Documente Academic
Documente Profesional
Documente Cultură
TRANSMISSION UNITS
Introduction
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.
Transmission Impairments
With analog transmission systems using copper cable there are three major
categories of impairments. They are attenuation, noise, and distortion.
1. 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.
2. Noise: Any random disturbance or unwanted signal on a transmission facility
that obscures the original signal. Noise is generally caused by the
environment in which the system is operating.
3. 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.
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;
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:
1W Net Work 2W
Example 2
Let us look at another network:
Net Work
1000 W 1W
Example 3
Consider a network with a 13 dB gain:
0.1W ?
Network 13 db gain
Example 4
Consider the following network
1W ?W
Network 27 dB loss
Example 5
10W 3W
Network 6 db gain
Example 6
Consider a network of 33 dB gain with an input level of 0.15W. What would be the
output?
30 dB represents multiplying the input power by 1000 and 3 additional dBs double it.
In this case the input power is multiplied by 2000.
The transmission unit normally used is the decibel. The other unit, however, is also
used in some East European countries.
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)
ex = V1/V2
Example
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 common of these. By definition dBm is a power level
related to 1 mw. The most important relationship to remember is:
0 dBm = 1mW.
Example
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.
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.
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.
Other Units
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.
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.
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
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, 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).
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.
Jitter, like BER, is another transmission impairment. It is not very significant in the
case of voice signal transmission but it has a great impact in the transmission of
data signals, especially with high-speed digital transmission. The present bit rates
are as high as 565 Mb/s and (140 x 16) Mb/s. Today Jitter is considered as a
performance parameter of any digital transmission system.
For example, Jitter due to unwanted phase change is called Phase Jitter. The
amount of change of phase, converted into time, is generally expressed in milli-
seconds or nano-seconds.
BER and Jitter are the unwanted by products of any transmission system and they
get associated with the transmission path and affect the quality of transmission. Bit
Errors beyond a limit, affect the communication and Jitter in the digital transmission
system, is a source of generation of errors.
Digital Transmission Analyser (DTA) is used for the measurement of both BER and
Jitter.
Digital Transmission - Performance Criteria ( General)
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.
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
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.
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.
LINE CHARACTERISTICS
Introduction
The propagation of elastic waves along any uniform and symmetrical transmission
line may be deduced in terms of the results for a hypothetical line of infinite length
having electrical constants per unit length identical to those of the line under
consideration. For this reason, the propagation of electric waves along an infinite line
will be considered first.
When an alternating voltage is applied to the sending end of an infinite length of line,
a finite current will flow due to the capacitance and the leakage conductance
between the two wires constituting the line.
The ratio of the voltage applied, to the current flowing, will give the input impedance.
This input impedance is known as the “ characteristic impedance” of the line, and
is denoted by ZO.
The characteristic impedance of any line is defined as the impedance looking into an
infinite length of the line.
Consider an infinite line having input terminals 1 and 2 as in fig 2(a). The impedance
looking in at terminals 1 and 2 will, by definition, be Zo.
Suppose that a short section AB at the near end of the line is now removed [fig 2(b)],
so that the line now starts at terminals 3 and 4. The impedance looking in at
terminals 3 and 4 will still be ZO, since the removal of the short section does not
affect the infinite nature of the line. This means that the short section AB, from the
electrical point of view, was originally terminated in impedance ZO at B. If the short
section AB is now terminated in actual impedance ZO, the current and voltage at all
points along its length will be exactly the same as if it were terminated in an infinite
length of line.
Therefore, it follows that any short line terminated in ZO behaves electrically, at all
points along its length, as if it were an infinite line.
Let the equivalent T section have series arms Z1/2, Z1/2 and shunt arm Z2 as in fig.3.
Hence for a short line, ZO can be determined if Z1 and Z2 can be found. This
will require two equations, which may be obtained by measuring the impedance
using two different terminating impedances. For convenience these termination will
be taken as zero and infinity.
Let the input impedance with an infinite- impedance termination i.e. open- circuit, be
Zoc. As in fig. 4(a)
Let the input impedance with a zero impedance termination i.e., short- circuit, be
Zsc. As in fig 4(b).
Note: The following measurements have been made on a line at 1600 Hz.
Zoc = 900Ω∠-30°
Zsc = 400Ω∠-10°
Zo = √Zoc x Zsc
= √900Ω∠-30° x 400Ω∠-10°
= 600 Ω∠-20°
Consider a current IS applied to the sending end A of an infinite line as in fig 5(a). At
the point B, at a distance of one mile down the line, let the current be I1.
Due to the loss introduced by the line, the current I1 will be less than IS and also a
phase-shift will be introduced. Therefore the ratio IS / I1 will be a vector quantity.
= eα∠β
α is known as the attenuation constant per mile of the line and is measured in
Nepers.
β is known as the phase constant or wavelength constant per mile of the line, and is
measured in radians per mile.
The attenuation of such a line is nα Nepers and the phase shift is nβ radians.
Line constants
The “ primary line constants’ (which, for the purpose of transmission theory, are
assumed to be independent of frequency) are R, G, L and C where
They are measured considering both conductors, i.e. per mile loop. These primary
constants may be obtained by measurements on a sample of the line.
Consider a short length of line, l mile long. This short section will have a
resistance Rl, a leakance Gl, an inductance Ll, and a capacitance Cl. Its
characteristic impedance will be Z0, the same as that of the complete line. Its
propagation constant will be γl, where γ is the propagation constant per mile of the
complete line. This short section of line may be represented as
If the length of the section is very small, Z1 will be approximately equal to the
series impedance of the section. i.e. Rl + jwLl; and Z 2 will be approximately equal to
the shunt impedance of the section, i.e. 1/Gl+jwCl.
Since R/G, in all cases, is greater than L/C, the variation of Z0 with frequency,
expected for a practical line, will be as in fig 6.
γ = √(R+jwL)(G+jwC)
Introduction
A signal is said to have suffered distortion if, after passing through a network,
(passive* or active **)
is not an exact replica of the original signal in respect of its amplitude wave shape.
* A passive network does not require power supply. It may consist of resistors, coils
and condensers either singly or in different combinations e.g. attenuators and filters
are passive networks.
** An active network is one that invariably requires power supply. It may consist of an
electron tube, a transistor or an IC chip in combination with other components like
resistors, coils, and condensers e.g. an amplifier or an oscillator.
Linear distortion takes place in passive networks. Different types of linear distortion
are;
1. Attenuation distortion
2. Phase distortion
Attenuation distortion
As is clear from the figure, higher frequencies are attenuated more than the
lower ones. An ideal band-pass filter should give same loss to all the frequencies of
the pass-band but in actual practice it is not so. The above figure shows a typical
attenuation distortion curve for a voice channel. Curve A shows the attenuation-
frequency response of an ideal band pass filter and curve B shows the attenuation
distortion in the case of a practical band pass filter. As such the network is designed
in such a manner that attenuation distortion caused by it remains within permissible
limits.
However, in the case of open wire carrier lines, equalizers are used to bring
amplitude distortion within limits.
Non-linear distortion
"Non linear distortion" is the general name given to a certain type of distortion
that occurs when the transmission properties of a system are dependent on the
instantaneous magnitude of the applied signal. It is further sub-divided as under:
a) Amplitude distortion.
b) Harmonic distortion.
c) Inter-modulation distortion.
Amplitude distortion
It is defined as the variation of gain or loss of a system with the amplitude of the
input. It is measured with the system operated under steady- state conditions with an
input of sinusoidal waveform.
Harmonic distortion
Inter-modulation distortion
It is due to the production of combination frequencies in the output when two or more
sinusoidal voltages of specified amplitude are applied at the input. For two parent
frequencies p and q, the output may contain frequencies such as (p± q), (1p± q), (p±
2q) etc. in addition to the frequencies p and q.
Introduction
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 figure 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.
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. 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 travelled to the far end.
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 intelligible cross talk, steps should be taken to see that
intelligibility of sentence articulation of the cross talk should be less than 10%.
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 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.
I - Disturbing Circuit.
II – Disturbed Circuit.
(a) Reflected near-end cross talk causes far- end cross talk.
(b) Near-end cross talk caused by reflected wave at far-end causes FEXT.
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
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.
Introduction
Impedance matching
Since it can be designed to have any desired impedance values looking in either
direction, a simple resistance pad can be used to match any two resistive-
impedance. It, however introduces high attenuation loss.
The reading of power level at a point in a circuit can be obtained in two ways.
The high impedance of the meter is essential to ensure that it will not disturb the
circuit under test. (Ex. a 5000 ohms meter will introduce a shunt loss of 0.5 dB). If
the impedance of the circuit under test is 600 ohms, the meter will give correct
reading in dBs.
Any variation in circuit impedance from 600 ohms will destroy the accuracy of the
measurement. In case, however the impedance is known, a correction factor may be
applied.
Reflection loss
The phenomenon of reflection is very common in our every day life. When we look
into a mirror or hear an echo, we know that these are due to reflection of light and
sound wave respectively. From these, it can be deduced that any wave suffers
reflection whenever there is an abrupt change of medium through which the wave is
propagating.
In the case when electromagnetic energy being propagated over a transmission line,
the wave motion is guided, between the two wires constituting the line and is called
a “ guided “ wave or a “ travelling “ wave. The transmission line may further be
connected to equipment or to another transmission line having different electrical
characteristics, thus causing change of medium and hence reflection of energy takes
place. The electrical characteristic that causes reflection is the impedance of
medium. Any variation in impedance will cause reflection. Similarly impedance
mismatch in networks (active or passive) causes reflection.
One direct consequence of reflection is that the amount of power transferred to the
load differs from that when matching conditions exist. All theoretical treatments of
networks and transmission lines are based on the condition of perfect matching,
which can only be approximated in practical applications. Hence the difference
between the matched and non-matched condition is expressed as a “ Reflection loss
“
Return loss
2.3.2 Each channel is sampled at a specified rate and transmitted for a fixed
duration. All channels are sampled one by, the cycle is repeated again
and again. The channels are connected to individual gates which are
opened one by one in a fixed sequence. At the receiving end also
similar gates are opened in unision with the gates at the transmitting
end.
2.3.3 The signal received at the receiving end will be in the form of discrete
samples and these are combined to reproduce the original signal. Thus, at
a given instant of time, onty one channel is transmitted through the medium, and
by sequential sampling a number of channels can be staggered in time as
opposed to transmitting all the channel at the same time as in EDM
systems. This staggering of channels in time sequence for transmission
over a common medium is called Time Division Multiplexing (TDM).
•Filtering
•Sampling
•Quantisation
•Encoding
•Line Coding
4.0 FILTERING
4.1 Filters are used to limit the speech signal to the frequency band 300-
3400 Hz.
5.0 SAMPLING
5.1 It is the most basic requirement for TDM. Suppose we have an
analogue signal Fig. 3 (b), which is applied across a resistor R through a
switch S as shown in Fig. 3 (a) . Whenever switch S is closed, an
output appears across R. The rate at which S is closed is called the
sampling frequency because during the make periods of S, the
samples of the analogue modulating signal appear across R. Fig. 3(d)
is a stream of samples of the input signal which appear across R. The
amplitude of the sample is depend upon the amplitude of the input
signal at the instant of sampling. The duration of these sampled pulses
is equal to the duration for which the switch S is closed. Minimum
number of samples are to be sent for any band limited signal to get a
good approximation of the original analogue signal and the same is
defined by the sampling Theorem.
5.3.3 Let us say our voice signals are band limited to 4 KHz and let sampling
frequency be 8 KHz.
If we have just one channel, then this can be sampled every 125 microseconds
and the resultant samples will represent the original signal. But, if we are to
sample N channels one by one at the rate specified by the sampling theorem,
then the time available for sampling each channel would be equal to Ts/N
microseconds.
5.4 The signals on the common medium (also called the common highway)
of a TDM system will consist of a series of pulses, the amplitudes of
which are proportional to the amplitudes of the individual channels at
their respective sampling instants. This is illustrated in Fig. 5
5.5 The original signal for each channel can be recovered at the receive end by
applying gate pulses at appropriate instants and passing the signals through
low pass filters. (Refer Fig. 6)
As seen in Fig. 9 (b), the first two segment in each polarity are collinear, (i.e. the
slope is the same in the central region) they are considered as one segment. Thus
the total number of segment appear to be 13. However, for purpose of analysis all
the 16 segments will be taken into account.
7.0 ENCODING
P ABC WXYZ
Polarity bit ‘1’ Segment Code Linear encoding
for + ve 'O' for - ve. in the segment
The first bit gives the sign of the voltage to be coded. Next 3 bits gives the segment
number. There are 8 segments for the positive voltages and 8 for negative
voltages. Last 4 bits give the position in the segment. Each segment contains
16 positions.
Referring to Fig. 9(b), voltage Vc will be encoded as 1 1 1 1 0101.
7.4 The reverse process is carried out at the receiving end to retreive the original
analogue signals. The digital combiner combines the encoded samples in the form of
"frames". The digital separator decombines the incoming digital streams into
individual frames. These frames are decoded to give the PAM (Pulse Amplitude
Modulated) samples. The samples corresponding to individual channels are
separated by operating the receive sample gates in the same sequence i.e. in
synchronism with the transmit sample gates.
9.0 SYNCHRONIZATION
9.1 The output of a PCM terminal will be a continuous stream of bits. At
the receiving end, the receiver has to receive the incoming stream of bits and
discriminate between frames and separate channels from these. That is, the
receiver has to recognise the start of each frame correctly. This operation is
called frame alignment or Synchronization and is achieved by inserting a fixed
digital pattern called a "Frame Alignment Word (FAW)" into the transmitted bit
stream at regular intervals. The receiver looks for FAW and once it is detected,
it knows that in next time slot, information for channel one will be there and so on.
9.2 The digits or bits of FAW occupy seven out of eight bits of Ts 0 in
the following pattern.
Bit position of Ts 0 B1 B2 B3 B4 B5 B6 B7 B8
FAW digit value X 0 0 1 1 0 1 1
9.3 The bit position B1 can be either ' 1 ' or '0'. However, when the PCM
system is to be linked to an international network, the B1 position is fixed at '1 ' .
The FAW is transmitted in the Ts O of every alternate frame.
Frame which do not contain the FAW, are used for transmitting supervisory
and alarm signals.
To distinguish the Ts 0 of frame carrying supervisory/alarm signals from those
carrying the FAW, the B2 bit position of the former are fixed at T. The FAW and
alarm signals are transmitted alternatively as shown in Table - 2.
TA B L E - 2
In frames 1, 3, 5, etc, the bits B3, B4, B5 denote various types of alarms. For
example, in B3 position, if Y = 1, it indicate Frame synchronisation alarm. If Y = 1 in
B4, it indicates high error density alarm. When there is no alarm condition, bits
B3 B4 B5 are set 0. An urgent alarm is indicated by transmitting "all ones".
The code word for an urgent alarm would be of the form.
X 111 1111
10.0 SIGNALLING IN PGM SYSTEMS
10.1 In a telephone network,-the signalling information is used for proper routing of a
call between two subscribers, for providing certain status information like dial
tone, busy tone, ring back. NU tone, metering pulses, trunk offering signal etc.
All these functions are grouped under the general terms "signalling" in PCM
systems. The signaling information can be transmitted in the form of DC pulses
(as in step by step exchange) or multifrequency pulses (as in cross bar systems)
etc.
10.2 The signalling pulses retain their amplitude for a much longer period than the
pulses carrying speech information. It means that the signalling information
is a slow varying signal in time compared to the speech signal which is fast
changing in the time domain. Therefore, a signalling channel can be digitized with
less number of bits than a voice channel.
10.3 In a 30 chl PCM system, time slot Ts 16 in each frame is allocated for carrying
signalling information.
10.4 The time slot 16 of each frame carries the signalling data
corresponding to two VF channels only. Therefore, to cater for 30 channels,
we must transmit 15 frames, each having 125 microseconds duration. For
carrying synchronization data for all frames, one additional frame is used.
Thus a group of 16 frames (each of 125 microseconds) is formed to make a
"multiframe". The duration of a multiframe is 2 milliseconds. The multiframe has
for channel 16. Similarly, time slot Ts16 of F2 carries signalling data of chls 2 .and 17.