line. To cover large areas, modern satellite communication systems use geosynchronous communication satellite as a repeater. The satellite is positioned 36,000 km above the earth equator. The satellite can 'see' a very large portion of the earth's surface and hence the signal transmitted from it reaches the Micro Earth Stations located at far off places. The technology used is spread spectrum multiple access (SSMA) method by which the size of the antenna is considerably reduced. This permits C–200 series to transmit and receive signals regardless of its location as long as the antenna physically points at the satellite and is not blocked by any obstructions. Fig. shows a typical data communication network.


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Fig. 9.7 Simplified Data Communication Network 5.1 Multi Channel per Carrier – VSAT (MCPC–VSAT) Department of Telecommunications has decided to provide reliable telecommunication services to sub divisional H.Q./Tahsil/Tourist Centre/Pilgrimage places as one of the major objective of its eighth five year plan, viz. 1992–1997. Most of such places are planned to be provided with C–DOT 126 RAX or 64 lines – MLT Local Telephone Exchange depending upon the local Telecommunication need. The local Exchange is to be interconnected to either district HQ or State capital through a reliable long distance trunk media. A good number of stations under the category of such sub divisional HQ Tahsil, tourist centre, pilgrimage places etc. are located in remote hilly and in accessible terrain which are neither feasible nor techno–economically viable on terrestrial media. Such station have been planned and engineered to be connected on Satellite media using Multichannel per carrier Very Small Aperture Terminal (MCPC–VSAT)


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technology. About 80 earth stations may be required to be linked with their respective state capital through satellite media during 8th Five Year Plan. 5.2 General Features of MCPC – VSAT The MCPC–VSAT shall operate in normal C–frequency band as given below:

• UPLINK 5.925–6.425 GHz • DOWNLINK 3.7–4.2 GHz

The MCPC–VSAT shall operate in a star configuration and shall provide connectivity on a pre assigned mode between a hub station located at main station (11 Mtr. dia. Antenna and G/T > 31.7 dB/degree K) of INSAT Network. A block schematic giving the system concept is given in Fig. The MCPC–VSAT shall be compact and it can be installed either in a container along with Local telephone exchange or in a separate building. A block diagram of MCPC–VSAT is given in Fig.2. The MCPC–VSAT shall consist of 7 channels each with voice digitised at the rate of 16 Kb/s and 1 low bit rate data channel. With the rate of ½ FEC coding transmission would be at the rate of 256 Kb/s. The MCPC–VSAT shall work with the power supply of –48V with 10% variation DC supply, available for local telephone exchanges at the site. 5.3 MCPC-VSAT Parameters

1. G/T = 15.0 dB/degree K 2. Transmit Frequency = 5.925-6.425 GHz 3. Receive Frequency = 3.700-4.200 GHz 4. Network Architecture = Star 5. Modulation = QPSK 6. Access = PAMA – Pre assigned 7. Digital Transmission = 7 clear voice channels and

one data channel. Voice digitised at 16 Kb/s rate.


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6. HVNET – DOT VSAT Network This is the first High Speed Satellite based VSAT network of Department of Telecom., Govt. of India. It provides for high speed data transfers and voice communication covering the entire country.

Fig. 8.8 HVNET Architecture

The DoT VSAT network consists of a HUB Station located at Yeur Earth Station of Dept. of Telecommunication near Thane (about 40 kms from Mumbai) and number of VSATs/Personal Earth Stations (PES) located throughout the country. The VSAT communicate to the Hub through the INSAT Satellite. All VSATs are connected in STAR topology and VSAT to VSAT communication is through HUB at Mumbai. The VSAT which is required to be installed at subscriber’s premises consists of three units, namely an Outdoor Unit, an Indoor Unit and Inter Facility Link (IFL). Cable interconnecting the two Units along with a 2.4 meter diameter Antenna assembly and can be installed easily in any open space and requires a floor area of about 4 mt x 4 mt. The IFL cable, which carries the telecom signals and power supply, the IFL cable can be up to 100 meters long.

YEUR

THANE HUB

STATION

GPSS

INTERNET

PSTN

RABMN

TELEX

I-NET

INSATINSAT

GUJARAT

VSAT

J & K

VSAT

NEWDELHI

VSAT

X.25

X.28

X.25

FAX

PC

HVNET SYSTEMHVNET SYSTEM

PersonalEarth

Stations

FEEDER

YEUR

THANE HUB

STATION

GPSS

INTERNET

PSTN

RABMN

TELEX

I-NET

INSATINSAT

GUJARAT

VSAT

J & K

VSAT

NEWDELHI

VSAT

X.25

X.28

X.25

FAX

PC

HVNET SYSTEMHVNET SYSTEM

PersonalEarth

Stations

FEEDER


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Module - 01


Chapter – 10

Introduction to Electronic Exchanges


• Development of Electronic Exchanges • Influence of Electronics in Exchange Design. • Facilities provided by Electronic Exchanges. • Constraints of Electronic Exchanges • Working principles of Electronic Exchanges • Software for SPC Exchanges


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1. Development of Electronic Exchanges To overcome the limitations of manual switching; automatic exchanges, having Electro-mechanical components, were developed. Strowger exchange, the first automatic exchange having direct control feature, appeared in 1892. Though it improved upon the performance of a manual exchange it still had a number of disadvantages, viz., a large number of mechanical parts, limited availability, inflexibility, bulky in size etc. As a result of further research and development, Crossbar exchanges, having an indirect control system, appeared in 1926. The Crossbar exchange improved upon many shortcomings of the Strowger system. However, much more improvement was expected and the revolutionary change in field of electronics provided it. A large number of moving parts in Register, Marker, Translator, etc., were replaced en-block by a single computer. This made the exchange smaller in size, volume and weight, faster and reliable, highly flexible, noise-free, easily manageable with no preventive maintenance etc. The first electronic exchange employing Space-Division switching (Analog switching) was commissioned in 1965. This exchange used one physical path for one call and, hence, full availability could still not be achieved. Further research resulted in development of Time-Division switching (Digital Switching) which enabled sharing a single path by several calls, thus providing full availability. The first digital exchange was commissioned in 1970 in France.

Table 2: Development of Electronic Exchanges (Major exchanges)

MODEL Digital Exchanges

Capacity (in thousands) Lines Trunks

Traffic Call Erlangs Attempts per second

E-10B 30 4 2,400 25 Mentaconta 10-60 - 10,000 28-60 System X 100 60 25,000 800000 AXE -10 64 - 26,000 800000 FETEX-150L 290 60 24,000 1800000 OCB-283 200 60 25,000 800000 EWSD 250 60 25,200 1000000 NEAX-61E 100 60 27,000 1000000


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TABLE 5- Advantages of Electronic Exchange over

Electromechanical Exchanges

Electromechanical Exchanges Electronic Exchanges

• Category, Analysis, Routing, translation, etc; done by relays.

• Any changes in facilities require addition of hardware and/or large amount of wiring change. Flexibility limited.

• Testing is done manually externally and is time consuming. No logic analysis carried out.

• Partial full-availability, hence blocking.

• Limited facilities to the subscribers.

• Slow in speed. Dialing speed is max. 11 digits and switching speed is in l milliseconds.

• Switch room occupies large volume.

• Lot of switching noise. • Long installation and testing

time. • Large maintenance effort and

preventive maintenance necessary.

• Translation, speech path Sub’s Facilities, etc., managed by MAP and other DATA. • Changes can be carried out by simple commands. A few changes can be made by Subs himself. Hence, highly flexible. • Testing carried out periodically automatically and analysis printed out. • Full availability, hence no blocking. A large number of different types of services possible very easily. • Very fast. Dialing speed up to 11 digits /sec possible. Switching is achieved in a few microseconds. • Much lesser volume required floor space of switch room reduced to about one-sixth. • Almost noiseless. • Short installation and testing period. • Remedial maintenance is very easy due to plug-in type circuit boards. Preventive maintenance not required.

2. Influence of Electronics in Exchange Design. When electronic devices were introduced in the switching systems, a new concept of switching evolved as a consequence of their extremely high operating speed compared to their former counter-parts, i.e., the Electro-mechanical systems, Relays, the logic elements in the electromechanical systems, have operate and release times which are roughly equal to the duration of telephone signals to maintain required accuracy. However, to achieve the requisite simultaneous call processing capacity, it became essential for such system to have number of such electrical control units (Called registers in a Cross-bar Exchange), in parallel, each handling one call at a time. In other words, it was necessary to have an individual control system to process each call.


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Electronic logic components on the other hand, can operate a thousand or ten thousand times during a telephone signal. This led to a concept of using a single electronic control device to simultaneously process a number of calls on time-sharing basis. Though such centralisation of control is definitely more economical it has the disadvantage of making the switching system more vulnerable to total system failure. This can; however be overcome by having a stand by control device. Another major consequence of using electronics in control subsystems of a telephone exchange was to make it technically and economically feasible to realize powerful processing units employing complex sequence of instructions. Part of the control equipment capacity could then be employed for functions other than call processing, viz., exchange operation and maintenance. It resulted in greatly improved system reliability without excessively increasing system cost. This development led to a form of centralized control in which the same processor handled all the functions, i.e., call processing, operation and maintenance functions of the entire exchange. In the earlier versions of electronic control equipment, the control system was of a very large size, fixed cost unit. It lacked modularity. It was economically competitive for very large capacity exchanges. Initially, small capacity processors were costlier due to high cost per bit of memory and logic gates. Therefore, for small exchanges, processor cost per line was too high. However, with the progressive development of the small size low cost processor using microprocessor, it became possible to employ electronic controls for all capacities. In addition control equipment could also be made modular aiding the future expansion. The impact of electronics on exchanges is not static and it is still changing as a function of advances in electronic technology. 3. Facilities provided by Electronic Exchanges. Facilities offered by electronic exchanges can be categorised in three parts.

a) Facilities to the Subscribers. b) Facilities to the Administration. c) Facilities to the Maintenance Personnel.

a) Facilities to the Subscribers. MFC Push-button Dialing. All subscribers in an electronic exchange can use push-button telephones, which use Dual Tone Multi- frequency, for sending the dialed digits. Sending of eleven digits per second is possible, thus increasing the dialing speed.


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Priority Subscriber Lines Priority Subscribers lines may be provided in electronic exchanges. These subscribers are attended to, according to their priority level, by the central processor, even during heavy congestion or emergency. Toll (Outgoing Call) Restriction All subscribers can avail of the facility of toll restriction or blocking of subscriber line for specific types of outgoing traffic, viz., long distance STD calls. This can be easily achieved by keying-in certain service codes. Service Interception Incoming calls to a subscriber can be automatically forwarded during his absence, to a customer service position or a recorded announcement. The customer service position answers the calls and forwards any message meant for the subscriber. Abbreviated Dialing Most subscribers very often call only limited group of telephone numbers. By dialing only prefix digit followed by two selection digits, subscribers can call up to 100 predetermined subscribers connected to any automatic exchange. This shortens the process of dialing all the digits. Call Forwarding The subscriber having the call forwarding facility can keep his telephone in the transfer condition in case he wishes his incoming calls to be transferred to another telephone number during his absence. Do Not Disturb This service enables the subscriber to free himself from attending to his incoming calls. In such a case, the incoming calls are routed to an operator position or a talking machine. This position or machine informs the caller that called subscriber is temporarily inaccessible. Conference Calls Subscribers can set up connections to more than one subscriber and conduct telephone conferences under the provision of this facility. Camp On Busy Incoming call to a busy subscriber can be “Camped on” until the called subscriber gets free. This avoids wastage of time in redialing a busy telephone number.


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Call Waiting The ‘Call Waiting’ service notifies the already busy subscriber of a third party calling him. He is fed with a special tone during his conversation. It is purely his choice either to ignore the third party or to interrupt the existing connection and have a conversation with the third party while holding the first party on the line. Call Repetition Instead of camp on busy a call can automatically be repeated. The calling party can replace his hand set after receiving the busy tone. A Periodic check is carried out on the called party’s status. When idle status is ascertained, the connection is set up and ringing current fed to both the parties. Third party Inquiry This system permits consultation and the transfer of call to other subscribers. Consultation can be initiated by means of a special signal from the subscriber telephone and by dialing the directory number of the desired subscriber without disconnecting the previous connection. Priority of calls to Emergency Positions Emergency calls such as ambulance, fire, etc., are processed in priority to other calls. Subscriber charge Indicator By placing a charge indicator at the subscriber’s premises the charges of each call made can be ascertained by him. Call Charge printout or immediate Billing The subscriber can request automatic post call charge notification in the printout form for individual calls or for all calls. The information containing called number, date and time, and the charges can be had on a Tele-type-write. Malicious Call Identification Malicious Call Identification is done immediately and the information is obtained in the printout from either automatically or by dialing an identification code. Interception or Announcement. In the following conditions, an announcement is automatically conveyed to calling subscribers.

• Change of a particular number of transferred subscriber. • Dialing of an unallocated cods. • Dialing of an unobtainable number.


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• Route congested or out of order. • Subscriber’s line temporarily out of order. • Suspension of service due to non-payment.

Connection Without Dialing. This allows the subscribers to have a specific connection set up, after lifting the handset, without dialing. If the subscriber wishes to dial another number, then he has to start dialing within a specified time period, say 10 seconds, after lifting the handset. Automatic Wake Up. Automatic wake up service or morning alarm is possible, without any human intervention. Hot Line or Private Wire. Hot line service enables the subscriber to talk to a specific subscriber by only lifting the handset. This service cannot be used along with normal dialing facility. The switching starts as soon as the receiver is lifted. Denied Incoming Call A Subscriber may desire that no incoming call should come on a particular line. He can ask for such a facility so that he can use the line for making only outgoing calls. Instrument Locking A few subscribers may like to have their telephone sets locked up against any misuse. Dialing of a secret code will extend such a facility to them. Free of charge Calls Calls free of charge are possible on certain special services such as booking of complaints, booking of telegrams, etc. Collect call If so desired, the incoming subscriber is billed for all the calls made to him, instead of the calling subscriber. b) Facilities to the Administration Reduced Switch Room Accommodation Reduction in switch room accommodation to about 1/6th to 1/4th as compared to Crossbar system is possible.


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Faster installation and Easy Extension The reduced volume of equipment, plug-in assemblies for interconnecting cables, printed cards and automatic testing of exchange equipment result in faster installation (about six months for a 10,000 line exchange) Due to modular structure, the expansion is also easier and quicker Economic Consideration The switching speed being much faster as compared to Crossbar system, the use of principle of full availability of trunk circuits and other equipment makes the system economically superior to electromechanical systems. Automatic test of Subscriber line Routine testing of subscriber lines for Insulation, capacitance, foreign potential, etc., are automatically carried out during night. The results of the testing can be obtained in the printout form, the next day. c) Maintenance Facilities Fault Processing Automatic fault processing facility is available for checking all hardware components and complete internal working of the exchange. Changeover from a faulty sub-system to stand-by sub-system is automatically affected without any human intervention. Only information is given out so that the maintenance staff is able to attend to the faulty sub-system. Diagnostics Once a fault is reported by the system, ‘on demand’ programs are available which help the maintenance staff to localise the fault, who can replace the defective printed card and restore the faulty sub-system. The faulty card is attended at a centralised maintenance centre specifically equipped for this purpose. Statistical programs Statistical programs are available to gather information about the traffic conditions and trunks occupancy rate to assess and plan the solutions in cases of anticipated problems. This facility helps the maintenance and administration personnel to maintain a specified level of grade of service. Blocking In case of congestion or breakdown of a specific route, facility of blocking such routes is available in modes, such as


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(i) Blocking of a specified percentage of calls in such a route either automatically or manually. (ii) Blocking a specific category of subscribers. Overloading Security Overloading of central processor in an electronic exchange can lead to disastrous results. To prevent this, central processor occupancy is measured automatically periodically, when it exceeds a specified percentage, audio-visual alarms are activated, in addition to printing out the message. Maintenance personnel have the following options. (i) Block some of the facilities temporarily, or (ii) Reduce the load by blocking some of the congested routes. 4. Constraints of Electronic Exchanges Though there are a number of definite advantages of Electronic exchanges, over the electromechanical exchanges, there are certain constraints, which should be considered, at the planning stage for deciding between the two systems. Traffic Handling Capacity Apparently, the traffic handling capacity of an exchange is limited by the number of subscriber lines and trunks connected to the switching network, and the number of simultaneous paths available through the switching network. However, in electronic exchanges, the prime limitation is the number of simultaneous calls, which can be handled by the control equipment, as it has to execute a number of instructions depending on the type of the call. Therefore the extent of loading of the exchange will be guided solely by the amount of processor loading. Moreover, the facilities to the subscribers will also have to be limited accordingly. Power Supply The power supply should be highly stable for trouble free operation as the components are sensitive to variations beyond +10%. It is almost essential to have a stand-by power supply arrangement. Total Protection from Dust All possible precautions should be observed for ensuring dust-free environment. Temperature and Humidity Control Due to the presence of quiescent current in the components and because of their compactness. Heat generated per unit volume is highest in electronic exchanges. Moreover, as the component characteristics drift substantially with the temperature and humidity, the air-conditioning


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load is higher. Obviously, the air-conditioning system should be highly reliable and preferably there should be a stand-by arrangement. The installation is also carried out in air-conditioned environment. Static Electricity and Electromagnetic interference. Due to the presence of static electricity on the body of persons handling the equipment, the stored data may get vitiated. Handling of PCB’s therefore, should be done with utmost care and should be minimised care should also be taken to protect the cards from exposure to stray electromagnetic fields. PCB Repair The repair of PCB’s is extremely complicated and sophisticated equipments are required for diagnosing the faults. This results in having costly inventory and a costly repair centre. With the frequent improvement and changes in the cards, proper documentation of cards becomes essential. 5. Working principles of Electronic Exchanges a) Introduction The prime purpose of an exchange is to provide a temporary path for simultaneous and bi-directional transmission of speech between (i) Subscriber lines connected to same exchange (local switching) (ii) Subscriber lines and trunks to other exchange (outgoing trunk call) (iii) Subscriber lines and trunks from other exchanges (incoming trunk calls) and (iv) Pairs of trunks towards different exchanges (transit switching) These are also called the switching functions of an exchange and are implemented through the equipment called the switching network. An exchange, which can setup just the first three types of connections. is called a Subscriber or Local Exchange. If an exchange can setup only the fourth type of connections, it is called a Transit or Tandem Exchange. The other distinguished functions of an exchange are

i) Exchange of information with the external environment (Subscriber lines or other exchanges) i.e. signaling. ii) Processing the signaling information and controlling the operation of signaling network, i.e. control, and iii) Charging and billing

All these functions can be provided more efficiently using computer controlled electronic exchange, than by the conventional electromechanical exchanges.


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b) Stored Programme Controlled (SPC) Exchange In electromechanical switching, the various functions of the exchange are achieved by the operation and release of relays and switch (rotary or crossbar) contacts, under the direction of a Control Sub-System. These contacts are hard - wired in a predetermined way. The exchange dependent data, such as, subscriber’s class of service, translation and routing, combination signaling characteristics, are achieved by hard-ware and logic, by a of relay sets, grouping of same type of lines, strapping on Main or Intermediate Distribution Frame or translation fields, etc. When the data is to be modified, for introduction of a new service, or change in services already available to a subscriber, the hardware change ranging from inconvenient to near impossible, are involved. In an SPC exchange, a processor similar to a general-purpose computer is used to control the functions of the exchange. All the control functions represented by a series of various instructions are stored in the memory. Therefore the processor memories hold all exchange-dependent data such as subscriber date, translation tables, and routing and charging information and call records. For each call processing step. e.g. for taking a decision according to class of service, the stored data is referred to, Hence, this concept of switching. The memories are modifiable and the control program can always be rewritten if the behavior or the use of system is to be modified. This imparts and enormous flexibility in overall working of the exchange. Digital computers have the capability of handling many tens of thousands of instructions every second, hence, in addition to controlling the switching functions the same processor can handle other functions also. The immediate effect of holding both the control programme and the exchange data, in easily alterable memories, is that the administration can become much more responsive to subscriber requirements, both in terms of introducing new services and modifying general services, or in responding to the demands of individual subscriber. For example, to restore service on payment of an overdue bill or to permit change from a dial instrument to a multi frequency sender, simply the appropriate entries in the subscriber data-file are to be amended. This can be done by typing- in simple instructions from a teletypewriter or visual display unit. The ability of the administration to respond rapidly and effectively to subscriber requirements is likely to become increasingly important in the future. The modifications and changes in services which were previously impossible are achieved very simply in SPC exchange, by modifying the stored data suitably. In some cased, subscribers can also be given the facility to modify their own data entries for supplementary services, such as on-demand call transfer, short code, (abbreviated) dialing, etc.


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The use of a central processor also makes possible the connection of local and remote terminals to carry out man-machine dialogue with each exchange. Thus, the maintenance and administrative operations of all the SPC exchanges in a network can be performed from a single centralised place. The processor sends the information on the performance of the network, such as, traffic flow, billing information, faults, to the centre, which carries out remedial measures with the help of commands. Similarly, other modifications in services can also be carried out from the remote centre. This allows a better control on the overall performance of the network. As the processor is capable of performing operations at a very high speed, it has got sufficient time to run routine test programmes to detect faults, automatically. Hence, there is no need to carry out time consuming manual routine tests. In an SPC exchange, a single processor can replace all control equipment. The processor must, therefore, be quite powerful; typically, it must process hundreds of calls per second, in addition to performing other administrative and maintenance tasks. However, totally centralised control has drawbacks. The software for such a central processor will be voluminous, complex, and difficult to develop reliably. Moreover, it is not a good arrangement from the point of view of system security, as the entire system will collapse with the failure of the processor. These difficulties can be overcome by decentralising the control. Some routine functions, such as scanning, signal distributing, marking, which are independent of call processing, can be delegated to auxiliary or peripheral processors. These peripheral units, each with specialised function, are often themselves controlled by small stored programmes processors, thus reducing the size and complexity at central control level. Since, they have to handle only one function, their programmes are less voluminous and far less subjected to change than those at central. Therefore, the associated programme memory need not be modifiable (generally, semiconductors ROM's are used). c) Block Schematic of SPC Exchange Despite the many difference between the electronic switching systems, and all over the world there is a general similarity between most of the systems in terms of their functional subdivisions. In it’s simplest form, an SPC exchange consists of five main sub-systems, as shown in Fig 10.1.

• Terminal equipment provides on individual basis for each subscriber line and for inter-exchange trunk.

• Switching network may be space- division or time-division, uni-directional or bi-directional.


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• Switching processor, consisting mainly of processors and memories.

• Switching peripherals (Scanner, Distributor and Marker), are Interface Circuits between control system terminal equipment and switching network.

• Signaling interfaces depending on type of signaling used, and • Data Processing Peripherals (Tele - typewriters, Printers, etc.) for

man-machine dialogue for operation and maintenance of the exchange.

Terminal Equipment. In this equipment, line, trunk, and service circuits are terminated, for detection, signaling, speech transmission, and supervision of calls. The Line Circuits carry out the traditional functions of supervising and providing battery feed to each subscriber line. The Trunk Circuits are used on outgoing, incoming and transit calls for battery feed and supervision. Service Circuits perform specific functions, like; transmission and reception of Decadic dial pulses or MF signals, which may be economically handled by a specialised common pool of circuits. In contrast to electromechanical circuits, the Trunk and Service circuits in SPC exchanges are considerably simpler because functions, like counting, pulsing, timing charging, etc, are delegated to stored programme. Switching Network. In an electronic exchange, the switching network is one of the largest sub-systems in terms of size of the equipment. Its main functions are

• Switching, i.e., setting up temporary connection between two or more exchange terminations

• Transmission of speech and signals between these terminations, with reliable accuracy.

There are two types of electronic switching system. viz. Space division and Time Division. d) Space Division switching System. In a Space Division Switching system, a continuous physical path is set up between input and output terminations. This path is separate for each connection and is held for the entire duration of the call. Path for different connections is independent of each other. Once a continuous path has been established, signals are interchanged between the two terminations. Such a switching network can employ either metallic or electronic cross-points. Presently, usage of metallic cross-points, viz., reed relay, mini-cross bar derivative switches etc, is favored. They have


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the advantage of compatibility with the existing line and trunk signaling conditions in the network. e) Time Division Switching System. In Time Division Switching, a number of connections (calls) share the same path on time division sharing basis. The path is not separate for each connection, rather, is shared sequentially for a fraction of a time by different calls. This process is repeated periodically at a suitable high rate. The repetition rate is 8 Khz, i.e. once every 125 microseconds for transmitting speech on telephone network, without any appreciable distortion. These samples are time multiplexed with staggered samples of other speech channels, to enable sharing of one path by many calls. Pulse Amplitude Modulation (PAM) switching initially accomplished the Time Division Switching. However, it still could not overcome the performance limitations of signal distortion noise, cross-talk etc. With the advent of Pulse Code Modulation (PCM), the PAM signals were converted into a digital format overcoming the limitations of analog and PAM signals. PCM signals are suitable for both transmission and switching. The PCM switching is popularly called Digital Switching.

Fig. 10.1 Functional sub-divisions of an SPC exchange f) Compatibility with Existing Network In this area, the application of electronic techniques has encountered the greatest difficulty. To appreciate the reasons, let us consider the basic requirements of a conventional switching network.


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• High OFF resistance and low ON resistance. • Sufficient power handling capacity for transmitting ringing current,

battery feed etc., on subscriber lines. • Good frequency response (300-3400 KHz) • Bi-directional path (preferable) • D.C. signaling path to work with existing junction equipment

preferable) • Economy • Easy to control. • Low power consumption, and • Immunity to extraneous noise, voltage surges.

The present day electronic devices cannot meet all these requirements adequately. Electronic devices can easily meet most requirements. These considerations show that substitutions of the analog mode of electromechanical switching network by fully electronic equipment are not, straight way practical. The main virtue of the existing electromechanical devices is their immunity to extraneous noise voltage surge, etc., which are frequently experienced in a telephone network. Moreover, metal contact switches offer little restriction on the voltages and currents to be carried. In the existing network and subscriber handsets, typically, 80 volt peak to peak ringing current is required to be transmitted on the line. This is difficult, if not impractical, for electronic switches to handle. Therefore, to avail of the advantages of the electronic exchanges, either of the two following alternatives may be adopted.

• Deploy a new range of peripherals/ equipments, suited to the characteristics of the electronic switching devices, on one hand, and requirements of telephone network on the other hand. i.e. employ Time Division Switching systems, or

• Continue to use metal contact switches, while other sub-systems may be changed to electronic. i.e. semi-electronic type of exchanges rather than fully electronic exchanges, to employ Space Division Switching Systems.

g) Switching Processor The switching processor is a special purpose real time computer, designed and optimised for dedicated applications of processing telephone calls. It has to perform certain real time functions (which have to be performed at the time of occurrence and cannot be deferred), such as, reception of dialed digits, and sending of digits in case of transit exchange. The block schematic of a switching processor, consisting of central control programme store is shown in Fig. 10.2.


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Fig. 10.2 Switching Processor Central Control (CC) is a high-speed data processing unit, which controls the operation of the switching network. In Programme store, sets of instructions called programmes are stored. The programmes are interpreted and executed by the central control. Data Store provides for the temporary storage of transient data, required in processing telephone calls, such as digits dialed by the subscriber, busy / idle states of lines and trunks etc. Translation Store contains information regarding lines. e.g. category of calling and called line routing code, charging information, etc. Data Stores is temporary memory, whereas Translation and Programme Stores are of semi-permanent type. The information in the Semi-permanent memories does not change during the processing of the call, but the information in Data Store changes continuously with origination and termination of each call. h) Switching Peripheral Equipment The time interval, in which the processor operates, is in the order of microseconds, while the components in the telephone switching section operate in milliseconds (if the switching network is of the analog type). The equipment, known as the switching peripheral, is the interface between these two equipments working at different speeds. The interface equipment acts as speed buffer, as well as, enables conversion of digital logic signals from the processor to the appropriate electrical signals to operate relays and cross-points, etc. Scanners, Signal distributors and Marker fall under this category of devices. Scanner Its purpose is to detect and inform CC of all significant events / signals on subscriber lines and trunks connected to the exchange. These

Central control Processor

Programme Store

Translation Store

Data Store

To Switching


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signals may either be continuous or discrete. The equipments at which the events / signals must be detected are equally diverse.

• Terminal equipment for subscriber lines and inter-exchange trunks and

• Common equipment such as DTMF (Dual - Tone Multi Frequency) or MFC digit receivers and inter-exchange signaling senders / receivers connected to the lines and trunks.

In view of this wide diversity in the types of lines trunks and signaling, the scanning rate, i.e. the frequency at which scan points are read, depends upon the maximum rate at which events / signals may occur. For example, on a subscriber line, with Decadic pulse signaling with 1:2 make -break ratio, the necessary precision, required for pulse detection, is of the order of ten milliseconds, while other continues signals (clear, off hook, etc.) on the same line are usually several hundred milliseconds long and the same high precision is not required. To detect new calls, while complying with the dial tone connection specifications, each line must be scanned about every 300 milliseconds. It means that in a 40,000 lines exchange (normal size electronic exchange) 5000 orders are to be issued every 300 milliseconds, assuming that eight lines are scanned simultaneously. Marker Marker performs physical setup and release of paths through the switching network, under the control of CC. A path is physically operated only when it has been reserved in the central control memory. Similarly, paths are physically released before being cleared in memory, to keep the memory information updated vis-a-vis switching network, depending upon whether the switching is Time division or Space division, marker either writes information in the control memory of time and space stages. (Time Division Switching), or physical operates the cross - points (Space Division Switching) Distributor It is a buffer between high - speed - low - power CC and relatively slow-speed-high-power signaling terminal circuits. A signal distributor operates or releases electrically latching relays in trunks and service circuits, under the direction of central control. Bus System Various switching peripherals are connected to the central processor by means of a common system. A bus is a group of wires on which data and commands pulses are transmitted between the various sub- units of a switching processor or between switching processor and switching peripherals. The device to be activated is addressed by sending


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its address on the address bus. The common bus system avoids the costly mesh type of interconnection among various devices. Line Interface Circuits To enable an electronic exchange to function with the existing outdoor telephone network, certain interfaces are required between the network and the electronic exchange. Analogue Subscriber Line Interface The functions of a Subscriber Line Interface, for each two-wire line, are often known by the acronym: BORSHT

B : Battery feed O : Overload protection R : Ringing S : Supervision of loop status H : Hybrid T : Connection to test equipment

All these functions cannot be performed directly by the electronic circuits and, therefore, suitable interfaces are required. Transmission Interface Transmission interface between analogue trunks and digital trunks (individual or multiplexed) such as, A/D and D/A converters, are known as CODEC, These may be provided on a per-line and per-trunk basis or on the basis of one per 30 speech channels. Signaling Interfaces A typical telephone network may have various exchange systems (Manual, Strowger, Cross bar, electronic) each having different signaling schemes. In such an environment, an exchange must accommodate several different signaling codes. Signaling Initially, all signaling between automatic exchanges was Decadic i.e. telephone numbers were transmitted as trains of 1to 10 pulses, each train representing one digit. To increase the speed at which the calls could be set up, and to improve the reliability of signaling, compelled sequence multi frequency signaling system was then introduced. In this system, each signal is transmitted as a combination of 2 out of a group of say 5 or 6 frequencies. In both Decadic and multi frequency methods, the signals for each call are sent over a channel directly associated with the inter-exchange speech transmission circuit used for that call. This is termed as channel associated signaling. Recently, a different technique has been developed, known as common channel signaling. In this technique, all the signaling information for a number of calls is sent over a signaling link independent of the inter-exchange speech circuits. Higher transmission rate can be utilised to enable exchange of much larger amount of information. This results in faster call setup,


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introduction of new services, e.g.. abbreviated dialing, and more retrials ultimately accomplishing higher call completion rate, Moreover, it can provided an efficient means of collecting information and transmitting orders for network management and traffic engineering. Data Processing Peripherals Following basic categories of Data Processing Peripherals are used in operation and maintenance of exchange.

• Man - machine dialogue terminals, like Tele-typewriter (TTY) and Visual Display Units (VDU), are used to enter operator commands and to give out low-volume date concerning the operation of the switching system. These terminals may be local i.e. within a few tense of meters of the exchange, or remotely located. These peripherals have been adopted in the switching Systems for their ease and flexibility of operation.

• Special purpose peripheral equipment is, sometimes employed for carrying out repeated functions, such as, subscriber line testing, where speed is more important than flexibility.

• High-speed large capacity data storage peripherals (Magnetic Tape Drives, magnetic Disc Unit) are used for loading software in the processor memory.

• Maintenance peripherals, such as, Alarm Annunciates and Special Consoles, are used primarily to indicate that automatic maintenance procedure have failed and manual attention is necessary.

6. Software for SPC Exchanges a) Introduction The SPC software is classified, as per functions less than two heads, viz., operational software and support software. The Operational Software is the set of all programs necessary for operations of the exchange. It may further be divided into two broad categories viz., System Programs and Application Programs. The System Programs are more or less equivalent to the operating system of a conventional computer. System software consists of programs that facilitate the operation and use of the processor by Application programs. The application Programs handle Call Processing, Administration and Maintenance of the exchange. The support Software comprising of assembler, loader and simulation programs, etc., is located at a centralized place, known as Software Centre, to serve a group of SPC exchanges.


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b) System Programs System Programs work as interface between the Exchange Hardware and the Application Programs. They manage and co-ordinate the activities and features of the hardware and Application programs. This is achieved by ensuring various functions. Managing l/O Operations The system programs for managing all input/output operations are some-times called Device Handlers. They are a collection of routines, which connect the system and the user programs with peripheral devices. There is one device handler for each type of peripheral device in the hardware configuration for transferring data between the device and the main memory.

• Assigning memory and peripheral devices to active processes. • Protecting the system from hardware and software faults and

errors. • Managing man-machine communication with Exchange personnel. • Providing access to the data describing the state of the system.

Application Programs The application programs can be divided into three main categories. Call Processing Programs Call processing programs are meant for setting up, supervising, releasing and charging calls in conformity with the telephone service specifications. Administration Programs Administration programs perform traffic supervision and measurement, line and trunk testing, modifying semi-permanent data concerning subscriber lines and trunks, defining the exchange hardware configuration, modifying data in translation and routing tables, e.g., changing the way a call is routed between two exchanges, or modifying a subscriber’s service entitlements, or storing a full length number corresponding to an abbreviated dialing code. Maintenance programs Maintenance programs carry out fault detection and fault localisation by testing the exchange hardware, including the processor itself.


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Operational Software Breakup The basic purpose of an exchange is to set up and release telephone calls. The most important function of the Operational Software is; therefore, call processing, which includes detection of a call origination, signal processing, path-finding through the switching network, address and digit translation charging, supervision etc., and finally releasing the call. Although call processing takes up the major part of the processor time, the corresponding programs constitute only a small fraction (around 1/5 th) of total instruction of the Operational Software (Fig. 10.3.) Typical operational software of an exchange comprises of 3 lakhs to 5 lakhs instructions. It may be seen that administration and maintenance programs represent about two-thirds of the software. Importance of administration and maintenance functions and hence the size of the corresponding software is growing day by day as the exchanges are becoming more sophisticated. Fig. 10.3 Software for SPC Exchanges Essential characteristics SPC software must have a ‘real time’ operating system. It must be capable of processing large number of calls, simultaneously, and should have special features to ensure telephone services without interruption even when maintenance of on-line capacity extension processes is in progress. Real time constraints The software of an exchange must meet the specified traffic handling Capacity and quality of service specifications. The traffic handling capacities of control processors of the exchange are expressed in terms of number of calls to be processed per second or per hour. c) Quality of service Quality of service is quantified mainly by two parameters.


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• Percentage of the calls failed, with respect to successful calls at a specified traffic load, due to internal problems of the exchange, viz., error in processing or congestion in the system, etc.,

• Percentage of calls, which wait for dial tone more than a specified time, should be less than a prescribed value.

It means that an exchange must, not only process a certain number of calls per second, but also ensure their processing within a prescribed time limit determined by the exchange specifications. A system, whose maximum response time is predefined in this way, is called a Real time System. Hence, the software of an electronic exchange must have this real time feature. An electronic exchange has several real time processing stages, each differing in terms of time constraints. For example, the most strict real time constraints concerns signal and signal processing. The specified accuracy for some signals is of the order of 10 ms, i.e., it should be able to process a signaling event lasting for just 10ms. The real time constraints for the remaining call processing programs, e.g., connections of dial tone, etc., are less strict. The system must respond to the subscriber action, say within a second. Real time constraints for administration and maintenance functions are least stringent. These may be between a few seconds (answering an operator command from TTY) or a few minutes (starting routine testing of some equipment). The corresponding programs are therefore, run with low priority. d) Multiprogramming The control processors in an electronic exchange operate in the multiprogramming mode, i.e., many tasks (mostly relating to call processing) are active simultaneously. For example, in a 30,000 lines exchanges there may be 3000 calls in the speech phase while another 500 calls being released or established at any instant of time. It means that 3500 tasks are being carried out simultaneously. Moreover, the system must monitor all the calls in the memory such that when any change occurs in the external telephone environment, relating to the call, its status is change accordingly. In addition, a few administration and maintenance tasks may be active, e.g., operator may issue a command through TTY for testing line, or modifying routing table, etc. On other occasion, automatic testing and traffic measuring may be simultaneously active. These requirements warrant multiprogramming mode of operation. It is not possible to set up a call in one continues processing sequence because call establishing involves several elementary processing actions, each lasting a few tens or hundreds of milliseconds, separated by idle waiting time period. These waiting periods can last long


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up to a few seconds. If it were to process one call at a time, control processor will also remain idle for the corresponding time period. Instead, each time a call processing is waiting for an external event, the corresponding call processing is de-activated and other tasks corresponding to other calls are run. When the awaited event for the preceding calls is detected, it is re-activated and processing of the call resumes. Thus, many tasks are performed simultaneously. It may also be noted that control processor is a sequential device, i.e., capable of executing only one instruction at a time. Continuity of Service Telephone service must be available to subscribers round the clock without any interruption. The cumulative duration of all the interruptions of service must be less than a few hours, during the life of the exchange. It is typically 2 hours, in 40 years. The system should ensure continuity of service, even on occurrence of a hardware or software fault. The philosophy of fault handling is completely different for an electronic exchange from a conventional scientific or data processing system. In a data processing system a false result is far more serious than a complete breakdown of the system, whereas in an electronic exchange system some proportion of incorrect call processing (say, one in 500) is acceptable instead of total system failure. This influences the design of SPC operational software in many ways, particularly the programs relating to faults and on line capacity extensions. Software design should enable maintenance of the processors, telephone peripherals and other exchange equipment on-line without interrupting service. To achieve this, a bit of degradation in the grade of service and total traffic handling capacity may be accepted, it should also be possible to extend the capacity of an operating exchange, i.e., add new telephone equipment, processor, memory etc., without interrupting the call processing. Similarly, correcting the software errors, adding new programs or new version of existing programs and other software maintenance and extension must not disturb telephone service. e) Program storages In most of the SPC systems, the total size (volume wise) of all the programs added together is much bigger than the size of main memory. It is therefore, not possible to make the entire programs main memory resident. However, a program can be executed only when it is resident in the main memory. In order to make best use of the limited resources, i.e., the capacity of the main memory, only vital parts of the system programs and application programs are kept permanently in main memory. All the other programs that are not active may be stored outside the main memory in auxiliary storage, also called mass memory.


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Hence, in any switching system, many programs, including the large ones, which are not being used very frequently, are stored in low cost mass memory (auxiliary storage) like magnetic tape or disk. Generally most of administration and maintenance programs are stored on magnetic disks and programs for system hardware and/or software extension are stored on magnetic tape. On-line and Off-line programs On line Program On line programs are run when the exchange is on-line, ie. while performing call processing, testing and other administration functions. The on-line programs can be either system programs or Application Programs. Off-line programs The off-line programs are used during initial commissioning of the SPC exchange, i.e., when it is off-line and has not started functioning. It is not possible to use the off-line programs during the operation of an SPC exchange because the main memory is normally made to accommodate on-line programmes only. However, if the standby processor, individual operational system, is put off-line programs can be run on it. Examples of such programs are text editor, linker, etc. f) Redundancy Methods A telephone exchange must guarantee service to the subscribers under all circumstances and as such it should never be at the mercy of any fault in any equipment. Ideally not more than 2 hr. of total interruption of service over a period of 40 years is admissible. Although the reliability of electronic components is very high, but certainly not absolute, Redundancy is always necessary at the processor level, to meet the exchange availability specifications. In other words, duplication must be provided, at least for all common control devices so that the exchange must continue to function even if one of these units are taken out of service, for any reason. Regarding the central processor unit, redundancy, in general, can be provided by any of the following techniques

• Load Sharing Method • Hot Standby Method • N+1 Redundancy Method

Load Sharing Load sharing is the frequently used mode of redundancy methods. Fig. 10.4 illustrates this concept.


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Fig.10.4 Load Sharing Redundancy Fig 10.5 Hot Standby Redundancy In a load-sharing environment, each processor has access to all inputs and all outputs, as in synchronous Replica Duplication. Calls are randomly distributed for processing to either of the two processors When a processor has accepted a call, it handles it right through to completion Two processors share the resources dynamically, but operate independently. They have to co-ordinate their operation for access to subscriber lines, trunks and switching network. There is an inter-processor link for exchanging the information for mutual co-ordination and for hardware or software exclusion mechanism to prevent both the processors accessing the same peripheral device. The processors have separate memories for storing temporary data for calls, because each processor handles its call, independently of the other-Although, rest of the memory can be shared for storing programmes and permanent data, it is preferable to have separate storages for each processor. This provides greater protection against software errors. Moreover, there is an arrangement to ensure that each command, sent from OBM, is executed by both the processors. This avoids issuance of commands separately to each processor. Each processor has access to all peripheral devices. During normal operation, both processors are on-line and share half of the calls each, on statistical basis. The exchange operator can however, send commands to distribute the traffic unevenly between the two processors. When a fault occurs, the fault processor is taken out of service without interrupting the telephone services, but the calls currently being established by the faulty processor are lost. The other processor while establishing its own calls, also handles all new calls and takes all calls in the ringing or speech phase, even though originally set up by the faulty processor.


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In this mode, error detection is left to self detection or detection by the other processor. This is one of the disadvantages of the load sharing. The greatest advantage is that total processing capacity in terms of busy hour call attempts is higher as compared to Synchronous Replica Duplication operation. This arrangement has also got a high tolerance to software faults because it is very rare that both the processors are executing the same programme at the same time. If an error appears in one of the software programmes that particular processor is taken out of service and system is safeguarded against total traffic interruption. Hot Standby System This is the simplest mode of providing redundancy in a single processor operation. Active processor handles the total traffic and other processor is on standby. It is totally decoupled from active processor both in hardware and in software. It is place on-line only if the active processor fails. Unfortunately, updating of standby processor before it is switched on-line is impossible. As the working processor has failed, it is incapable of carrying out any operation. A fault can therefore cause an interruption in service and requires total re-initialization, with release of all calls already established. To avoid this problem, the processor must have a common memory, as shown in the Fig. 10.5. In such a system, current state of the active processor is copied into, the back up storage, every 5 seconds. After switch over between the two processors the new working processor loads the last stored state of the environment into the main memory. This stored state differs only slightly from the real state of calls in the ringing and speech phases and most of these calls are saved. However, because of common memory provision for both the processors, many maintenance operations are unduly complicated, restricting the popularity of such an arrangement. g) N+1 Redundancy Methods Some multiprocessor systems, as shown in Fig. 10.6 have the N+1 system configuration, in which one processor is added as a redundant unit. This unit can, under normal conditions, take part of the traffic handling so that the total N+1 configuration has a larger capacity than the engineered value, based on N processors. However, in case of failure of any processor, the redundant processor takes up the entire load of the failed processor.


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Fig 10.6 N+1 Redundancy Method In modern SPC exchanges, the control system is a combination of various systems depending upon the type and complexity of the function and requirements of the network. However, the load sharing redundancy method is generally employed, to avoid a complete collapse of the services, with the advantage of higher traffic handling capacity in normal conditions.


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Module - 01


Chapter – 11

Switching Concepts


• Telecom Network Model • Circuit Switching • Packet Switching • Telephone Switching Hierarchy • Local Exchange • Tandem Exchange • Call flow • Interconnections


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1. Telecom Network Model The telecommunications network can be described by a layered model consisting of the following layers:

• Switching and Services layer: This consists of all the switching nodes, local as well as transit. It also consists of any other equipment and like computers and software used to provide services to the customers.

• Transport Layer: This represents the links among the nodes and provides the medium and systems to carry the information from one node to the other. These are junctions and trunks. Junctions are links between the local switches and local and national switches. Trunks are the links between the national switches, the national and international switches and between the international switches i.e. the long distance network. The long distance or trunk network is composed of multiplexed channels of varying capacity connecting the National Switches and the International Switches. The trend has been to move from point-to-point links using Plesiochronous Digital Hierarchy (PDH) towards advanced networks with built in controllability based on Synchronous Digital Hierarchy (SDH) technique. The two most important trends in the long distance networks are digitization and introduction of fiber-optic technologies. These developments have reduced the transmission cost per channel-kilometer and improved the quality.

• Access Layer: This represents the access network that links the customers to the local switch.

2. Switching concepts In telecommunication, switching of voice or data calls are established in two modes:

• Circuit Switching • Packet Switching

3. Circuit Switching (CS) In telecommunications, a CS network is one that establishes a fixed circuit (or channel) between exchanges and terminals before the users may communicate, as if the exchanges were physically connected with an electrical circuit. The delay is constant during the connection. Other callers cannot use each circuit until the circuit is released and a new connection is set up. Even if no actual communication is taking place in a dedicated circuit then, that channel still remains unavailable to other users. Channels that are available for new calls to be set up are said to be idle. The copper wire used for the connection


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could not be used to carry other calls at the same time, even if the subscribers were in fact not talking and the line was silent. Virtual CS is a packet switching technology that may emulate CS, in the sense that the connection is established before any packets are transferred, and that packets are delivered in order. For call setup and control, it is possible to use a separate dedicated signaling channel. The method of establishing the connection and monitoring its progress and termination through the network may also utilize a separate control channel to communicate the call setup and control information and use TDM to transport the actual circuit data. With CS, a route is reserved from source to destination. The entire message is sent in order so that it does not have to be reassembled at the destination. CS can be relatively inefficient because capacity is wasted on connections, which are set up but are not in continuous use. On the other hand, the connection is immediately available and capacity is guaranteed until the call is disconnected.

Fig. 11.1 Circuit Switched Network In circuit Switched Network- Path or pipe between two ends in communication is opened and stays open for the duration of the call 4. Packet Switching (PS) CS contrasts with PS, which splits traffic data (for instance, digital representation of sound, or computer data) into packets that are routed over a shared network. PS is the process of segmenting a message/data to be transmitted into several smaller packets. Each packet is labeled with its destination and the number of the packet, precluding the need for a dedicated path to help the packet find its way to its destination. Each is dispatched and many may go via different routes. At the destination, the original message is reassembled in the correct order, based on the packet number. PS networks do not require a circuit to be established and allow many pairs of nodes to communicate almost simultaneously over the same channel.


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Telephone switches A switch, in telephony jargon refers to a telephone switch or exchange located at the local telephone company's central office, directly serving subscribers. The switch services include basic dial tone, calling features, and additional digital and data services to subscribers using the local loop. These switches were slower to convert from circuit switching technologies to time division multiplexing. 5. Telephone Switching Hierarchy In order to organize Direct Dialing, switches in the Public Switched Telephone Network (PSTN) are arranged into a hierarchy containing different levels.

• International gateways - handing off and receiving traffic from outside the country’s networks.

• TAX exchanges - exchanges connecting the various areas of a city or towns in country.

• Tandem exchanges - which interconnect whole regions of the network in a multi exchange environment or exchanges connecting major population centres within particular region of the network.

• Local exchanges – directly serves subscribers and end-users. 6. Local Exchange The fundamental difference between a local exchange and the other exchanges is that a local exchange provides telephone service to customers, and as such is concerned with "subscriber type" activities: generation of dial-tone and handling of network services such as advice of duration and charge etc. Specifically, a local switch provides dial tone, local switching and access to the rest of the network, Fig 11.2. Typically a local switch will cover an area of a city, an individual town, or several villages and could serve from several hundred to 100,000 subscribers. Some form of remote switch often performs the function of a local switch in rural areas or Remote Digital Terminal installed at the original switch site to handle local switching or concentration, respectively. The local switching infrastructure is then physically located in a larger population center. Urban areas with extensive underground plant tend to keep the classic local office architecture. The telephone line from a subscriber runs underground or overhead on poles to the local BT building. While this building is often known as "the exchange", in actual fact it might well not be. All the lines in an area terminate on a Main Distribution Frame (MDF). A second set of wires run from the MDF to the switching unit in the building. For the majority of lines this unit is a Remote Line Unit (RLU), while for the remainder it is the Digital Local Exchange (DLE).


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Fig. 11.2 Direct Local Exchange connections

7. Tandem (TDM) Exchange TDM switches do not provide dial tone - they simply route calls between other switches, so they are more concerned with efficient switching and signaling, Fig 11.3. Where a call is being made to a subscriber on a different DLE from the caller, it needs to travel over some kind of link between DLEs. In theory all DLEs could be connected to one another, but this would be wasteful of links. Alternatively calls could be relayed from DLE to DLE, but this would take up a lot of the processing power of the DLEs. Instead there is a second network of switches, known as the TDM Network. This network is made up of four kinds of switch: In the following diagram, the lettered circles represent DLEs (and their associated RLUs), the numbered squares tandem switches, and the triangle a Direct Junction Switching Unit (DJSU). The tick lines show the tandems all connected to one another, the thin lines are DLE-TDM links.

Fig. 11.3 Tandem Connections


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8. Call flow There are four kinds of call:

• Own exchange calls, where the call remains within a single DLE. • Linked exchange calls that go over an inter-DLE link, such as from

B to C (these two categories are together known as Local exchange calls).

• Single tandem calls that go DLE-TDM-DLE, such as from A to B. • Double TDM calls such as from A to E.

In fact, a call is normally described in terms of the most optimal route, so B to C is linked exchange even if the call doesn't go over the link, A to B is single TDM even if TDM1 has failed and the call is going via TDM2 and TDM3, and A to E is double tandem even if congestion means the call goes A-1-2-3-4-E. Each switch (both DLE and TDM) will have a routing table giving up to four routes, in order of preference, for each possible call. 9. Interconnections There may be ranges of other licensed operators who also carry telephone calls. In order that calls can be transferred from one operator to the other or vice versa, it is obviously necessary for the two networks to be connected. This is done at Points of Interconnection (POIs). POI can be at any tandem switch or, in some circumstances, at a DLE. An operator is not required to connect to every TDM switch. Instead, will route calls through the network to a convenient POI. In the above example, there are three POIs, two at tandems and one at a DLE. Each DLE and tandem will have entries for the other operators in its routing tables, just as with calls to other DLEs.


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Module - 01


Chapter – 12

Digital Switching


• Digital Switching • Time and Space Switching • Digital Space Switching Principle • Practical Space Switch • Digital Time Switching Principle • Two Dimensional Switching • TST Network


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1. Digital Switching A Digital switching system, in general, is one in which signals are switched in digital form. These signals may represent speech or data. The digital signals of several speech samples are time multiplexed on a common media before being switched through the system. To connect any two subscribers, it is necessary to interconnect the time-slots of the two speech samples, which may be on same or different PCM highways. The digitalised speech samples are switched in two modes, viz., Time Switching and Space Switching. This Time Division Multiplex Digital Switching System is popularly known as Digital Switching System. 2. Time and Space Switching Generally, a digital switching system several time division multiplexed (PCM) samples. These PCM samples are conveyed on PCM highways (the common path over which many channels can pass with separation achieved by time division.). Switching of calls in this environment requires placing digital samples from one time-slot of a PCM multiplex in the same or different time-slot of another PAM multiplex. For example, PCM samples appearing in TS6 of I/C PCM HWY1 are transferred to TS18 of O/G PCM HWY 2, via the digital switch, as shown in Fig 12.1.

FIG 12.1 Digital Switch The interconnection of time-slots, i.e., switching of digital signals can be achieved using two different modes of operation. These modes are:

• Space Switching • Time switching

Usually, a combination of both the modes is used.


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In the space-switching mode, corresponding time-slots of I/C and O/G PCM highways are interconnected. A sample, in a given time-slot, TSi of an I/C HWY, say HWY1, is switched to same time-slot, TSi of an O/G HWY, SAY HWY2. Obviously there is no delay in switching of the sample from one highway to another highway since the sample transfer takes place in the same time-slot of the PCM frame. Time Switching, on the other hand, involves the interconnection of different time-slots on the incoming and outgoing highways by re-assigning the channel sequence. For example, a time-slot TSx of an I/C Highway can be connected to a different time-slot. TSy, of the outgoing highway. In other words, a time switch is, basically, a time-slot changer. 3 Digital Space Switching Principle The Digital Space Switch consists of several input highways, X1, X2,...Xn and several output highways, Y1, Y2,.............Ym, inter connected by a cross point matrix of n rows and m columns. The individual cross point consists of electronic AND gates. The operation of an appropriate cross point connects any channel, a, of I/C PCM highway to the same channel, a, of O/G PCM highway, during each appropriate time-slot which occurs once per frame as shown in Fig 12.2. During other time-slots, the same cross point may be used to connect other channels. This cross point matrix works as a normal space divided matrix with full availability between incoming and outgoing highways during each time-slot. Each cross point column, associated with one O/G highway, is assigned a column of control memory. The control memory has as many words as there is time-slot per frame in the PCM signal. In practice, this number could range from 32 to 1024. Each cross point in the column is assigned a binary address, so that only one cross point per column is closed during each time-slot. The binary addresses are stored in the control memory, in the order of time-slots. The word size of the control memory is x bits, so that 2x = n, where n is the number of cross points in each column. A new word is read from the control memory during each time-slot, in a cyclic order. Each word is read during its corresponding time-slot, i.e., Word 0 (corresponding to TSO), followed by word 1 (corresponding to TS1) and so on. The word contents are contained on the vertical address lines for the duration of the time-slot, thus the cross point corresponding to the address, is operated during a particular time-slot. This cross point operates every time the particular time-slot appears at the inlet in successive frames, normally, a call may last for around a million frames. As the next time-slot follows, the control memory is also advanced by one step, so that during each new time-slot new corresponding words are read from the various control memory columns. This results in operation of a completely different set of cross points being activated in


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different columns. Depending upon the number of time-slots in one frame, this time division action increases the utilisation of cross point 32 to 1024 times compared with that of conventional space-divided switch matrix. Example Consider the transfer of a sample arriving in TS7 of I/C HWY X1 to O/G HWY Y3. Since this is a space switch, there will be no reordering of time i.e., the sample will be transferred without any time delay, via the appropriate cross point. In other words, the objective is to connect TS7 of HWY X1 and TS7 of HWY Y3. The central control (CC) selects the control memory column corresponding output highway Y3. In this column, the memory location corresponding to the TS7 is chosen. The address of the cross point is written in this location, i.e., 1, in binary, is written in location 7. This cross point remains operated for the duration of the time-slot TS7, in each successive frame till the call lasts. For disconnection of call, the CC erases the contents of the control memory locations, corresponding to the concerned time-slots. The AND gates, therefore, are disabled and transfer of samples is halted. 4. Practical Space Switch In a practical switch, the digital bits are transmitted in parallel rather than serially, through the switching matrix. In a serial 32 time-slots PCM multiplex, 2048 Kb/s are carried on a single wire sequentially, i.e., all the bits of the various time-slots follow one another. This single wire stream of bits, when fed to Serial to Parallel Converter is converted into 8-wire parallel output. For example, all 8 bits corresponding to TS3 serial input are available simultaneously on eight output wires (one bit on each output wire), during just one bit period, as shown in fig. This parallel output on the eight wires is fed to the switching matrix. It can be seen that during one full time-slot period, only one bit is carried on the each output line, whereas 8 bits are carried on the input line during this period. Therefore, bit rate on individual output wires, is reduced to 1/8th of input bit rate=2048/8=256Kb/s


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Fig. 12.2 Space Switch Due to reduced bit rate in parallel mode, the cross point is required to be operated only for 1/8th of the time required for serial working. Eight times more channels, i.e., 32 x 8 = 256 channels, in the same frame, can thus, share it. However, since the eight bits of one TS are carried on eight wires, each cross point has eight switches to interconnect eight input wires to eight output wires. Each cross point (all the eight switches) will remain operated now for the duration of one bit only, i.e., only for 488 ns (1/8th of the TS period of 3.9 µs)

Fig 12.3 Serial parallel converter


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For example, to connect 40 PCM I/C highways, a matrix of 40x 40 = 1600 cross points each having a single switch, is required in serial mode working. Whereas in parallel mode working, a matrix of (40/8 x 40/8) = 25 cross point is sufficient. As eight switches are required at each cross point 25 x 8 = 200 switches only are required. Thus, there is a reduction of the matrix by 1/8th in parallel mode working, hence reduction in size and cost of the switching matrix. 5 Digital Time Switching Principle A Digital Time Switch consists of two memories, viz., a speech or buffer memory to store the samples till destination time-slots arrive, and a control or connection or address memory to control the writing and reading of the samples in the buffer memory and directing them on to the appropriate time-slots. Speech memory has as many storage locations as the number of time-slots in input PCM, e.g., 32 locations for 32 channel PCM system. The Control Memory controls the writing/reading operation in the speech memory. It has same number of memory locations as for speech memory, i.e., 32 locations for 32-channel system. Each location contains the address of one of the speech memory locations where the channel sample is either written or read during a time-slot. These addresses are written in the control memory of the CC of the exchange, depending upon the connection objective. A Time-Slot Counter, which usually is a synchronous binary counter, is used to count the time-slots from 0 to 31, as they occur. At the end of each frame, it gets reset and the counting starts again. It is used to control the timing for writing/reading of the samples in the speech memory. Example Consider the objective that TS4 of incoming PCM is to be connected to TS6 of outgoing PCM. In other words, the sample arriving in TS4 on the I/C PCM has to be delayed by 6 - 4 = 2 time-slots, till the destination time-slot, viz., TS6 appears in the O/G PCM. The required delay is given to the samples by storing it in the speech memory. The I/C PCM samples are written cyclically i.e. sequentially time-slot wise, in the speech memory locations. Thus, the sample in TS4 will be written in location 4, as shown in Fig. 12.4. The Control Memory controls the reading of the sample. The Control Memory location corresponding to output time-slot TS6 is 6. In this location, the CC writes the input time-slot number, viz., 4, in binary. These contents give the read address for the speech memory, i.e., it indicates the speech memory locations from which the sample is to be read out, during read cycle.


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When the time-slot TS6 arrives, the control memory location 6 is read. Its content addresses the location 4 of the speech memory in the read mode and sample is read on to the O/G PCM. In every frame, whenever time-slot 4 comes a new sample will be written in location 4. This will be read when TS6 occurs. This process is repeated till the call lasts. For disconnection of the call, the CC erases the contents of the control memory location to halt further transfer of samples. Time switch can operate in two modes, viz.

• Output associated control • Input associated control

Output associated control In this mode of working, 2 samples of I/C PCM are written cyclically in the speech memory locations in the order of time-slots of I/C PCM, i.e., TS1 is written in location 1, TS2 is written in location 2, and so on, as discussed in the example. The contents of speech memory are read on output PCM in the order specified by control memory. Each location of control memory is rigidly associated with the corresponding time-slot of the O/G PCM and contains the address of the TS of incoming PCM to be connected to. The control memory is always read cyclically, in synchronism with the occurrence of the time-slot. The entire process of writing and reading is repeated in every frame, till the call is disconnected.

Fig 12.4 Output Associated Control Switch


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It may be noticed that the writing in the speech memory is sequential and independent of the control memory, while reading is controlled by the control memory, i.e., there is a sequential writing but controlled reading. Input associated control Here, the samples of I/C PCM are written in a controlled way, i.e., in the order specified by control memory, and read sequentially. Each location of control memory is rigidly associated with the corresponding TS of I/C PCM and contains the address of TS of O/G PCM to be connected to. The previous example with the same connection objective of connecting TS4 of I/C PCM to TS6 of O/G PCM may be considered for its restoration. The location 4 of the control memory is associated with incoming PCM TS4. Hence, it should contain the address of the location where the contents of TS4 of I/C PCM are to be written in speech memory. A CC writes the number of the destination TS, viz., 6 in this case, in location 4 of the control memory. The contents of TS4 are therefore, written in location of speech memory, as shown in Fig 12.5. The contents of speech memory are read in the O/G PCM in a sequential way, i.e., location 1 is read during TS1, location 2 is read during TS2, and so on. In this case, the contents of location 6 will appear in the output PCM at TS6. Thus the input PCM TS4 is switched to output PCM TS6. In this switch, there is sequential reading but controlled writing.

Fig 12.5 Input Associated Controlled Time Switch


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6 Two Dimensional Switching Though the electronic cross points are not so expensive, the cost of accessing and selecting them from external pins in a Space Switch, become prohibitive as the switch size increases. Similarly, the memory location requirements rapidly go up as a Time Switch is expanded, making it uneconomical. Hence, it becomes necessary to employ a number of stages, using small switches as building blocks to build a large network. This would result in necessity of changing both the time-slot and highway in such a network. Hence, the network, usually, employs both types of switches viz., space switch and time switch; and. therefore, is known as two Dimensional network. These networks can have various combinations of the two types of switches and are denoted as TS, STS, TSST, etc. Though to ensure full availability, it may be desirable to use only T stages. However, the networks having the architecture of TT, TTT, TTTT, etc., are uneconomical, considering the acceptability of tolerable limits of blocking, in a practical network. Similarly, a two-stage two-dimensional network, TS or ST, is basically suitable for very low capacity networks only. The most commonly used architecture has three stages, viz., STS or TST. However, in certain cases, their derivatives, viz., TSST, TSSST, etc., may also be used. An STS network has relatively simpler control requirements and hence, is still being favored for low capacity networks, viz., PBX exchanges. As the blocking depends mainly on the outer stages, which are space stages, it becomes unsuitable for high capacity systems. A TST network has lesser blocking constraints, as the outer stages are time stages, which are essentially non-blocking, and the space stage is relatively smaller. It is, therefore, most cost-effective for networks handling high traffic, However, for still higher traffic handling capacity networks, e.g., tandem exchanges, it may be desirable to use TSST or TSSST architecture. The choice of a particular architecture is dependent on other factors also, viz., implementation complexity, modularity, testability, expandability, etc. As a large number of factors favor TST structure, it is most widely used. 7. TST Network As the name suggests, in a TST network, there are two time stages separated by a space stage. The former carry out the function of time-slot changing, whereas the latter performs highway jumping. Let us consider a network having n input and n output PCM highways. Each of the input and output time stages will have n time switches and the space stage will consist of an n x n cross point matrix. The speech memory as well as the control memory of each time switch and each column of a control memory of the space switch will have m locations, corresponding


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to m time-slots in each PCM. Thus, it is possible to connect any TS in I/C PCM to any TS in O/G PCM. In the case of a local exchange, the network will be of folded type, i.e., the O/G PCM highways, via a suitable hybrid. Whereas, for a transit exchange, the network will be non-folded, having complete isolation of I/C and O/G PCM highways. However, a practical local exchange will have a combination of both types of networks. For the sake of explanation, let us assume that there are only four I/C and O/G PCM highways in the network. Hence, there will be only four time switches in each of the T-stages and the space switch will consist of 4x4 matrixes. Let us consider an objective of connecting two subscribers through this switching network of local exchange, assuming that the CC assigns TS4 on HWY0 to the calling party and TS6 on HWY3 to the called party The speech samples of the calling party have to be carried from TS4 of I/C HWY 0 and to TS6 of O/G HWY3 and those of the called party from TS6 of I/C HWY 3 to TS4 of O/G HWY 0, with the help of the network. The cc establishes the path, through the network in three steps. To introduce greater flexibility, it uses an intermediate time-slot, TSx, which is also known as internal time-slot. The three switching steps for transfer of speech sample of the calling party to the called party are as under:

Step 1 Input Time Stage (IT) TS4 HWY0 to TSx HWY0 Step 2 Space stage (S) Tsx HWY0 to Tsx HWY3 Step 3 Output Time Stage (OT) Tsx HWY3 to TS6 HWY3

As the message can be conveyed only in one direction through this path, another independent path, to carry the massage in the other direction is also established by the CC, to complete the connection. Assuming the internal time-slots to be TS10 and TS11, the connection may be established as shown in Fig 12. 6.

FIG 12.6 T S T Switch


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Let us now consider the detailed switching procedure making some more assumptions for the sake of simplicity. Though practical time switches can handle 256 time-slots in parallel mode, let us assume serial working and that there are only 32 time-slots in each PCM. Accordingly, the speech and control memories in time switches and control memory columns in space switch will contain 32 locations each. To establish the connection, the CC searches for free internal time-slots. Let us assume that the first available time-slots are TS10 and TS11, as before. To reduce the complexity of control, the first time stage is designed as output-controlled switch, whereas the second time stage is input-controlled.

FIG 12.7 T S T Switch Structure For transfer of speech samples from the calling party to the called party of previous example, CC orders writing of various addresses in


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location 10 of control memories of IT-10, OT-3 and column 3 of CM-S of corresponding to O/G highway, HWY3. Thus, 4 corresponding to I/C TS4 is written in CM-IT-0, 6 corresponding to O/G TS6 is written in CM-OT-3 and 0 corresponding to I/C HWY 0 is written in column 3 of CM-S, as shown in Fig. 12.7. As the first time switch is output-controlled, the writing is done sequentially. Hence, a sample, arriving in TS4 of I/C HWY 0, is stored in location 4 of SM-IT-0. It is readout on internal HWY 0 during TS10 as per the control address sent by CM-IT-0. In the space switch, during this internal TS10, the cross point 0 in column 3 is enabled, as per the control address sent by column 3 of CM-S, thus, transferring the sample to HWY3. The second time stage is input controlled and hence, the sample, arriving in TS10, is stored in location 6 of SM-OT-3, as per the address sent by the CM-OT-3. This sample is finally, readout during TS6 of the next frame, thus, achieving the connection objective. Similarly, the speech samples in the other direction, i.e., from the called party to the calling party, are transferred using internal TS11. As soon as the call is over, the CC erases the contents in memory locations 10 and 11 of all the concerned switches, to stop further transfer of message. These locations and time-slots are, then, available to handle next call.


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Module - 01


Chapter – 13

Signaling in Telecom Systems


• Signaling in Telecommunication • Types of signaling information • Subscriber Line signaling • Inter-exchange Signaling • In-Band and Out-of-Band Signals • E & M and R2 Signaling • Channel Associated Signaling • Common Channel Signaling System No 7 • Signaling Network Architecture • Signaling Components • SS7 Link Types • Basic signaling Procedure • Advantage of CCS


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1. Signaling in Telecommunication Introduction A telecommunication network establishes and realizes temporary connections, in accordance with the instructions and information received from subscriber lines and inter exchange trunks, in form of various signals. Therefore, it is necessary to interchange information between an exchange and it external environment i.e. between subscriber lines and exchange, and between different exchanges. Though these signals may differ widely in their implementation they are collectively known as telephone signals. A signaling system uses a language, which enables two switching equipments to converse for the purpose of setting up calls. Like any other language it possesses a vocabulary of varying size and varying precision, ie. a list of signals, which may also vary in size and syntax in the form of a complex set of rules governing the assembly of these signals. 2. Types of signaling information The signaling information can be categorized under four main heads. Call request and Release information Call request information i.e. calling subscriber off hook or seizure signal or an incoming trunk, indicates a new call. On its receipt, the exchange connects appropriate equipment for receiving address information (called number). Release information i.e. on hook or release signal on a trunk indicates that the call is over. The exchange releases all the equipment held out for the call, and clears up any other information used for setting up at including the call. Selection (Address) information. When the exchange is ready to receive the address information. It sends back a request, which is known as proceed to send (PTS) signal in trunk signaling and dial tone in subscriber signaling. Address information essentially comprises of full or part of the called subscribers number and possibly additional service data. End of selection information This information indicates the status of the called line, or the reason for non-completion of the call attempt, essentially indicating called line free or busy.


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Supervisory information It specifies the on/off hook condition of a called subscriber after the connection has been setup Called subscriber off hook called subscriber has answered and charging may commence. Called subscriber on hook Called subscriber has hung up to terminate the call, and the call is disconnected after a time delay if the calling subscriber does not hang up. The on/off-hook conditions of the calling subscriber are covered by call request and release information. Call connection The interchange of signaling information can be illustrated with the help of a typical call connection sequence.

• A request for originating a call is initiated when the calling subscriber lifts the handset.

• The exchange sends dial tone to the calling subscriber to indicate to him to start dialing.

• The called number is transmitted to the exchange, when the calling subscriber dials the number.

• If the number is free, the exchange sends ringing current to him. • Feedback is provided to the calling subscriber by the exchange

by sending. • Ring-back tone, if the called subscriber is free • Busy tone if the called subscriber is busy, or • Recorded message, if provision exists, for non-completion of call

due to some other constraint. • The called subscriber indicates acceptance of the incoming call

by lifting the handset • The exchange recognizing the acceptance terminates the ringing

current and the ring-back tone, and establishes a connection between the calling and called subscribers.

• The connection is released when either subscriber replaces the handset. When the called subscriber is in a different exchange, the following inter-exchange trunk signal functions are also involved, before the call can be set up.

• The originating exchange seizes an idle inter exchange trunk, connected to a digit register at the terminating exchange.

• The originating exchange sends the digit. The steps 4 to 8 are then performed to set up the call.


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3. Subscriber Line signaling Calling Subscriber Line Signaling In automatic exchanges the power is fed over the subscriber’s loop by the centralized battery at the exchange. Normally, it is 48 V. The power is fed irrespective of the state of the subscriber, viz., idle, busy or talking. Call report When the subscriber is idle, the line impedance is high. The line impedance falls, as soon as, the subscriber lifts the handset, resulting in increase of line current. This is detected as a new call signal and the exchange after connecting an appropriate equipment to receive the address information sends back dial-tone signal to the subscriber. Address signal After the receipt of the dial tone signal, the subscriber proceeds to send the address digits. The digits may be transmitted either by decade dialing or by multifrequency pushbutton dialing. Decadic Dialing The address digits may be transmitted as a sequence of interruption of the DC loop by a rotary dial or a Decadic push-button keypad. The number of interruption (breaks) indicates the digit, exept0, for which there are 10 interruptions. The rate of such interruptions is 10 per second and the make/break ration is 1:2. There has to be an inter-digital pause of a few hundred milliseconds to enable the exchange to distinguish between consecutive digits. This method is, therefore, relatively slow and signals cannot be transmitted during the speech phase. Multifrequency Push-button Dialing This method overcomes the constraints of the Decadic dialing. It uses two sets of four voice frequencies. Pressing a button (key) generates a signal comprising of two frequencies, one from each group. Hence, it is also called Dual-Tone Multi-frequency (DTMF) dialing. The signal is transmitted as long as the key is kept pressed. This provides 16 different combinations. As there are only 10 digits, at present the highest frequency, viz., 1633 Hz is not used and only 7 frequencies are used, as shown in Fig.13.1. By this method, the dialing time is reduced and almost 10 digits can be transmitted per second. As frequencies used lie in the speech band, information may be transmitted during the speech phase also, and hence, DTMF telephones can be used as access terminals to a variety of systems, such as computers with voice output. The tones have been so


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selected as to minimize harmonic interference and probability of simulation by human voice.

Figure 13.1 Tone-Dialing Frequency Groups. End of selection signal The address receiver is disconnected after the receipt of complete address. After the connection is established or if the attempt has failed the exchange sends any one of the following signals.

• Ring-back tone to the calling subscriber and ringing current to the called subscriber, if the called line is free.

• Busy-tone to the calling subscriber, if the called line is busy or otherwise inaccessible.

• Recorded announcement to the calling subscriber, if the provision exists, to indicate reasons for call failure, other than called line busy.

Ring back, tone and ringing current are always transmitted from the called subscriber local exchange and busy tone and recorded announcements, if any, by the equipment as close to the calling subscriber as possible to avoid unnecessary busying of equipment and trunks. Answer Back Signal As soon as the called subscriber lifts the handset, after ringing, a battery reversal signal is transmitted on the line of the calling subscriber. This may be used to operate special equipment attached to the calling


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subscriber, e.g., short-circuiting the transmitter of a CCB, till a proper coin is inserted in the coin-slot. Release signal When the calling subscriber releases i.e., goes on hook, the line impedance goes high. The exchange recognizing this signal, releases all equipment involved in the call. This signal is normally of more than 500 milliseconds duration. Permanent Line (PG) Signal Permanent line or permanent glow (PG) signal is sent to the calling subscriber if he fails to release the call even after the called subscriber has gone on-hook and the call is released after a time delay. The PG signal may also be sent, in case the subscriber takes too long to dial. It is normally busy tone. Called subscriber line signals.

• Ring Signal • Answer Signal • Release Signal

4. Inter-exchange Signaling Inter-exchange signaling can be transmitted over each individual inter exchange trunk. The signals may be transmitted using the same frequency band as for speech signals (inband signaling), or using the frequencies outside this band (out-of-band signaling). The signaling may be

• Pulsed: The signal is transmitted in pulses. Change from idle condition to one of active states for a particular duration characterizes the signal, e.g., address information

• Continuous: The signal consists of transition from one condition to another; a steady state condition does not characterize any signal.

• Compelled: It is similar to the pulsed mode but the transmission is not of fixed duration but condones till acknowledgement of the receiving unit is received back at the sending unit. It is a highly reliable mode of signal transmission of complex signals.

Line signals The simplest cheapest, and most reliable system of signaling on trunks, was DC signaling, also known as metallic loop signaling, exactly the same as used between the subscriber and exchange, i.e. Circuit seizure/release corresponding to off/on-hook signal of the subscriber.


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5. In-Band and Out-of-Band Signals Exchanges separated by long distance cannot use any form of DC line signaling. Suitable interfaces have to be interposed between them, for conversion of the signals into certain frequencies, to enable them to be carried over long distance. A signal frequency (SF) may be used to carry the on/off hook information. Pulsing of the states can also transmit the dialing pulses. The number of signals is small and they can be transmitted in-band or out-of band. The states involved are shown in Table 1. Table 1. Single Frequency Signaling States Tone Signal Condition

State Forward Backward

Idle (On hook) FORWARD Seizure (off hook) Release (on hook) BACKWARD Answer (off hook) Clear Back (on hook) Blocking (off hook)

On Off On Off Off On

On On Off/on Off On Off

For in band signaling the tone frequency is chosen to be 2600Hz. or 2400 Hz. As the frequency lies within the speech band, simulation of tone-on condition indicating end-of call signal by the speech has to be guarded against, for pre-mature disconnection. Out-of- Band signaling overcomes the problem of tone on condition imitation by the speech by selecting a tone frequency of 3825 Hz which is beyond the speech band. However, this adds up to the hardware costs. 6. E & M Signals E & M lead signaling may be used for signaling on per-trunk basis. An additional pair of circuit, reserved for signaling is employed. One wire is dedicated to the forward signals ((M-Wire for transmit or mouth) which corresponds to receive or R-lead of the destination exchange, and the other wire dedicated to the backward signals (E-wire for receive or ear) which corresponds transmit or send wire or S-Lead of the destination exchange. This type of signaling is normally used in conjunction with an interface to change the E & M signals into frequency signal to be carried along with the speech. 7. R2 Signaling CCITT standardized the R2 signaling system to be used on national and international routes. However, the Indian environment requires


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lesser number of signals and hence, a slightly modified version is being used. There is a provision for having 15 combinations using two out of six frequencies viz., 1380, 1500, 1620, 1740, 1860 and 1980 Hz, for forward signals and another 15 combination using two out of six frequencies viz., 1140,1020, 900, 780, 660 and 540 Hz, for backward signals. In India, the higher frequency in the forward group i.e., 1980 Hz, and the lower frequency in the backward group, i.e., 540 Hz are not used. Thus, there are 10 possible combinations in both the directions. The weight codes for the combinations used are indicated in Table 3. Table 3- Signal Frequency Index and Weight Code

Signal Frequency (Hz)

Forward 1380 1500 1620 1740 1860

Backward 1140 1020 900 780 660

Index f 0 f 1 f 2 f 3 f 4

Weight Code

0 1 2 4 7

8. Digital Signaling All, the systems discussed so far, basically, are on per line or per trunk basis, as the signals are carried on the same line or trunk. With the emergence of PCM systems, it was possible to segregate the signaling from the speech channel. Inter exchange signaling can be transmitted over a channel directly associated with the speech channel, channel-associated signaling (CAS), or over a dedicated link common to a number of channels, common channel signaling (CCS). The information transmitted for setting up and release of calls is same in both the cases. Channel associated signaling requires the exchanges, to have access to each trunk via the equipment which may be decentralised, whereas, in common channel signaling, the exchange is connected to only a limited number of signaling links through a special terminal. 3. Channel- Associated Signaling (CAS) CAS is composed of line signal and register signal. For CAS, the signaling channel is combined with the bear information channel (refers to register signal) or the two have fixed correspondence (refers to line signal)


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Fig. 13.2 CAS In the PCM systems the signaling information is conveyed on a separate channel, which is rigidly associated with the speech channel. Hence, this method is known as channel associated signaling (CAS). Though the speech sampling rate is 8 KHz, the signals do not change as rapidly as speech and hence, a lower sampling rate of 500 Hz, for digitisation of signals can suffice. Based on this concept, TS 16 of each frame of 125 microseconds is used to carry signals of 2 speech channels, each using 4 bits. Hence, for a 30 channel PCM system, 15 frames are required to carry all the signals. To constitute a 2 millisecond multiframe of 16 frames. F 0 to F 15 TS 16 of the frame F 0 is used for multiframe synchronisation. TS 16 of F1 contains signal for speech channels 1 and 16 being carried in TS 1 and TS 17, respectively, TS16 of F2 contains signals of speech channels 2 and 17 being carried in TS2 and TS 18, respectively and so on, Both line signals and address information can be conveyed by this method. Although four bits per channel are available for signaling only two bits are used. As the transmission is separate in the forward and backward direction, the bits in the forward link are called af and bf, and those in the backward link are called ab and bb. Values for these bits are assigned as shown in Table below. As the dialing pulses are also conveyed by these conditions, the line state recognition time is therefore, above a threshold value. The bit bf is normally kept at 0, and the value 1 indicates a fault. However, the utilisation of such a dedicated channel for signaling for each speech channel is highly inefficient, as it remains idle during the speech phase. Hence, another form of signaling known as common-channel signaling evolved.


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Forward Backward

af bf ab bb

Idle 1 0 1 0 Seizure 0 0 1 0 Seizure

acknowledge 0 0 1 1

Answer 0 0 0 1 Clear Forward 1 0 0/1 1

Clear Back 0 0 1 1 4. Common Channel Signaling System No. 7 (CCS#7) History Common Channel Signaling (CCS) is a technique that enables stored program control exchanges, network databases, and other nodes in a network to exchange messages related to call setup, call supervision, call release (connection control information), information needed for distributed application processing and network management information. From the point of view of plain old telephony service (POTS), CCS is a technique that separates the physical channel used for signaling from that, which is used to carry the end user’s telecommunication traffic. One signaling channel is able to carry the signaling control information of up to 3,000 trunks, depending on the implemented protocol; hence, the term “common channel signaling” is used. The CCITT has specified the common channel signaling system no.7 (CCS-7). CCS-7 is optimised for application in digital networks. It is characterised by the following main features:

• internationally standardized (national variations possible). • suitable for the national, international and intercontinental

network level. • suitable for various communication services such as telephony,

text services, data services digital network (ISDN). • high performance and flexibility along with a future-oriented

concept, which well meet new requirements. • high reliability for message transfer.


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• processor-friendly structure of messages (signal units of multiples of 8 bits).

• signaling on separate signaling links; the bit rate of the circuits is, therefore, exclusively for communication.

• signaling links always available, even during existing calls. • use of the signaling links for transferring user data also. • used on various transmission media

o cable (copper, optical fiber) o radio relay o satellite (up to 2 satellite links)

• use of the transfer rate of 64 kbit/s typical in digital networks. • used also for lower bit rates and for analog signaling links if

necessary. • automatic supervision and control of the signaling network.

Signaling Network In contrast to channel-associated signaling, which has been standard practice until now, in CCS7 the signaling messages are sent via separate signaling links. One signaling link can convey the signaling messages for many circuits.

Fig. 13.3 CCS The CCS7 signaling links connect signaling points (SPs) in a communication network. The signaling points and the signaling links form an independent signaling network, which is overlaid over the circuit network. 5. Definition

Signaling System 7 (SS7) is architecture for performing out-of-band signaling in support of the call-establishment, billing, routing, and


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information-exchange functions of the public switched telephone network (PSTN). It identifies functions to be performed by a signaling-system network and a protocol to enable their performance.

SS7 is a means by which elements of the telephone network exchange information is conveyed in the form of messages. SS7 messages can convey information such as:

• Forwarding a call placed from XXXXXXXX to YYYYYYYY. Look for it on trunk 067.

• The called subscriber for the call on trunk 11 is busy. Release the call and play a busy tone.

• The route to ABC is congested. Don’t send any messages to ABC unless whose priority is 2 or higher.

• Taking trunk 143 out of service for maintenance.

SS7 is characterized by high-speed packet data and out-of-band signaling.

6. Out-of-Band Signaling

Out-of-band signaling is signaling that does not take place over the same path as the conversation.

We are used to thinking of signaling as being in-band. We hear dial tone, dial digits, and hear ringing over the same channel on the same pair of wires. When the call completes, we talk over the same path that was used for the signaling. Traditional telephony used to work in this way as well. The signals to set up a call between one switch and another always took place over the same trunk that would eventually carry the call. Signaling took the form of a series of multi-frequency (MF) tones, much like touch tone dialing between switches.

Out-of-band signaling establishes a separate digital channel for the exchange of signaling information. This channel is called a signaling link. Signaling links are used to carry all the necessary signaling messages between nodes. Thus, when a call is placed, the dialed digits, trunk selected, and other pertinent information are sent between switches using their signaling links, rather than the trunks which will ultimately carry the conversation. Today, signaling links carry information at a rate of 56 or 64 kbps. It is interesting to note that while SS7 is used only for signaling between network elements, the ISDN D channel extends the concept of out-of-band signaling to the interface between the subscriber and the switch. With ISDN service, signaling that must be conveyed between the user station and the local switch is carried on a separate


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digital channel called the D channel. The voice or data which comprise the call is carried on one or more B channels.

Out-of-band signaling has several advantages that make it more desirable than traditional in-band signaling.

• It allows for the transport of more data at higher speeds (56 kbps can carry data much faster than MF out pulsing).

• It allows for signaling at any time in the entire duration of the call, not only at the beginning.

• It enables signaling to network elements to which there is no direct trunk connection.

7. Signaling Network Architecture

If signaling is to be carried on a different path from the voice and data traffic it supports, then what should that path look like? The simplest design would be to allocate one of the paths between each interconnected pair of switches as the signaling link. Subject to capacity constraints, all signaling traffic between the two switches could traverse this link. This type of signaling is known as associated signaling, and is shown below in Figure.

Fig. 13.4 Associated Signaling

Associated signaling works well as long as a switch’s only signaling requirements are between itself and other switches to which it has trunks. If call setup and management was the only application of SS7, associated signaling would meet that need simply and efficiently. In fact, much of the out-of-band signaling deployed in Europe today uses associated mode.

The North American implementers of SS7, however, wanted to design a signaling network that would enable any node to exchange signaling with


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any other SS7–capable node. Clearly, associated signaling becomes much more complicated when it is used to exchange signaling between nodes which do not have a direct connection. From this need, the North American SS7 architecture was born.

8. Signaling Components

The signaling architecture defines a completely new and separate signaling network. The network is built out of the following three essential components, interconnected by signaling links: Signal Switching Points (SSPs)—SSPs are telephone switches (end offices or tandems) equipped with SS7-capable software and terminating signaling links. They generally originate, terminate, or switch calls. Signal Transfer Points (STPs)—STPs are the packet switches of the SS7 network. They receive and route incoming signaling messages towards the proper destination. They also perform specialized routing functions. Signal Control Points (SCPs)—SCPs are databases that provide information necessary for advanced call-processing capabilities.

Once deployed, the availability of SS7 network is critical to call processing. Unless SSPs can exchange signaling, they cannot complete any interswitch calls. For this reason, the SS7 network is built using a highly redundant architecture. Each individual element also must meet exacting requirements for availability. Finally, protocol has been defined between interconnected elements to facilitate the routing of signaling traffic around any difficulties that may arise in the signaling network.

To enable signaling network architectures to be easily communicated and understood, a standard set of symbols was adopted for depicting SS7 networks. Figure shows the symbols that are used to depict these three key elements of any SS7 network.

Fig. 13.5 Signaling Network Elements

STPs and SCPs are customarily deployed in pairs. While elements of a pair are not generally co-located, they work redundantly to perform the same logical function. When drawing complex network diagrams, these pairs may be depicted as a single element for simplicity, as shown in Figure.


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Fig. 13.6 STP and SCP Pairs 9. Basic Signaling Architecture Figure shows a small example of how the basic elements of an SS7 network are deployed to form two interconnected networks.

Fig. 13.7 Sample Network

The following points should be noted:

1. STPs W and X perform identical functions. They are redundant. Together, they are referred to as a mated pair of STPs. Similarly, STPs Y and Z form a mated pair.

2. Each SSP has two links (or sets of links), one to each STP of a mated pair. All SS7 signaling to the rest of the world is sent out over these links. Because the STPs of a mated pair are redundant, messages sent over either link (to either STP) will be treated equivalently.

3. The STPs of a mated pair are joined by a link (or set of links).


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4. Two mated pairs of STPs are interconnected by four links (or sets of links). These links are referred to as a quad.

5. SCPs are usually (though not always) deployed in pairs. As with STPs, the SCPs of a pair are intended to function identically. Pairs of SCPs are also referred to as mated pairs of SCPs. Note that they are not directly joined by a pair of links.

6. Signaling architectures such as this, which provide indirect signaling paths between network elements, are referred to as providing quasi-associated signaling.

10. SS7 Link Types

SS7 signaling links are characterized according to their use in the signaling network. Virtually all links are identical in that they are 56–kbps or 64–kbps bidirectional data links that support the same lower layers of the protocol; what is different is their use within a signaling network. The defined link types are shown in Fig. 13.8 and defined as follows:

Fig. 13.8 Link Types


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A Links

A links interconnect an STP and either an SSP or an SCP, which are collectively referred to as signaling end points ("A" stands for access). A links are used for the sole purpose of delivering signaling to or from the signaling end points (they could just as well be referred to as signaling beginning points). Examples of A links are 2–8, 3–7, and 5–12 as in Figure.

Signaling that an SSP or SCP wishes to send to any other node is sent on either of its A links to its home STP, which, in turn, processes or routes the messages. Similarly, messages intended for an SSP or SCP will be routed to one of its home STPs, which will forward them to the addressed node over its A links.

C Links

C links are links that interconnect mated STPs. As will be seen later, they are used to enhance the reliability of the signaling network in instances where one or several links are unavailable. "C" stands for cross (7–8, 9–10, and 11–12 are C links). B links, D links, and B/D links interconnecting two mated pairs of STPs are referred to as either B links, D links, or B/D links. Regardless of their name, their function is to carry signaling messages beyond their initial point of entry to the signaling network towards their intended destination. The "B" stands for bridge and describes the quad of links interconnecting peer pairs of STPs. The "D" denotes diagonal and describes the quad of links interconnecting mated pairs of STPs at different hierarchical levels. Because there is no clear hierarchy associated with a connection between networks, interconnecting links are referred to as either B, D, or B/D links (7–11 and 7–12 are examples of B links; 8–9 and 7–10 are examples of D links; 10–13 and 9–14 are examples of interconnecting links and can be referred to as B, D, or B/D links).

E Links

While an SSP is connected to its home STP pair by a set of A links, enhanced reliability can be provided by deploying an additional set of links to a second STP pair. These links, called E (extended) links provide backup connectivity to the SS7 network in the event that the home STPs cannot be reached via the A links. While all SS7 networks include A, B/D, and C links, E links may or may not be deployed at the discretion of the network provider. The decision of whether or not to deploy E links can be made by comparing the cost of deployment with the improvement in reliability. (1–11 and 1–12 are E links.)


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F Links

F (fully associated) links are links which directly connect two signaling end points. F links allow associated signaling only. Because they bypass the security features provided by an STP, F links are not generally deployed between networks. Their use within an individual network is at the discretion of the network provider. (1–2 is an F link.)

11. Basic Call Setup Example

Before going into much more detail, it might be helpful to look at several basic calls and the way in which they use SS7 signaling.

Fig. 13.9 Call Setup Example

In this example, a subscriber on switch A places a call to a subscriber on switch B.

1. Switch A analyzes the dialed digits and determines that it needs to send the call to switch B.

2. Switch A selects an idle trunk between itself and switch B and formulates an initial address message (IAM), the basic message necessary to initiate a call. The IAM is addressed to switch B. It identifies the initiating switch (switch A), the destination switch (switch B), the trunk selected, the calling and called numbers, as well as other information beyond the scope of this example.

3. Switch A picks one of its A links (e.g., AW) and transmits the message over the link for routing to switch B.

4. STP W receives a message, inspects its routing label, and determines that it is to be routed to switch B. It transmits the message on link BW.


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5. Switch B receives the message. On analyzing the message, it determines that it serves the called number and that the called number is idle.

6. Switch B formulates an address complete message (ACM), which indicates that the IAM has reached its proper destination. The message identifies the recipient switch (A), the sending switch (B), and the selected trunk.

7. Switch B picks one of its A links (e.g., BX) and transmits the ACM over the link for routing to switch A. At the same time, it completes the call path in the backwards direction (towards switch A), sends a ringing tone over that trunk towards switch A, and rings the line of the called subscriber.

8. STP X receives the message, inspects its routing label, and determines that it is to be routed to switch A. It transmits the message on link AX.

9. On receiving the ACM, switch A connects the calling subscriber line to the selected trunk in the backwards direction (so that the caller can hear the ringing sent by switch B).

10. When the called subscriber picks up the phone, switch B formulates an answer message (ANM), identifying the intended recipient switches (A), the sending switch (B), and the selected trunk.

11. Switch B selects the same A link it used to transmit the ACM (link BX) and sends the ANM. By this time, the trunk also must be connected to the called line in both directions (to allow conversation).

12. STP X recognizes that the ANM is addressed to switch A and forwards it over link AX.

13. Switch A ensures that the calling subscriber is connected to the outgoing trunk (in both directions) and that conversation can take place.

14. If the calling subscriber hangs up first (following the conversation), switch A will generate a release message (REL) addressed to switch B, identifying the trunk associated with the call. It sends the message on link AW.

15. STP W receives the REL, determines that it is addressed to switch B, and forwards it using link WB.

16. Switch B receives the REL, disconnects the trunk from the subscriber line, returns the trunk to idle status, generates a release complete message (RLC) addressed back to switch A, and transmits it on link BX. The RLC identifies the trunk used to carry the call.

17. STP X receives the RLC, determines that it is addressed to switch A, and forwards it over link AX.

18. On receiving the RLC, switch A idles the identified trunk.


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12. Database Query Example

People generally are familiar with the toll-free aspect of 800 (or 888) numbers, but these numbers have significant additional capabilities made possible by the SS7 network. 800 numbers are virtual telephone numbers. Although they are used to point to real telephone numbers, they are not assigned to the subscriber line itself.

When a subscriber dials an 800 number, it is a signal to the switch to suspend the call and seek further instructions from a database. The database will provide either a real phone number to which the call should be directed, or it will identify another network (e.g., a long-distance carrier) to which the call should be routed for further processing. While the response from the database could be the same for every call (as, for example, if you have a personal 800 number), it can be made to vary based on the calling number, the time of day, the day of the week, or a number of other factors.

The following example shows how an 800 call is routed see Figure.

Fig. 13.10 Database Query Example

1. A subscriber served by switch A wants to reserve a rental car at a company's nearest location. She dials the company's advertised 800 number.

2. When the subscriber has finished dialing, switch A recognizes that this is an 800 call and that it requires assistance to handle it properly.

3. Switch A formulates an 800 query message including the calling and called number and forwards it to either of its STPs (e.g., X) over its A link to that STP (AX).


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4. STP X determines that the received query is an 800 query and selects a database suitable to respond to the query (e.g., M).

5. STP X forwards the query to SCP M over the appropriate A link (MX). SCP M receives the query, extracts the passed information, and (based on its stored records) selects either a real telephone number or a network (or both) to which the call should be routed.

6. SCP M formulates a response message with the information necessary to properly process the call, addresses it to switch A, picks an STP and an A link to use (e.g., MW), and routes the response.

7. STP W receives the response message, recognizes that it is addressed to switch A, and routes it to A over AW.

8. Switch A receives the response and uses the information to determine where the call should be routed. It then picks a trunk to that destination, generates an IAM, and proceeds (as it did in the previous example) to set up the call.

13. Layers of the SS7 Protocol

As the call-flow examples show, the SS7 network is an interconnected set of network elements that is used to exchange messages in support of telecommunications functions. The SS7 protocol is designed to both facilitate these functions and to maintain the network over which they are provided. Like most modern protocols, the SS7 protocol is layered.

Fig. 13.11 SS7 Protocol Stack


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Physical Layer

This defines the physical and electrical characteristics of the signaling links of the SS7 network. Signaling links utilize DS–0 channels and carry raw signaling data at a rate of 56 kbps or 64 kbps (56 kbps is the more common implementation).

Message Transfer Part—Level 2

The level 2 portion of the message transfer part (MTP Level 2) provides link-layer functionality. It ensures that the two end points of a signaling link can reliably exchange signaling messages. It incorporates such capabilities as error checking, flow control, and sequence checking.

Message Transfer Part—Level 3

The level 3 portion of the message transfer part (MTP Level 3) extends the functionality provided by MTP level 2 to provide network layer functionality. It ensures that messages can be delivered between signaling points across the SS7 network regardless of whether they are directly connected. It includes such capabilities as node addressing, routing, alternate routing, and congestion control.

Collectively, MTP levels 2 and 3 are referred to as the message transfer part (MTP).

Signaling Connection Control Part

The signaling connection control part (SCCP) provides two major functions that are lacking in the MTP. The first of these is the capability to address applications within a signaling point. The MTP can only receive and deliver messages from a node as a whole; it does not deal with software applications within a node.

While MTP network-management messages and basic call-setup messages are addressed to a node as a whole, other messages are used by separate applications (referred to as subsystems) within a node. Examples of subsystems are 800 call processing, calling-card processing, advanced intelligent network (AIN), and custom local-area signaling services (CLASS) services (e.g., repeat dialing and call return). The SCCP allows these subsystems to be addressed explicitly.


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Global Title Translation

The second function provided by the SCCP is the ability to perform incremental routing using a capability called global title translation (GTT). GTT frees originating signaling points from the burden of having to know every potential destination to which they might have to route a message. A switch can originate a query, for example, and address it to an STP along with a request for GTT. The receiving STP can then examine a portion of the message, make a determination as to where the message should be routed, and then route it.

For example, calling-card queries (used to verify that a call can be properly billed to a calling card) must be routed to an SCP designated by the company that issued the calling card. Rather than maintaining a nationwide database of where such queries should be routed (based on the calling-card number), switches generate queries addressed to their local STPs, which, using GTT, select the correct destination to which the message should be routed. Note that there is no magic here; STPs must maintain a database that enables them to determine where a query should be routed. GTT effectively centralizes the problem and places it in a node (the STP) that has been designed to perform this function.

In performing GTT, an STP does not need to know the exact final destination of a message. It can, instead, perform intermediate GTT, in which it uses its tables to find another STP further along the route to the destination. That STP, in turn, can perform final GTT, routing the message to its actual destination.

Intermediate GTT minimizes the need for STPs to maintain extensive information about nodes that are far removed from them. GTT also is used at the STP to share load among mated SCPs in both normal and failure scenarios. In these instances, when messages arrive at an STP for final GTT and routing to a database, the STP can select from among available redundant SCPs. It can select an SCP on either a priority basis (referred to as primary backup) or so as to equalize the load across all available SCPs (referred to as load sharing).

ISDN User Part (ISUP)

ISUP user part defines the messages and protocol used in the establishment and tear down of voice and data calls over the public switched network (PSN), and to manage the trunk network on which they rely. Despite its name, ISUP is used for both ISDN and non–ISDN calls.


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In the North American version of SS7, ISUP messages rely exclusively on MTP to transport messages between concerned nodes.

Transaction Capabilities Application Part (TCAP)

TCAP defines the messages and protocol used to communicate between applications (deployed as subsystems) in nodes. It is used for database services such as calling card, 800, and AIN as well as switch-to-switch services including repeat dialing and call return. Because TCAP messages must be delivered to individual applications within the nodes they address, they use the SCCP for transport.

Operations, Maintenance, and Administration Part (OMAP)

OMAP defines messages and protocol designed to assist administrators of the SS7 network. To date, the most fully developed and deployed of these capabilities are procedures for validating network routing tables and for diagnosing link troubles. OMAP includes messages that use both the MTP and SCCP for routing.

14. Information over the Signaling Link

Signaling information is passed over the signaling link in messages, which are called signal units (SUs).

Three types of SUs are defined in the SS7 protocol.

1. Message signal units (MSUs) 2. Link status signal units (LSSUs) 3. Fill-in signal units (FISUs)

SUs are transmitted continuously in both directions on any link that is in service. A signaling point that does not have MSUs or LSSUs to send will send FISUs over the link. The FISUs perform the function suggested by their name; they fill up the signaling link until there is a need to send purposeful signaling. They also facilitate link transmission monitoring and the acknowledgment of other SUs.

All transmission on the signaling link is broken up into 8-bit bytes, referred to as octets. SUs on a link are delimited by a unique 8-bit pattern known as a flag. The flag is defined as the 8-bit pattern "01111110". Because of the possibility that data within an SU would contain this pattern, bit manipulation techniques are used to ensure that the pattern does not occur within the message as it is transmitted over the link. (The SU is reconstructed once it has been taken off the link, and any bit manipulation is reversed.) Thus, any occurrence of the flag on the


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link indicates the end of one SU and the beginning of another. While in theory two flags could be placed between SUs (one to mark the end of the current message and one to mark the start of the next message), in practice a single flag is used for both purposes.

15. Addressing in the SS7 Network

Every network must have an addressing scheme, and the SS7 network is no different. Network addresses are required so that a node can exchange signaling nodes to which it does not have a physical signaling link. In SS7, addresses are assigned using a three-level hierarchy. Individual signaling points are identified as belonging to a cluster of signaling points. Within that cluster, each signaling point is assigned a member number. Similarly, a cluster is defined as being part of a network. Any node in the American SS7 network can be addressed by a three-level number defined by its network, cluster, and member numbers. Each of these numbers is an 8-bit number and can assume values from 0 to 255. This three-level address is known as the point code of the signaling point. A point code uniquely identifies a signaling point within the American SS7 network and is used whenever it is necessary to address that signaling point.

Network numbers are assigned on a nationwide basis by a neutral party. Regional Bell operating companies (RBOCs), major independent telephone companies, and interexchange carriers (IXCs) already have network numbers assigned. Because network numbers are a relatively scarce resource, companies' networks are expected to meet certain size requirements in order to be assigned a network number. Smaller networks can be assigned one or more cluster numbers within network numbers 1, 2, 3, and 4. The smallest networks are assigned point codes within network number 5. The cluster to which they are assigned is determined by the state in which they are located. The network number 0 is not available for assignment and network number 255 is reserved for future use.


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16. Basic Signaling Procedure SPA & SPB – Signaling Points A & B IAM – Initial Address Message ACM – Address Complete Message ANC – Answer Signal Charge CLF – Clear Forward Message RLG – Release Guard Signal

Fig 2.12 Signaling procedure

17. Advantages of CCS

The other advantages of CCS, in addition to space saving are: - 1. Faster call set up by cutting down the post dialer delay. In SPC

environment setting up a call via two transit centres takes just 0.8 second with CCS, compared to 3.5 seconds with MF signaling.

2. New services can be made available with a better quality. For example, setting up a call with abbreviated dialing facility and routed via two transit centres, takes just 3 seconds with CCS, as compared to 12 seconds required by the network using CAS, moreover it is also possible to use additional services, as it is possible to transmit signals during speech phase also.

3. More call completion is possibly by re routing the call without increasing the call set up time to an unacceptable level.


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4. In MF signaling system it is possible for a clever subscriber to access the system by generating of generally used signaling tones. By generating tones of the correct frequency and at the correct time can make long distance calls without being charged thus resulting in loss of revenue, however, such calls are not possible in CCS, as the signaling link is totally separate from the speech link.

5. Unified signaling system is possible to provide all existing and envisage services as required under the integrated services Digital Network (ISDN).

6. Modern network management will be possible by provision of an efficient means of collecting information and transmitting orders for technical operation and maintenance of the network.

7. Traffic engineering becomes more efficient. The speech circuit’s requirements will go down because of substantial reduction of ineffective traffic. This advantage itself is sufficient to make additional cost of signaling link cost effective. Moreover, as large amount of data is available in shorter time span, the real time load on the processor will come down resulting in increase in its efficiency by almost 20%

Constraints of CCS As in CCS more processing of the signaling is required, the cost of hardware and software for the signaling interface will be more. In addition to this, there would be following constraints of the network.

• As a single data link carries signaling information of a large number of speech circuits, its failure would result in immobilisation of all these speech circuits.


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Module - 01


Chapter – 14

ISDN


• Integrated Service Digital Network (ISDN)- Introduction

• ISDN Service • ISDN Channel Classification • Interface Structures


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1. Integrated Services Digital Network (ISDN) - Introduction The ISDN is an abbreviation of Integrated Services Digital Network. The current communications networks vary with the type of service, such as telephone network, telex network, and digital data transmission network. On the other hand, the ISDN is an integrated network for various types of communications services handling digitized voice (telephone) and non-voice (data) information. ISDN Definition The CCITT defines the ISDN as follows: “A complete, terminal-to-terminal digital network. Fig. shows the end-to-end digital connectivity”.

Fig. 14.1 End-to-End Digital Connectivity

ISDN is a network that provides both telephone and non-telephone services in the same network, which utilizes Signaling System No. 7 (SS7) for signaling between switching systems.

Fig. 14.2 The Signaling Connection between Switching Systems

A network offers standard user network interface. Fig. above shows the standard user network interface.


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Fig. 14.3 Standard User Network Interface

2. ISDN Services The ISDN provides the following functions, as shown in Fig.

• Packet switching service • Circuit switching service • Leased circuit service

Fig. 14,4 A Wide Range of Services

Circuit switching service includes both telephone and data circuit switching. As shown in the Figure, ISDN can interface with various terminals, such as a telephone set, FAX, Video terminal or personal computer to provide a wide range of services. Two statements can summarize the ISDN concept:

• ISDN offers a variety of services, such as telephone, data and image transmission through one network.

• ISDN handles all information digitally.


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Standard user-network interface.

• The subscriber line is connected with an NT (Network Termination) installed at the customer premises.

• Various terminals are connected to the NT. These terminals can include digital telephones, multi media terminal, digital facsimile machines, personal computers, etc. as shown in the figure.

• The NT and terminals are connected by S or T interface (S/T interface), as recommended by the CCITT. Up to 8 terminals are connected to one S/T interface. The NT and terminals are connected using an 8-pin connector, which is also recommended by the CCITT.

• As shown in this figure, the personal computer uses the RS232C interface that is different from the ISDN S/T interfaces, so a TA (Terminal Adapter) is provided to adapt the RS232C interface for use with the ISDN interfaces.

The interface between the user and the network. Telephone service makes use of two wires for the subscriber line between the switching system and customer’s premises. These same two wires can be used by ISDN to receive ISDN services. An NT (Network Termination) is installed at the subscriber’s home and connected to the subscriber line.

Fig. 14.5 The Interface between the User and Network


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• The Interface between the NT and the ISDN exchange (switching system) is called U interface. This interface has not been defined in the CCITT Recommendations because circumstances are different in each country. The point between the NT and the on-premises terminals is called the S or T reference point. The ISDN user/network interface refers to these S/T points, and is defined in the CCITT Recommendations.

• The S/T interface uses four wires, two for sending and two for receiving. Since U interface uses two wires, the NT provides a two-wire/four-wire conversion function.

• CCITT recommends the use of AMI (Alternative Mark Inversion) code at the S/T point. AMI code is a bipolar waveform.

• As shown in the figure, the ISDN Terminal provides S/T interface that follows the CCITT Recommendations, and can be connected directly to the NT. Since the personal computer and the analog FAX utilize a different interface from S/T interface, they require protocol conversion by a TA (Terminal Adapter).

3. ISDN Channel Classification Various channels can be used to transmit information between a terminal and the switching system. These include B, D and H channels. Each channel has a different bit rate and information carrying attributes. B- Channel The B-channel carries user information such as voice and packet data at a rate of 64 kbps. However, the B-channel does not carry signaling information. D- Channel The D-channel interface carries mainly signaling information such as originating or terminating subscriber number, call origination and disconnect signals for circuit switching and packet switched user data at 16 kbps or 64 kbps. The D-channel also permits multiple logical channels to be established for use in packet communications. H-channel The H-channel carries high-speed user information such as high-speed facsimile, video, high-speed data, etc. H channels do not carry


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signaling information for circuit switching by the ISDN. Table outlines these three channel types and characteristics.

Channel Type

Bit Rate Function

B 64 kbps To carry user information Circuit switching mode and packet switching mode

D 16 kbps 64 kbps

To carry signaling information for circuit switching

H H0 : 384 kbps H11 : 1536 kbps H12 : 1920 kbps

To carry high-speed packet data such as facsimile and video An H channel does not carry signaling information for circuit switching by the ISDN

Note: H0 : 64K X 6 = 384 kbps H11 : 64K X 24 = 1536 kbps H12 : 64K X 30 = 1920 kbps

4. Interface Structures Basic Interface This interface is primarily for home use. The basic interface is set at a transmission speed of 144 kbps. This provides two (2) 64 kbps B-channels for user information exchange and a 16 kbps D-channel for signaling and control. The interface is thus referred to as 2B+D. Primary Group Interface These interfaces are primarily for business use. The primary group interface for ATT system consists of a 1.544 Mbps line. This line can thus provide up to 23 B-channels at 64 kbps and a single D-channel at 64 kbps. In Europe and other countries using CEPT system standards, the primary group is 2.048 Mbps and the interface is 30B-channels and single 64 kbps D-channel (30B+D). This line is used for PABX etc.


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Module - 01


Chapter – 15

Telephone Traffic Engineering


• Basic Concepts of Telephone Traffic • Parameter Definitions • Traffic Load Measurement (TLM): • Sampling Methods • Call Arrival Patterns • Traffic Models • Measurement of Telephone Traffic. • Grade of service. • Erlangs formula • Traffic offered • Traffic Forecasting and Planning • Demand Forecast Procedure • Traffic Forecasting • Traffic forecasting methods


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1. Basic Concepts of Telephone Traffic Traffic Theory Network designers need a way to properly size network capacity, especially as networks grow. Traffic theory enables network designers to make assumptions about their networks based on past experience. Traffic is defined as either the amount of data or the number of messages over a circuit during a given period of time. Traffic also includes the relationship between call attempts on traffic-sensitive equipment and the speed with which the calls are completed. Traffic analysis enables you to determine the amount of bandwidth you need in your circuits for data and for voice calls. Traffic engineering addresses service issues by enabling you to define a grade of service or blocking factor. A properly engineered network has low blocking and high circuit utilization, which means that service, is increased and your costs are reduced. Telephone traffic is originated by the individual needs of different subscribers and so it is beyond the control of telephone administration. Any and every subscriber can originate a call at any and every moment without giving any previous information and the duration of calls is also not previously known. Although the individual telephone traffic originates at random, the average telephone traffic for a particular exchange follows the general pattern of activity in the exchange area. Normally there is a peak in morning, a dip during lunch period followed by an afternoon peak. In some localities the traffic has seasonal characteristic, for example at a holiday resort.

Fig. 15.1 Switching links


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Fig. 15.2 Trunks or Circuits Whatever be the nature of variation of traffic, a telephone engineer is interested in maximum traffic that occurs in an exchange. The hour in which maximum traffic usually occurs in an exchange is known as Busy Hour. Busy Hour Traffic is the average value of maximum traffic in the busy hour. In computing Busy Hour Traffic the seasonal effects are also taken into account. Sometimes it is convenient to refer to busy hour calling rate. Busy hour calling rate is the number of calls originated per subscriber in the busy hour. This provides a simple means for designing the exchange with respect to the number of subscribers. It also provides probable growth of traffic to the estimated growth in number of subscribers. The busy hour calling rate may vary about 0.3 for a small country exchange and 1.5 or more for a busy exchange in business area in a city. In addition to Busy Hour calling rate day calling rate is also useful for example to estimate the daily current drain on the exchange battery. When the volume of traffic is quoted in terms of number of calls originated in a given time, this is insufficient to determine the consequent occupancy of lines and equipment. Therefore, measurement of traffic should not only consider number of calls but also their duration. The duration during which equipments and circuits are held when a call is made is called HOLDING TIME. Normally, it is average holding time per call for the particular item of equipment that is taken into account, so far as the caller is concerned the useful time is during the conversation only. However, the total time during which equipments and circuits are held when a call is made also includes, the period during which call is being established and time taken to release the equipment after the call has concluded. 2. Parameter Definitions Trunk or Circuit: Trunk or Circuit describes the resources required, or resources available, to carry traffic as shown in Fig. In wireless communication, for analogue traffic, a trunk is usually equal to one radio channel and for digital traffic; a circuit may be a dedicated radio channel or one slot within multiple time slots or one code within a list of codes.


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Busy Hour Traffic (BHT): Network traffic load is measured during the busiest hour because this period represents the maximum traffic load that a network must support. The result gives a traffic load measurement commonly referred to as the Busy Hour Traffic (BHT). There are times, one cannot do a thorough sampling or and can have only an estimate of how many calls are handled daily. In such a circumstance, usually make assumptions about environment, such as average number of calls per day and the AHT. In the standard business environment, the busy hour of any given day accounts for approximately 15 to 20 percent of the traffic for that day. In computations, generally use 17 percent of the total daily traffic to represent the peak hour traffic. In many business environments, an acceptable AHT is generally assumed to be 180 to 210 seconds and used to estimate trunking requirements without having more complete data. Busy Hour Call Attempt (BHCA): BHCA is the number of times a telephone call, or a telephony session request, is attempted or received during the busiest hour of the day. BHCA refers to the transmitted signaling requests per hour and it is the product of the number of users (Nu) multiplied by the 2.4 Requests/hour. A call attempt means an attempt to achieve a connection to one or more devices attached to a telecommunications service. HOLDING TIME (HT): HT is the period of time a trunk or circuit is busy on a call. The holding time includes the channel request and seizes time, the message length, and the hang time. AVERAGE HOLDING TIME (AHT) Total number of actual call seconds within an hour Average holding time = ----------------------------------------------------------- Number of calls for same hour AHT is the total time of all calls in a specified period divided by the number of calls in that period. Example: (3976 total call sec.)/(23 calls) = 172.87 sec per call = AHT of 172.87 sec. AVERAGE BUSY PERIOD: Average busy period is a continuous period of time during which the highest usage occurs. 3. Traffic Load Measurement (TLM): In traffic theory, you measure traffic load. Traffic load is the ratio of call arrivals in a specified period of time to the average amount of time


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taken to service each call during that period. These measurement units are based on Average Hold Time (AHT). The two main measurement units used to measure traffic load are Erlangs and Centum Call Seconds (CCS). ERLANG: Unit commonly used to measure call intensity or telephone traffic volumes, over a time period, normally 1 hour. It is generally accepted that when referring to traffic usage, one Erlang represents a single trunk or circuit occupied for 1 hour. One Erlang is 3600 seconds of calls on the same circuit, or enough traffic loads to keep one circuit busy for 1 hour. (Number of hourly calls) (Average holding time in sec) Traffic Load in Erlang = --------------------------------------------------------- 3600 Traffic in Erlang is the product of the number of calls times of the AHT divided by 3600. Example: If a group of user made 30 calls in 1 Hour, and each call had an average call duration (AHT) of 5 minutes, then the number of Erlangs this represents is worked out as follows:

Minutes of traffic in the hour = number of calls x duration Minutes of traffic in the hour = 30 x (5 x 60 sec)

Minutes of traffic in the hour = 9000 Hours of traffic in the hour = 9000/3600 Hours of traffic in the hour = 2.5

Traffic figure = 2.5 Erlangs Erlang traffic measurements are made in order to help telecommunications network designers understand traffic patterns within their voice networks. This is essential to successfully design their network topology and establish the necessary trunk group sizes. Erlang traffic measurements or estimates can be used to work out how many lines are required between a telephone system and a central office (PSTN exchange lines), or between multiple network locations. CENTUM (HUNDRED) CALL SECONDS (CCS): One CCS is 100 seconds of calls on the same circuit. Voice switches generally measure the amount of traffic in CCS. (Number of hourly calls) (Average holding time in sec) Traffic Load (CCS)= --------------------------------------------------------------- 100


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Traffic in CCS is the product of the number of calls times of the AHT divided by 100. Example: (23 calls * 172.87 AHT)/100 = 39.76 CCS CCS TO ERLANG: Erlang = CCS / 36 ACCESS DELAY (AD): AD is the amount of time it takes to get access to the first available trunk. This time delay is expressed as multiples of the average Holding Time (HT). GRADE OF SERVICE (GOS) GOS for systems with queues is the probability of a call being delayed by busy servers and is associated to an access delay. It is expressed as a decimal or percentage. MESSAGE LENGTH (ML): ML is the time taken by the trunk to send the information portion of a transmission associated with a call. This time varies widely depending on the business type or radio service category. TOTAL TRAFFIC VOLUME OR TRAFFIC INTENSITY (TI): TI is the volume of traffic the radio system is expected to carry during peak periods. This traffic is usually expressed in Erlangs. TRAFFIC OFFERED (TO): TO per user is the volume of traffic each user is expected to be offered by the system during peak periods, i.e. traffic per phone / mobile. This traffic is usually expressed in Erlangs. 4. Sampling Methods The accuracy of traffic analysis depends on the accuracy of sampling methods. The following parameters will change the represented traffic load:

• Weekdays versus weekends • Holidays • Type of traffic • Apparent versus offered load • Sample period • Total number of samples taken • Stability of the sample period

Probability theory states that to accurately assess voice network traffic, at least 30 samples of the busiest hours of a voice network to be taken during the sampling period. To get the most accurate results, we need to take as many samples of the offered load as possible. The ITU-T recommends that public switched telephone network (PSTN) connections measurement or read-out periods are 60 minutes


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and/or 15-minute intervals. These intervals are important because they summarize the traffic intensity over a period of time. If measurements are taken throughout the day, one can find the peak hour of traffic in any given day. There are two recommended ways to determine the peak daily traffic, as follows:

• Daily Peak Period (DPP) records the highest traffic volume measured during a day. This method requires continuous measurement and is typically used in environments where the peak hour may be different from day to day.

• Fixed Daily Measurement Interval (FDMI) requires measurements only during the predetermined peak periods. It is used when traffic patterns are somewhat predictable and peak periods occur at regular intervals.

Business traffic usually peaks around 10:00 a.m. to 11:00 a.m. and 2:00 p.m. to 3:00 p.m. In the example in Table 1, using FDMI sampling, the hour with the highest total traffic load is 10 a.m., with a total traffic load of 60.6 erlangs. Table 1 Daily Peak Period Measurement Hour Monday Tuesday Wednesday Thursday Friday Total

Load

9:00 a.m.

12.7 11.5 10.8 11.0 8.6 54.6

10:00 a.m.

12.6 11.8 12.5 12.2 11.5 60.6

11:00 a.m.

11.1 11.3 11.6 12.0 12.3 58.3

12:00 p.m.

9.2 8.4 8.9 9.3 9.4 45.2

1:00 p.m.

10.1 10.3 10.2 10.6 9.8 51.0

2:00 p.m.

12.4 12.2 11.7 11.9 11.0 59.2

3:00 p.m.

9.8 11.2 12.6 10.5 11.6 55.7

4:00 p.m.

10.1 11.1 10.8 10.5 10.2 52.7


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The example in Table 2 uses DPP to calculate total traffic load. Peak Traffic

12.7 12.2 12.5 12.2 12.3 61.9

Peak Time

9:00 a.m.

2:00 p.m.

10:00 a.m. 10:00 a.m.

11:00 a.m.

It is also required to divide the daily measurements into groups that have the same statistical behavior. The ITU-T defines these groups as: Workdays, Weekend days, and Yearly exceptional days. Grouping measurements that have the same statistical behavior becomes important because exceptionally high call volume days (such as New Year Day and Festival Day) might skew the results. 5. Call Arrival Patterns The first step in choosing the proper traffic model is to determine the call arrival pattern. Call arrival patterns are important in choosing a traffic model because different arrival patterns affect traffic facilities differently. The three main call arrival patterns are as follows:

• Smooth Call Arrival Pattern • Peaked Call Arrival Pattern • Random Call Arrival Pattern

Fig. 15.3 Smooth Call Arrival Pattern Fig. 15.4 Peaked Call Arrival Pattern 6. Traffic Models After the determination of the call arrival patterns and determined the blocked calls, number of sources, and holding times of the calls, one need to select the traffic model that most closely fits the environment. Although no traffic model can exactly match real life situations, these models assume the average in each situation.


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There are many different traffic models—the key is to find the model that best suits the environment. The traffic models that have the widest adoption are Erlang B, Extended Erlang B, and Erlang C. ERLANG B This is the most commonly used traffic model, and is used to work out how many lines are required if the traffic figure (in Erlangs) during the busiest hour is known. The model assumes that all blocked calls are immediately cleared. The Erlang B traffic model is based on the following assumptions:

• An infinite number of sources • Random traffic arrival pattern • Blocked calls cleared • Hold times exponentially distributed

In Erlang B traffic model, three variables involved such as Busy Hour Traffic (BHT), Blocking and Lines:

• BHT (in Erlangs) is the number of hours of call traffic there are during the busiest hour of operation of a telephone system.

• Blocking is the failure of calls due to an insufficient number of lines being available. E.g. 0.02 mean 2 calls blocked per 100 calls attempted.

• Lines are the number of lines in a trunk group. If two of the figures are known, the third can be obtained from Erlang B Table (shown for 24 trunks only).


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The Erlang B model is used when blocked calls are rerouted, never to come back to the original trunk group. This model assumes a random call arrival pattern. The caller makes only one attempt; if the call is blocked, then the call is rerouted. The Erlang B model is commonly used for first-attempt trunk groups where you need not take into consideration the retry rate because callers are rerouted, or expect to see very little blockage. Extended ERLANG B The Extended Erlang B traffic model is used by telephone system designers to estimate the number of lines required for PSTN connections (CO trunks) or private wire connections and takes into account the additional traffic load caused by blocked callers immediately trying to call again if their calls are blocked. This traffic model may be used where no overflow facilities are available from the trunk group being designed. The four variables involved are Busy Hour Traffic (BHT), Blocking and Lines:

• Recall factor is the percentage of calls which immediately retry if their calls are blocked.

• BHT (in Erlangs) is the number of hours of call traffic there are during the busiest hour of operation of a telephone system.

• Blocking is the failure of calls due to an insufficient number of lines being available. E.g. 0.03 mean 3 calls blocked per 100 calls attempted.

• Lines are the number of lines in a trunk group. � Of the Busy Hour Traffic, Blocking and Lines values, if two of the figures are known, the third can be worked out. 7. Measurement of Telephone Traffic. The total cost of providing telephone service can be roughly divided into those charges, which are constant and independent of volume of traffic and those, which are determined by the amount of traffic. The cost of subscriber’s line and instrument and certain individual equipment in the exchange is totally independent of the volume of traffic. The quantity of common switching equipment required is almost entirely dependent by volume of traffic. The quantity of such equipment is dependent not only on number of calls but also on duration of calls. Therefore to determine the quantity of switching equipment in automatic exchange or staffing in manual exchange telephone traffic may be measured in terms of both the number of calls and the duration of calls. For certain purpose it is sufficient to specify a Traffic Volume which is product of number of calls occurred during the time concerned


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by their average duration. However for the purposes of automatic exchange a more precise unit of traffic flow is required. This is called traffic intensity. Traffic intensity is the average number of calls simultaneously in progress. The unit of traffic intensity has been named the Erlang.

A traffic intensity of one Erlang is obtained in any specified period when the average number of calls simultaneously in progress during that period in unity. The specified period is always one hour and is taken as being the busy hour unless some other period is indicated. There is a more precise way to define traffic intensity. The average Traffic Intensity during a specified period T carried by a group of circuits or equipments is given by the sum of the holding times divided by T the holding times and period T all being expressed in the same unit. Sometime it is stated that the average traffic intensity is equal to the average number of calls that originate during the average holding time. All the above three definitions give the same numerical result. The foregoing relationships may be expressed symbolically as follows. Let S be sum of holding times during a given period T. both expressed in hours. Then by definition

A = S/T Where A is the average traffic intensity. Let C be the total number of calls during the period T then the average holding time ‘t’ hours per call, is given by t=S/C Then A = S/T Can also be written as A = Ct/T It also follows that when the average call duration is known, the average call intensity can be obtained by determining the number of calls occurring during the period T. Also because A is equal to average number of calls simultaneously in progress, an approximate value of A can be obtained by counting the number of occupied circuits or equipments at uniform interval during the time T and finding the average value. For example a group of selectors is examined at 5 minute intervals for one hour with following result-

Period = 1 2 3 4 5 6 7 8 9 10 11 12 Number of selectors held = 3 2 4 6 7 5 9 6 2 0 1 0 Total number of selectors held = 45 Average number of Selector held = 45/12 =3.75

The approximate traffic carried by the group was therefore 3.75 E. This figure is not necessarily accurate because between actual counts there may be greater or lesser number of simultaneous calls in progress. Making the period not more than twice the average holding time can reduce the error. A Commonly used interval is 3 minutes.


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8. Grade of service. Owing to the fact that calls originated in a pure chance manner, it is likely that during the busy hour some calls may fail to mature due to insufficiency of switching equipment. To ensure that the number of calls so lost is reasonably small, it is the standard practice switching equipment such that on the average not more than one call out of every 500 in the busy hour is lost at each switching stage, with the provision that loss does not fall below 1in 100 with a 10 percent increase of traffic. This allowable loss is termed the grade of service and is usually represented by the symbol ‘B’ with one lost call in 500 the grade of service is written as

B = 1/500 or B = 0.002 The Grade of service is a factor employed for dimensions of the exchange equipment. A few typical problems are worked out below to illustrate how the terms and definitions of telephone traffic are actually applied in practice. Example 1 If the calling rate per line per day in an exchange of 5000 lines is 6.0 and proportion of the traffic that occurs in the busy hours is 12 percent, what is the busy hour’s traffic in Erlangs, assuming an average holding time of 2.5 minutes per call? Calling rate per line per day = 6.0 Capacity of the exchange = 5000 lines Total number of calls made in a day = 5000 x 6 = 30000 Number of calls originated in busy hours = 30,000 x 12/100 Holding time of a call (t) = 2.5 minutes Busy hour traffic = C x t/60

= 3600 x 2.5/60 = 150 Erlangs or TUs.

Example 2 A group of trunks observed for 10 busy hours carried an average of 20 Erlangs and the total number of calls lost was 12. The calls had an average duration of 2 minutes. What grade of service was given?

Traffic carried by the selectors in one busy hour = 20 Erlangs

Average holding time = 2 minutes Total number of calls carried in one busy hour = 20 x 60/2 = 600 Number of calls lost in ten busy hours = 12 Average number of calls lost in one busy hour = 12/10 = 1.2 Total number of calls offered in busy hour = 600 + 1.2 = 601.2


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Grade of service = Number of calls lost Number of calls offered

Grade of Service = 1.2/601.2 = 0.001996, Say- 0.002 9. Erlangs formula Erlangs formula, which is given below, furnishes a method of computing the number of trunks (arranged in a full availability group) required to carry a given volume of traffic with a specified grade of service

AN

N! B = ..... (1) 1 + A + A 2 + A3 .....+.....An

1! 2! 3! N! When B = grade of service N = Number of trunks in a full availability group and A = average traffic offered.

The above formula is based on a number of assumptions, which approximate nearly to the full availability conditions actually obtained in practice. These assumptions are as follows...

(a) That calls occur individually and collectively at random. (b) That full availability conditions exist. (c) That the average traffic is the average of a large number of busy hours. (d) That calls which originate when all trunks are busy are lost and that such lost calls have a zero holding time. (e) That the condition known as statistical ‘equilibrium’ exists i.e. there is no tendency for the traffic as a whole to rise or fall.

10. Traffic offered Traffic offered to each trunk in a full availability group By definition, the grade of service (B) is the ratio of traffic lost to traffic offered. Thus if “A” is the traffic offered and ‘a’ the traffic lost, then B = a/A or a = A/B Traffic lost = Traffic offered X grade of service Substituting in Erlangs formula

AN N ! Lost traffic (a) = A x 1 + A + A2--------------- + AN 1! 2 ! N! Traffic offered to 1st trunk = total traffic = A Traffic offered to 2nd trunk = Traffic lost off the first trunk


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= Traffic offered to 1st trunk X GoS A 1 1! = A x 1 + A 1 1! A A = A x 1 + A = A2 1 + A .................... (2) Traffic offered to 3rd trunk = Traffic lost off the second trunk

A2 2! = A x 1 + A1 + A2 1 ! 2 ! A3 = ----------------- (3) 2 (1 + A + A2 ) 2 Similarly, the traffic offered to the Nth trunk AN = --------------- (4) N-1 (1+ A + A2 +.......+ AN-1) 1 ! 2 ! N-1 ! Traffic carried by each trunk in a full availability group Traffic carried by individual trunks is calculated by direct application of Erlangs formula as follows.

(i) Traffic carried by the 1st Trunk = Traffic offered to the 1st trunk-Traffic lost off the 1st trunk. Now traffic offered = Total traffic originated to 1st trunk = A and Traffic lost off 1st Trunk A2 = (Vide equation ......2) 1 + A Therefore Traffic carried by the 1st Trunk


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A = A - 1 + A A = ......................... (5) 1 + A (ii) Traffic carried by 2nd Trunk = Traffic offered to 2nd Trunk - Traffic lost off 2nd Trunk = Traffic offered to 2nd Trunk - Traffic offered to the 3rd Trunk A2 A3 = 1 + A 2 ( 1 + A + A2 /2 ) A2 ( 2 + A ) = ---------------- (6) 2 ( 1+A ) ( 1+A+A2/2 ) (iii) Similarly the Traffic carried by the Nth Trunk = Traffic offered to the Nth trunk - Traffic lost off Nth Trunk = Traffic lost off the (N-’) th Trunk - Traffic lost off the Nth Trunk = AB1 - AB --------------------- (7) Where “A” is traffic offered, B and B1 are the grades of service with N and (N-1) Trunks respectively. A N N! Now B = ----- (8) 1 + A + A 2 + ----------------+ A N-1 + A N 1! 2! N-1! N! A N-1 N-1! and B1 = ------- (9) 1 + A + A 2 + .........................+ A N-1 1! 2! N-1! Transposing equation (8)


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B (1 + A + A 2 + ................................+ A N-1 + A N ) = A N 1! 2! N-1! N! N! i.e. B ( 1 + A + A2 +........................+ A N-1 + BAN-1 = A N 1! 2! N-1! N! N! B ( 1 + A + A2 + A 3 --------- + A N-1 ) = A N ( 1-B ) ............. (10) 1! 2! 3! N-1! N! Similarly B1 ( 1 + A + A 2 + A 3 ................. A N-1) = A N-1 ............. (11) 1! 2! 3! N-1! N-1! B 1 = A N-1 x N! . B N-1! AN (1-B) N = A (1-B) BN or AB1 = (1 - B ) Now last trunk Traffic = AB1- AB BN - AB = 1- B B ( N A ) = 1-B When the grade of service is reasonable good, (1-B) is nearly equal to unit with this approximation. Last Trunk traffic = B(N-A) -------------------------------------- (12)

Typical Example

Calculate (a) The Lost Traffic and (b) The Grade of Service given by 4 switches arranged in a full availability group, when offered 0.53 Erlang With usual notation by Erlang formula A N N! B = 1 + A + A 2 + ...................................+ A N 1! 2! N! Substituting A = 0.53 Erlang and N = 4


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The GoS (0.53) 4 4! B = 1 + 0.53 + (0.53) 2 + (0.53) 3 + (0.53) 4 1! 2! 3! 4! 0.0789 24

= 1 + 0.53 + .1404 + .0245 + .003 0.003 = 1 = 1.698 566 = 0.00177 (a) Lost Traffic = traffic offered x grade of service = 0.53 x 0.00177 = 0.000928 Erlang

11. Traffic Forecasting and Planning In the development of a telecommunications network the time lag between identification of the need to provide subscribers’ equipment, lines and exchange plant, and the ability to meet those needs may be quite considerable. To augment the network we must accurately forecast these needs so that plant arrives and is installed before existing capacity is exhausted. In an ideal telecommunications network, with no restrictions, forecasting and planning would ensure that demand for services are accurately foreseen and satisfied as they arise. Types of Forecasts Demand Forecast To forecast the number of subscribers in a well defined area. For access network planning this forecast may be done for each section first and then this forecast may be combined to form forecasts for bigger areas like blocks and exchange areas. For the purpose of planning the junction network forecasts of subscribers are required in some of the methods. Traffic Forecast For access network planning forecast of traffic per subscriber and per block (originating and terminating) may be required. This would help in deciding the number optical terminals and number of subscribers per optical terminal when concentration is used. In case of design of ring structures where a number of optical terminals would be put in a ring, inter-block traffic may be of some importance.


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For junction network planning, total exchange originating and terminating traffics, traffics for different traffic zones and traffic dispersion (traffic interest) between different zones and exchanges will have to be projected. 12. Demand Forecast Procedure Period of Forecast The demand forecast would usually be made annually in short term, say for 3-5 years and then at an interval of 5 years such that the last forecast is for 20 year period. Method of Forecasting Forecast made at the corporate level follows "Top-Down" methodology using macro economic parameters and mathematical models to arrive at a country level forecast which is then appropriated to the regions, exchange areas etc. Forecast at the regional level or exchange area level (or below) will involve surveying and field studies. Information required for forecasting Forecasting studies are facilitated by availability of good geographical maps, network drawings and subscriber database. Geographical maps that would be useful are

• A city area map showing all the exchange location and physical features

• A detailed map of the exchange areas showing all the plots/buildings. This is usually in the scale of 1:1000

• Network drawings showing exchange areas, exchange boundaries, blocks, cabinet locations, cabinet/block boundaries would also be required.

Zoning of the area For ease of survey and correctness of forecasting a new area needs to be subdivided into manageable units. A city would have a number of exchange areas, an exchange area would have a number of blocks or cabinet areas and each block would have a number of sections or DP areas. To zone a new area, the area can be divided into survey units and teams of surveyor could then survey the assigned survey units and record information about the types of tenancies, residential or business that may exist, under construction or planned. From this information, tenancies/plots are grouped into sections. Sections should be predominantly residential or business areas or they could be areas where residential and business tenancies are uniformly mixed. Size of a section in governed by the size of DPs normally used. This could be an area that can be served by a DP of size 10 or 20 pairs. Multistoried buildings could


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make separate sections while in a low density area the size of section could be large. The section would then be grouped into Blocks in such a way that it should be possible to serve each block by a cabinet. In theory at least the cabinet and block areas are synonymous but in practice there may be times when one block has more than one cabinet. Section becomes the smallest unit of area for which the forecast is made. Section forecasts can be combined to make block forecast and the block forecasts are grouped to make exchange forecast. Classification of sections Detailed survey would be carried out in each section to identify the types of tenancies. Since all kind of tenancies do not have the same growth potential, the tenancies would need to be classified on some basis. A common method of classification is based on types of tenancies. Broad classification of tenancies used is:

• Residential • Business

Sub-classification is then done in each of the above categories. R1 could be detached houses, R2 could be Condominiums or luxury apartments, R3 could be Low cost housing and so on. Similarly for business, B1 could be big office complexes, B2 could be big shopping malls, B3 could be detached shops, B4 could be factories/workshops, B5 could be hospitals, government offices, schools etc., B6 could be restaurant, cinema, petrol station, parks, mosque/church/temple, museums etc. 13. Traffic Forecasting Traffic consideration has rarely been important for dimensioning the access network. The reason for this has been exclusive rights of a subscriber on the pair allocated to him. The maximum traffic that a pair can carry is 1 Erlang. Also that is the maximum a subscriber can generate. [Simply put, a circuit continuously busy during the observation period is said to be carrying 1 Erlang traffic]. With the advent of new technologies, use of concentration in the access network (a la V5.2 interface), subscriber will no longer have exclusive right over a channel from his phone to the exchange. A subscriber does not use his phone all 24 hours a day and channels can be more efficiently utilized if shared among many subscribers. This would need measurement and forecasting of traffic. Traffic data The production of traffic forecasts and the subsequent application of traffic theory to the dimensioning and administration of a telephone network depend on the availability and quantity of reliable reference


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data. These data must be systematically checked during collection and processing to ensure their integrity. Traffic data for planning purposes are of three main types: a) Exchange Data These are the general statistics which specify the traffic generating capabilities of part or all of the exchange and include measured data as well as data supplied from other sources, together with information derived from these data which includes:

• Number of subscribers of various categories. • Total originating and terminating traffics from groups of

subscribers within the exchange • Call holding times. • Usage rates (erlangs per subscribers)

b) Traffic Route Data These data usually include established routes which are dimensioned for a good grade of service and contain information about route size, identity, and the traffic carried by it. This would be required for dimensioning junction network c) Dispersion Data These data are held in the form of a set of row vectors for each originating exchange. They may contain call dispersion and/or traffic dispersion and associated mean holding time statistics. This would be required for dimensioning junction network 14. Traffic forecasting methods 1. Intuitive forecasting is the systematic assessment of informed opinion and is often the basis of subscriber surveys which are developed to produce forecasts of subscriber development. 2. Trend methods assume that the future will have a predictable relationship with past performance. Their application depends on the existence of a database of past statistics which can be analysed to determine past trends. Trend projection is a frequently used traffic forecasting method for the short to medium term. 3. Goal-oriented (Normative) forecasting assumes that there will exist needs in the future which will have some effect on the parameter being studied. This is of importance in longer term traffic forecasting, (typically in excess of 5 years), since the total traffic is dependent on a variety of other parameter such as population, subscribers and calling rates, etc.


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4. Comparison methods, where traffic patterns in a particular area are forecasted on the basis of known historical developments in another area. Economic Planning Periods When cable is laid it is necessary to provide some spare capacity to meet the future requirements. The number of years for which provision is made in advance is called the planning period. If a cable is provided for a long planning period, the initial cost per pair becomes less but considerable cable capacity remains idle for a long time period and such provision becomes uneconomical. On the other hand, if the cable is provided for too short a planning period, the cost per pair becomes very high and the cable provision again becomes uneconomical. There is an optimum period of planning for provision of cables. The economical planning periods for the primary network is short to medium term while that for distribution network is long term. Generally for bigger systems where the demand/rate of growth is high, it is not practicable to have long planning periods in view of uncertainty in forecast, the high capital cost involved, large scale introduction of fiber in the network and technological advances in the access network. It is therefore becoming increasingly common to plan the primary network for 3 to 5 years and distribution network for a 5-10 year forecast. Duct Planning The system of laying cables in pipes laid underground with provision of manholes/ joint boxes at specified distances so that, without resorting to repeated digging, these cables may be operated for rectification of faults or for joining with other cables. Additional cables may be laid subsequently on the same route in near future. Ducts are planned for long term. Planning New Technologies in the Access Network The term access network refers to the network between the local exchange and the subscriber. This network is still predominantly made up of the copper cable based point-to-point connections. This has kept the network in large proportions passive, inflexible and relatively unreliable. With the advent of digital technology, the process of installation, maintenance has become less cumbersome and quality of services has improved. It is therefore felt that the any cause for dissatisfaction, among customers about present services, is predominantly due to the frequent failures in the access network and the time taken for restoring them. One of the most fundamental and remarkable of the driving technologies in access network is the optical fiber.


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Telephone operators keep pace with the changing technologies and take the following steps:

• Provide infrastructure prepared for higher bandwidth, like fiber to the curb (FTTC) solutions.

• Move from a passive to a very active access network. • Provide network units, which will enable them to flexibly

provision a mix of services with minimum impact on network management and installed equipment base.

• Develop ring structures within the network to increase the subscriber loop reliability.


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Module - 01


Chapter – 16

Long Distance Switching Plans


• Charging Plan • National Switching Plan. • National Routing Plan • Transmission Plan • National Numbering Plan • Synchronous Plan • Signaling Plan


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1. Charging Plan The Charging plan was introduced on 1.1.71. The whole country was divided into 278 areas called as “Long Distance Charging Areas” (LDCAs). Each charging area comprised one or more revenue district with a nominated “Long Distance Charging Centre” (LDCC), which was an important town, or headquarters of one of the revenue district. The trunk calls were categorised into two types- Short Distance Trunk Calls and Long Distance Trunk Calls. The trunk calls between two telephone exchange systems in the same LDCA or the contiguous LDCAs were termed as Short Distance Trunk Calls. And the trunk calls between two telephone exchange systems in the two noncontiguous LDCAs were termed as Long Distance Trunk Calls. The Short distance Trunk Calls were charged on point to point basis as per the radial distance between the calling and called exchange system while the long distance trunk calls were charged on area to area basis as per the radial distance between the LDCC of the calling and called exchange systems LDCAs 1.1 Need for the provision of the this charging plan In this method of charging for each new exchange installation (i.e. new exchange system) calculation of radial distance of this exchange system from other exchange systems in the same LDCA as well as contiguous LDCAs is required for the determination of trunk call rates. Then this information has to be sent to all the concerned exchange systems. The country has now been divided into 321 Secondary Switching Areas. Therefore, the existing LDCAs have to be revised to meet this new concept. To further simplify the charging plan so that minimum calculation works is involved whenever a new exchange is installed. 1.2 Revised Charging Plan 1.3 Revision of Long Distance Charging Areas The country has been divided into 321 Secondary Switching Areas (SSAs), which are generally co-terminus with a revenue district. The technical and administrative infrastructure of the department has been changed on the basis of these SSAs. Therefore, the 278 Long Distance Charging Areas (LDCAs) have now been increased to 321 LDCAs to be co-terminus with the 321 SSAS. An important town in each LDCA has been nominated as Long Distance Charging Centre (LDCC) 1.4 Identification of Short Distance Charging Areas Each LDCA has been further divided into short Distance Charging Areas (SDCAs). One important town in each SDCA has been nominated as Short Distance Charging Centre (SDCC). Generally each SDCA


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comprise of one tehsil or revenue sub-division of a district. This has been done as per the following guidelines:

• As far as possible a whole Tehsil is made as a SOCA. • The area of SDCA is generally between 800 and 2000 Sq. Kms. • Where the area of a Tehsil has been below 800 Sq. Kms, it has

been combined with an adjacent Tehsil to form a single SDCA provided the distance between the farthest exchanges is up to 40 Kms.

• Where the area of a Tehsil has been above 2000 Sq. Kms. or where the distance between any two exchanges exceeds 40 Kms, it has been split into two more SDCAs as warranted by the existing or anticipated telephone facilities. There are a few exceptions where the splitting has not been done and the respective heads of circles have been requested to reconsider them for splitting then into smaller SDCAs.

• In the case of hilly areas the distance between the farthest exchanges in an SDCA can be up to 50 Kms. instead of 40 Kms.

Fig. 16.1 Charging areas


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1.5 Charging: The charging of calls is to be done on the basis of the following guidelines: -

• Call between the exchanges within the same SDCA: The STD call within the same SDCA is to be charged at the rate of one pulse every 180 seconds.

• Calls between the Exchanges in SDCAs Belonging to the same LDCA or Adjacent LDCAs:

The trunk calls (manual or STD) between exchanges in different SDCAs situated in the same LDCA or adjacent LDCAs are to be charged on the basis of radial distance between nominated Short Distance Charging Centre appropriate to the calling and called exchanges. In case of two exchanges systems whose local areas have a common border, one call unit per 180 seconds will apply for STD calls.

• Calls between the Exchanges in Non-Adjacent LDCAs: Trunk calls (manual or STD) between exchanges in non-adjacent LDCAs are to be charged on the basis of radial distance between the Long Distance Charging Centres appropriate to each exchange subject to a minimum of 51 Kms.

• Charging of Local Calls. The local calls emanating from electronic exchange in all telephone systems with an equipped capacity of 30,000 Lines or more are to be metered at the rate of one call unit every five minutes. The local calls in non-electronic exchanges of such systems or Local calls in telephone systems less than 30,000 lines will continue to be unit fee untimed. 1.6 Advantage of the Revised Charging Plan.

• The revised charging plan will once for all define the charging areas in the country, which will not undergo any change thereafter.

• Whenever a new exchange will be opened, for the calculation of charges from this exchange to other exchanges, only identification of the SDCA of this new exchange will be required.

• The telephone exchanges located in the periphery of bigger cities like Delhi, Mumbai, Chennai, and Pune etc. demand for the inclusion of their exchanges within the local area of the bigger systems on account of present high tariff of one unit for 36 seconds. The introduction of three minutes pulse per unit call will benefit these peripheral exchanges also and their demand for inclusion of these exchanges within the local area of the bigger systems will reduce.


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2. National Switching Plan. 2.1 General The objective of the National Traffic Routing and Switching Plan is to decide the structure of the network and also to define the manner of traffic routing. The first Plan was defined in 1965, which was subsequently revised in 1972 and 1987. 2.2 Switching Plan (PSTN). There will be 3-level TAX hierarchy for 321 TAX switches located at each Secondary Switch Centre (SSC). Level of a particular TAX is decided based on the hierarchy plan as well as the traffic emanating from the area served by TAX concerned. A terminal exchange will preferably be directly connected to the parent TAX switch. But in some cases it may be connected to the TAX through a Tandem exchange.

• 21 level I TAXs will be completely mesh connected. Overflow of traffic through other Level I TAXs (without circulating traffic) is to be provided.

• 160 Level II TAXs will be connected to the respective Level I TAXs as a backbone route. The level III TAXs, which were shown directly, parented to level I TAX stations and is being redesignated as Level II TAXs and hence the number of Level II TAXs has increased.

• 140 Level II TAXs will be connected to the respective Level II TAXs as a backbone route.

• Direct (or High usage) route will be opened from one TAX to another TAX (other than the backbone route) if the total traffic (outgoing and incoming) exceeds 12 Erlang for digital route and 8 erlangs for analog route.

• Tandem exchange may be provided in a large multi-exchange area to cater to mostly local calls. Terminal exchange preferably will be directly connected to the TAX for STD calls. However, small terminal exchanges any be connected to the TAX through a Tandem exchange.

• Rural Tandem exchanges may be provided in SDCAs with more than four exchanges to cater to intra SDCA calls and also STD calls via TAX. In other SDCA, terminal exchanges may be parented to the TAX directly. Exchanges of more than 512 lines may be directly parented to the TAX switch in the SSA.

• Terminal exchanges will be parented to Level I.II or level III TAXs. RLU / RSU/ Concentrator are to be treated as part of the terminal exchange.

• Parenting of Local Exchanges to more than 1 TAX would purely be governed by traffic and reliability consideration.


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• For international traffic each Level I TAX will be connected to at least one international Gateway Switch. For that Level I TAX where (outgoing and incoming) traffic is high (at least 30 erlangs), these will be connected to two Gateway Switch.

• Level II/III will be connected to Gateway Switch if the traffic (outgoing and incoming) is more than 15 erlangs.’

• All TAXs and Tandem exchanges should be 4 W switches.

At some locations, the requirement of TAX capacity makes it necessary to use two or more TAXs due to limitations of the TAX switches presently available. It is desirable to have larger capacity TAX so that multiple TAXs in same location are avoided. In the interim period if more that one TAX is used in a location, their configuration and use (e.g. functional separation, geographical usage separation or integrated approach) may have to be decided on case by case basis with due consideration of economics and security. Merits and demerits of an integrated TAX vis a vis the functionally separate TAXs of various levels at a particular location was examined and it was found that this is an involved exercise and deserves a separate study on the subject, hence, it was decided that an additional study can be taken up for studying the merits of an integrated TAX completely meshed with other level I TAX vis a vis the functionally separator TAX of various categories of various level at a particular location. The question of merger of level II and level III TAX can also similarly be referred as terms of reference to the future study indicated above. 3. National Routing Plan (PSTN) The principle of traffic routing showing choice of direct (or high usage) route, alternate route and backbone route is shown in Fig 16.2. In outgoing chain, direct route to the lowest chain where available is the first choice. This is followed by direct route to the next higher level in the terminating chain and so forth. And the final choice is the backbone route.


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Fig: 16.2 Traffic Routing

LEGEND: I - Level 1 TAX : II - Level 2 TAX : III - Level 3 TAX

TE --- Terminal Exchange T --- Tandem Exchange

Backbone Routes ______ Direct (or High Usage Route)

Example of Call between A4 and B4

First Choice

A4-A3-B3-B4

Second Choice

A4-A3-B2-B2-B4

Third Choice A4-A3-B1-B2-B3-B4

Fourth Choice

A4-A3-A2-B3-B4

Fifth Choice A4-A3-A2-B2-B3-B4

Sixth Choice A4-A3-A2-B1-B2-B3-B4

Seventh Choice

A4-A3-A2-A1-B3-B4

Eighth Choice A4-A3-A2-A1-B2-B3-B4

Final Choice A4-A3-A2-A1-B1-B2-B3-B4 (Backbone Route)

Note: Many of these routes may not be available.


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The grade of service will be 1: 200 for backbone route for each switching stage. High usage routes will be dimensioned according to economic criterion. Calls from a Terminal exchange should be routed to the TAX directly or through a Tandem exchange it is preferable to have direct route to the TAX except in

• Small exchanges in large multi-exchange area. • SDCA with large number of small exchanges.

The grade of service for connection between a terminal exchange and TAX/ Tandem will be 1: 200. The grade of service for connection between a Tandem and TAX will be 1:200. Traffic between terminal exchanges in a multi-exchange area will be routed direct where economically possible. In other cases, the traffic will be routed through a Tandem Exchange. Traffic between small terminal exchanges in SDCA will be routed through the Rural Tandem Exchange located at SDCC or the TAX located at SCC. Traffic from the Cellular will be routed via Level TAXs. International traffic to neighboring countries will flow through designated Level I TAX. International traffic to other countries will be sent from Level I and other TAXs connected to the International Gateway Switch. International traffic from other TAXs will flow through the backbone route to Level I (or Level 2 TAX if connected to Gateway Switch) and then to the International Gateway Switch. Similarly, incoming international traffic will also flow to level and other TAXs connected to Gateway Switch and through these to other TAXs. For high traffic countries international traffic from those TAXs, which are connected to two Gateway switches, may be routed to a designated Gateway Switch in consultation with VSNL. Overflow of such traffic will go to the other Gateway Switch. 3.1 Switching / Routing for Mobile Communication For Routing of calls meant for Mobile Communication Digital Tandem switches will be receiving the calls for Mobile Switching Centre (MSC) from either the local exchanges or distant TAXs. All incoming trunk groups from MSC and PSTN in the Digital Tandem switches will be required to be put under detailed observation. Following routes are identified for interworking between PSTN and MSC (PLMN.) 1. Traffic PSTN (Local) to PLMN (Local) Fig. 16.3 indicates that after analysis of ’98 PQR from S x S, Xbar and Digital Exchanges, call is routed to a Digital Tandem switch and then it routes the same with all 9 digits to the local MSC provided PQR is the valid code for the local network under consideration. For other invalid PQRs, NU tone will be fed.


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2. Traffic between PSTN (Local) and PLMN (Distant) Fig. 16.4 indicates that after analysis of ‘0’ in the local exchanges, calls are routed to the superior TAX. The same after analyzing 98PQR will route the call to distant TAX and the same will switch it to the desired MSC through distant Digital Tandem switch. 3. Traffic between PLMN and PLMN Subscribers (Same City) PLMN to PLMN subscribers call gets routed through MSC itself if more than 1 operator is there in the same city both MSCs are required to be linked up Digital Tandem will not be required for routing purpose. 4. Traffic between PLMN and PSTN (Local) Fig. 16.5 indicates that Local tandem on analysis of dialed digits other than ‘0’ and 98 routes the call to the desire PSTN exchange. 5. Traffic between PLMN and PSTN (Distant) Fig. 16.5 indicates that MSC (PLMN Exchange) traffic for distant PSTN station gets routed through Local Digital Tandem and Local TAX for all codes starting with ‘0’ Appropriate distant TAX and distant PSTN exchange are also then involved in the switching of the call. 6. Traffic between PLMN (Local) and PLMN (Distant) Fig. 16.6 indicates that MSC traffic for distant gets MSC routed through Digital Tandem Switches at both ends and also Digital TAXs if necessary for all codes starting with ‘098PQR’.

Fig: 16.3 Traffic between PSTN (Local) and PLMN (Local)


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Fig: 16.4 Traffic between PSTN (Local) and PLMN (Distant)

Fig: 16.5 Traffic between PLMN and PSTN (Local) & PSTN (Distant)


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Fig: 16.6 Traffic from PLMN (Local) and PLMN (Distant) 4. Transmission Plan 4.1 General The objective of the Transmission Plan is to define the required transmission quality of the telecommunication network. The National Transmission Plan was earlier issued in 1971-72. A revised Transmission Plan was issued in 1988. The transmission plan (1988) is not very explicit on some points. Switching Plan has also been changed which affects some aspects of Transmission Plan. In addition, it is also possible to improve upon some of the parameters so that the Transmission Plan can meet the CCITT objectives. The new Transmission Plan is given the following paragraphs. The Loss Plan, Noise Plan and Error Performance Plan are the most critical elements of the transmission Plan. 4.2 Transmission Loss Plan Telephone Instrument should meet the following Reference Equivalents.

� Minimum SRE (send reference equivalent) 3 dB � Minimum ORE (Overall reference equivalent) 2 dB � Maximum ORE 13 dB

Junction from local Exchange to TAXs should be on digital transmission media (4W). Junction loss (Loss from output 2W local Exchange analog switches or loss from 2W analog subscriber interface in case of digital local exchange to input of TAX switch) should be 3.5 dB.


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Junction from tandem exchange to TAX / Tandem should be on transmission media (4W) with 0 dB loss. The maximum Junction loss between two local exchanges (for local calls) should not exceed 9 dB. The maximum Junction loss between local exchange and tandem exchange (including hybrid loss) should not exceed 4.5 dB. Local PCM system should be lined for following losses.

� 2W switch to 2 switch 3 dB � 2W switch to 4 W switch 3.5 dB

Analog long distance circuits should be lined up for 4W to 4W with 0 dB loss. The loss between MUX center and local exchange / TAX should not exceed 1 dB (preferably they should be collocated). Relative level of channel at 4W points is – 3.5 dB. The traffic weighted mean ORE for the connection with these will be 14-16 dB and will meet the CCITT objectives.

Fig. 16.7 Transmission Loss Plans


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4.3 Transmission Noise Plan The following noise limits for analog circuits should be used: Distance Commissioning Maintenance Usable Limit (dBmop) Limit (dBmop) Limit (dBmop) a) Terrestrial Circuits 1 – 160 Km -68 -55 -52 161 – 320 km -55 -52 -49 321 – 640 km -53 -50 -47 641 – 1000 km -51 -48 -45 1001 – 2500 km -49 -46 -43 2501 – 5000 km -46 -43 -41 b) Satellite Circuit Maintenance -50 -48 -46 Others -47 -45 -43 4.4 Error Performance Plan (Digital System) The digital transmission will have following three grades for different applications.

• High grade – Above level TAXs (Digital Microwave, Coaxial and Optical Fiber)

• Medium grade – Between level I TAX and local exchanges (Digital Microwave Coaxial, Optical Fiber, cable PCM and UHF).

• Local grade – Between local exchange and subscriber (Digital subscriber, VHF and subscriber PCM)

The reference performance objective, commissioning limits and maintenance limits 64 kb/s digital section should be met for different grade of circuits. At other measuring bit rates DM and SES are almost same but ES may be correction as given by CCITT Recommendation G. 821 (Annexure 3.3). 4.5 Transmission Loss Plan The CCITT now has given recommendations for the loss plan in terms of loudness rating (LR). The corresponding reference equivalent (RE) is also indicated by CCITT. For telephones used in our network, information is available in terms of RE only and information in terms of LR is not available. Hence, the present transmission plan is with reference to RE. The considerations for required assignment of loss values in different parts of network are: Over all reference equivalent of the connection should not be very high or should not be very low CCITTs recommendations for ORE are:

• Long term objectives of 13-18 dB traffic weighted mean and • Short term objectives 13-23 dB weighted mean


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The maximum value of ORE is recommended as 34 dB. CCITT has recommendations that the maximum and minimum of SRE (send reference equivalent) up to send virtual switching point (-3.5dB) should not be lower than 9.6 dB and 6.dB respectively. This is to prevent overloading of the transmission systems. A minimum of 3 dB losses can be accepted between local exchange and 3-5 dBr points. Thus the minimum SRE of telephone instrument should be at least 3dB. For cross talk and echo consideration the minimum ORE of the subscriber circuit should not be less than 2 dB. The loss assignment should be such that the probability of singing or hollowness is very low (S…. occurs when loop loss gets reduce below 0.dB. Hollowness occurs when loop loss gets 3 dB.) Singing with one end open occurs during switching and fault conditions and its probability is much less). The stability of the channel and the return loss at the 2/4W point contributes to this stability. For long connection echo level to be controlled. The CCITT has given recommendations in terms of echo reference equivalent for different round trip delay. In mixed analog and digital network the stability of analog and digital channels are different and needs to be taken in consideration. The following points may be noted. Super Group Regulating Equipment (SGPR) has been introduced in the Indian network. With use of SGPR the stability of the channel is likely to meet the CCITTs recommendations of 1 dB standard deviation. The stability of channel derived from the digital medium will be much better. Each A to D converter is likely to give stability with a standard deviation of 0.4dB. Very few measurements on balance return loss has been done at the 2W/4W point. Limited measurements indicate that the balance return loss can be improved by changing balancing network from 600 ohms to 900 ohms. The existing value may have a mean of dB with standard deviation of 1.5dB. The above ranges of permissible noise in a connection when converted to pwop/Km may appear too large compared to the 2 to 3 pwop/Km quoted as design limits for the line noise of coaxial and microwave systems. However, it is interesting to note that the results of measurements made by other administrations on actual international connections reveal similar figures. The margin against misoperation of signaling receivers with these noise limits would be better than 18 dB (with MFC receiver sensitivity changed to 29 dBm against 36 dBm originally). The domestic communication satellite is planned to have commissioning limits of -50 dBmop for circuits between level I Tax and -47 dBmop for other circuits (e.g. main to primary and main to remote). The variations on such circuits are much smaller and these would give noise in connections similar to those of terrestrial circuits.


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Fixed services to remote and rural areas (mostly VHF and MARR) are being planned with noise values ranging from -46 to -50 dBmo. 4.6 Error Performance The Hypothetical Reference Circuit (HRX) of 27500 km is divided into 25000 km of High Grade and 1250 km of Medium Grade at each end and two local grades at each and International Model link. The performance objective of HRX (in percentage) is: DM 10 ES 8 SES 0.1 + 0.1 for propagation. Note: Errored Seconds (ES) is a second with at least one defect or anomaly. Severely Errored Seconds (SES) is a second with Binary Error Ratio (BER) of 10-3 or worse. Degraded Minutes (DM) is a group of 60 consecutive seconds after excluding SES with a BER of 10-6 or worse. The objectives are allocated as: High Grade 40% Medium Grade 15% (Each end) Local Grade 15% (Each end) (Propagation margin is not given for Local Grade System as per CCIR) 5. National Numbering Plan 5.1 Introduction The last few years have seen tremendous growth all around and particularly in the field of cellular mobile services. In some of the countries, these services have already exceeded the traditional basic services. In India too, the cellular mobile services have seen a growth of almost 100% during the last one year. Further, the existing Numbering Plan was meant to address monopolistic environment in national and international long distance dialing. The Government of India has since introduced unlimited competition in basic, National Long Distance (NLD) and International Long Distance (ILD) Services and licensed four service providers in respect of cellular mobile services in most of the licensed service areas. As such, it was felt to review the existing Numbering Plan and to formulate a plan, which will be futuristic, flexible and could cater to the numbering needs for about next 30 years in respect of the existing and likely new services. Keeping this in view, the new Numbering Plan has been formulated for a projected forecast of 50% tele-density by the year 2030 and thus making numbering space available for 75 crore telephone connections in the country comprising of 30 crore basic & 45 crore cellular mobile connections.


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5.2 Objectives of the plan

• To plan in conformity with relevant and applicable ITU standards to the extent possible.

• To meet the challenges of the changing telecom environment. • To reserve numbering capacity to meet the undefined future needs. • To support effective competition by fair access to numbering

resources. • To meet subscriber needs for a meaningful and user-friendly

scheme. • To standardise number length wherever practical. • To keep the changes in the existing scheme to the minimum.

Only the decimal character set 0-9 has been used for all number allocations. Letters and other non-decimal characters shall not form part of the National (Significant) Number [N(S)N]. Dialing procedure as per ITU Recommendation E.164 has been followed. The Short Distance Charging Area (SDCA) based linked numbering scheme with 10-digit N(S)N has been followed. This would expand the existing numbering capacity to ten times. 5.3 Salient features of the National Numbering Plan

• It is a SDCA based linked numbering scheme. • N(S)N is 10-digit for both the basic as well as cellular mobile

services. • The Subscriber Number (SN) for basic services will be of 6, 7 or 8

digits depending upon the length of SDCA code. • Basic to cellular mobile service calls shall use prefix ‘0’ only if Point

of Interconnect (POI) is not available in the same Long Distance Charging Area (LDCA) from where the call is originated.

• Cellular mobile using ‘0’ shall access basic services. • There is no change in the cellular mobile numbering structure. • Levels 0, 1, 7, 8 and 9 shall not be used as first digit for telephone

exchange codes in basic services. • There is no change in the numbering structure for paging services. • Carrier Access Code (CAC) for NLD and ILD has been defined as

‘10’. • Separate Carrier Identification Codes (CIC) has been earmarked for

toll and non-toll quality NLD and ILD services. • All the service providers shall use ‘100’, ‘101’ and ‘102’ for Police,

Fire and Ambulance services respectively. • ‘107X’ has been defined for emergency information services like

earthquake, floods, air and train accident etc.


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• Intelligent Network service access codes on ‘16XX’ are shifted to ‘18XX’.

• Trunk services codes are shifted from ‘18X’ to ‘150X’. • Certain level ‘1’ codes are earmarked for all service providers to

offer various subscriber related services, as per their choice, within their network.

• Voice Mail Service (VMS) access code is shifted from ‘93’ to ‘170’. • Enough spare levels/ codes are reserved for future needs.

5.4 Planning Period A Planning period of 50 years has been assumed for the National Number Plan presented here. 5.5 Population Growth Based on past growth, the population is expected to increase at a rate of about 2% every year. The National Numbering Plan is based on a Growth Factor of 2.5% for population growth during the 50-year period. 5.6 Telephone Growth It has been assumed that the National Numbering Plan should permit minimum telephone densities of 50% in large urban areas and 10% in non-urban areas, though in practice all areas may not develop equally or to those limits. 5.7 Local Charge Area As per present practice, charging distance for unit fee is 20 kilometer and Local Charge (Unit Fee) Area approximately 400 Sq. Kms. Each such Local Charge Areas of 400 Sq. Kms. has to be distinguished by a unique National Code. 5.8 Number Length The theoretical capacity of an 8 digit-numbering scheme is 100 million telephones and that of a 9 digit numbering scheme 1,000 million telephones. In actual practice, because of normal wastage in numbering scheme, necessary for leaving spare codes for specific purpose (such as digit 00’ for international trunk working, digit 01’ in local numbering scheme for special services, etc) the utilization of the total capacity of any Numbering Plan would be below 50%. Therefore a nine digit-numbering scheme has been adopted. 5.9 National Numbering Plan In brief, the National Numbering Plan allots.

(a) 4 two digit (AB) Codes for the major telephone systems at Mumbai, Kolkata, Delhi and Chennai.


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(b) 137 three-digit (ABC) codes for systems with present day telephone potential (working connections PLUS waiting list) of 500 and over.

(c) 314 three-digit (ABC) codes for Group areas up to 25,000 sq. Kms. areas and or two million populations, whichever is higher.

5.10 2-Digit (AB) Codes 2- digit (AB) codes have been allotted to Mumbai, Kolkata, Delhi and Chennai, which would enable the telephone systems in each of these large metropolitan areas to develop up to 5 million, telephones each. “ABC” codes with capacity of only 0.5 million would not be sufficient for these areas. A-digit codes “1” “2” “3” and “4” have been allotted to Delhi, Mumbai Kolkata and Chennai regions respectively, and accordingly following “AB” codes have been allotted to these metropolis for simplicity in remember double-digit numbers. 11 Delhi 22 Mumbai 33 Kolkata 44 Chennai 5.11 3-digit (ABC) codes for other Large telephone system To other large telephone systems with over 500 telephone potential today, i.e., working connections PLUS waiting list of over 500, one more 3-digit (ABC) codes have been allotted, ensuring that the ultimate date, the codes now allotted should permit a telephone density of at least 50%. In cases where with one 3-digit (ABC) code, the number of telephone would be restricted, to below 50% telephone density at the end of plan period, a second (or even a third 3 digit (ABC) code has been allotted, thus doubling (or trebling) the telephone potential of the area. Such exchange areas would be of the National code at a future data in the sense that the 3 digit (ABC) national code would become a 2 digit (AB) national code by absorption of the last digit, i.e., the ‘C’ digit, of the national code into the local numbering scheme. 5.12 3-Digit (ABC) Code for group Areas One 3-digit ABC code has been allotted to each Group area with up to 25,000 sq. Kms. areas or 2,000,000 populations at present, whichever is higher. In a group Area of up to 25,000 Sq. Kms. there can be up to 64 charging areas of 400 Sq. Kms. Theoretically, on each ABC code there can be 100 ABCDE codes. Therefore, even 64% utilization of such codes would cover all the charging areas. In practice, the Group areas are generally much smaller than 25,000 Sq. Kms. and therefore, the availability of National codes for the individual charging areas would be more than adequate. In some very special cases, such as the desert districts of Rajastan and Ladakk, Kashmir and NEFA, one 3-digit ABC code has been allotted


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for an area of more than 25,000 Sq. Kms. It has been assumed that in cases of such very sparsely populated areas, the limit of 20 Kms. for unit fee calls would itself need relaxation and a much higher distance limit for unit fee areas would be adopted, thus enabling 65 codes or less to be sufficient even for that Group Area. The limit of 2 million population for a Group Area (which excludes population of any large station within the Group Area which has been allotted separate National Code) enables each Group Area to expand up to 500, 000 telephones giving an ultimate date telephone density of 10% or over. Where the present population in the proposed Group Area is beyond 2 million or the area is more than 25,000 Sq. Kms., a second ABC code has also been allotted. In such cases, when doing detailed allocation of ABCDE National Codes to individual local charging areas within each Group Area, Group Area would be first divided into two sub-Group Areas, each with a separate 3-digit (ABC) code of its own. 5.13 Allocation of Codes With 100% utilization of all 10 digits ‘1’ to ‘0’ as ‘A’ ‘B’ and ‘C’ code digits, it would be possible to make available 1000 ABC codes. The actual requirements for ABC codes are much less and therefore only the following have been used as ‘A’ ‘B’ and ‘C’ digits. ‘A’ digit : 2,3,4,5,6,7 and 8 ‘B’ digit : 2,3,4,5,6,7,8 and 9 ‘C’ digit : 2,3,4,5,6,7,8 and 9 This leaves ‘9’ as spare ‘A’ digit for future allocation either within India by rearrangement or to adjacent countries, if this becomes feasible at a future date ‘0’ is also available spare as ‘B’ and ‘C’ digit for future allotment. The Country has been divided into 8 ‘A’ digit areas or region as follows. Digit ‘1’ Punjab, J & K., U.P.(Part) and Rajasthan (Part). Digit ‘2’ Maharashtra (Part), Gujarat and Rajasthan (Part) Digit ‘3’ West Bengal, Assam, NEFA, Sikkim and A & N Islands. Digit ‘4’ Chennai, Kerala and Lacadive and Minicoy Islands. Digit ‘5’ U.P (Part) Digit ‘6’ Bihar and Orissa. Digit ‘7’ Madhya Pradesh, Maharashtra (Part) and Rajasthan (Part). Digit ‘8’ Mysore and Andhra Pradesh. Nine Regions 1. Delhi 4. Chennai 7. Gauhati 2. Mumbai 5. Jullender 8. Bangalore 3. Kolkata 6. Ahmedabad 9. Nagpur Each ‘A’ digit area is sub-divided into nine ‘AB’ areas each ‘AB’ area comprising up to 9 continuous ‘ABC’ Group Areas. In grouping of


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‘ABC’ areas into ‘AB’ areas, Telephonic community of interest as well as present and proposed major trunk routing have been kept in view. The delineation of the Group area is present based on the political district boundaries and are tentative at this stage. They would be confirmed after delineating individual charging areas within each Group are for allocation of ‘DE’ codes. 5.14 National Numbering Scheme The prefix ‘000’ shall be used for home country direct service (Bilateral) and international toll free service (Bilateral). The format used is: ‘000 + Country Code + Operator Code’ except ‘000800’ which is used for bilateral international toll free service. 5.15 Sub level ‘00’ - International Prefix: The prefix ‘00’ shall be used for International dialing. It will be followed by country code and the N(S)N of the country to which that call is attempted. The format is as per ITU Recommendation E.164:

Prefix Country Code National (Significant) Number

00 CC N(S)N 5.16 Sub level ‘010’ - National Carrier Access (Prefix) Code: The prefix ‘010’ shall be used for selection of national long distance carrier. It will be followed by (National) Carrier Identification Code (CIC) and N(S)N. The format shall be as under:

Prefix Carrier Identification Code

National (Significant) Number

010 CIC N(S)N

Initially CIC shall be a two-digit code. This will be sufficient for allotment to 40 NLDOs (including NLDOs licensed for basic services) and 10 BSOs licensed only for basic services, considering that maximum of two codes may be allotted to each service provider depending upon toll quality and non-toll quality network. However to take care of all possible future requirements, length of CIC may be reviewed and changed to 3-digit code in future. The allotment of CIC may start from ‘10’ and codes ‘00’ to ‘09’ may be kept reserved. For intra circle long distance service, the carrier access code shall be the same as applicable for NLD service. The CIC from ‘10’ to ‘79’ shall be allotted to NLD service providers. For the NLD service providers, who


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are also Basic Service Operators (BSOs), same CIC shall be applicable for intra circle (service area) calls. CIC from ‘80’ to ‘99’ shall be allocated to the BSOs who are not licensed to provide NLD service. 5.17 Sub level ‘0’ - National Prefix: The prefix ‘0’ shall be used for national long distance calls (cellular mobile as well as basic services), intra service area (Circle) long distance calls of basic services, cellular mobile to basic services calls and calls from basic services to cellular mobile (depending upon point of interconnect). The format shall be as under:

• For basic services (PSTN) long distance calls:

Prefix National Destination Code Subscriber Number

0 2/ 3/ 4-digit trunk (SDCA) code

8/ 7/ 6-digit

• For basic services to cellular mobile calls if Point of Interconnect is

not available in the same LDCA from where the call is originated:

Prefix PLMN Access Code

MSC Code Subscriber Number

0 2-digit e.g. 98, 94 etc.

3-digit 5-digit

• For cellular mobile to cellular mobile calls outside the service area

from where the call is originated:

Prefix PLMN Access Code

MSC Code Subscriber Number

0 2-digit e.g. 98, 94 etc

3-digit 5-digit

• For cellular mobile to basic services calls:

Prefix National Destination Code Subscriber Number

0 2/ 3/ 4-digit trunk (SDCA) code

8/ 7/ 6-digit


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5.18 Sub level ‘011’ to ‘089’ - Geographical Number Range: These codes are also called trunk codes and identify a specified geographical area where a call is to be terminated. The national telecom network in India has been divided as under:

• SDCA: Short Distance Charging Area also called local area • LDCA: Long Distance Charging Area comprising of one or

several SDCAs ITU Recommendation E.164 provides four options for National Destination Code (NDC) structure. India has adopted type-2 structure for PSTN where NDC is the trunk (Area) code assigned to each SDCA. Each SDCA is allotted a unique trunk code. There are at present 2645 SDCAs distributed in 322 LDCAs. Accordingly, 2645 codes are required to identify the complete country based on SDCA linked numbering scheme. The length of the Trunk Code (TC) shall vary from 2 to 4 digits depending upon the size and telephone density requirement of the SDCA. Details regarding SDCA linked numbering scheme for PSTN, SDCA trunk codes (geographical) and the spare 2/ 3/ 4-digit codes are given at Annex-I, II and III respectively. 5.19 Level ‘1’ Special Services: Level ‘1’ is used for accessing special services like emergency services, supplementary services, inquiry and operator-assisted services. Some sub levels have been allocated for use by access providers (operators). These levels can be used for providing the services within their network. 5.20 Level ‘2’ to ‘8’ - PSTN Number: The numbers starting from ‘2’ to ‘8’ are reserved for PSTN within SDCA. At present the PSTN numbers in SDCAs vary between 5 to 7 digits so that the total N(S)N is 9-digit, except in a few cases where it is already 10-digit. It has been observed that the total telephone number requirement in some SDCAs with 5 digits numbers would be more than 50,000 in next 25 to 30 years. Similarly, in many SDCAs with 6 digits local number length, projected demand is likely to go beyond 5,00,000. Therefore, the subscriber number in SDCAs is increased by one digit so that the N(S)N is uniform and of 10-digit length. 5.21 Level ‘9’ Services: The range of numbers in level ‘9’ except ‘90’, ‘95’ and ‘96’ are reserved for cellular mobile services. These will be allocated in 2-digit format.


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5.22 Access to adjacent Area ‘95’ shall be used for accessing adjacent areas in the following format:

95 SDCA Code Subscriber Number Table of National Numbering Scheme

Number /Prefix Sub

level

Services

Structure

Remarks

0 National Long Distance Service

11-digit number ‘0+NDC+SN’ NDC+SN is also called as N(S)N

As per ITU Rec. E.164

11XX to 89XX range

National Destination Code

NDC=2 to 4-digit Also known as SDCA Code

0900 IN Service - Premium Rate

‘0900+XXX + IN Number’ Where ‘XXX’ is SCP code from ‘000’ to ‘999’

‘xxx’ to be allocated to IN provider

0901 IN Service - Universal Number (Long Distance)

‘0901+XXX + IN Number’ where ‘XXX’ is SCP code from ‘000’ to ‘999’

‘xxx’ to be allocated to IN provider

0902 UPT Service Not allocated1

0903 to

Reserved for IN Services

Not allocated1

091 to 093

Reserved for PLMN

Not allocated1

094 For dialling PLMN

11-digit number ‘0+94+MSC Code+SN' MSC Code=’000’ to ‘999’, SN=5 digits

.

0950 Reserved for Mobile Satellite Services

‘0950+XX+SN’ ‘XX’= service provider

0951 Reserved for Mobile Satellite Service

Not allocated1

0

0952 HVNET ‘0952+X+SN’ X is service provider


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Number /Prefix

0953 INMARSAT Mini-M

‘0953+X+SN’ ‘X’ is service provider

0954 Digital Satellite Phone Terminal

‘0954+X+SN’ ‘X’ is service provider

0955 ~

Reserved Not allocated1

096 Spare Not allocated2

097 Reserved for PLMN

Not allocated1


11-digit Number ‘0+98+MSC code+ SN MSC code =’000’ to ‘999’ SN= 5 digits


Not allocated1

1 Special Services 3 to N digits depending on service.

2 to 8 PSTN Subscriber Number

SN= 6 to 8 digits

9 Services 90 Spare Not allocated2

91 to 93

Reserved for PLMN

Not allocated1


10-digit Number ‘94+MSC code+ SN’ MSC code =’000’ to ‘999’ SN= 5 digits

95 Access to adjacent SDCA

‘95+SDCA code+ SN’

96 Paging Service 10-digit ‘96+XY+ Pager Number’ ‘XY’ is service provider code Where ‘XY’= ‘00’ to ‘99’

‘XY’= ‘00’ to ‘99’ to be allocated. Accessible from Outside Service area.


Not allocated1


10-digit Number ‘98+MSC code+ SN’ MSC code =’000’ to ‘999’ SN= 5 digits


Not allocated1


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5.23 Numbering Format The PSTN Numbering format shall be as per the table 1: Trunk Code (SDCA code)

+ Telephone Exchange Code

+ Last n digits of Subscriber Number

ABCD + EF + PQRS ABCD + EFG + PQR ABC + EF + PQRST

ABC + EFG + PQRS AB + EFG + PQRST AB + EFGH + PQRS

• Digit A can have any value from 1 to 8. • Digit B, C and D can have any value between 0 and 9. • Digit E can have any value between 2 to 8. [0, 1, 9 are NOT

allowed, since level ‘0’ is used as trunk prefix, level ‘1’ is used for special services, levels ‘7’ & ‘8’ are kept spare for future services and level ’9’ is used for cellular mobile services, paging services & access to adjacent areas.]

• Digit F, G and H can have any value from 0 to 9. • Digit P, Q, R, S and T can have any value from 0 to 9.

Dialing

• Dialing within SDCA For a call within a local area i.e. SDCA, subscriber number only will need to be dialed. The number of dialed digits will thus be 6, 7 or 8.

• Dialing outside SDCA For calls outside the SDCA, ‘0+N(S)N’ as per ITU E.164 or ‘0+10+CIC+N(S)N’ needs to be dialed. However, access to adjacent areas can also be on level ‘95’ followed by N(S)N. All such adjacent areas shall have to be accessible by dialing N(S)N with ‘0’ prefix or through carrier selection procedure. 5.24 General Guidelines

• The allocation of SDCA codes in the SDCA based linked numbering scheme shall be as per Annex-II.

• Spare codes, which are not allocated to any SDCA at present, are listed at Annex-III.


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• Two digits spare codes are reserved for allocation to those SDCAs in which numbering requirements may become more than 40 Lakhs in the long run.

• Certain three digit spare codes like 555, 666 and 888 are not to be used as SDCA codes. These are reserved for future services as indicated in Annex-III.

• 10-digit N(S)N is considered sufficient for more than 30 years. 5.25 General requirements for migration

• First digit of the telephone exchange codes can have any value between ‘2’ to ‘6’. Presently ‘2’, ‘3’ and ‘5’ are being used. ‘4’ and ‘6’ have been reserved for future use.

• Digit ‘2’ has been allocated as the first digit for BSNL/ MTNL numbers.

• For the private BSOs, digit ‘3’ or ‘5’ is to be used as first digit. • First digit in subscriber number should not be ‘0’, ‘1’, ‘8’ and ‘9’.

5.26 Numbers for Special Services (10) Emergency Services

100 POLICE 3-digit number

Restricted

101 FIRE 3-digit number

Restricted

102 AMBULANCE 3-digit number

Restricted

103 SPARE Not Allocated1

104 SPARE Not Allocated1

116 Wakeup call Registration

3-digit number

117 Wakeup call Cancellation

3-digit number

Restricted

118 Call waiting Registration

3-digit number

Restricted

119 Call Wailing Cancellation

3-digit number

Restricted

123

Dynamic STD Password Setting

3-digit number

Restricted

124

Class of Service Registration

3-digit number

Restricted


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180 IN Services

180X+YYY+ IN number, Where ‘X’=IN service ‘0’ to ‘9’ ‘YYY’= SCP code

‘YYY’ code to be allocated

5.27 Numbering for Cellular Mobile Network Allotment of Mobile Switching Centre (MSC), Signaling Point (SP) and Mobile Network Codes (MNC) for Cellular Operators The cellular mobile network has been divided into 19 service areas and 4 metropolitan cities for which licenses were issued. Accordingly Numbering Plan was formulated for allotment of MSC code for each operator with spare codes for future allotment. The format of cellular mobile service Numbering Plan is given below: ACCESS CODE MOBILE SWITCHING

CENTRE (MSC) CODE SUBSCRIBER NUMBER

2-digit 3-digit 5-digit 98/94 etc. ABC XXXXX

• Presently level ‘98’/’94’ have been allocated as access code for mobile networks.

• ABC=’000’ to ‘999’. However, ABC=’000’ to ‘099' have not been allocated and are reserved.

• XXXXX= 00000 to 99999. Initially in each service area/metro, two operators were given license to operate. With further opening up of mobile services, two more operators are permitted to operate in each service area/metro areas. The Signaling Point (SP) codes for mobile operators are given in 9000/10000 series. The MSC codes and SP codes allotted to the cellular mobile operators in different circles/metros.


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5.28 Guidelines for allotment of MSC codes

• MSC codes shall be allotted to the service providers from the series from which they are already issued the MSC codes.

• Additional MSC codes shall be allotted only when the subscriber base of 60% has been achieved with the allotted codes.

• When all the codes are exhausted from the same series, MSC codes from other series, which are not used or expected to be less used, may be allocated.

5.29 Mobile Number allotment All mobile numbers in India have the prefix 9 (This includes pager services, but the use of pagers is on the decline). Each zone is allowed to have multiple private operators (earlier it was 2 private + BSNL, subsequently it was changed to 3 private + BSNL in GSM 900/1800, now it also includes 2 private + BSNL in CDMA). All cell phone numbers are 10 digits long, (normally) split up as OO-AA-NNNNNN where OO is the operator code, AA is the zone code assigned to the operator, and NNNNNN is the subscriber number. 5.30 Mobile Numbering plan

• 92-xx-yyyyyy - TATA Indicom Numbers • 93-xx-yyyyyy - Reliance Mobile Numbers • 94-xx-yyyyyy - BSNL CellOne Numbers • 97-xx-yyyyyy - Various operators except Reliance, TATA and BSNL • 98-xx-yyyyyy - Various operators except Reliance, TATA and BSNL • 99-xx-yyyyyy - Various operators except Reliance, TATA and BSNL

Example: Mumbai Bharti Airtel - Pre fix: 98 , 99 MNC: 92 , 67 , 87 Vodafone Essar - Pre fix: 98 , 99 MNC : 20 , 19 , 33 , 30 MTNL - Pre fix : 98 , 99 MNC : 69 BPL Mobile - Pre fix: 98 MNC : 21 , 70


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6. Synchronisation Plan

A synchronization plan is the determination of the distribution of synchronization in a network. It involves the selection and location of a master clock or clocks, the distribution of primary and secondary timing, and the selection of the clocks and reference facilities. To achieve the best performance and most robustness from a synchronization network, several rules and procedures must be followed when developing a synchronization plan. Some of the most important rules are avoiding timing loops, maintaining a hierarchy, following the BITS concept, using the best facilities for synchronization reference transport, and minimizing the cascading of the timing reference. Timing loops occur when a clock uses a timing reference that is traceable to it. When such loops occur, the reference frequency becomes unstable. The clocks in a timing loop will swiftly begin to operate at the accuracy of the clock’s pull-in range. This will result in the clock exhibiting performance many times worse than it does in free-run or holdover mode. Therefore, it is important that the flow of timing references in a network be designed such that timing loops cannot form under any circumstance. No combination of primary and/or secondary references should result in a timing loop. Timing loops can always be avoided in a properly planned network.

Slips

If the node clocks in a telecommunication network operate asynchronously then, transmit and receive rates of telecommunication systems in each node would be different to the other nodes. In this case, the input buffers of the telecommunication systems would frequently overflow or underflow, causing data errors commonly referred to as slips. The object of network synchronisation is therefore to avoid and to minimise slips. This can only be achieved by synchronising all the node clocks, and hence all the telecommunication systems, to the same master clock or to a number of pseudosynchronous (very closely matched, nearly synchronous) master clocks. In practice, master clocks or Primary Reference Clocks (PRCs) are Cesium beam oscillators, and slave node clocks are usually Ovenised Crystal Quartz Oscillators (OCXOs).

Table 1: Effect of slips on services

Service Effect of slips

Voice (uncompressed) Only 5% of slips will lead to audible clicks

Voice (compressed) A slip will cause a click

Facsimile A slip can wipe out several lines


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Modem A slip can cause several seconds of drop out

Compressed video A slip can wipe out several lines. More slips can freeze frames for several seconds

Encrypted data protocol

Slips will reduce transmission throughput

The slip rate between systems can be calculated by equation below, and slip rates for 8k frames per second signals under various frequency differences are shown in Table 2.

Equation 1: Slips per day =

Frequency difference x Traffic frames/second x seconds/day (86400)

Table 2: Frequency differences and slip rates

Frequency difference between systems

Slip rate for 8k frames per second signals

0 0

10-11 1 slip in 4.8 months

10-10 1 slip in 14.5 days

10-9 1 slip in 1.45 days

10-8 6.9 slips per day

10-7 2.9 slips per hour

10-6 28.8 slips per hour

10-5 4.8 slips per minute

Apart from frequency difference, wander levels that exceed the input tolerance of telecommunication systems would also cause slips. Wander is slow modulation of the clock or traffic signals from their ideal positions in time and very low frequency (mHz) wander is impossible to filter out in a synchronisation network. Contraction and expansion of transmission cables under varying temperatures generate very low frequency wander on the traffic/ synchronisation signals. The levels of wander generated by optical fiber are lower than the copper cables.


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6.1 Maintaining a hierarchy It is the policy of our department to develop a national integrated digital network (IDN). To ensure satisfactory performance of the IDN, it is necessary to accurately control the rate at which digital signals are transmitted throughout the network. This can be achieved by synchronizing the clocks controlling digital switches and transmission systems. Further, Synchronisation of the digital network is an essential prerequisite to the introduction of high-speed data services and integrated services digital network (ISDN). The objective of the Synchronisation plan is to create a wholly synchronised national network meeting the CCITT recommendations on controlled slip rate (Recommendation G822).

6.2 Methods of Controlling Slips

(a) By Plesynchronous Operation

(b) By Synchronous Operation

6.3 Plesynchronous Operation In this method of operation, each exchange has its independent clock with very high stability i.e. the long-term frequency departure of 1x10E- 11 or more per year. There is no need to establish a synchronous network of the exchange clocks. The cost of such accurate clocks is very high and hence this method is no suitable for controlling slips in the national digital network.

6.4 Synchronous Operation In this method of operation the synchronisation of nodal clock is done in a continuous manner. There are two ways of synchronisation: Master-Slave Method: In this method, there is one master or Primary Reference Clock (PRC) that supplies the timing reference for the entire network. Almost all carriers with digital networks rely on Primary Reference Source (PRS) for timing since ITU recommends that networks operate with a long-term accuracy of 1×10–11. The PRS times all the equipment in the location in which it resides. This equipment, in turn, will time the rest of the network. The slip rate contribution of a PRS is usually negligible. A network, which derives timing from two PRS clocks, will experience at most, five slips per year, caused by the inaccuracy of the two clocks. This is negligible compared to the performance of receiver clocks. Receiver clocks typically operate with a daily performance that is 10 to 100 times worse than the PRS to which they are slaved. Therefore, it has been the trend of telecommunication network operators to rely more heavily on PRS clocks and to use multiple PRS clocks to time their network. This reduces the cascading of timing in the synchronization network.


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The locations in which the PRSs are used are determined by network topology. PRSs are usually placed in locations that will minimize the cascading of timing in the network. In this manner, the best performance can be achieved in the network. Additional sites that may require the use of PRSs are international switching locations. It is at these locations that one administration interfaces with another and all signals are transferred plesiochronously. It is important, therefore, to guarantee that these locations operate with the 1×10–11-frequency accuracy necessary for plesiochronous operation. Fig. 16.8 Synchronous operation through PRS The digital network is conceived as a hierarchy of digital nodes arranged in layers or levels. The PRS (Master Clock) supplies timing reference to nodes at the highest level (i.e. level I TAX), which uses this reference to phase lock, its own clock. Each node in the network receives the timing reference from a higher order node. The master clock is of high standard of accuracy and reliability say with long term frequency departure of 1x10E-11 per year. This method is shown in Fig. 16.9. Mutual Synchronisation Method: This is a concept for achieving a synchronous interconnected digital network without a master. When mutually synchronised, every exchange clock in the network is locked to the average of all incoming clock rates. A common system frequency is thus obtained by forcing a number of clocks to be interdependent on each other. This method of synchronisation is shown in Fig. 16.10.


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Fig. 16.9 Master-Slave Method Fig. 16.10 Mutual Synchronisation 6.5 Proposed Synchronisation Plan On the basis of technical and economic considerations, the Committee set up for this purpose has recommended the Master-Slave method on a hierarchical basis. The details of their recommendations are given in the following paragraphs.

• National reference clock centres There will be two National reference clock centres (NRC), one to be designated as Main National reference clock centre (MNRC) and the other as the backup NRC (BNRC). The MNRC with normally control the entire network. In the event of failure of MNRC, the Back-up National reference clock will take over the synchronisation function. The MNRC will be at Bombay established by VSNL. The BNRC will be installed at Delhi, dedicated and duplicated links will be provided between MNRC and BNRC.

• Clock Hierarchy

The clock hierarchy for the proposed synchronisation network closely follows the natural topology of the telecom traffic hierarchy. This is shown in Fig. 16.11. The MNRC/BNRC forms layer 1, the level 1 TAXs or layer 2, the level II TAXs form layer 3, the level III TAXs from layer 4 and local exchanges from layer 5.


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Fig. 16.11 Synchronous Hierarchy vs. Traffic Hierarchy

• Synchronisation Equipment Digital Network Synchronisation Equipment (DNSE) will be provided at each node i.e. each digital exchange in the network. The DNSE will be of the same type for all level of exchanges i.e. layer 2 to 5, to provide economies of scale and permit uniform maintenance practice. The synchronisation equipment will accept timing reference over 2 Mb/s PCM links. To account for possible link failures, a number of links (typically three) can be terminated with a pre-assigned priority, so that if the priority link 1 fails reference timing is derived from priority 2 link and so on. The equipment will phase lock the clock of its node to a timing reference supplied from another predetermined node. The timing output obtained from the DNSE will then be supplied to the digital exchange.

• Synchronisation links

Normally the synchronisation links will from part of the traffic circuits except for links from the two NRCs to the four metro centres. These will be dedicated links. The links from the MNRC will be the main links ‘M’ and those from BNRC the standby links ‘S’.


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Both the NRCs will monitor each other’s timing references received on the duplicated dedicated links between them. In normal operation, the BNRC will not utilise its own master clock output; instead, it will divert the received reference from the MNRC on the standby links. Thus both links ‘M’ and ‘S’ will carry timing reference from MNRC. In the event of failure of the timing reference of the MNRC the BNRC will feed its own timing reference on the ‘S’ links. In this case, the MNRC since it is faulty diverts the received reference from BNRC on the ‘M’ links. This scheme provides an extremely secure arrangement of feeding primary timing reference, adequately taking care of both link failures and failure of an entire NRC. The synchronisation information shall be passed between clocks in a strict hierarchical manner. All synchronisation links are unidirectional i.e. timing information flows in only one direction and only form a higher level to a lower level or at the same level. Each exchange in the network will have up to three independent inputs for improved reliability. These will be independent with respect to availability. The layer 2 exchanges will have timing inputs through ‘M’ and ‘S’ links. In addition they may also receive a third reference input from other layer 2 exchanges. This arrangement is shown in Fig. 16.12.

Note: Only one layer 2 exchange shown in figure Fig. 16.12 Reference Timing inputs to Layer 2


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Each exchange at layer 3 will have up to three reference inputs. One will be from its own parent layer 2 nodes, second from one of the non-parent layer 2 node and third will be from one of the designated layer 3 node. Links will be provided taking diversity of the media into account. A typical connection is shown in Fig. 16.13.

Ref. timing is from layer 2 to layer 3 and links are provided with media diversity

Fig. 16.13 Reference inputs to Layer 3 (Example) Each exchange at layer 4 will have up to three reference inputs – one form its own parent layer 3 nodes, second form one of the non-parent layer 3 node and third form one of the layer 4 node designated for this purpose or from layer 2 node if possible. Links will be provided taking diversity of the media into account. A typical connection is shown in Fig. 16.14.

Media Diversity to be ensures Fig. 16.14 Reference inputs to Layer 4 (Example)


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The layer 5 exchanges (local exchanges) will also have three reference inputs from exchanges from higher layers or the same layer. In a multi exchange building, the distribution of synchronisation information to the exchanges within the building may be by coaxial or similar cable. One Digital Network Synchronisation equipment can be used to supply timing to several exchanges. 6.6 Clock Quality The clock characteristics at different layers as recommended by the Committee are:

• Layer 1: Primary timing reference standard with a minimum stability of

1x10E-11 (for the life of caesium beam tube). • Layer 2: Clock with a minimum stability of 2x10E-10 per day. • Layer 3: Clock with a minimum stability of 2x10E-10 per day. • Layer 4: Clock with a minimum stability of 6x10E-8 per day. • Layer 5: Clock with a minimum stability of 10E-6 per day.

Stringent Synchronization planning must be done for all networks if performance objectives and service needs are to be met. In carrier networks, the major focus of synchronization planning lies in the determination of timing distribution and the selection of clocks and facilities used to time the network. Careful attention must be made to the selection of clocks and reference facilities, and to the minimization of reference cascading. In private networks, the major goal of synchronization planning is to reduce errors caused and propagated by poor CPE clocks. This requires limited use of stratum 4 CPE clocks, limited cascading, use of as many carrier-timing sources as possible, and the use of BITS architectures. 7. Signaling Plan The worldwide signaling network has two functionally independent levels: the international level and the national level. This provides for a clear division of responsibility for signaling network management and allows identification plans of signaling points in the international network and the different national networks to be independent of one another. Within the International Signaling System No. 7 network, an International Signaling Point Code (ISPC) identifies a signaling point while within the National signaling system No.7 network, a signaling point is identified by a National Signaling Point Code (NSPC). The Point Code Management function ensures that the current and future signaling address requirements for the telecommunications


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industry are properly met. The detailed texts and definitions are available at ITU-T recommendations in Q.70x series recommendations. 7.1 Definitions

• Signaling Point (SP): A node in a signaling network that originates and receives signaling messages, or transfers signaling messages from one signaling link to another, or both.

• Signaling relation: An association between two signaling points that allows interexchange of Signaling System No. 7 messages.

• Signaling Point Code (SPC): A code used to identify a signaling

point and processed within the Message Transfer Part (MTP) of each signaling point and within users of the MTP.

• International Signaling Point Code (ISPC): A signaling point code

with a unique 14-bit format used at the international level for signaling message routing and identification of signaling points involved. The ISPC is used in signaling messages containing the Network Indicator NI=00.

• Member State: Country" (or geographical area), and/or "Regulator"

shall be considered as Member State.

• Signaling links: Signaling links are basic components in a signaling network connecting together signaling points. The signaling links encompass the level two functions, which provide for message error control (detection and subsequent correction).

7.2 Signaling Links CCS7 messages are exchanged between network elements over 64kbps or higher speed bi-directional channels called Signaling links. Signaling occurs out-of-band on dedicated channels rather than in-band on voice channels. Compared to in-band signaling, out-of-band signaling provides:

• Faster call setup times (compared to in-band signaling using multi-frequency (MF) signaling tones)

• More efficient use of voice circuits • Support for Intelligent Network (IN) services which require

signaling to network elements without voice trunks (e.g., database systems)

• Improved control over fraudulent network usage


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7.3 Signaling Points Each signaling point in the CCS7 network is uniquely identified by a numeric point code. Point codes are carried in signaling messages exchanged between signaling points to identify the source and destination of each message. Each signaling point uses a routing table to select the appropriate signaling path for each message. There are three kinds of signaling points in the CCS7 network as shown in Fig. 5.14.

• SSP (Service Switching Point) • STP (Signal Transfer Point) • SCP (Service Control Point)

Fig. 16.15 SS7 Signaling Points SSPs are switches that originate, terminate, or tandem calls. An SSP sends signaling messages to other SSPs to setup, manage, and release voice circuits required to complete a call. An SSP may also send a query message to a centralized database (an SCP) to determine how to route a call (e.g., a free 800 call). An SCP sends a response to the originating SSP containing the routing number(s) associated with the dialed number. An alternate routing number may be used by the SSP if the primary number is busy or the call is unanswered within a specified time. Actual call features vary from network to network and from service to service. Network traffic between signaling points may be routed via a packet switch called an STP. An STP routes each incoming message to an outgoing signaling link based on routing information contained in the SS7 message. Because it acts as a network hub, an STP provides improved utilization of the SS7 network by eliminating the need for direct


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links between signaling points. An STP may perform global title translation, a procedure by which the destination signaling point is determined from digits present in the signaling message (e.g., the dialed 800 numbers, calling card number, or mobile subscriber identification number). An STP can also act as a "firewall" to screen SS7 messages exchanged with other networks. Because the SS7 network is critical to call processing, SCPs and STPs are usually deployed in mated pair configurations in separate physical locations to ensure network-wide service in the event of an isolated failure. Links between signaling points are also provisioned in pairs. Traffic is shared across all links in the link set. If one of the links fails, the signaling traffic is rerouted over another link in the link set. The SS7 protocol provides both error correction and retransmission capabilities to allow continued service in the event of signaling point or link failures. 7.5 Structure of international and national signaling networks The national and international networks are considered structurally independent and, although a particular, signaling point may belong to both networks, signaling points are allocated signaling point codes according to the rules of each network. The most elementary signaling network consists of originating and destination signaling points connected by a single signaling link. To meet availability requirements this may supplement by additional links in parallel, which may share the signaling load between them. If, for all signaling relations, the originating and destination signaling points (OSP and DSP) are directly connected in this way in a network, then the network operates in the associated mode. The worldwide signaling network is structured into two functionally independent levels, namely the international and national levels, as illustrated in Fig. 16.17. This structure makes possible a clear division of responsibility for signaling network management and allows numbering plans of signaling points of the international network and the different national networks to be independent of one another. A signaling point (SP), including a signaling transfer point (STP), may assign to one of three categories:

• National signaling point (NSP) (signaling transfer point) which belongs to the national signaling network only (e.g. NSP1) and is identified by a signaling point code (Originating Point Code –OPC


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or Destination Point Code-DPC) according to the national numbering plan of signaling points;

• International signaling point (ISP) (signaling transfer point) which

belongs to the international signaling network only (e.g. ISP3) and is identified by a signaling point code (OPC or DPC) according to the international numbering plan of signaling points;

• A node that functions both as an international signaling point

(signaling transfer point) and a national signaling point (signaling transfer point) and therefore belongs to both the international signaling network and a national signaling network and accordingly is identified by a specific signaling point code (OPC or DPC) in each of the signaling networks.

Fig. 16.17 Structural of international and national signaling networks 7.6 International and national signaling networks Criteria The signaling network structure must be selected to meet the most stringent availability requirements of any User Part served by a specific network. The availability of the individual components of the network signaling links (signaling points and signaling transfer points) must be considered in determining the network structure.


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In order to take account of signaling message delay considerations, regard should be given, in the structuring of a particular signaling network, to the overall number of signaling links (where there are a number of signaling relations in tandem) related to a particular user transaction (e.g. to a specific call in the telephone application). For all messages for the same transaction (e.g. a telephone call), the MTP will maintain the same routing if the same signaling link selection code is used in the absence of failure. However, a transaction does not necessarily have to use the same signaling route for both forward and backward messages. The number of signaling links used to share the load of a given flow of signaling traffic typically depends on

• The total traffic load; • The availability of the links; • The required availability of the path between the two signaling

points concerned; and • The bit rate of the signaling links.

Load sharing requires at least two signaling links for all bit rates, but more may be needed at lower bit rates. When two links are used, each of them should be able to carry the total signaling traffic in case of failure of the other link. In the international signaling network, the number of signaling transfer points between an originating and a destination signaling point should not exceed two in a normal situation. In failure situations, this number may become three or even four for a short period. This constraint is intended to limit the complexity of the administration of the international signaling network A 14-bit code shall be used for the identification of signaling points. For National Signaling Networks, no specific structures are required; however, Administrations should cater for the requirements imposed on a national network for the protection of international services in terms of network related user requirements, such as availability and performance of the network perceived by users. The signaling points and the signaling transfer points, which are involved in a signaling of cross–border traffic, should belong to the international hierarchical level. When those signaling points or signaling transfer points are also involved in signaling of national traffic, they should belong to their national hierarchical level as well. Therefore, the double numbering of signaling point codes based on both the international and national numbering schemes should be required. The Network Indicator in the service information octet as described in section 6 makes discrimination between international and national point codes.


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7.7 National Signaling Point Code (NSPC) Format This section describes the format of the code used to identify National Signaling Points Codes in the national Signaling System No. 7 network, which is identified within the signaling system by the Network Indicator (NI) as follows:

• NI=10 is national network (currently used) • NI=11 is national network (Reserved for other national networks, if

needed. The National Signaling Point Codes are the numbers that uniquely identify a network (NE) in a CCS7 network. The format of the 14-bit binary code used for the identification of national signaling points. The 14 bits of NSPC are first converted to a five- digit decimal number denoted ABCDE. That will be range from 00000 to 16383. The NSPC (ABCDE) decimal number will be divided into two fields. The first field will be consisting of three decimal digits (ABC) representing the Network Identity. The network identity will have 164 blocks, 163 of which have the capacity of 100 codes and one (#164) with a capacity of 83 codes. The second field will be consisting of two decimal digits (DE) representing the signaling point code. Each block of (DE) will have a capacity of 100 signaling point codes. The NSPC structure is illustrated below: Network Identity Signaling Point Code ABC DE 3 digits 2 digits Where ABC= 000 to 163 Where DE has a range from 00 to 99 for all

ABC values except for ABC=163, DE has a range 00 to 83.

7.8 International Signaling Point Code (ISPC) Format This section describes the format of the code used to identify international signaling points in the international Signaling System No. 7 network which is identified by the Network Indicator NI=00. The format of the 14-bit binary code used for the identification of international signaling points is illustrated below. Three (3) decimal numbers represent the binary code as follows:

• The first indicating the three (3) most significant bits (NML), with a range of 0 to 7;

• The second indicating the following eight (8) bits (K-D), with a range of 000 to 255; and.

• The third consisting of the three (3) least significant bits (CBA), with a range of 0 to 7.


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The combination of the fields containing bits NML and bits K-D is regarded as the Signaling Area/Network Code (SANC). The three (3) bits (CBA) identify a specific signaling point which when combined with the SANC forms the 14-bit ISPC (e.g. 2-068-1). 7.9 Standards SPCs shall conform to relevant and applicable international standards. Particular attention is drawn to the following ITU-T Recommendations:

• Q.704: Signaling Network Functions and Messages

• Q.705: Signaling Network Structure

• Q.708: Numbering of International Signaling Point Codes


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Module - 01


Chapter – 17

Network Management


• Introduction • Objectives of Network Management • Present Scenario • Success Factors for Network Management • Network Management Process and Procedures • Network Management Strategy • Network Traffic Management • Functions within the NMC • NTM Principles and Objectives • NTM measurements and Parameters


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1. Introduction Network management means deploying and coordinating resources in order to plan, operate, administer, analyze, evaluate, design and expand communication networks to meet service level objectives at all times at a reasonable cost and with optimum capacity. Most organizations have recognized the strategic importance of their communication network and its management. In most cases, better control ensures a higher level of performance and this performance corresponds with higher productivity. In addition, higher productivity often translates into bottom line financial improvements. This leads us to consider what the principal driving forces are for investing and spending more on network management. • Controlling strategic assets • Controlling complexity • Improving service • Balancing various needs • Reducing downtime • Controlling costs 2. Objectives of Network Management Based on surveys with a number of users including large, medium and small environments, what they required and want, may be listed as follows: • Assurance of continued end-user service, characterized by availability and quick response time, despite growth and change: End-users are interested in maintaining a certain agreed-upon level of services, which may necessitate connecting end-user devices to local and or remote computing facilities by point-to-point, multipoint communication links, or by using LAN's, MANs, or specific communication solutions. Manageability at this level requires powerful measurement techniques in both logical and physical components of the networks. Changing technology and growth rates must not impair the level of service. • Capability to heal bypass, or circumvent failed network elements as automatically as possible by integrating physical and logical network management. Early detection and powerful alarm correlation techniques should help diagnose problems quickly. As a result, the right strategy to repair, bypass or circumvent failed components may be rapidly selected and implemented. Artificial intelligence will pay a key role in this activity. Constant availability of individual diagnostic tests, preferably from any point in the network is desired as well.


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• Capability to operate fully even when important network elements have failed: Powerful backup components and procedures for both the physical and logical segments of the network are expected to help in resuming service with no or minimal performance impairment while trouble shooting failed components. Network management is expected to guide and supervise this activity as part of fault management. • Capability to monitor and diagnose unsatisfactory conditions in the entire network, including systems software and applications: Performance monitoring is expected to embrace a broader scope than ever before. Not only physical and logical network components, but also a portion of the server's software, database activities and applications need to be included. By doing so, real end-to-end network management may be offered to users. • Real-time or near-real-time analysis of network performance: users request not only historical data for performance trending and thresholding, but also real-time or near-real-time information. Early recognition of performance bottlenecks, and selecting alternate routes, may help to return the level of service to the expected range. Powerful measurement techniques with low overhead data collection combined with efficient real-time processing are the key prerequisites. • Statistics and historical data automatically saved in a standard data base with vendor-provided formats for analysis and also provision for users to support their own analysis formats, first, to offer portability, a general purpose data base is recommended for use as a repository; second for satisfying specific needs, flexibility is requested in terms of processing, presentation services and operational procedures. Users expect a general-purpose platform, supported by the majority of vendors; first, in further steps, event and alarm consolidation, correlation and finally integration across different systems are on users request lists. Increasing operations productivity by controlling quantity of staff, number of operations sites, and skill levels: Due to a serious shortage of human resources, users are looking for solutions that can help to stabilize the number of employees required despite growth and new applications support. Staff reduction is high priority in some voice-only networking environments. People are also looking to artificial intelligence as a significant means of supporting human work that can provide some consistent assistance in problem determination. One of the scenarios for the next century is a lights-out network control center. The path to such a reality is via automated and unattended operations. • Providing a powerful network management database for supporting operations, administration, analysis, and planning: Users are looking for a central information repository as an ultimate solution. This repository or management information base (MIB) is expected to store all relevant information about components, procedures, operational rules, projects, and so on. Users expect this to be a relational or object-


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oriented database. They expect this database to store objects instances, conductivity data, and status. The database then forms the basis of the documentation system. • Rapid continual response to changing network applications, subscribers, devices, tariffs, and services: Dynamic adoption to the ever-changing environment is a key item that drives network management expenditures. While awaiting the development of a central information repository, users will very likely compromise on a directory device, with other word; acting as an umbrella connecting all existing databases and files. Performance expectations are modest all the beginning. X.500 seems to be an interesting alternative in this respect. • Dynamic expansion and reconfiguration of network capacity using bandwidth management techniques: Co-operation between users and suppliers is expected to improve substantially. Users want to gain insight into networking segments they do not own or control. To remain competitive, leading suppliers will very likely offer the opportunity in other form of importing/exporting data. Electronic data interchange is considered the real target. • Use of one network-generation source: By consistent naming and network addressing conventions, there is no need to have too many 'views' of a network. Different views have grown historically as different management products have set other specific definitions of management depth and scope. If the configuration database is considered the single and only source of generating the network, the development of unnecessary contradictory names and addresses can be avoided. • Increasingly accurate and simplified accounting data: To supervise spending and control costs, users request timely access to accounting information for both data and voice networks. It is envisioned that suppliers on a near real-time basis will provide station detail maintenance records to users. • Implementation of generic applications: Users are actually looking for generic applications across single functions, groups of functions, and instrument. Such application packages may be purchased or leased by the user. • Integrated network management: Users are requesting increasing technical levels of integration. This means integrated solutions across consolidated forms, multiple network architecture's processing systems and network elements, geographical areas, multiple vendors and private, virtual, and public network management instruments. • Centralization with distributed implementation: Users tend to prefer a central network-management solution, with implementation of certain "local" functions controlled by the central facility. In particular, users generally request a solution for managing remote local area networks by means of some distributed monitoring features.


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• Practical implementation of solutions based on international standards: To avoid the confusion of using too many proprietary solutions, international standards are preferred, but not at any price. Users generally request the co-existence of leading de facto standard solutions, such as IBM's Net View with OSI-based solutions or AT&Ts Accumaster. In addition, SNMP may be considered for peer-to-peer connections. • Integrated reporting of data flows and capacity trends, as well as standard performance metrics: Users are expecting powerful reporting features with a minimal number of periodic paper-based reports, but with the opportunity of accessing data elements and generating adhoc reports on indicators needed for a specific application area. Standard database and access features are highly ranked. 3. Present Scenario There is a serious gap between what users require or want and the present status of network-management applications, instruments, and use of human resources. When analyzing data and voice networks costs, the results show a continuous decrease in equipment cost and an increase in communication costs. The equipment cost decrease is due partially to large-scale integration at the hardware level and partially to standardization results at the software level. The increase in communication costs is explainable as part of the progress of distributing computing power and databases with the consequent push to connect surprisingly high numbers of stand-alone user devices. Surveys confirm annual growth rates at the end-user device level in the range of 35% to 45%. The trend of ever-increasing communication costs is expected to continue as non-networked personal computers and local area networks continue being hooked together with the increasing variety of networking options. The alarming fact, however, is that people costs are rapidly increasing. The reasons may be found in one or more of the following: networks never get smaller or less complex, resulting in a demand for more human resources: higher skill levels of analysts and planners are paving the way for higher salaries; expanding the scope and depth of service to users requires still more people, especially in the operating area. There is a push from company’s management for budgets that are more evenly distributed between data processing and communication resources. In terms of the basic communication forms, the current situation may be characterized as follows: • Voice is strictly centrally managed and not yet ready for integration. • Data is both centrally and decent rally managed: there is interest to include voice and image management. � Image including word is


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strictly decent rally managed: image is beginning to be ready for electronic mail. • Video is strictly centrally managed: at the moment this communication form is limited to innovators. In summary, the integration process over multiple communication forms will very likely take many years. Data and voice integration will happen first, driven by shared communication form facilities and integrated equipment. 4. Critical success Factors for Network Management Critical success factors are those few key areas of activity in which favorable results are absolutely necessary for an organization to reach its goals. The goal for managing networks is to maintain end-user service levels and thus ensure that the network is operating effectively and efficiently at all times in order not to cause any problems in the corporation's short-middle and long-range operations. Critical success factors for network management are: Processes and procedures: Sequence of application steps including guidelines for how to use tools necessary to execute network-management functions. Instruments: Hardware and software or both, for collecting, compressing data basing information and predicting future performance of network components. Human resource addresses the first of these factors i.e. process & procedure. The following sections will address each of these critical success factors individually. 5. Network Management Process and Procedures Fig. shows the most important network-management subsystems based on the recommendations of international standard organizations. The subsystems consist of a number of well-defined functions that are supported by many processes and procedures in practical implementations.


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Fig. 17.1 Network Management Sub-systems Configuration management is a set of middle and long-range activities for controlling physical, electrical, and logical inventories, maintaining vendor files and trouble tickets, supporting provisioning and order processing managing changes and distributing software. Directory service and help for generating different network generations are also provided. Fault management is a collection of activities required to dynamically maintain the network service level. These activities ensure high availability by quickly recognizing problems and performance degradation, and by initiating controlling functions when necessary, which may include diagnosis, repair test, recovery, work around, and backup. Log control and information distribution techniques are supported as well. Performance management defines the ongoing evaluation of the network in order to verify that service levels are maintained, identify actual and potential bottlenecks and establish and report on trends for management decision making and planning. Building and maintaining the performance database and automation procedures for operational control are also included. Security management is a set of functions to ensure the ongoing protection of the network by analyzing risks, minimizing risks, implementing a network security plan, and by monitoring success of the strategy. Special functions include the surveillance of security indicators, partitioning password administration, and warning or alarm messages on violations. Accounting management is the process of collecting, interpreting processing and reporting counting-and charging-oriented information of resource usage. In particular, processing of SMDRs, (Stations Message


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Detailed Record) bill notification, and charge-back procedures are included for voice and data. Network planning is the process of determining the optimal network, based on data for network performance, traffic flow, resource utilization, networking requirements, technological trade-offs and estimated growth of present and future applications. Sizing rules and interfaces to modeling devices are also parts of the planning process. 6. Network Management Strategic Directions and Benefits The principal directions of network management are integration, centralization, automation and repository support. Integration has to be accomplished across multiple communication forms, multiple vendors, multiple network architecture's, private, public, and virtual networks, LANs, MANs, and WANs across multiple processors, applications databases, and network-management products. Centralization offers the opportunity of central control supported by shared or dedicated processors in combination with distributed implementation of certain network-management functions, such as flittering, problem detection, data compression, and change management. Automation aims for simplification of the operator’s tasks by improving productivity, error minimization, problem prediction and prevention and speeding up recovery using various tools techniques, and facilities. Artificial intelligence is expected to play an important role in future automation. Repositories are expected to be everything to everybody. Besides configuration databases, performance data, vendor data, and trouble tickets may also be integrated. In addition to inventory data, connectivity information and dynamic indicators are expected to be included. If network-management function, procedures, and instruments are properly implemented, and human resources are properly assigned responsibilities, the Information Service Organization will benefit in several ways: Visibility of the networking topology • Correlated alarm management • Facts for service and utilization indicators • Visibility of cost elements • Sizing facilities and equipment on the basis of quantified demand By doing so, unnecessary costs, excess capacity, inefficient network management applications, inappropriate topologies, and unsatisfied users will be avoided.


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7. Network Traffic Management

Network Traffic Management (NTM) is the function of supervising the performance of the network and taking actions to control the flow of traffic, when necessary, to ensure the maximum utilization of network capacity in all situations. This function is performed in 'real time' that is the time between the occurrence of an event and the reporting of it is almost negligible. The aim is to control unusual traffic flow in the network, almost as it occurs. A total network traffic management system consists of four component parts. These are: 1. Monitoring: means of measuring the network data 2. Controlling: means of changing the network configuration 3. Transport : means of transporting, monitoring & controlling information 4. Support: means through which decision making at the man/machine interfaces are aided. General Network Traffic Management is the function of supervising the performance of the network and taking action to control the flow of traffic, when necessary, to insure the maximum utilisation of network capacity in all situations. A telecom/PTT faces mounting congestion problems as the network grows in size and complexity. The factors that were influential in considering adoption of network traffic management are • A growing concern with the consequences of total failure of SPC exchanges. • Evidence that the network was less robust, as efficiency improved. • Studies showing that the congestion effects of plant shortfalls due to various practical problems, and traffic surges due to major events, either man-made or natural, would be partly offset by NTM. • Increasing recognition of the network as a single entity. It is for all the above reasons and much more, that network traffic Management (NTM) was formed as a discipline within the Network Management Operations. Other disciplines within the Network Management Operations are • Common Channel Signaling • Network restoration coordination • Planned Event coordination • Real Time Outage Reporting • Major Outage Management


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• Customer Traffic Management 8. Functions within the NMC The Network Management Center can be broadly categorized into two main areas:

� Network Management Operations � Network Management Support

The function of the operational group is to take action as close to real time as possible to reduce the effect of any problems in the network using the facilities available in each discipline, and to report to management on any major unplanned outage. The primary function of the support group is to investigate and provide all the information and necessary tools for the operators to carry out their function efficiently, while, the service restoration Planning section of the support group develops all the strategies and controls for the operators to use. Operator Functions: The action taken by the NTM operator is based on a NTM Service Restoration Plan which consists of a set of NTM strategies, each relating to a network overload or failure situation and each consisting of a set of NTM controls which are available for use depending on the prevailing conditions. The NTM staff within the Support SRP section prepares these strategies and controls after considerable investigations into exchange data trunking and charging. Once they are prepared, the control exchange data is loaded into exchanges and tested. If tested satisfactory the operator then approves them for use. The specification and testing of NTM support systems and the management of the network database of these systems are functions of the support section. The NTM staff within the Support section also performs the following functions: • Develop and Maintain operational guidelines • Coordinate all adhoc investigations, reports and system software and hardware modifications. • Develop and monitor NTM performance parameters • Provide high-level support to operations staff 9. NTM Principles and Objectives The objective of NTM is to ensure that as many calls as possible are successfully completed in all situations. This objective is reached by using the following principles.


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NTM Principles If one or more of the following NTM principles are not used in a strategy plan then it is not a valid NTM strategy • Keep all circuits fill with successful calls. • Utilise all available circuits. • Give priority to calls requiring a minimum number of links to form a connection when all available circuits are in use. • Inhibit switching congestion and prevent its spread. The SRP section takes all these objectives and principles into account when developing strategies and controls for use by the operator, the operator must also consider the following basic operational principles. Operational Principles • An NTM 'hot spot' must exist before a control action is considered. • Solve problem in local area before involving distant areas. • Use expansive controls before protective controls. 10. NTM measurements and Parameters The following measurements are used by NTM: Bid: A signal attempt to obtain the service of a resource i.e. a call attempt on a circuit is a bid. Seizure: A successful bid, i.e. a call carried in a circuit. Overflow: An unsuccessful bid to obtain a resource (Circuit) on the selected route. Usage (Occupancy): A measurement of the load carried by a server (Circuit or item of equipment) or by a group of servers. Expressed as a percentage in use. (Traffic intensity): A measurement of the load carried by a server (circuit or item or equipment) or by a group of servers expressed in Erlangs. (Time): A measurement of the duration of a successful seizure. Timeouts: A measurement of delay in obtaining a resource. Rejected: A seizure on incoming circuits, which is refused by the central processor. Answer: A signal in a backward direction indicating a call is answered. Terms and Definitions Busy Condition: of a resource which is in use following a seizure. Normally applied to a customer’s service. Congestion: The state when the immediate establishment of a new connection is impossible owing to inaccessibility of any the resources.


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Normally applies to the failure to successfully seizure a circuit from an exchange towards the destination exchange. Destination: A destination with a LOW Answer Bid Ratio Answer Bid Ratio: Number of answered calls per number of bids ABR = Answered Calls / Bids Answer Seizure Ratio: Number of answered calls per numbers seizures ASR = Answered Calls / Seizures Seizures Per Circuit per Hour: Number of seizures per circuit expressed per hour Mean Holding Time: Average holding time of equipment in second MHTS = Total Usage in Seconds Percentage Overflow: Number of bids overflowing expressed as a percentage. %OFL = Bids - Seizures X 100 Bids These measurements and calculated parameters provide the following: Exchange Parameters

� Rejected Calls � Central Processor % Occupancy � Common Control Equipment % Occupancy � Common Control Equipment Timeouts � Exchange Status Signal

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M01- Communication Basics