Module 1 - Part2

• A four wire circuit, in telecommunications is a 2-way circuit using 2 paths

arranged, that the respective signals are transmitted in one direction only by
one path and in the other direction by the other path.

Telecommunications Transmission( 4 wire circuit)

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Basic transmission

• The o/p voltage from a receive amplifier induces equal voltages to be

induced in the secondary of transformer T1

• If Impedance of 2-wire line and line balance is equal , then equal currents
flow in the Transformer T2,but the windings are in antiphase,so no EMF is
induced in secondary of T2.

Therefore no signal at the input of send amplifier

• Note:

• When signal is applied from a two wire line at the transformer windings , it
divides equally b/w send amplifier and o/p of receive amplifier(no use or
effect)(half power is lost or there is 3dB loss)

Zero current flows in the line balance in this case

If both paths of the four wire circuit were connected directly to the two wire circuit
,a signal could circulate round the loop thus created. This would result in
continuous oscillations called the ‘Singing’, unless sum of gains in the circuit is
less than zero.

Echoes in 4 wire circuit

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Echo and signalling paths in a four wire circuit

The signal reflected to the speaker’s end of the circuit is called Talker Echo.

The signal reflected to listener’s end is listener Echo.

The attenuation between a two wire line and four wire line or four wire line and
two wire line is 3dB.

The total attenuation from one two wire circuit to another is

L2 = 6 – G4 dB -----------------(1)

In eqn(1) G4 is the net gain on one side of the four wire circuit .

The attenuation through the hybrid transformer from one side of the four wire
circuit to the other is called transhybrid loss.

This loss is B= 20 log│(N+Z)/(N-Z)│ dB ----------------(2)

In (2) Z- impedance of a two wire line and

N – impedance of the balance network.

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The loss ‘B’ is the part of the transhybrid loss which is due to impedance
mismatch between the two wire line and balance network.

B is also called as Balance –return loss(BRL).

The attenuation Lt of the echo that reaches the talker’s two wire line round the path
is given by

Lt = 3 - G4 + (B+6)- G4 + 3 dB = 2L2 + B dB ---------------(3)

The echo is delayed by a time Dt = 2T4 ---- it is the delay of four wire
circuit.

The attenuation L1 of the echo that reaches the listener’s two wire line (relative to
the signal received directly ) is

L1 = (B+6) - G4 + (B+6) - G4 dB = 2L2 + 2B ---------------(4)

Effect of echoes :

For a speaker , it interrupts his or her conversations

For a listener, it reduces intelligibility of speech

Stability:

If the balance return loss(BRL) of the terminations of a four wire circuit are
sufficiently small and gains of its amplifiers are sufficiently high, the net gain
round the loop may exceed zero and singing will occur.

The net loss Ls of the singing path is given by :

Ls = 2(B+6 – G4 ) dB --------(5)

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Substituting (1) in (5) we get

Ls = 2(B + L2) ---------------(6)

Condition for stability:

Ls> 0

ie. From (6)

L2 +B > 0

ie. G2< B ( where G2 = -L2)--------------this means that gain G2 is limited by

B(BRL)

Definitions :

1)Singing point: It is the maximum gain ‘S’ that can be obtained without
producing singing.

That is S= B --------(7)

2) Stability Margin:

This defined as the maximum amount of additional gain that can be

introduced(equally and simultaneously) in each direction of transmission without
causing singing.

That is Ls – 2M =0 -----(8)

Or

M = B+L2 --------------(9)

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For long distance transmissions a simple is adopted in some countries.

That is

L2 = 4.0 + 0.5n dB --------(10)

Where ‘n’ is the number of four wire circuits in tandem on the switched
connection.

Echo suppressor:

It is a voice operated attenuator

An Echo suppressor consists of a voice activated attenuator, which is fitted on one

of the four paths of a four wire circuit operated by speech signals on the other path.

This allows speech transmission only in one direction , any transmission in

opposite direction is attenuated, thus stopping the echo path.

Note:

1)Echo suppressors are disabled in intermediate links of long distance

communications , where they may lead to lock out conditions since they are
connected in tandem

2) During data transmission also the Echo suppressors are to be disabled , since
return channel in such systems may be used for re- transmission of information,
when errors are detected.

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Echo canceller:

The echo is cancelled by subtracting a replica of it.This replica is generated by

means of a filter by a feedback loop, which adapts to the transmission
characteristic of the echo path and tracks any variation in it that may occur during a
conversation.

Introduction to digital Tranmission:

Transmission systems are required to provide circuits between nodes of

telecommunications N/W.If they use seperate transmission path for each direction
then they are called channels.

The typical arrangement is the terminal station(telephone), transmission link,

alongwith repeaters at intermediate stations.Both channnel and signals can be
classified as Analog and digital.

Analog signal -varies with time - Eg : speech signal

Digital signal -constant with time - Eg : Telegraph signals

Bit rate: number of bits transmitted /sec

Baud rate: number of symbols transmitted/sec, symbols are group of bits.

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Analog signals over digital circuits : Communication signals sent over digital
circuits, where A to D converters are required (Eg. PCM to transmit speech)

Digital signals over Analog circuits: Data communication and voice frequency
telegraphy over telephone lines over analog circuits.

Advantages of digital Txm over Analog Txm:

1. Immunity to noise and interference

2.Regeneration of signal is possible, provided that received signal is not so

corrupted.

Mutiplexing and Demultiplexing :

Multiplexing: at the sending terminal the signals from different channels are
combined to form a composite signal of wider badwidth.

Demultiplexing:At the receiving terminal the signals are seperated and

retransmitted to different channels.

Baseband channel:It is a single channel which enters or leaves a terminal station.

Broadband or Bearer channel: This is a channel that carries the multiplex signals

Methods of Multiplexing:

FDM(Frequency division multiplexing): Each baseband channel uses the bearer

channel for all of the time but only a fraction of bandwidth.

TDM(Time division multiplexing): Each baseband channel uses the bearer channel
entire bandwidth but only a fraction of time.

Power levels :

This is expressed in dB.

Gain = G = 10log10(P2/P1) dB ; P2> P1 then it is called gain.

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where P2 - output power

P1- input power

If P2< P1, then it is loss or attenuation in dB

Relative power level :

dB : this is also relative power level since we are comparing two power levels to
declare the power ratio as gain or loss.

dBm : 10log10(Given power level/1mW) - the reference power level is 1mW.

dBW: 10log10(Given power level/1W)- the reference power level is 1W.

Examples :

power to dBm or dBW

1)Convert 1W to dbm : 10log10(1W/1mW) = 30 dBm

2) Convert 1mW to dbm: 10log10(1mW/1mW)= 0 dBm

3) Convert 1mW to dBW; 10log10(1mW/1W) = -30dBW

dBm or dBW power to power in watts:

1) 30 dBm = 10log10(power /1mW)

power = antilog ( 3) *1mW = 1W

2) -30dBW= 10log10(power /1W)

power = antilog(-3)*1W= 1mW

Digital Transmission

Bandwidth and Equalization :

• Minimum bandwidth needed to transmit a digital signal at ‘B’ bauds was

given by Nyquist as Wmin = (1/2)B( with this condition there is no
ISI(intersymbol interference))

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 • Equalization is needed (gain &phase)to obtain negligible ISI.

• Time domain equalizers can also be used on basis of impulse response.

• Time domain equalizer is also called as Transversal Filter.

• If characteristics of the transmission path change with time , such equalizer

can be made to adjust itself automatically.

• This is called as Adaptive Equalizer

Equalizer is utilized as a compensation circuit to correct for a loss slope

created by other

elements within a circuit [such as in amplifier stages].

• Phase equalizer:

A linear-phase equalizer uses linear-phase filters. This means that when

a signal goes through the filter, all frequencies should experience the same
time delay (known as “pure time delay”), which preserves the wave shape as much
as possible.

Noise and Jitter

Jitter is the deviation from true periodicity of a presumably periodic signal, often in
relation to a reference clock signal.

Note:

• For telephony which has speech signal transmission we employ PCM

PCM helps to obtain satisfactory transmission in presence of severe crosstalk and

noise

Error occurences in digital signal Txn:

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 Regenerative Repeater:

• Samples the received

waveform at intervals
corresponding to the digit
rate.

• Generates the same knowing

the thresholds in case of
Bipolar and unipolar signals

If there is any periodic variation of the times of the regenerated pulses it is called
jitter

Frequency division multiplexing (FDM):

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Note:

Each individual channel occupies a finite frequency range, typically some multiple
of a given base frequency.

FDM:

In telephony 12 telephone channels are to be multiplexed

At the sending end:

• Each incoming signal from audio freq circuit(fm) is applied to a balanced

modulator with a carrier (fc)

• The O/P is the DSBSC signal (fc±fm)

• Signal goes through Band pass filters and transmits lower side band(LSB)
(fc-fm)

• The outputs of these filters are commoned to give a composite signal

containing signal of each telephone channel translated to different frequency
spectrum.

At the receiving end :

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 • The incoming signal is applied to bank of bandpass filters

• Each filter selects the band containing the signal of one channel.

• The signal is applied to a modulator with proper ‘fc’ value and the O/P of
modulator = base band signal + unwanted frequency components.

The O/P above is passed through a low pass filter and the base band signal is given
to A.F. circuit.

The standard telephony voice band [300 – 3400 Hz] is heterodyned and stacked on
high frequency carriers by single sideband amplitude modulation. This is the most
bandwidth efficient scheme possible.

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In the North American system, there are:

• 12 channels per group

• 5 groups per supergroup

• 10 super groups per mastergroup

• 6 master groups per jumbogroup

CCITT standards for North America

Hierarchy of FDM channel assemblies

Basic super group

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Basic Time division multiplex transmission

Pulse transmission in TDM

The above pulses have the same repetitive frequency (fr) but staggered in time

During Transmission:

• Each baseband channel is connected to the transmission path via a

sampling gate

sampling gate: It opens for short intervals by means of train of pulses

• On the transmission path interleaved of pulses are present which are

modulated by signals of different channels

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TDM reception

• At the receiving terminal, gates open with respect to pulses, same as in

transmission path(sync)

• The specific channels are obtained by filter and demodulated during the
alloted time intervals and amplified.

Note:

• PAM is used. The baseband signal are sent at regular intervals by means of
PAM.

• To accommodate telephone channels from 300Hz to 3.4 KHz, the standard

sampling frequency is 8KHz.

• Pulse generators at the Tx and RX must be in Sync.

• Always an additional pulse signal is sent in each repetition period along with
channel pulses(sync signal).

• The above together is called as a Frame

Why use PCM?

• Because PAM due to delay distortion and attenuation can cause dispersion
of pulses.

• Due to the above, they will cause interchannel crosstalk

PCM(Pulse code modulation) system

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PCM system description

• In PCM, each analog signal is applied to A/D converter(analog to

digital)which produces group of pulses representing voltage in binary code.

• At the receiving end, a digital to analog converter(D/A converter)performs

the inverse process.

• A/D and D/A converter operate only during time duration of time slot of one
channel, they are common to all channels of a TDM system

Quantizing distortion in PCM

• This arises because the system can only transmit finite number of sample
values.

• For 8-bit code 2^8=256 different sample values.(small diff b/w o/p and i/p
signal)

• This process of quantizing introduces non linear distortion.

If amplitude of signal is large as compared to quantizing step, then errors in

successive

samples are random.

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PCM encoded signal

Some terms :

• Word or byte: group of bits representing one sample.

• Octet: 8-bit word or byte

• For telephony :

• Sampling is carried out 8KHz and 8-bit encoding

• Quantization noise: It is a rounding error between the analog input voltage to

the ADC and the output digitized value. The noise is non-linear and signal-
dependent.

Types of quantization

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Uniform and non uniform quantization

• If step size in quantization is uniform , that is a large amplitude signal is

represented by large number of steps and is reproduced with little distortion.

• The effect of quantizing noise can be reduced by using smaller steps for
small input voltages and larger steps for large input voltages, this process is
called as instantaneous companding.

This helps to have the quantizing signal to noise ratio nearly constant over a range
of

input levels.

Companding Laws :

Mu-Law:

A-Law

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PCM primary multiplex group
30 channel PCM format

24 channel PCM format

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Plesiochronous Digital Heirarchy(PDH)

Note : primary multiplex group of 24 0r 30 channels is used as building block for

higher order multiplex systems.

• At each level the bit streams are called tributaries combined by a

mutiplexer.

• Although the bit streams have same nominal bit rate, they are derived from
different crystal oscillators and hence can vary with clock tolerance.(called
plesiochronous )

• If inputs to multiplexer are synchronous(they have same bit rate and

phase)they can be interleaved by taking a bit or group of bits from each in
turn.

• This is done by a switch, that samples each input by control of multiplex

clock

Interleaving :

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European PDH

Frame length : 125µs corresponding to basic channel sampling rate of 8KHz

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North American PDH

Need for overhead bits

• For frame alignment

• For performing justification

1)Frame alignment: for the demux to correctly route the digits to outgoing
channels, a frame alignment word (FAW)is used.hence synchronisation is
maintained.

2)Justification :for correct operation of mux &demux

 If tributary is slow , justification digits are added.

 If tributary is fast , no bits are added

Synchronous digital heirarchy(SDH)

• Introduced by CCITT during 1990

• It is a standard technology for synchronous data transmission on optical

media.

• SDH are standardized multiplexing protocols that transfer multiple digital

bit streams over optical fiber using lasers or light-emitting diodes (LEDs).
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 • In the U.S. it is called as SONET(Synchronous Optical Network)

Note: interfaces used are optical

• The digital rate of SDH is 155.52Mb/s and multiples of this by factors 4n

Eg. 622.08 Mb/s and 2488.32Mb/s

• It includes management channels and hence can act as transmission bearer

network- allots transmission capacity flexibly to different services

SDH – standardized bit rates

SDH frame representation

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 • The basic SDH signal is called the synchronous transport module(STM)
(STM-1)

• It has nine equal segments with overhead bytes at the start of each.
Remaining bytes contain a mixture of traffic and overhead bytes.

• The total length is 2430 bytes (270X9)with each overhead using nine bytes.

• Each column contains 9 bytes (one from each row)

• Each byte having 64 k b/s capacity

• Three columns(27 bytes) can hold 1.5Mb/s with 24 channels and some
overheads

• STM-1 frame can also give payloads at European rates of 8,34 and 140 Mb/s

North American rates of 6 and 45 Mb/s will also work for STM-1

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Multiplexing process in SDH

• The bytes from a tributary are assembled into a container and a path
overhead is added to form a virtual container(VC)

• In the North America , VC is called virtual tributary synchronous payload

envelope.

• The VC is travels through the network as a complete package until

demultiplexed

• VC is not fully synchronised with STM-1,hence start point is indicated by a

pointer, the value of these pointers show where exactly the demuxing should
start.

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