TELECOMMUNICATION SWITCHING LAB

LAB FILE

Submitted to
Department of Electronics and Communication Engineering
Amity School Of Engineering And Technology
Amity University Uttar Pradesh

Submitted To

Mrs. Garima Mahendru

Submitted by

Radhika Raghav
A2305118020
6ECE 1-X
 Index

S.NO. Aim Date Date

Performed Submitted

1 To study slope overload in delta modulation 22-12-20 9-01-21

2 To study adoptive delta modulation and demodulation 12-01-21 18-01-21

with CVSD modulation

3 To study Pulse Code Modulation using a digital link 19-01-21 26-01-21

4 (a) To study Time Division Multiplexing (a) 2-02-21 (a) 9-02-21

(b) To simulate Pulse Code Modulation and Time (b) 9-02-21 (b) 13-02-21
Division Multiplexing

5 (a) To study data coding and decoding for NRZ format (a) 16-02-21 (a) 23-02-21
(NRZ-L and NRZ-M)

(b) To study data coding and decoding for unipolar

NRZ, unipolar RZ and Alternate Mark Inversion (b) 23-02-21 (b) 9-03-21
format through simulation.

6 To study data coding and decoding for unipolar to 2-03-21 9-03-21

bipolar and vice versa (RZ-AMI, RZ)

7 To study data coding and decoding for Biphase 9-03-21 16-03-21

Manchester and Biphase Mark

8 To study speech circuit using IC and its interfacing to 16-03-21 23-03-21

the line (EPABX)

9 To study DTMF and pulse dialling (EPABX) 16-03-21 26-03-21

10 To study busy and dial tone using EPABX 16-03-21 23-03-21

 EXPERIMENT 1

Aim : To study slope overload in delta modulation

Apparatus: Delta Modulation/Demodulation Trainer Kit (ST2105), Digital Storage

Oscilloscope (DSO), Power supply and Connecting wires

Theory:

Delta modulation is an analog to digital signal conversion technique employed to realize high
signal to noise ratio during transmission where quality of the received signal is not of much
importance. In this modulation, the sampling rate is much higher and the stepsize after
quantization is of a smaller value Δ.

It is the simplest form of differential pulse coded modulation where the difference between
successive samples is encoded into n-bit data streams. This modulation is also called one bit
pulse code modulation. With delta modulation, one bit is transmitted, that merely indicates
whether or not the sample is larger or smaller than the previous sample. It provides a staircase
approximation of over-sampled base-band signal as the difference between the present sample
and previous approximated sample is quantized into two levels ( ±δ). If the previous
approximation is below the present sample value, then the error is quantized by +δ. If the
approximation is above the present sample value then the error is quantized by -δ. If the delta
is positive, the modulator records a 1 else the it records a 0.

Fig 1.1: Staircase Waveforms of Delta modulation

 Fig 1.2: Delta Modulated signal

In delta modulation system, the transmitter circuit is composed of a Summer, Quantizer,

Accumulator and an Encoder interconnected with each other. At the receiver circuit,
demodulation is done using an integrator and a low pass filter.

Delta modulation uses 1-bit per sampling period (TS ) and requires a sampling rate much
greater than the Nyquist rate.

This type of modulation is used in voice transmission systems or in systems where timely data
delivery at the receiver is more important than the data quality. It is preferred for real time
signal processing.

Some of the advantages of Delta Modulation are

Delta modulation requires simple circuitry and hardware which reduces processing time
and the cost.

Since, the delta modulation transmits only one bit for one sample, therefore the
signaling rate and transmission channel bandwidth is quite small for delta
modulation compared to PCM .

As the sampling interval is reduced, the signal correlation will be higher.

There are two types of quantizing errors that affect this Modulation system. They are the slope
overload and granular noise.

Fig 1.3: Noises in Delta Modulation

Slope overload error occurs when the step size is too small relative to local slope characteristics
of the input waveform to track the original waveform and fails to keep up with the rate-of
change of the input signal. To avoid this, step size is increased.

If an input signal changes more than half of the step size within a sampling interval, there will
be slope-overload distortion. So, the desired limiting condition on the input signal x(t) for
avoiding slope-overloading is

𝑑𝑥(𝑡)/𝑑𝑡 |𝑚𝑎𝑥 ≤ ∆/ 𝑇s
If the signal is sinusoidal in nature, x(t) = 𝑎𝑚cos(2𝜋𝑓𝑡) where ‘f’ represents the maximum
frequency of the signal and ‘𝑎𝑚 ‘ its peak amplitude, slope overload can be avoided if

∆/ 𝑇s ≥ 2𝜋𝑎mf,
where 𝑇s = 1/fs and fs is the sampling frequency

In contrast to slope overload error, granular noise occurs when step size is too large. Thus, to
compromise between slope overload distortion and granular noise errors, an optimum step size
should be selected that can minimize the mean-square value of the quantizing error in a linear
Delta modulator.
Observation

Fig 1.4: ST2105 Kit

Fig 1.5: ST2105 Kit connected for Delta Modulation/ Delta Modulation circuit diagram
 Fig 1.6: Input Signal and its parameters

Vpp = 75.2 V
Vmax = 37.6 V
Vmin = -37.6 V
Amplitude = 73.6 V
RMS = 27.2 V
Frequency = 2.002 KHz

Fig 1.7: Clock Signal and its parameters

Vpp = 92 V
Vmax = 45.6 V
Vmin = -45.4 V
Amplitude = 49.6 V
RMS = 24.8 V
Frequency = 32.000 KHz
 Fig 1.8: Integrated Signal and its parameters

Vpp = 38.4 V
Vmax = 19.2 V
Vmin = -19.2 V
Amplitude = 28 V
RMS = 10.4 V
Frequency = 2.000 KHz

Fig 1.9: Demodulated Signal with output

 Fig1.10: Output with integrated signal

Simulation:

Code:
A=input('enter amplitude l');
F = input('enter frequency');
del= input('enter step size');
t = 0:1/F:2;
x = A*sin(2*pi*t);
plot(x)
hold on
y=[0];
xr=0;
for i=1:length(x)-1
if xr(i)<=x(i)
d=1;
xr(i+1)=xr(i)+del;
else
d=0;
xr(i+1)=xr(i)-del;
end
y=[y d];
end
stairs(xr)
hold off
MSE=sum((x-xr).^2)/length(x);
Outputs:

1. Amplitude = 5; Frequency = 100; Step size = 0.1

Fig 1.11: Simulation output showing slope overload noise

2. Amplitude = 2; Frequency = 100; Step size = 0.1

Fig 1.12: Simulation output showing smaller slope overload noise

3. Amplitude = 2; Frequency = 150; Step size = 0.1

Fig 1.13: Simulation output with no slope overload noise

Conclusion: Step size should be small in delta modulation to avoid slope overload.
Mathematically, the following condition should be satisfied:

∆/ 𝑇s ≥ 2𝜋𝑎mf, where 𝑇s = 1/fs and fs is the sampling frequency

We simulated delta modulation for 3 different parameters value, keeping step size constant =
0.1. For amplitude= 5 and frequency = 100 Hz, slope overload was drastic. Adjusting
amplitude and taking it as 2 reduced but not eliminated this error. For amplitude= 2 and
frequency = 150 Hz, slope overload was removed.

Result: Slope overload in delta modulation was studied and a sinusoidal signal was delta
modulated and demodulated to achieve the original signal. Delta Modulation was simulated in
MATLAB.

CRITERIA TOTAL MARKS MARKS OBTAINED REMARKS

Concept(A) 2
Implementation(B) 2
Performance(C) 2
Total marks 6
 EXPERIMENT 2

Aim : To study adoptive delta modulation and demodulation with CVSD modulation

Apparatus: Delta Modulation/Demodulation Trainer Kit (ST2105), 20 MHz dual channel

Digital Storage Oscilloscope (DSO), Power supply and Connecting wires

Theory

A larger step-size is needed in the steep slope of modulating signal and a smaller step size is
needed where the message has a small slope. The minute details get missed in the process. The
concept of Adaptive Delta Modulation is to control the adjustment of step-size according to
requirement in order to obtain the sampling in a desired fashion

Adaptive Delta Modulation is a modified form of delta modulation. This method was
introduced to solve the granular noise and slope overload error caused during Delta modulation.
It is similar to Delta modulation except that the step size is varies according to the input signal.

Advantages of Adoptive Delta modulation method are:

 Adaptive delta modulation decreases slope error present in delta modulation.

 The slope overload error and granular error present in delta modulation are solved
using this modulation. Because of this, the signal to noise ratio of this modulation is
better than delta modulation.
 This modulation utilizes bandwidth more effectively than delta modulation.
 The signal to noise ratio of ADM is better than that of DM because of the reduction
in slope overload distortion and idle noise.

Disadvantages of Adaptive Delta Modulation are:

 For a relatively constant magnitude of input signal x(t), the system will produce high
granular noise.
 Lower bit rate than PCM

First the difference between the present sample value and previous approximation is calculated.
This error is quantized; if the present sample is smaller than the previous approximation,
quantized value is high, otherwise it is low. The output of the one-bit quantizer is given to the
Logic step size control circuit where the step size is decided. If the quantizer output is high,
then the step size is doubled for the next sample. If the quantizer output is low, the step size is
reduced by one step for the next sample.
 Fig 2.1 : Waveform for adaptive delta modulation

The transmitter circuit consists of a summer, quantizer, Delay circuit, and a logic circuit for
step size control. The baseband signal X(nTs) is given as input to the circuit. The feedback
circuit present in the transmitter is an Integrator. The integrator generates the staircase
approximation of the previous sample. ADM quantizes the difference between the value of the
current sample and the predicted value of the next sample. It uses a variable step height to
predict the next values, for the faithful reproduction of the fast varying values. At the summer
circuit, the difference between the present sample and staircase approximation of previous
sample e(nTs) is calculated. This error signal is passed to the quantizer, where a quantized
value is generated. The step size control block controls the step size of the next approximation
based on either the quantized value is high or low. If the quantizer output is high, then the step
size is doubled for the next sample. If the quantizer output is low, the step size is reduced by
one step for the next sample. The quantized signal is given as output.

Fig 2.2: ADM Transmitter

At the receiver end Demodulation takes place. The receiver has two parts. First part is the step
size control. Here the received signal is passed through a logic step size control block, where
the step size is produced from each incoming bit. Step size is decided based on present and
previous input. In the second part of the receiver, the accumulator circuit recreates the staircase
signal. This waveform is then applied to a low pass filter which smoothens the waveform and
recreates the original signal.

Fig 2.3: ADM Receiver

Observations

Fig 2.4 : ST2105 Kit connected for Adoptive Delta Modulation/ Adoptive Delta Modulation circuit diagram
 Fig 2.5: Input Signal and its parameters

Vpp = 76 V
Vmax = 36.8 V
Vmin = -39.2 V
Amplitude = 64 V
RMS = 23.2 V
Frequency = 1.003 KHz

Fig 2.6: Clock Signal and its parameters

Vpp = 49.6 V
Vmax = 24.8 V
Vmin = -24.8 V
Amplitude = 48.8 V
RMS = 23.2 V
Frequency = 32 KHz
 Fig 2.7: Integrated Signal and its parameters

Vpp = 138 V
Vmax = 77.6 V
Vmin = -60 V
Amplitude = 78.4 V
RMS = 35.2 V
Frequency = 1.053 KHz

Fig 2.8: Adaptive delta modulation output and integrated signal with dual channel
 Fig 2.9: Input and demodulation signal with dual channel and their parameters

Vpp = 108 V
Vmax = 54 V
Vmin = -54 V
Amplitude = 102 V
RMS = 36 V
Frequency = 986.4 Hz

Result: Adoptive delta modulation was studied and performed. A sinusoidal signal was modulated and
demodulated to achieve the original signal.

Adoptive delta modulation helps in overcoming disadvantages of delta modulation. ADM removes
granular noise as slope overload by using variable step size and adjusting it at every step according to
the input signal.

CRITERIA TOTAL MARKS MARKS OBTAINED REMARKS

Concept(A) 2
Implementation(B) 2
Performance(C) 2
Total marks 6
 EXPERIMENT 3

Aim : To study Pulse Code Modulation using a digital link

Apparatus: TDM and PCM Transmitter and Receiver Trainer Kit, 20 MHz dual channel Digital
Storage Oscilloscope (DSO), Power supply, Connecting wires and Probes

Theory

Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals so
that it can be transmitted through the digital communication network. In a PCM stream,
the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample
is quantized to the nearest value within a range of digital steps. The output is a binary sequence.

Advantages of Pulse Code Modulation

 Multiple use of one transmission-channel by multiplexing

 Low susceptibility to crosstalk
 Higher transmitter efficiency and noise immunity
 Uniform transmission quality.
 Efficient snr and bandwidth trade off.
 Encoders provides secure data transmission and offer efficient regeneration.

The Pulse Code Modulation process is done in three steps - Sampling, Quantization, and Enoding.

The low pass filter prior to sampling prevents aliasing of the message signal by eliminating the high
frequency components present in the input analog signal.

Fig 3.1 PCM transmitter

 Sampling

 Sampling is defined as, “The process of measuring the instantaneous values of continuous-time
signal in a discrete form.” It helps to collect the sample data at instantaneous values of message
signal, so as to discretize the analog input for digitization and to be able toreconstruct the original
signal at receiver side.
 The sampling done here is the sample-and-hold process. It generates flat- top Pulse Amplitude
Modulated (PAM) signal.
 Sampling frequency is the number of average samples taken per second. According to the Nyquist
Theorem, sampling rate should be at least 2 times the upper cutoff frequency of the input message.

Fig 3.2 Sampling an analog signal

Quantization

 Quantization is a process of representing the sampled values of the amplitude by a finite set.
 Quantization is done by dividing the range of possible values of the analog samples into different
levels of equal or variable step size.
 The analog sample values are then approximated to the nearest quantization values thus assigning
it a particular discrete level.
 The message signal is now represented by a sequence of coded pulses.
 Quantizing reduces the redundant bits and compresses the data.

Fig 3.3 Quantizing samples of an input signal

Encoding

 Encoding process translates a discrete set of sample values to a more appropriate form called
codeword.
 Encoder designates each sampled quantized level by generation of a unique and identifiable
binary code.
 The binary symbol 1 is represented by a pulse of constant amplitude for a one bit duration and
symbol 0 is represented by switching off the pulse for one bit duration.
 Encoding minimizes the bandwidth used and makes the transmitted signal more robust to noise
interference.

Fig 3.4: Pulse Code Modulation steps

Fig 3.5: Stages of a pulse code modulated signal

Observation

Fig 3.6: TDM Pulse Code Modulation Receiver Trainer Kit

Fig 3.7: Circuit Diagram/ Trainer Kit Connections For Pulse Code Modulation
 Fig 3.8: Input Signal and its parameters

Vpp = 4.12 V
Vmax = 2.08 V
Vmin = -2.04 V
Amplitude = 4.04 V
RMS = 1.44 V
Frequency = 1.00 KHz

Fig 3.9: Input signal and sampling with dual channel DSO

Fig 3.10: PCM output

 Fig 3.11: Demodulated Signal and its parameters

Vpp = 4.12 V
Vmax = 2.08 V
Vmin = -2.04 V
Amplitude = 4.04 V
RMS = 1.44 V
Frequency = 1.00 KHz

Result: Pulse Coded Modulation (PCM) was studied and performed successfully. Waveforms of all the
steps of pulse code modulation were observed using DSO.
The input waveform went through sampling, quantising and encoding to produce a pulse code modulated
signal. This waveform was demodulated to obtain the original signal.
 Sampling produced discrete time pulses of variable amplitude
 Quantizing approximated the variable amplitude to discrete levels, producing binary pulses of
variable width. Different widths of these pulses signify different code words assigned to the
individual pulse.
 Encoding will assign a unique binary symbol to each width.
 The output binary sequence will then be divided into frames and identification bits will be added to
separate each frame. This frame will be sent into the channel by pulse code modulation transmitter.

CRITERIA TOTAL MARKS MARKS OBTAINED COMMENTS

CONCEPT (A) 2
IMPLEMENTATION (B) 2
PERFORMANCE (C) 2
TOTAL 6
 EXPERIMENT 4

Aim : To study Time Division Multiplexing

Apparatus: TDM and PCM Transmitter and Receiver Trainer Kit, 20 MHz dual channel Digital Storage
Oscilloscope (DSO), Power supply, Connecting wires and Probes

Theory

The transmission medium can handle have one signal at a time. If there are multiple signals to share over a
medium, then the medium must be divided in such a way that each signal is given some portion of the
available resources. Multiplexing is the process of combining multiple signals into one composite signal so
that these can be transmitted over a common channel, in such a manner that each individual signal can be
retrieved at the destination. The advantage of multiplexing is that we can transmit a large number of signals
to a single medium and reduce the cost of transmission.

Multiplexing may be conducted through the interleaving of time segments from different signals onto a
single transmission path. Time-division multiplexing (TDM) is a method of transmitting and receiving
independent signals over a common signal path so that each signal appears on the line only a fraction of time
in an alternating pattern. TDM divides the overall time domain into several recurrent time slots of fixed
length and allocates these to each source in an alternating manner which will be multiplexed onto the line
by the transmitter. The input signals can occupy the total bandwidth of the channel while being transmitted
and each signal is allotted a definite amount of time. Time-division multiplexing is possible only when the
available data rate of the channel exceeds the data rate of individual signals to be transmitted.

Fig 4.1: Time Division Multiplexing System

Disadvantages of time division multiplexing

 Synchronization is required in time division multiplexing. If the multiplexer and the demultiplexer are not
properly synchronized, a bit belonging to one channel may be received by another channel.
 Due to slow narrowband fading, all the TDM channels may get wiped out.

 TDM systems suffers from some latency. As only one channel can transmit at a given time, some data would
be delayed.
 Advantages of time division multiplexing
 Time division multiplexing systems are flexible. The scheme allows for variation in the number of signals
being sent along the line, and constantly adjusts the time intervals to make optimum use of the
available bandwidth.
 Time division multiplexing circuitry is not complex.
 Intermodulation distortion is not present in TDM.
 Entire channel bandwidth can be utilized for each channel.
 TDM provides higher efficiency and throughput than FDM.
 TDM system does not need any carrier wave or carrier signal.
 Pulse overlapping can sometimes cause crosstalk but it can be reduced by utilizing guard time.

Working of time division multiplexed system

Fig 4.2 : TDM Block Diagram

 The transmission system consists of multiple anti-aliasing Low Pass Filters, one for each data inputs. The
output of the LPF is then fed to the commutator.
 Multiplexer accepts the input from each individual end user, breaks each signal into segments, and assigns
the segments to the composite signal in a rotating, repeating sequence.
 As per the rotation of the commutator the samples of the data inputs are collected.
 The data flow of each input stream is divided into units. During transmission, one unit of each of the input
streams is allotted one-time slot, periodically, in a sequence, on a rotational basis.

Fig 4.3 : TDM Working

 Suppose we have n data inputs, then one after the other, according to the rotation, these data inputs get
multiplexed. A data block of source 1 is transmitted during time slot 1, source 2 during time slot 2, and so
on till first data blocks of all sources are sent. This makes the first frame One TDM frame consists of one
time slot per source. After the last source, the cycle starts all over again with a new frame, starting with the
second or data block from source 1.
 Synchronization bits are added to the beginning of each frame to allow the demultiplexer to synchronize
with the incoming stream so that it can separate the time slots accurately.
 At the receiver end, the individual signals are separated out by means of a de-commutator, reassembled and
routed to the proper end users.
 The commutator and de-commutator must have same rotational speed so as to have accurate demultiplexing
of the signal at the receiving end. According to the rotation performed by the de-commutator, the samples
are collected by the LPF and the original data input is recovered at the receiver.

Types of Time Division Multiplexing

1. Synchronous TDM is a technique in which time slot is preassigned to every device and each device is
given some time to transmit, irrespective of the fact that the device contains the data or not. If there are n
sources, then a single frame consists of n time slots, and each time slot is dedicated to exactly one source for
carrying data from the corresponding input. Each source places its data to the link only when the
corresponding slot arrives. If the source does not have any data, then the slot will remain empty. In such
manner, the channel capacity cannot be fully utilized when some source do not want to send the data.

Fig 4.4: Synchronous TDM

2. Asynchronous TDM is a technique in which time slots are flexible. It dynamically allocates the time slots
on the demand only to those input channels which have the data to send, thus saving the channel capacity.
In case the source doesn’t have data to send when its turn, that source is skipped and data from next source
in line is transmitted. Slot needs to carry address of the destination along with data to differentiate at receiver
side as there is no separate slots assigned for each source.

Fig 4.5: Asynchronous TDM

Observation

Fig 4.6.: TDM Pulse Code Modulation Receiver Trainer Kit

Fig 4.7: Circuit Diagram/ Trainer Kit Connections For Time Division Multiplexing
Signal Vpp Vmax Vmin Amplitude RMS Frequency
Analog Input 1 7.84 V 3.92 V - 3.92 V 7.68 V 2.72 V 2.49 Hz
Analog Input 2 8.16 V 4.08 V - 4.08 V 8.08 V 2.88 V 500 Hz
Analog Input 3 7.20 V 3.60 V - 3.60 V 7.04 V 2.48 V 1 KHz

Fig 4.8: Input Signals and their parameters

 Fig 4.9: Time Division Multiplexed Signal

Fig 4.10: Demultiplexed Signals

Result: Time Division Multiplexing (TDM) was studied and performed successfully. Waveforms of all
the steps of Time Division Multiplexing were observed using DSO.
 The input waveforms were multiplexed and transmitted during different time slots, using a single
channel. A part of each input was transmitted for a fraction of a second and since the time slots
were so small, the data appears to be transmitting parallelly, instead off simultaneously.
 The waveforms used the same frequency while transmitting, determined by the highest frequency
component of all the signals, according to Nyquist theorem.
 This multiplexed waveform was demultiplexed to obtain the original signals.

CRITERIA TOTAL MARKS MARKS OBTAINED COMMENTS

CONCEPT (A) 2
IMPLEMENTATION (B) 2
PERFORMANCE (C) 2
TOTAL 6
 EXPERIMENT 4- SIMULATION

Aim : To simulate Pulse Code Modulation and Time Division Multiplexing

Software Used: MATLAB (2015a)

Theory

Pulse Code Modulation

Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals so that it can
be transmitted through the digital communication network. In a PCM stream, the amplitude of the analog
signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a
range of digital steps. The output is encoded binary sequence.

Code
clc; close all; clear all;
t = 0:0.0001:20; %sampling at nyquist rate
c=input('Enter Bit Depth Of PCM Coding:');
part = -1:0.1:1;%A quantization partition defines several contiguous, nonoverlapping ranges of values within the
set of real numbers.
codebook = -1:0.1:1.1;%A codebook tells the quantizer which common value to assign to inputs that fall into
each range of the partition.
msg = cos(t);
[~,quants] = quantiz(msg,part,codebook);%returns a vector that tells which interval each input is in
subplot(3,1,1);
plot(t,msg);
title('Message Signal');
subplot(3,1,2);
plot(t,quants);
title('Quantized Signal');
y = uencode(quants,c);
ybin=dec2bin(y,c); %converting it to final binary form to make it transmit ready
subplot(3,1,3);
plot(t,y);
title('PCM PLOT');
Simulation Output
Time Division Multiplexing

Time-division multiplexing (TDM) is a method of transmitting and receiving independent signals over a
common signal path so that each signal appears on the line only a fraction of time in an alternating pattern.
TDM divides the overall time domain into several recurrent time slots of fixed length and allocates these to
each source in an alternating manner which will be multiplexed onto the line by the transmitter. Time-division
multiplexing is possible only when the available data rate of the channel exceeds the data rate of individual
signals to be transmitted.

Code
clc; clear all; close all;
x=0:.5:4*pi;
sig1=8*sin(x);
l1=length(sig1);
sig2=8*triang(l1);
subplot(2,2,1); plot(sig1);
title('Sinosoidal Signal'); ylabel('Amplitude'); xlabel('Time');
subplot(2,2,2); plot(sig2)
title('trangular Signal'); ylabel('Amplitude'); xlabel('Time');
subplot(2,2,3); stem(sig1);
title('Sampled Sinosoidal Signal'); ylabel('Amplitude'); xlabel('Time');
subplot(2,2,4); stem(sig2)
title('Sampled trangular Signal'); ylabel('Amplitude'); xlabel('Time');
l2=length(sig2);
for i=1:l2
sig(1,i)=sig1(i);
sig(2,i)=sig2(i);
end
tdmsig=reshape(sig,1,2*l1);
figure; stem(tdmsig);
title('Tdm Signal'); ylabel('Amplitude'); xlabel('Time');
demux=reshape(tdmsig,2,l2);
for i=1:l1
sig3(i)=demux(1,i);
sig4(i)=demux(2,i);
end
figure;
subplot(2,1,1); plot(sig3);
title('Recovered Sinosoidal Signal'); ylabel('Amplitude'); xlabel('Time');
subplot(2,1,2); plot(sig4);
title('Recovered Triangular Signal'); ylabel('Amplitude'); xlabel('Time');
Simulation Output
Result : To simulate Pulse Code Modulation and Time Division Multiplexing were simulated using MATLAB.
Waveforms of all the steps were observed.
1. PCM: The input waveform went through sampling, quantising and encoding to produce a pulse code
modulated signal. We observed that quantised signal was able to trace the original signal better for higher step
size. However, this will require greater channel bandwidth.
2. TDM: Sinusoidal and triangular waveforms were multiplexed. A part of each input was transmitted for
a fraction of a second but since the time slots were so small, the data appears to be transmitting parallelly, instead
off simultaneously. This multiplexed waveform was demultiplexed to obtain the original signals.

CRITERIA TOTAL MARKS MARKS OBTAINED COMMENTS

CONCEPT (A) 2
IMPLEMENTATION (B) 2
PERFORMANCE (C) 2
TOTAL 6
 EXPERIMENT 5

Aim: To study data coding and decoding for NRZ format (NRZ-L and NRZ-M)

Apparatus: Data Formatting and carrier modulation transmitter kit (ST2106), 8 bit variable data generator
(ST2111), DSO, power supply and connecting wires

Theory:

Line coding refers to the process of converting digital data into digital signals. At the sender side digital data are
encoded into a digital signal and at the receiver side the digital data are recreated by decoding the digital signal.
This process of coding is chosen so as to avoid overlap and distortion of signal such as inter-symbol interference.
If chosen correctly, line codes can detect error in transmission without any additional algorithm.

In telecommunication, a non-return-to-zero (NRZ) line code is a binary code in which ones are represented by
one significant condition, while zeros are represented by some other significant condition with no other neutral or
rest condition. A high in data is represented by a positive pulse and low in data either has no pulse (in unipolar
codes) or a negative pulse (for polar coding). All pulses have duration equal to the symbol bit duration and signal
does not return to zero at the middle of the bit. Voltage level remains constant during entire bit interval. The end
or start of a bit will not be indicated and it will maintain the same voltage state, if the value of the previous bit and
the value of the present bit are same.

The advantages of NRZ are −

It is simple line coding technique. NRZ is convenient because computer circuits use unipolar NRZ
internally, and it requires little effort to expand this system outside the computer.

For unipolar NRZ signaling less bandwidth is needed.

In polar NRZ and bipolar NRZ, no low frequency components are present in the signaling waveforms
after mapping.

For a given data signaling rate, NRZ code requires only half the baseband bandwidth required by
the Manchester code.

The disadvantages of NRZ are −

No error correction is done and the signal droop is caused at places where signal is non-zero at 0 Hz.

The pulses in NRZ have more energy than a return-to-zero code.

If the encoded data contains long 'runs' of logic 1's or 0's, this does not result in any bit transitions. The
lack of transitions prevents the receiver from detecting the boundaries of the received bits which lead
to loss of synchronization between clocks of transmitter and receiver. Hence separate clock line is
required between transmitter and receiver.

In decoding a digital signal, the receiver calculates a running average of the received signal power.
This average is called the baseline. The incoming signal power is evaluated against this baseline to
determine the value of the data element. A long string of 0s or 1s can cause a drift in the baseline
(baseline wandering) and make it difficult for the receiver to decode correctly.
NRZ-L stands for ‘Not Return to Zero - Level’. It is a simple scale and level shift of the input TTL waveform.
There is a change in the polarity of the signal, only when the incoming signal changes from 1 to 0 or from 0
to 1. 1 forces a high level and 0 forces a low level. NRZ L appears as raw binary bits without any coding
except the first bit of the input signal has a change of polarity. This is the standard positive logic signal format
used in digital circuits.

NRZ-M stands for ‘Not Return to Zero -inversion of Mark. It is a type of NRZ- Invert where Mark forces a
transition. Mark represents a high in data or binary 1. If a 1 occurs at the incoming signal, then there occurs a
transition at the beginning of the bit interval and physical level changes. For a 0 at the incoming signal, there
is no transition and physical level remains constant.

The Baseline wandering and self-synchronization are problems for both of them, but for NRZ-L they are more
severe because of transition at the boundary for NRZ-I.

Fig 5.1: NRZ-L and NRZ-M waveforms for data input 101100011010

Observations

Fig 5.2: 8 bit variable data generator (ST2111)

Fig 5.3: Data Formatting and carrier modulation transmitter kit (ST2106) connected for NRZ

Fig 5.4: NRZ L data format connected with dual channel and NRZ L data format output
 Fig 5.5: NRZ M data format connected with dual channel and NRZ M data format output

Result: NRZ coding was studies and NRZ-L and NRZ-M waveforms were plotted.

We transmitted data input 10100110 and observed that NRZ-L went through level transition when input bit
changed from 1 to 0 or from 0 to 1. For two similar consecutive bits, physical levels remained constants.
NRZ-M levels toggled only when the next bit was 1 and the level remained constant for input bit 0.
 Experiment -5 - Simulation

Aim: To study data coding and decoding for unipolar NRZ, unipolar RZ and Alternate Mark
Inversion format through simulation.
Software Used: MATLAB (R2015a)

Unipolar NRZ
This line coding technique follows pattern of raw data as level remains constant during entire
bit interval. A high in data is represented by a positive pulse and low in data has 0 magnitude
pulse. All pulses have duration equal to the symbol bit duration and signal does not return to
zero at the middle of the bit.

Code:
bits = [1 0 1 0 0 0 1 1 0];
bitrate = 1; % bits per second
figure;
T1 = length(bits)/bitrate; % full time of bit sequence
n1 = 200;
N1 = n1*length(bits);
dt1 = T1/N1;
t1 = 0:dt1:T1;
x1 = zeros(1,length(t1)); % output signal
for i = 0:length(bits)-1
if bits(i+1) == 1
x1(i*n1+1:(i+1)*n1) = 1;
else
x1(i*n1+1:(i+1)*n1) = 0;
end
end
plot(t1,x1,'LineWidth',3);
axis([0 t1(end) -0.1 1.1])
grid on;
title(['Unipolar NRZ: [' num2str(bits) ']']);
AMI
Alternate Mark Inversion (AMI) has three voltage levels namely +, - and 0. For a 1, the
voltage level gets a transition from + to – or from – to +, having alternate 1s to be of equal
polarity. A 0 will have a zero voltage level. All pulses have duration equal to the symbol bit
duration and signal does not return to zero at the middle of the bit.

Code
h=[1 0 1 0 0 0 1 1 0]; n=1; l=length(h); h(l+1)=1;
ami=-1;
while n<=length(h)-1; t=n-1:0.001:n;
if h(n) == 0
if h(n+1)==0
y=(t>n);
else
if ami==1
y=-(t==n);
else
y=(t==n);
end
end
d=plot(t,y); grid on; title('Line code AMI NRZ'); set(d,'LineWidth',2.5); hold on;
axis([0 length(h)-1 -1.5 1.5]); disp('zero');
else
ami=ami*-1;
if h(n+1)==0
if ami==1
y=(t<n);
else
y=-(t<n);
end
else
if ami==1
y=(t<n)-(t==n);
else
y=-(t<n)+(t==n);
end
end
%y=(t>n-1)+(t==n-1);
d=plot(t,y); grid on; title('Line code AMI NRZ'); set(d,'LineWidth',2.5); hold on;
axis([0 length(h)-1 -1.5 1.5]); disp('one');
end
n=n+1;
end
Unipolar RZ
A high in data is represented by a positive pulse and low in data has 0 magnitude pulse. All
pulses have duration equal to half symbol bit duration and signal does returns to zero at the
middle of the bit. For logical 1, level remains high for half bit duration and returns to zero
during the remaining half of the bit duration. For logic 0, it remains 0 constantly.

Code:
bits = [1 0 1 0 0 0 1 1 0]; bitrate = 1; % bits per second
figure;
T2 = length(bits)/bitrate; % full time of bit sequence
n2 = 200; N2 = n2*length(bits); dt2 = T2/N2; t2 = 0:dt2:T2;
x2 = zeros(1,length(t2)); % output signal
for i = 0:length(bits)-1
if bits(i+1) == 1
x2(i*n2+1:(i+0.5)*n2) = 1;
x2((i+0.5)*n2+1:(i+1)*n2) = 0;
else
x2(i*n2+1:(i+1)*n2) = 0;
end
end
plot(t2,x2,'LineWidth',3); axis([0 t2(end) -0.1 1.1]); grid on; title(['Unipolar RZ: [' num2str(bits) ']']);

Result: Line coding for input 101000110 has been simulated using MATLAB in three formats-
Unipolar Non Return to Zero, Unipolar Return to Zero and Alternate Mark Inversion.

CRITERIA TOTAL MARKS OBTAINED REMARKS

MARKS
Concept(A) 2
Implementation(B) 2
Performance(C) 2
Total marks 6
 Experiment-6

Aim: To study data coding and decoding for unipolar to bipolar and vice versa (RZ-AMI,
RZ)

Tool/equipment used: Data formatting trainer kit, 20 MHz dual channel DSO, connecting
wires, patch cord.

Theory

Line coding refers to the process of converting digital data into digital signals. At the sender side
digital data are encoded into a digital signal and at the receiver side the digital data are recreated
by decoding the digital signal. This process of coding is chosen so as to avoid overlap and distortion
of signal such as inter-symbol interference. If chosen correctly, line codes can detect error in
transmission without any additional algorithm.

The symbols '0' and `l' in digital systems can be represented in various formats with different
levels & waveforms. The selection of particular format for communication depends on the
system bandwidth, system's ability to pass DC level information, error checking facility, ease
of clock regeneration & synchronizations at receiver, system complexity & cost etc.

In telecommunication, a unipolar return-to-zero (RZ) line code is a binary code which is

analogous to on-off keying in modulation. All the signal levels are on one side of the time axis,
either above or below.
Zeros are mapped by absence of pulse for entire bit period while ones are mapped using positive
pulse of amplitude 'V' in the first half bit period and returns to zero in the next half bit duration.
All high pulses have duration equal to half the symbol bit duration and pulse transitions to 0
exactly at the middle of the bit period. The level remains there until the beginning of the next
logic 1.

Fig 6.1: Unipolar Return to Zero

The advantages of Unipolar RZ are −
It is simple.
The spectral line present at the symbol rate can be used as a clock.
DC component is half in RZ as compared to NRZ
The disadvantages of Unipolar RZ are −
No error correction.
The signal droop is caused at the places where signal is non-zero at 0 Hz.
It requires two signal changes to encode a bit and therefore occupies greater bandwidth.
It has a significant DC component
Loss of synchronization occurs due to long string of zeros in the binary data. Such data
does not produce any transitions which may create problems in error detection and
recovery.

In telecommunication, bipolar encoding is a type of line code, where three values of levels are
used; positive, negative and zero. Such a signal is called a duobinary signal. Alternate Mark
Inversion (AMI) is a synchronous clock encoding technique that uses dual polarity pulses to
represent logical 1. A neutral zero voltage represents binary 0. Binary 1s are represented by
alternating positive and negative voltages. The next logic 1 is represented by a pulse of polarity
opposite to the last logic 1. For RZ – AMI, the pulse duration is half the symbol bit duration
and the signal level goes back to zero at middle of bit interval.

Fig 6.2: AMI coding

The advantages of AMI are −

If a long string of ones occurs, there will be no loss of synchronization.

Provides error detecting; every bit error results in bipolar violation.
High noise immunity and low crosstalk
It requires low bandwidth.
Signal droop issue is not observed
It prevents a significant build-up of DC. The cable may then be used for longer
distances and to carry power for intermediate equipment.

The disadvantages of AMI are −

No clock is present.
Long sequences of zero bits result in no transitions and a loss of synchronization.
Observations:

Fig 6.3: Data Formatting Carrier Modulation Transmitter Trainer Kit

Fig 6.4: input data with clock

 Fig 6.5: Input data and output data in RZ format with dual channel

Fig 6.6: Input data and AMI output data with dual channel

Conclusion: Two data inputs were taken- 10011010 and 10101010

We observed that, in RZ coding, 1 was represented by positive level and the level
went back to zero level at middle of bit period. 0 was represented by zero level.

For AMI, 0 was mapped by zero level and level of 1 depended on level of previous
1. If first 1 is mapped by positive level, second 1 was shown by a negative level and
third by positive level again. Level returns to zero in middle of bit.
RZ coding solved problems of NRZ. It can be thought of as on – off keying.
It is simple.
The spectral line present at the symbol rate can be used as a clock.
DC component is half in RZ as compared to NRZ

AMI coding was used extensively in first-generation PCM networks, and is still
commonly seen on older multiplexing equipment today, but successful transmission
relies on no long runs of zeroes being present.

The alternating coding prevents the build-up of DC voltage level down the
cable so it is suitable for AC coupled lines
It requires lower bandwidth
It provides error detection
Signal drooping doesn’t occur

Results: Data coding and decoding for unipolar return to zero and return to zero-
alternate mark inversion was successfully performed and the outputs were
obtained using DSO.

Criteria Total marks Marks obtained Remarks

Concept(a) 2
Implementation(b) 2
Performance(c) 2
Total marks 6
 Experiment-7

Aim: To study data coding and decoding for Biphase Manchester and Biphase Mark

Tool/equipment used: Data formatting trainer kit, 20 MHz dual channel DSO, connecting
wires, patch cord.

Theory

Line coding refers to the process of converting digital data into digital signals. At the sender side
digital data are encoded into a digital signal and at the receiver side the digital data are recreated
by decoding the digital signal. This process of coding is chosen so as to avoid overlap and distortion
of signal such as inter-symbol interference. If chosen correctly, line codes can detect error in
transmission without any additional algorithm.

The symbols '0' and `l' in digital systems can be represented in various formats with different
levels & waveforms. The selection of particular format for communication depends on the
system bandwidth, system's ability to pass DC level information, error checking facility, ease
of clock regeneration & synchronizations at receiver, system complexity & cost etc.

In Bi-phase Encoding, the signal level is checked twice for every bit time, both initially and in
the middle. The clock rate is double the data transfer rate and thus the modulation rate is also
doubled. The bandwidth required for this coding is greater.

Advantages of Biphase coding

It provides guaranteed mid bit transitions. Hence it offers synchronization

facility at receiver.
As they are bipolar signals, ideally no DC components are present.
It provides error detection by detecting absence of expected transitions.
Absence of transition can indicate error in transmission.

Disadvantage of Biphase coding

It maps at least one transition per bit time. Its modulation rate is two times that
of NRZ and it requires more bandwidth
Bit rate is double the baud rate so it requires more bits for encoding
Manchester coding is a special case of binary phase-shift keying (BPSK), where the data
controls the phase of a square wave carrier whose frequency is the data rate. It is used by the
physical layer of the Open System Interconnection to encode the clock and data of a
synchronous bit stream. Manchester encoding is a combination of the RZ and NRZ-L schemes.
Manchester encoding splits each bit period into two, and ensures that there is always a
transition between the signal levels in the middle of each bit and encoding of each data bit is
either low then high, or high then low, for equal time. Two consecutive bits of the same type
force a transition at the beginning of a bit period. The duration of the bit is divided into two
halves. The voltage remains at one level during the first half and moves to the other level in
the second half. For logical 1, the first half consists of a negative voltage and the second-half
is represented as positive voltage, and for logical 0, the first half consists of a positive voltage
and the second-half is represented as negative voltage.

Fig 7.1: Manchester Coding

Biphase mark code is a line code in which data and clock signals are combined to form a
single 2-level self-synchronizing data stream of two levels. It combines the ideas of RZ and
NRZ-I. There is always a transition at the middle of the bit, but the bit values are determined
at the beginning of the bit. If the next bit is 0, there is a transition; if the next bit is 1, there is
none. It uses the presence or absence of transitions to indicate logical value. Only the presence
of a transition is important, not the polarity. BMC transitions on every positive edge of the
clock signal and transitions on negative edge of the clock signal when the data is a 1.

Fig 7.2: Biphase mark coding

Observations:

Fig 7.3: Data Formatting Carrier Modulation Transmitter Trainer Kit

Fig 7.4: input data with clock

 Fig 7.5: Input Data Parameters

Vpp = 14.4 V
Vmax = 8 V
Vmin = -6.4 V
Amplitude = 5 V
RMS = 3.6 V
Frequency = 120.1 KHz

Fig 7.6: Clock Parameters

Vpp = 9.2 V
Vmax = 6.4 V
Vmin = -2.8 V
Amplitude = 5.2 V
RMS = 3.6 V
Frequency = 239.5 KHz
 Fig 7.7: Input data and output data in Biphase Mark format with dual channel

Fig 7.8: Input data and output data in Biphase Manchester format with dual channel

Conclusion: Two data inputs were taken for both formats.

In Biphase Mark, 00001011 was encoded as 10 10 10 10 11 01 00 11 and 01100011 was

encoded as 10 11 00 10 10 10 11 01

In Manchester, 00001011 was encoded as 00 10 10 10 10 11 00 11 and 01011001 was encoded

as 00 11 00 11 01 00 10 11.
 Manchester code derives its name from its development at the University of
Manchester, where the coding was used for storing data on the magnetic drums of
the Manchester Mark 1 computer. It was used in early Ethernet physical layer standards
and is still used in consumer IR protocols, RFID and near-field communication.
Manchester encoding is particularly helpful in binary data transfer based on analog, RF,
optical, high-speed-digital, or long-distance-digital signals.

Biphase Mark coding is specified in the IEEE 802.5 standard for Token Ring local
area networks, and is used for many other applications, including magnetic and optical
storage. Many magnetic stripe cards also use BMC encoding,

Results: Data coding and decoding for Biphase Mark and Manchester was
successfully performed and the outputs were obtained using DSO.

Criteria Total marks Marks obtained Remarks

Concept(a) 2
Implementation(b) 2
Performance(c) 2
Total marks 6
 Experiment-8

Aim: To study speech circuit using IC and its interfacing to the line (EPABX)

Tool/equipment used: Telecommunication EPABX Trainer Kit, 3 Telephone Instruments, 20 MHz

Dual Channel Digital Storage Oscilloscope, Connecting Wires, Power Supply

Theory

A business telephone system is any of a range of a multiline telephone systems typically used in business
environments, encompassing systems ranging from small key systems to large scale private branch.
EPABX stands for Electronic Private Automatic Branch Exchange which is a private telephone network
used by the organizations and the companies for various types of communication, either between the
employees or outside the clients. This is a switching system which has enabled both internal and external
stitching functions for any organization. With the help of this system, businesses aren’t required to run
a line from every phone in the building to the telephone company’s central office. It is independent and
can work without any trunk lines. The main features offered by any EPABX system are call transfer,
call pick up and back up, conference, barge-in etc.

Speech circuit refers to a circuit designed to transmit speech (or any other data) in the form of a digital
or analog signal. All electronic devices that transmit or record speech signal data use a speech circuit to
store the signal data in the appropriate form.

The Telephone system as it presently exists to a discussion of how it is changing and improving as
mechanical and conventional electrical devices are replaced by electronic devices, usually in the form
of integrated circuits These devices have most of the required components on the chip with connections
provided for outboard components such as resistors and capacitors which are used to "program" the
chip. The emphasis is on the circuits that provided two-way speech in the telephone set.

The Speech Circuit –

A simplified block diagram of a speech circuits suitable for implementation as an integrated circuit is
shown in built-in transmit, receive, and side tone circuits, as well as a DC loop interface regulator, and
equalizer circuit. It is connected to the telephone line by a conventional rectifier bridge. The bridge is
dynamically equivalent to a small resistance in series with the signal path, and a high resistance in
parallel to it. External components are used to adjust transmit, receive, and side tone gains and
frequency responses; The DC Line int ace controls the voltage and current characteristics of the entire
speech network depending upon the value of loop current in the subscriber line. Regulator circuitry
sets the operating voltage in the integrated circuit and biases the speech circuits.

Fig 8.1: Speech Circuit

Ring tone is a brief signal of about 100 ms duration sent during an established connection on
a trunk line between central offices by the calling operator to recall the operator at the receiving
office.
The ringback signal is originated from the receiving or called end of a trunk line during an
established connection, to recall the originating operator. The signal is also sent by a coin line
operator to recall a customer at a pay station after the customer hangs up, for example to inform
the customer of time and charges of the completed call.

Observations:

Fig 8.2: Ringing Signal

Fig 8.3: Ring Back Signal

 Fig 8.4: Voice Signals

Conclusion: Speech circuit is an integral part of all electronic devices that have a voice sending
or receiving feature. These include microphones, telephones, cellphones, voice and video
recorders, digital cameras, radio broadcasting equipment and electronic gadgets.

 Ringback is a ringing signal in telephony used to recall an operator or customer

 Ringing tone is the audible ringing that is heard by the calling party after dialing

Results: The study of speech circuit using IC and its interfacing to the line (EPABX) is
successfully completed and the waveforms for ringtone, ring back tone and voice signals have
been plotted using DSO.

Criteria Total marks Marks obtained Remarks

Concept(a) 2
Implementation(b) 2
Performance(c) 2
Total marks 6
 Experiment-9

Aim: To study DTMF and pulse dialling (EPABX)

Tool/equipment used: Telecommunication EPABX Trainer Kit, 3 Telephone Instruments, 20 MHz

Dual Channel Digital Storage Oscilloscope, Connecting Wires, Power Supply.

Theory

A business telephone system is any of a range of a multiline telephone systems typically used
in business environments, encompassing systems ranging from small key systems to large scale
private branch. EPABX stands for Electronic Private Automatic Branch Exchange which is a
private telephone network used by the organizations and the companies for various types of
communication, either between the employees or outside the clients. This is a switching system
which has enabled both internal and external stitching functions for any organization. With the
help of this system, businesses aren’t required to run a line from every phone in the building to
the telephone company’s central office. It is independent and can work without any trunk lines.
The main features offered by any EPABX system are call transfer, call pick up and back up,
conference, barge-in etc.

Pulse dialling

Pulse dialing is a signaling technology in which a direct current local loop circuit is interrupted
according to a defined coding system for each signal transmitted, usually a digit. Pulse dialing
indicates each digit in the phone number by a series of clicks that corresponds only to that digit.
The most common version decodes the digits 1 through 9, as one to nine pulses, respectively,
and the digit 0 as ten pulses. It would then need a short pause in order to clearly identify one
digit from the next.

Pulse dialling telephone sets have a rotary dial which opens and closes the local loop circuits
at a timed rate. The number of dial pulses resulting from one operation of the dial is determined
by how far the dial is rotated before releasing it.

The pulsing switch on the rotary dial of the conventional telephone set is in series with the
induction coil (hybrid) and interrupts the loop current as number of times equal to the number
dialled: one time for the number 1. Five times for the number 5 etc. at 10 pulses per second this
is called loop disconnect dialling. The impulse dialler electronic circuit that replaces the
mechanical dial must do the same function interrupt the loop current at 10 pulses per second.

Pulse dialing has two problems:

 It is very slow — a 0 is not zero pulses, but ten pulses, so to signal a 0 including the
inter-digit intervals takes 1.7 seconds.

 The signaling only goes to the line card on the switch.

 Fig 9.1: Pulse dialling System and function

Dual-tone multi-frequency Dialling

Dual-tone multi-frequency signalling (DTMF) is a telecommunication signalling system using

the voice-frequency band over telephone lines between telephone equipment and other
communications devices and switching centres. Other multi-frequency systems are used for
internal signalling within the telephone network.

It uses different combinations of two sine wave tones to indicate a different number. Instead of
sending multiple signals for each digit, it would only need to send one for each. These tones
are called row and column frequencies as they correspond to the layout of a telephone keypad.
The row frequencies are low group frequencies. The column frequencies belong to high group
frequencies. This prevents misinterpretation of the harmonics. Also the frequencies
for DTMF are so chosen that none have a harmonic relationship with the others and that mixing
the frequencies would not produce sum or product frequencies that could mimic another valid
tone.

These can be used only if the central office is equipped to process the tones. Instead of a rotary
dial, these telephone sets have a push button keypad with 12 keys for the numbers 0 through 9
and the symbols • (asterisk) and $ (pound sign). Pressing one of the keys causes an electronic
circuit in the keypad to generate two output tones that represent the number.

DTMF Dialing Using Integrated Circuits Dialing also may be accomplished by sending dual
tones onto the line. In which a low-frequency and high-frequency sine wave oscillator feed the
speech network, the integrated circuit DTMF generator has a counter and decoder that counts
pulses from a crystal-controlled oscillator and provides output codes that correspond to the low
frequency tone required and the high-frequency tone required. Each of the two outputs from
the counter feed into its own digital-to-analog (D/A) converter. The D/A converter, as the name
implies, converts the digital code out of the counter to a sinewave tone. The low frequency tone
and the high frequency tone are summed in an operational amplifier and are fed to the speech
as a combined signal by the output stage.
 Fig 9.2: DTMF Dial Pad

Fig 9.3: Pulse Dialing vs Tone Dialing

Fig 9.4: Pulse Dialing and DTMF dialing waveforms

Observations:

Fig 9.5: Pulse Dialling Signal

Fig 9.6: DTMF Dialling Signals

Conclusion:

1. Tone dialing uses specific tones to indicate the number that was entered while pulse dialing
uses a number of signal pulses

2. All tone dialing phones have numeric keypads while most pulse dialing phones have rotary
dials

3. Tone dialing is very quick and easy to use while pulse dialing can take a long time

4. Tone dialing is fast and doesn’t require make and break circuit.

5. Tone dialing is the prevailing method right now while pulse dialing is already obsolete

DTMF was first developed in the Bell System in the United States. DTMF tones are mainly
used at the telephone switching centres for detection of dialled/called number. They are also
used by certain radio and cable TV networks to signal a network station or local cable operator
when a local advertisement is to be inserted or for station identification. In broadcasting, this
is known as local insertion. DTMF tones were also used by terrestrial stations for turning on
and shutting off remote transmitters. DTMF is applicable in IVR systems, home automation,
call centers, security systems, as well as industrial applications
Results: The study of DTMF and Pulse Dialling using IC and its interfacing to the line
(EPABX) is successfully completed and the graphs have been plotted using DSO

Criteria Total marks Marks obtained Remarks

Concept(a) 2
Implementation(b) 2
Performance(c) 2
Total marks 6
 Experiment-10

Aim: To study busy and dial tone using EPABX

Tool/equipment used: Telecommunication EPABX Trainer Kit, 3 Telephone Instruments, 20

MHz Dual Channel Digital Storage Oscilloscope, Connecting Wires, Power Supply

Theory

A tone generator is a signal generator circuit which artificially creates sounds frequencies by
converting applied electrical signals into audible compression waves. It can be used to produce
sirens in emergency vehicles, generate melody tunes in toys, in door bells, testing audio
equipment and signalling tones in telephones. Different type of tone generators generates
different audio signals depending on the source and application.

Several different techniques for tone generation are used in multi-tone ringers. The most
common circuit produces two tones. For generating a two-tone ringing signals two oscillators
(or counters) are used- one set to operate at a low frequency (10 to 20 Hz) and other at higher
frequency (440 to 480 Hz). These analog signals are either continuous tones or tone bursts.

A major component of any telephone system is signaling. The electric pulses or audible tones
are used for alerting, addressing, supervision and information. Signalling Tones are involved
in establishing, maintaining and releasing a telephone conversation.

Dial Tone

 A dial tone is a telephony signal sent by a to a subscriber device when an off-hook condition
is detected. It indicates that the exchange is working and is ready to accept dialed digits from
the subscriber to initiate a telephone call.

 Most often the dial tone is sent out by the exchange even before the handset is brought near the
ear. Sometimes, a few seconds may elapse before the dial tone is heard. This happens
particularly in common control exchanges which use shared resources for user interfaces.

 The dial tone is a 33 Hz or 50 Hz or 400 Hz continuous tone. The 400 Hz signal is usually
modulated with 25 Hz or 50 Hz.

Fig 10.1: Dial Tone Wave

 Busy Tone
 A busy tone is sent to the calling subscriber whenever the switching equipment or junction line
is not available to put through the call or the called subscriber line is engaged.
 It is a bursty 400Hz signal with silence period of same duration in between.
 The burst and silence durations have the same value of 0.75s or 0.375s.

Fig 10.2: Busy Tone wave

Observations:

Fig 10.3: Dial Tone on DSO Fig10.4: Busy tone on DSO

Conclusion: The tone generator's function is to produce different frequency sine wave for each digital
word applied to its input. They are commonly used in home security setup, to produce dial tone in
telephonic devices and and melodic tunes in devices such as toys & doorbells.
 Dial tone inform the calling subscriber that the exchange is ready to establish the call
 Busy tone inform the calling subscriber if the called party is busy.

Results: The study of Tone generator and dialler circuit is successfully completed and the dial tone and
busy tone were plotted using DSO

Criteria Total marks Marks obtained Remarks

Concept(a) 2
Implementation(b) 2
Performance(c) 2
Total marks 6