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Lab Manual Mobile Computing

The document outlines the vision and mission of the Department of Electronics & Telecommunication Engineering, emphasizing hands-on activities and the production of skilled engineers. It includes a preface discussing cellular network planning and various experiments related to telecommunications, such as PSTN switch configuration and traffic analysis. Additionally, it covers theoretical concepts like Erlang's formula and mathematical modeling for telecommunication traffic analysis.

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0% found this document useful (0 votes)
17 views46 pages

Lab Manual Mobile Computing

The document outlines the vision and mission of the Department of Electronics & Telecommunication Engineering, emphasizing hands-on activities and the production of skilled engineers. It includes a preface discussing cellular network planning and various experiments related to telecommunications, such as PSTN switch configuration and traffic analysis. Additionally, it covers theoretical concepts like Erlang's formula and mathematical modeling for telecommunication traffic analysis.

Uploaded by

aniket thete
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
You are on page 1/ 46

ATC,CHAS,AHMEDNAGAR

Department of Electronics & Telecommunication


Engineering

Vision of the Department

To be a premier department in the field of Electronics and Telecommunication engineering with


emphasis on hands-on activities.

Mission of the Department

M1: Producing professional engineers to handle the state-of-the-art technologies in the field.

M2: Imparting research and innovative aptitude with high moral for overall development of students.

M3: Establish leadership skills, team spirit and high ethical values among the students.

M4: Encourage students to become entrepreneur.

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Preface
“A practical approach is probably the best approach to mastering a subject and gaining a
clear insight.”

Cellular Network Planning begins with an introduction to the subject, covering


conventional and contemporary wireless systems. Spectral allocation and the frequency
plan are discussed, along with the essential characteristics of wireless systems. The design
of mobile cellular systems includes cell planning, traffic and channel problems. The
manual presents a review of existing models, considering both green field dimensioning
and network expansion strategies, and discusses multi-objective optimization and base
station deployment based on artificial immune systems. It also discusses a cost-effective
base station deployment approach based on artificial immune systems, and introduces the
Link-Budget analysis for a wireless communication system .

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INDEX

Expt. No. Name of the Experiment Page No.


1. To study of PSTN-TST Switch 1
To Study & Implement Lost call system/delay system used in the
2. 7
analysis of voice/ Data Traffic.
3. To Study AT commands. 11
To Write a program to simulate experiment on
4. GMSK/QPSK/QAM modulation over AWGN to evaluate Bit 14
error rate.
To write a program to measure bit error rate in presence of Hata
5. 18
propagation model and establish Link budget
6. To implementation of VOIP Protocols 23
To Perform an experiment to elaborate the operation of Multiple
7. 26
access technique such as CDMA.
8. To Visit Mobile Telephone switching office 33
Compute and compare the median loss by employing HATA
9. model for various distance for carrier frequencies of 2.1 GHz and 34
6 GHz.
Simulate BER performance over a wire line AWGN channel with
10. 35
BPSK transmission for SNR: 0 to 50 dB.
Compute the RMS delay spread for a given Power profile and
11. 38
plot the graph of Power VS Delay.

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Expt.No:1 Date:
Study of PSTN-TST Switch

Aim: To study the simulation of the PSTN switch configuration T-S-T switch.

Apparatus: PSTN T-S-T switch hardware trainer kit, connecting probes, CRO, CRO
probes.

Theory: The PSTN T-S-T system consists of the following sections.


i) I/p signal generators
ii) Sampling pulse generators
iii) Time division switching section (T-switch)
iv) Space division switching section (S-Switch)
v) Time division switching (Demultiplexing) section: (T –Switch)
vi) Power Supply

i) I/p signal generators :


IC (IC L 8088) is waveform generator. DC is used to generate sine wave
signal, Preset is used to fix it‟s frequency to 250 Hz, 500Hz, 1 GHz,Pr1 & Pr2 are
adjusted for proper peaks of sine wave signal Pr3 is used to adjust duty cycle This
sine wave output signal is available at pin2 of IC & it is then amplified at pin 2 of
IC & then amplified by IC4. The amplified output terminal Pot P1 is used to vary
the amplitude of sine wave Signal. The output amplitude varies from 0 to 5 Vp-p.

ii) Sampling Pulse generator section :


To generate time division switching (multiplexing) sample and hold circuit is
used the sampling pulse generator is required to provide sampling signal to S/H
circuit. This section is based on voltage controlled oscillator (VCO) IC
CD4046B.The pulse frequency is determined by varying control voltage is varied
by frequency POT (10K).This section generates the sampling signal of variable
frequency 2 to 128KHz width.

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iii) Time Division Switching (Multiplexing) Section (T-Switching) :


To generate TDM signal S/H circuit & analog switches used .The sampling
pulse generator is required to provide multiplexing of four different signals. The 4
i/p signals are given to 4 different event i/ps of multiplexers. IC CD4053B.This IC
passes signal-I during “00” counting pulse of counter which counts sampling
pulse signal-II is sampled during “01” count .similarly signals-III counting .Thus
at the o/p the multiplexed signal is available.

iv) Space division switching section (S-Switch) :


To transfer time division switch into space switching cross point DC is used.
The IC-22100 is 4 x 4 space switching circuit is accepts inputs at x1,x2,x3 & x4
input points. The out points y1,y2,y3 & y4.The control input consists of 4 address
inputs A,B,C & D and strobe inputs. As shown in truth table .

v) Time Division Switching (Demultiplexing) Section (T-Switching) :


This section is based on demultiplexer IC CD 4053B & LPF. The TDM
multiplexed signal is applied to the o/p this IC. Then this IC passes signal 1 to the
o/p-1 during “00” count of sampling pulse while it passes signal-2 to o/p-2 during
“01” count of sampling pulse. Similarly signal-3 & signal-4 are passed during o/p
“10” & “11” respectively. Thus de-mux 4 different LPF which pass only low
frequency .Thus, this removes high frequency noise of PAM signal.
This section is based on 3 T4, LC, RC circuits by removing high frequency
.We recover original modulating signal.

vi) Power Supply Section :


The regulated power supply is used for different supply voltages following
o/p DC voltages are required to operate trainer.
+15 V, 250 mA
-15 V, 250 mA
+5 V, 250 mA
3-terminal regulators are used for different o/p voltages.
These IC‟s are supplied by different DC i/p voltages by 2 half wave rectifiers
consisting of D1 to D4 & C1, C2, C3& C4. The capacitors at each i/p & each o/p
are for filtering purpose SW301 is main AC ON/OFF switch.
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Procedure:
i) Connect signal i/p-1 terminal of multiplexer section to sine wave signal of sine
wave generator with frequency & amplitude 2 Vp-p w/f (W1).
ii) Connect i/p-2 terminal of multiplexer section to sine wave signal of sine wave
generator with frequency & amplitude 1 Vp-p w/f (W2).
iii) Connect i/p-3 terminal of multiplexer section to sine wave signal of sine wave
generator with frequency & amplitude 2 Vp-p w/f (W3).
iv) Connect i/p-4 terminal of multiplexer section to sine wave signal of sine wave
generator with frequency & amplitude 2 Vp-p w/f (W4).

v) Keep frequency POT of sampling pulse generator on mid position.


vi) Connect CRO channel at TDM o/p signal (T-Switching).Observe TDM signal w/f
(W4).
vii) Keep all control i/p (A, B, C, D) to low value, keep data control i/p to low value.
Apply strobe pulse by pushing pulse switch. This will reset all o/p i.e. all gates are
blocked & no o/p are available at signal-1 & signal-2 to test point. This is the T-S-
T switching.

Conclusion:
Thus, we have studied, designed& simulated PSTN switch configuration T-S-T
Switch.

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Fig. 1 TDMA Multiplexing/Demultiplexing

Fig. 2 Multiplexing / Demultiplexing.

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Design:
Address Select
A B C D D0 D1
0 0 0 0 x1 y1
1 0 0 0 x2 y1
0 1 0 0 x3 y1
1 1 0 0 x4 y1
0 0 1 0 x1 y2
1 0 1 0 x2 y2
0 1 1 0 x3 y2
1 1 1 0 x4 y2
0 0 0 1 x1 y3
1 0 0 1 x2 y3
0 1 0 1 x3 y3
1 1 0 1 x4 y3
0 0 1 1 x1 y4
1 0 1 1 x2 y4
0 1 1 1 x3 y4
1 1 1 1 x4 y4

Oral Questions:
1. What are the different kinds of switching systems?
2. Define a. Circuit Switching b. Manuel Switching c. Message Switching
d. Electronic Switching
3. Explain the principle of crossbar switching?
4. What is combination switching? Describe combinational switching?
5. Explain two stage combinational switches? a. TS switch b. ST switch
6. Explain three stage combinational switches? a. TST switch b. STS switch
7. Compare TST & STS networks.
8. Define the terms a. Busy Hour(BH) b. Call Completion Rate(CCR)
c. Busy Hour Calling Rate(BHCR) d. Day to busy hour traffic ratio
e. Traffic Intensity(Ao)
9. What is the unit of traffic? What is the Relation between E, CCS, CS & CM.

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10. What is Probability of delay


11. Define a.GOS b. Blocking probability
12. Explain the Lost System? What are the three important models of loss system?
a. LCC b.LCR c.LCH
13. Explain the Erlang Distribution system?
14. Explain the Markaw chain?

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Expt. No. 2 Date :

Lost call system/delay system used in the analysis of voice/ Data Traffic.

Aim: To study the mathematical analysis of telecommunication traffic.

Objective:
1. To derive the derivation of Erlang‟s formula.
2. To solve the numerical based on telecommunication traffic.

Theory:
1. The unit of traffic: The telecommunication traffic is defined as avg. no calls in
progress. The quality traffic is measure in Erlang or in CCS.
It is also denoted is
1 Erlang = 36 CCS = 3600CS = 60CM
Where,
CS = Call Second
CM = Call Minute
CCS = Centum Call Seconds

2. Grade of Service: It is defend as a proportion of call that lost or delay due to


congestion.
No.of call lost
GOS =
Traffic Offered
Traffic Lost
GOS =
Traffic Offered
A−A0
GOS =
A

Where A= traffic offered


A0= carried traffic
A-A0 = lost traffic
3. A Mathematical model: Mathematical model is analytical solution for telegraphic
problem. Mathematical model is base on following assume
i) Pure chance traffic.
ii) Statistical Equilibrium

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The number of call arrival rate in a given period of time has a Poisson distribution
P(x) = µx e-µ
𝑥!

Where x= No. of call arrives


µ= mean no of call

4. Lost calls system:

Fig. 1 Lost-call System


Lost calls system is based on assumption given as
a) Full availability: Full availability mean each cell arrival connect to any of
outgoing trunks.
b) Pure Chance Traffic: Pure chance traffic means that call arrival & call
termination are independent triangle event.
c) Statistical equilibrium: Statistical equilibrium means probability remain
unchanged
d) Calls that encounter congestion are lost: Call which encounter congestion
are lost means any call which encounter congestion is immediately remove
from system here it is assume that the traffic offered is total of successful &un
successful calls. “x” calls are in progress are given as
P(x) = µx e-µ for 0 < x < N ---------------------- (1)
𝑥!
There cannot be more than N calls nor there can be negative number of calls.
Thus we can conclude that
∑𝑁 (𝑥) = 1 = ∑𝑁 µx p(0) ---------- (2)
𝑥=0 𝑥=0 𝑥!

Hence, P(0) = 1 ---------- (3)


µx
∑ p(0)
𝑥=0 𝑥!

Equation (3) substituting in equation (1)


µx
P(0) = 𝑥!
µx
∑ p(0)
𝑥=0 𝑥!

That is known as 1st Erlang‟s distribution for queuing system called 2nd
Erlang‟s distribution

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Numerical:
1) A subscriber makes 3 phone calls of 3,4 & 2 min duration in 1hr period.
Calculate the subscriber traffic in Erlang, CS, CCS & CM
2) A group of 3 trunks is offered 4E of traffic ,Find
a) The probability that only one trunk is free
b) The probability that only one trunk is busy
c) The probability that at least one trunk is free

Algorithm: A group of 5 trunks is offered 2 E of traffic. Find the Grade of service, the
probability that only one trunk is busy, the probability that only one trunk is free, the
probability that at least one trunk is free.

A mathematical model To obtain analytical solutions to teletraffic problems it is necessary


to have a mathematical model of the traffic offered to telecommunication system.
A simple model is based on the following assumptions
• Pure chance traffic
• Statistical equilibrium
Pure chance traffic: The assumption of pure chance traffic means that call arrivals and call
terminations are independent random events. Sometimes it is also called as Poissonian traffic.
If call arrivals are independent random events, their occurrence is not affected by previous
calls. Sometimes traffic is called as memoryless traffic. This assumption of random call
arrivals & termination leads to the following results

Statistical equilibrium: The assumption of Statistical equilibrium means that the generation
of traffic is a stationary random process i.e., probabilities do not change during the period
being considered. Consequently the mean number of calls in progress remains constant.
Statistical equilibrium is not obtained immediately before the busy hour, when the calling rate
is increasing nor at the end of the busy hour, when calling rate is falling.
Conclusion:
Hence we have studied the mathematical analysis of telecommunication traffic & solve
the numerical on telecommunication traffic.

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Matlab Code
% Write a program to simulate LCC model
% Input for this pgm - A, N
% Output of this pgm - B
%
clc;
clearall;
n=input('Enter the no of servers=');
c=input('Enter the no. of calls per hour=');
th=input('Enter the holding duration of eah call=');
A=c*th/60;
A=round(A);
fprintf('\n The value of Offered traffic A=%d',A);
% Specified A and N
for N=1:1:n
Num=A^N/factorial(N)
fprintf('\n The value of Numerator Num=%d',Num);
sum=0;
for k= 1:1:n
Den=A^k/factorial(k)
sum=sum+Den;
end
Den=sum+1;
fprintf('\n The value of Denominator Den=%d',Den);
B=Num/Den;
gridon;
plot(N,B,'r+:')
end
xlabel('No of trunks = ');
ylabel(' Blocking probability=');
title('Program to simulate LCC model');

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Expt. No: 3 Date


AT commands

Aim: To Set up and carry out experiment on AT commands for voice and data
operation.

Apparatus: GSM trainer kit.

Theory:
GSM (Global System for Mobile Communication):
It has been the backbone of the phenomenal success in mobile communication
over last decade. Now at down of era of the broadband service, GSM continue to
involve to meet new demand.
In GSM system frequency increases, the call handling capability of a system
without increasing the no. of available channel, when using identical frequency in a
small area. The GSM system can tolerate higher level of co-channel interference for
FEC and equalization.
Mobile station, mobile equipment are used for mobile terminals supporting
GSM services. A call from Mobile originated call & a call from fixed network to GSM
mobile station is called a Mobile terminated call (MTC) or incoming call.

AT command features:
i) Command line: Command always start with AT (which means attention) and
finish with a character (CR).
ii) Info. Response & Result codes: Response start and end with <CR> LF> except
for AT0 ( DCE response format ) & AT (Result code suppression) commands. If
command syntax is corrected for but with some incorrect parameter the CME
error <ERR> or +CMS ERROR <SMS ERR> string.
If command line has been performed successfully on “OK” string is returned. In
some cases, such as “AT+CPRN” or incoming events it does not return OK string
or response.

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Some of the important AT commands are as follows.


1] General Commands:
a) AT + CGMI: Gives the manufacturer identification.
b) AT+CGMM: Gives the model information inters of supported freq. band & it
multiband then there is a combination of diff. bands.
c) AT+CGMR: Gives the received s/w version.
d) AT+CGSN: Gives the IMEI i.e. International Mobile Equipment Identity of
product.
e) AT+CGCS: Used to inform ME which character set is used by TE.

2] Phone Book Commands :


a) AT + CPBS: “SIM” = Selects SIM phonebook.
b) AT + CPBR: 12:14 = Read entries from 12 to 14.
c) AT + CPBW: 3 = Erase third location
d) AT + CPBF: “N” = Read entries with „N‟
e) AT + CPBP: “9245364864” = Search corresponding entry.

3] Call Control Commands:


a) ATA: Answer to incoming calls.
b) ATH: Disconnect the call.
c) ATDC: Redial the dialed number.
d) ATSO: „2‟ = Automatic answer after „2‟ rings.

4] Security commands:
a) AT + CPRN: “1234” = Enter PIN code i.e. 1234 here.
b) AT + CPIN: <PUK> <NEW PIN> = Enter PUK code & new PIN code.
c) AT + CPIN2: < PIN2> = Enter PIN2 code.
d) AT + CPIN2: <PUK 2> < NEW PIN2> = Enter PUK2 & New PIN2 code.

5] Network Service Commands :


a) AT + COPS = 2 = Ask for cuttenr PLMN
b) AT + COPS = 2 = Ask for PLMN list
c) AT + COPS = ? = Ask for registration on home s/w
d) AT + COPS = 0 = Set <format> to long alphanumeric
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e) AT + COPS = 2 = Ask for de-registration on n/w.

6] SMS Command :
a) AT + CSMS = 0 = SMS AT commands are compatible with GSM 7.05 phase 2
version 4.7.
b) AT + CSMS = 1 = SMS AT commands compatible with phase 2.

7] Supplementary service commands :


a) AT + CLCK = “A0”,1,1234 = Locks the facility for barring outgoing calls.
b) AT + CLCK = “AT”,0,1234 = Unlocks the facility for barring incoming
calls.
c) AT + CPW1 > = “AO”, 1234, 5555 = Used to change password for barring
outgoing calls.

Conclusion:
Hence, we have studied GSM, trainer kit & their related AT commands.

Oral Questions
1. What is GSM?
2. What is the maximum data rate supported by a GSM system?
3. What are the different GSM technologies?
4. Which uplink/ downlink spectrum is allocated to a.) GSM – 900: b.) DCS – 1800?
5. Explain a.) Equalization b.) Interleaving c.) Speech coding d.) Channel coding
6. What is frequency re-use?
7. What are the GSM interfaces
8. Explain Cell Splitting ?
9. Explain: a.) Synthesized Frequency Hopping, b.) Base Band Frequency Hopping, c.)
Cycling Frequency Hopping
10. What do you mean by following? a.) Erlangs b.) GoS
11. Explain: a)FDD b)TDD

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Expt. No: 4 Date


Experiment on GMSK/QPSK/QAM

Aim: Write a program to simulate experiment on GMSK/QPSK/QAM modulation over


AWGN to evaluate Bit error rate

Resources required/Apparatus: PC with MATLAB

Theory: The move to digital modulation provides more information capacity, compatibility
with digital data services, higher data security, better quality communications and quicker
system availability. Developers of communication systems face following constraints:
1. available bandwidth
2. permissible power and
3. inherent noise level of the system
The RF spectrum must be shared, yet everyday there are more users for this spectrum as
demand for communication services increases. Digital modulation schemes have greater
capacity to convey large amounts of information than analog modulation schemes.
Phase shift keying (PSK) involves the switching of the instantaneous phase of the carrier
between 2 or more levels according to the baseband digital data. A MATLAb script to
generate Binary PSK is given below :
clearall;
closeall;
clc;
fc=1000; % Frequency for "0" bits
t=linspace(0,1/1000,50);
e0=cos(2*pi*fc*t); % BPSK output for "1"
e1=-cos(2*pi*fc*t); % BPSK output for "0"
b=mod(randperm(16),2);
bnot=1-b;
n=['The binary data is ',num2str(b)];
bpsk1=[];bpsk2=[];bin=[];
for i=1:length(b)
bpsk1=[bpsk1,b(i)*e0];
bpsk2=[bpsk2,bnot(i)*e1];
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bin=[bin,b(i)*ones(1,50)];
end;
bpskout=bpsk1+bpsk2;
tm=[0:length(bpsk1)-1];
plot(tm,bin,'r--');
axis([0 length(bin) 0 1.5]);
holdon;
plot(tm,bpskout,'b');
axis([0 length(tm) -1.5 1.5]);
holdoff;
xlabel('Time index');
ylabel('Amplitude');
legend('Random binary','BPSK output');
title('Simulation of Binary Phase Shift keying');
gtext(n);

Transmitter:
For the QPSK modulation, a series of binary input message bits are generated. In
QPSK, a symbol contains 2 bits. The generated binary bits are combined in terms of two bits
and QPSK symbols are generated. From the constellation of QPSK modulation the symbol
„00‟ is represented by 1, „01‟ by j (90 degrees phase rotation), „10‟ by -1 (180 degrees phase
rotation) and „11‟ by -j (270 degrees phase rotation). In another form of QPSK, these phase
rotations are offset by 45 degrees. So the effective representation of symbols in this form of
QPSK is „00‟=1+j (45 degrees), ‟01′=-1+j (135 degrees), „10‟ = -1-j (225 degrees) and „11‟=
1-j (315 degrees). Here we are simulating a 45* rotated QPSK system. Once the symbols are
mapped, the power of the QPSK modulated signal needs to be normalized by 1/sqrt(2).
For the channel model “randn” function in Matlab is used to generate the noise term.
This function generates noise with unit variance and zero mean. In order to generate a noise
with standard deviation - σ for the given Eb/N0 ratio, use the above equation, find σ, multiply
the “randn” generated noise with this σ, add this final noise term with the transmitted signal
to get the received signal. For a pi/4 rotated QPSK system, since the modulated signal is in
complex form (due to sine and cosine basis functions),the noise should also be in complex
form.

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Receiver:
QPSK receiver employs two threshold detectors that detect real (in phase arm) and
imaginary parts(quadrature arm). The detected signals are sent through a parallel to serial
converter (implemented by“reshape” function in MATLAB).

% Demonstration of Eb/N0 Vs BER for QPSK modulation scheme


clear;
clc;
% Input Fields
N=1000000;%Number of input bits
EbN0dB = -4:2:10; % Eb/N0 range in dB for simulation
%
data=randn(1,N)>=0; %Generating a uniformly distributed random 1s and 0s
oddData = data(1:2:end);
evenData = data(2:2:end);
qpskModulated = sqrt(1/2)*(1i*(2*oddData-1)+(2*evenData-1)); %QPSK Mapping
M=4; %Number of Constellation points M=2^k for QPSK k=2
Rm=log2(M); %Rm=log2(M) for QPSK M=4
Rc=1; %Rc = code rate for a coded system. Since no coding is used Rc=1
BER = zeros(1,length(EbN0dB)); %Place holder for BER values for each Eb/N0
index=1;
for i=EbN0dB,
%
%Channel Noise for various Eb/N0
%
%Adding noise with variance according to the required Eb/N0

EbN0 = 10.^(i/10); %Converting Eb/N0 dB value to linear scale


noiseSigma = sqrt(1./(2*Rm*Rc*EbN0)); %Standard deviation for AWGN Noise

%Creating a complex noise for adding with QPSK modulated signal


%Noise is complex since QPSK is in complex representation
noise = noiseSigma*(randn(1,length(qpskModulated))+1i*randn(1,length(qpskModulated)));
received = qpskModulated + noise;
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% %Threshold Detector
detected_real = real(received)>=0;
detected_img = imag(received)>=0;
estimatedBits=reshape([detected_img;detected_real],1,[]);
% %Bit Error rate Calculation
BER(index) = sum(xor(data,estimatedBits))/length(data);
index=index+1;
end

%Plot commands follows


plotHandle=plot(EbN0dB,log10(BER),'r--');
set(plotHandle,'LineWidth',1.5);
title('SNR per bit (Eb/N0) Vs BER Curve for QPSK Modulation Scheme');
xlabel('SNR per bit (Eb/N0) in dB');
ylabel('Bit Error Rate (BER) in dB');
gridon;
holdon;
theoreticalBER =0.5*erfc(sqrt(10.^(EbN0dB/10)));
plotHandle=plot(EbN0dB,log10(theoreticalBER),'b*');
set(plotHandle,'LineWidth',1.5);
legend('Simulated','Theoretical-QPSK','Theoretical-QPSK');
gridon;

Conclusions : Hence we simulated experiment on QPSK modulation over AWGN to


evaluate Bit error rate

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Expt. No: 5 Date :


Hata propagation model

Aim : Write a program to measure bit error rate in presence of Hata propagation model
and establish Link budget

Resources required/Apparatus: PC with MATLAB

Theory: In order to plan the installation and deployment of a wireless network, one
needs to characterize the performance of the communication system in terms of the
transmitted power and total load in terms of users that can be supported by the network.
Link-budget of wireless link is a systematic listing of power losses and gains of
different intermediate components in the transceiver chain.

Hata Model
The Hata model is a popular model for signal strength prediction proposed initially by
the Japanese engineer MasaharuHata in his 1980 paper titled “ Empirical Formula for
Propopgation Loss in Land Mobile Radio Services “. The Hata model presents an
analytical approximation for the graphical-information based on another Okumura
model. Parameters required for simulations :
Hbts= Height measured from the base of the BTS tower to the radiation centerline
Tbts = Terrain elevation at the location of the BTS
Htav= Height of the average terrain (from 3 Km to 15 km distance from the BTS)
Hm=Height of the mobile antenna in meters
f= Range of frequencies in MHz
d=Range of Tx-Rx separation distances in Kilometers
Pt = Power transmitted by the BTS antenna in Watts
Gt= BTS antenna gain in dBi
Hb= Hbts+ Tbts - Htav = Effective Height of the BTS antenna in meters
models =
Big City (Urban model
aHm=3.2*(log10(11.75*Hm))^2-4.97;
C=0
Small & Medium City (Urban model)
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aHm = (1.1*log10(f)-0.7)*Hm-(1.56*log10(f)-0.8);
Sub-urban environment
aHm = (1.1*log10(f)-0.7)*Hm-(1.56*log10(f)-0.8);
C=-2*(log10(f/28))^2-5.4;
Open Rural environment
aHm = (1.1*log10(f)-0.7)*Hm-(1.56*log10(f)-0.8);
C=-4.78*(log10(f))^2+18.33*log10(f)-40.98;
A = 69.55 + 26.16*log10(f) - 13.82*log10(Hb)-aHm;
B = 44.9 - 6.55*log10(Hb);

Path Loss (dB)=PL=A+B*log10(d)+C;


Received Signal Level(dB) =Pr = 10*log10(Pt*1000)+Gt-PL

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%Matlab code to simulate Hata-Okumura Models


clc;clearall;
% Input Section
Hbts= 50 ;%Height measured from the base of the BTS tower to the radiation centerline
Tbts = 350 ;%Terrain elevation at the location of the BTS
Htav= 300;%Height of the average terrain (from 3 Km to 15 km distance from the BTS)
Hm=3 ;%Height of the mobile antenna in meters
f=900 ;%100:100:3000; %Range of frequencies in MHz
d=0.5:0.5:15; %Range of Tx-Rx separation distances in Kilometers
Pt = 0.020; %Power transmitted by the BTS antenna in Watts
Gt= 10; %BTS antenna gain in dBi
%
Hb=Hbts+Tbts-Htav ;%Effective Height of the BTS antenna in meters
%Cell array to store various model names
models = {'Big City (Urban model)';'Small & Medium City (Urban model)';'Sub-urban
environment';'Open Rural environment'};
display('Hata-Okumura Model');
display(['1 ' models{1,1}]);
display(['2 ' models{2,1}]);
display(['3 ' models{3,1}]);
display(['4 ' models{4,1}]);
reply = input('Select Your choice of environment : ','s');
if 0<str2num(reply)<4
modelName = models{str2num(reply),1};
display(['Chosen Model : 'modelName])
else
error('Invalid Selection');
end
switch reply
case'1',
C=0;
if f<=200
aHm=8.29*(log10(1.54*Hm))^2-1.1;
else
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aHm=3.2*(log10(11.75*Hm))^2-4.97;
end
case'2',
C=0;
aHm = (1.1*log10(f)-0.7)*Hm-(1.56*log10(f)-0.8);
case'3',
aHm = (1.1*log10(f)-0.7)*Hm-(1.56*log10(f)-0.8);
C=-2*(log10(f/28))^2-5.4;
case'4',
aHm = (1.1*log10(f)-0.7)*Hm-(1.56*log10(f)-0.8);
C=-4.78*(log10(f))^2+18.33*log10(f)-40.98;
otherwise ,
error('Invalid model selection');
end
A = 69.55 + 26.16*log10(f) - 13.82*log10(Hb)-aHm;
B = 44.9 - 6.55*log10(Hb);
PL=A+B*log10(d)+C;
subplot(2,1,1)
plot(d,PL,'r','LineWidth',2);
title(['Hata-Okumura Path Loss Model for : 'modelName]);
xlabel('Distance - Kilometers');
ylabel('Path Loss (dB)');
%Compute Received Signal Level
Pr = 10*log10(Pt*1000)+Gt-PL
subplot(2,1,2)
plot(d,Pr,'r','LineWidth',2);
title(['Hata-Okumura Model for : 'modelName]);
xlabel('Distance - Kilometers');
ylabel('Received Signal Level (dBm)');

Conclusions: Hence we have simulated program to measure bit error rate in presence
of Hata propagation model and establish Link budget.

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Oral Questions
1. Write expression for net signal power at the receiver?
Ans : The link-budget expression for the SNR required is given as:
SNRreq  Pt (dB)  Gt (dB)  L(dB)  M (dB)  Gr (dB)  Lc (dB)  (N  I )dB
SNRreq  net signal power at the receiver
Gt  Transmit  antenna gain
L  Median  link  propogation Loss
M=Margin
Gr  Mobile  receive antenna gain
Lc  Cabling losses
N  I  Re ceiver(noise  interference)
The above equation can be recast to compute the required transmit power as :
Pt (dB)  SNRreq  Gt (dB)  L(dB)  M (dB)  Gr (dB)  Lc (dB)  (N  I )dB

2. Derive an expression of received power for Free space propagation model?


3. What is meant by Small Scale Fading?

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Expt. No: 6 Dates:


Implementation of VOIP Protocols

Aim: To study VOIP protocol implementation.

Apparatus: VOIP Phone, Modem 56K, VOIP Subscription with Skype, VOIP software
etc.

Theory: VOIP refers to the information of telephone conversion over a packet


switched RF network. It could be as small as single subnet LAN or as large as public
internet.

Fig. 1 Component of H-323 VOIP Network.

Fig. 2 VOIP Application Scenarios.


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Why use VoIP?


1. Better bandwidth utilization by:
i. Using compression
ii. Exploiting silence periods during conversations
iii. Sharing of equipment for voice and data traffic (unified processing)
2. Introduction of new services: Conferences, distance learning, etc.

Elements of a VoIP network


1. Terminals: LAN-based communication end-points.

2. Gateway (media and/or signaling):


Interface between packet- and circuit-switched networks
i. Media gateway: voice transcoding, protocol conversion
ii. Media gateway controller: call handling, call state
iii. Signaling gateway: signaling mediation

3. Gatekeeper: Admission control, SNMP services, addresses translation.

4. MCU:(Multipoint Control Unit) Handling of broadcasts / conference calls VoIP


context.
5. Int. Telecomm. Union (ITU) VoIP:
i. International Telecommunication union (ITU-T) recombination H.32 is a global
standards for packet based multimedia communication including VOIP.H-233
can also be implemented on WAN‟s or LAN‟s.
ii. H-233 is referred as an “Umbrella” standard because it supports multimedia
communication.
iii. H-233 is a set of network components & protocols that supports real time
audios, videos & data communication.

6. Hardware Overview:
i. Voice Skype Cyber phone-K:
It is 2-piece USB phone with keypad & a built in ringer that connects to PC
USB port apart from usual buttons ,there are some special buttons provided.

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ii. Connecting Cyber phone-K:


To install ,imply plug the USB of phone into a USB socket of PC or a powered
USB hub.
iii. VOIP voice Contact:
It is a small compact USB telephone adaptor that connects your PC USB port
telephone lines. It enables you to make calls using Skype or ordinary telephone
calls from the same card or cordless telephones.
iv. USB/PSTN Switch:
Set this switch to USB/PSTN, if you have connected your

Features of VOIP
1. User benefits by new applications based on converged voice & data n/w.
2. Service provides benefits with more efficient use of their n/w
3. Equipment makers will benefit from the new types of software.

Application of VOIP:
1. Back of carrier networks
2. Converged data & voice networks
3. On cellular Phones.

Conclusion:
Hence, we have studied Implementation of VOIP commands.

Oral Questions
1. What is VOIP ?
2. What are cyber phone technology?
3. What is Skype ?
4. What is the use of Skype?
5. What are applications of VOIP ?
6. What are features of VOIP?
7. What are Hardware specifications of VOIP ?
8. What are the elements of VOIP ?
9. What is PSTN ?
10. What is ISDN ?
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Expt. No: 7 Date:


Multiple access technique (CDMA)
Aim:
a. To Perform an experiment to elaborate the operation of Multiple access
technique such as CDMA.
b. Generation and detection of DS-SS with coherent BPSK
Objective:
1. To study the CDMA
2. To study the Generation of PN sequence and it‟s Properties.
3. To study the DS-SS with BPSK transmitter & Receiver.
4. To study the performance characteristics of DS-SS with BPSK.
Equipments:
1. CDMA trainer kit no.2115
2. DSO.
Theory:
SS stands for Spread spectrum. There are some applications in
communication where it is necessary for the system to resist external interference and
to make it difficult for unauthorized receivers to receive the message being
transmitted. For e.g. In military applications. In such cases SPREAD SPECTRUM
MODULATION is used. The SS occupies a larger bandwidth than that of a normal
s/g is „pseudo random‟ in nature.
Because of its random nature when the original data to be transmitted
modulates this a encoded or encrypted data is obtained. This data can‟t be decoded on
the receiver side w/o the knowledge of the same „special code (PR sequence)‟ which
is independent of the message s/g. Thus this prevents the undesirable hacking of data.
Since the random sequence occupies larger BW after modulation, the modulated s/g
too occupies a large BW. i.e. it‟s spectrum is spreaded . Hence the name, Spread
Spectrum.
In direct sequence modulation, two stages of modulation are used. In the first
stage the incoming data sequence modulates the wideband code. This transforms the
narrowband incoming data sequence into wideband s/g. This then shifts the phase of
the carrier i.e. it then undergoes second modulation using PSK. This is called as
Direct Sequence Spread Spectrum (DSSS) with BPSK.

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DS-BPSK Transmitter:
Transmission of DS-SS with BPSK :
The binary sequence d(t) is converted into NRZ s/g, b(t) by using the
NRZ encoder. The NRZ s/g b(t) is then used to modulate the PN sequence c(t)
generated by the PN code generator. The transmitter uses two stage of modulation.
The first stage uses a product multiplier with b(t) and c(t) as it‟s i/p‟s and the
second stage consists of a BPSK modulator. The modulated s/g at the o/p of the
product modulator i.e. m(t) is used to modulate the carrier for BPSK modulation.
The transmitted s/g, x(t) is thus a direct sequence spread BPSK i.e. DS-BPSK s/g.
The BPSK carrier is given by,
Vcarrier(t) =√ (2Ps) sin(2∏fct)
The o/p of the BPSK modulator i.e. x(t) is terminated. x(t) is given mathematically
as
x(t) = m(t) x Vcarrier(t) = m(t) x √2Ps sin( 2∏fct )
but m(t) = + 1
hence x(t) = √2Ps sin( 2∏fct )

thus the phase shift of x(t) is 0 ْ˚ corresponding to a positive m(t) and it is 180 ْ˚
corresponding to a negative m(t).

DS-BPSK Receiver:
The received s/g x(t) and the locally generated replica of the PN sequence are
applied to a multiplier. This is the first stage of multiplication. The multiplier
performs the dispreading operation. O/p of multiplier is then applied to a coherent
BPSK detector with a local carrier applied to it. At the o/p of the coherent BPSK
detector we get back the original data sequence.

Spectrum of DS-SS with BPSK :


The o/p of the DS-BPSK is transmitter is spreaded s/g because of the
modulation of the PN sequence. The PN sequence has a spreaded spectrum. Hence
DS-BPSK also has a spread spectrum because of modulation of the PN sequence.

Performance Characteristics of DSSS-BPSK:


1. Processing Gain: The processing gain of DS-SS represent as
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B.W of spread spectrum s/g


PG =
B.W of unspread s/g

From we can write as


B.W of spread spectrum s/g = 1/Tb
B.W of unspread s/g = 1/Tc
Hence,
PG = Tb/Tc

2. Probability of error:
Pe = ½(erfc√Eb/JTc)
Where,
J = Avg. interference power.
Tc =Chip duration.
Eb = Energy per bit.

3. Jamming margin:
( Jamming margin) db = (PG)db – 10log10( Eb/No)min
Where,
( Eb/No)min = Minmum bit energy to noise density ratio.
PG = processing Gain.

Procedure:
Refer to the above figure while configuring setup for the experiment.
1. Switch data switches to 1 to 0 as per your choice of binary data pattern.
2. Connect any two of the four taps viz. A, B, C, or D to the inputs of EX-OR
gate of PN Sequence generator. Connect 240 KHz clock signal on board to the
input of the PN Sequence generator.
3. Now switch on the power supply and observe the output of Binary data
generator and PN Sequence generator. Since the data generator frequency
used here is 30 KHz and that of PN Sequence generator is 240 KHz, and hence
there are 8 PN sequence bits per Data bits for spreading the binary signal.

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4. There are 2 outputs of PN Sequence generator shown on the board. One of the
outputs is for spreading the Binary data signal and the other one is for
dispreading the coded signal to recover back the original data(when BPSK
modulation is used for RF modulation of spreaded signal).
5. Connect binary data and PN sequence outputs to the Ex-OR gate of DSSS
block. Connect the output of DSSS block to the input of unipolar to bipolar
converter. Take the output of this converter to input of BPSK modulator.
Connect sinusoidal carrier from Carrier Generator to the carrier output of
BPSK modulator. This completes the modulator connections.
6. Now connect output of BPSK modulator to the input of BPSK demodulator
block. Connect output of this block to the comparator input. Here we should
receive original chipped data.
7. Connect the recovered chipped data(output of comparator) to one of the inputs
of data recovery block. Connect „PN sequence for spreading‟ output of PN
sequence generator block to the other input of data recovery gate.
8. Now turn power supply ON. Observe data and PN sequence at their respective
output pins. Press load button if data is not appearing.
9. Observe the output of DSSS block. This is called „Chipped Data‟.
10. Observe the output of BPSK modulator. This is RF modulated chipped data.
11. Observe the output of comparator and then Data Recovery Block. Adjust
phase of recovered carrier in BPSK modulator section and bias of comparator
until you see a complete replica of original binary data.
12. Change data pattern and repeat the whole procedure with this new data. Again
adjust phase and bias of comparator so as to recover the data completely.
13. Change chip (PN sequence) pattern and observe the results.

Conclusion:
1. Hence we have studied the DS-SS with BPSK transmitter and receiver
by using CDMA trainer.
2. Also we have studied the spreading and dispreading the sequence.

Oral Questions
1. Explain concept of DSSS?
2. What is the use of DSSS in CDMA technology ?
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3. Design PN sequences?
4. Compare CDMA, FDMA and CDMA
5. With the help of mathematical expression and block diagram explain direct
sequence Spread spectrum.
6. Explain its application in CDMA with relevant diagram.
7. What is PN sequence?
8. Draw a suitable PN sequence generator and prove the properties of PN sequence
and sketch autocorrelation function of same.
9. Draw a 5bit sequence generator and write the PN sequence generated by
10. Explain Spread Spectrum? What is requirement of Spread Spectrum? Classify
Spread Spectrum ?
11. What is difference between multiplexing and multiple accesses technique?
12.What is multiple accesses technique? Compare it with the relevant
diagram.(8mks)

Fig.1 DS-BPSK transmitter.

Fig.2. Receiver of the DS-SS with BPSK receiver.

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Fig.3: Code Division Multiple Access

Fig.4 O/p of transmitter signal.

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Waveforms & observation:

Fig.5 Waveforms of DS-SS with BPSK.

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Expt. No : 8 Date :
Visit to Mobile Telephone switching office

Aim : Visit Mobile Telephone switching office

Objective:
a) To visit Mobile Telephone switching office
b) To study different sections of MTSO.

Date of visit:

Study report of visit: It should be submitted by the students after the visit.

Conclusion:
1. Hence we have visited to MTSO and understand the different operations of
Mobile Telephone switching office.
2. Hence we have understand the different sections of MTSO.

Oral Questions
1. Where you have visited for MTSO?
2. What are the different sections you vested to MTSO?
3. What is the procedure you have seen at MTSO?
4. What is the practical approach of mobile phone?
5. Explain MTSO in detail?

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Expt. No: 9 Date:

Aim: To Compute and compare the median loss by employing Hata model for various
distance for carrier frequencies of 2.1 GHz and 6 GHz. Assume transmit and receive antenna
heights of 40 m and 2 m in a large city. Plot the graph of path loss vs distance.

Design & Solution:


To find :
Median loss by employing Hata model for various distances( 4km,6km,8km,10km) L = ?
Given:
fc= 2.1 GHz and 6 GHz
Case 1)
For fc= 2.1GHz
The median Path Loss for Large City employing HATA Model is given by ,
L50(dB) = 69.55 +26.16log10fc(MHz) - a(hre) – 13.82 log10(hte) + (44.9-6.55
log10(hte)log10d km
30m < hte < 200m
1m < hre < 10 m
The correction factor for a large city is given as,
a(hre) = 8.29 (log10 1.54 hre)2 – 1.1 dB , fc ≤ 300 MHz
a(hre) = 3.2(log10 11.75 hre)2 – 4.97 dB , fc ≥300 MHz

Carrier frequency fc= 2.1 GHz


fc (MHz) = 2.1 x 109 / 106 = 2100
a(hre) = 3.2(log10 11.75 hre)2 – 4.97 dB
= 1.04 dB
L50(dB) = 69.55 +26.16log102100 -1.04 – 13.82 log1040 + (44.9-6.55
log1040)log108 km
L50(dB) = 69.55 +86.90 -1.04 – 22.14 + 31.07
L50(dB) = 164.97 dB

Case 2)
For fc= 6GHz
The median Path Loss for Large City employing HATA Model is given by ,
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L50(dB) = 69.55 +26.16log10fc(MHz) - a(hre) – 13.82 log10(hte) + (44.9-6.55


log10(hte)log10d km
30m < hte < 200m
1m < hre < 10 m
The correction factor for a large city is given as,
a(hre) = 8.29 (log10 1.54 hre)2 – 1.1 dB , fc ≤ 300 MHz
a(hre) = 3.2(log10 11.75 hre)2 – 4.97 dB , fc ≥300 MHz
Carrier frequency fc= 2.1 GHz
fc (MHz) = 6 x 109 / 106 = 6000
a(hre) = 3.2(log10 11.75 hre)2 – 4.97 dB
= 1.04 dB
L50(dB) = 69.55 +26.16log106000 -1.04 – 13.82 log1040 + (44.9-6.55
log1040)log108 km
L50(dB) = 69.55 +98.83 -1.04 – 22.14 + 31.07
L50(dB) = 176.27 dB

Conclusion:
Hence we have compared the median loss by employing HATA model for various distance
for carrier frequencies of 2.1 GHz and 6 GHz.

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Expt. No: 10 Date:

Aim: Write a program to simulate experiment on BPSK modulation over AWGN to


evaluate Bit error rate

Resources required/Apparatus: PC with MATLAB

Theory: The move to digital modulation provides more information capacity, compatibility
with digital data services, higher data security, better quality communications and quicker
system availability. Developers of communication systems face following constraints:
1. Available bandwidth
2. permissible power and
3. inherent noise level of the system
The RF spectrum must be shared, yet everyday there are more users for this
spectrum as demand for communication services increases. Digital modulation
schemes have greater capacity to convey large amounts of information than analog
modulation schemes. Phase shift keying (PSK) involves the switching of the
instantaneous phase of the carrier between 2 or more levels according to the
baseband digital data. A MATLAB script to generate Binary PSK is given below :

Mat Lab Code:


clear all;
close all;
clc;
fc=1000; % Frequency for "0" bits
t=linspace(0,1/10,50);
e0=cos(2*pi*fc*t); % BPSK output for "1"
e1=-cos(2*pi*fc*t); % BPSK output for "0"
b=mod(randperm(16),2);
bnot=1-b;
%disp('Bit error rate is"');
%disp(bnot);
n=['The binary data is ',num2str(b)];
bpsk1=[];bpsk2=[];bin=[];
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for i=1:length(b)
bpsk1=[bpsk1,b(i)*e0];
bpsk2=[bpsk2,bnot(i)*e1];
bin=[bin,b(i)*ones(1,50)];
end;
bpskout=bpsk1+bpsk2;
tm=[0:length(bpsk1)-1];
plot(tm,bin,'r--');
axis([0 length(bin) 0 1.5]);
hold on;
plot(tm,bpskout,'b');
axis([0 length(tm) -1.5 1.5]);
hold off;
xlabel('Time index');
ylabel('Amplitude');
legend('Random binary','BPSK output');
title('Simulation of Binary Phase Shift keying');
gtext(n);
eb=input('Enter bit energy:');
eb=input('Enter no/2');
%eb=input('Enter bit energy:');
%eb=100;
%no=4
x=sqrt(2*eb/no);
Y = erfc(x);
disp('The bit error rate is:');
disp(Y);

Conclusion:
Hence we have simulated experiment on BPSK modulation over AWGN to
evaluate Bit error rate (BER)

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Expt. No: 11 Date:

Aim : Compute the RMS delay spread for a given Power profile and plot the graph of Power
vs Delay.

Apparatus: Octave, PC and Printer

Theory: In radio systems with low antenna heights, there are often multiple indirect paths
between the transmitter and receiver due to reflections from surrounding objects, in
addition to the direct path when there is line-of-sight. Such multipath propagation is
particularly significant in urban environments, where the sides of buildings and paved
road surfaces provide strong reflections. As a result, the received signal consists of the
summation of several components having various amplitudes, phase angles and directions
of arrival.

Figure: Illustration of reflections of various kinds

The resulting spatial variability of signal strength can be viewed as having two regimes:
a) rapid fading which varies over distances of the order of a wavelength due
primarily to changes in phase angles of different signal components.
b) slow fading which varies over larger distances due primarily to changes in
shadowing loss by surrounding objects.
In addition, the various signal components can be Doppler shifted by different amounts
due to the movement of the mobile or of reflecting objects such as vehicles.
The multipath mobile channel can be characterized in terms of its impulse response which
varies at a rate dependent on the speed of the mobile and/or the scatterers. Therefore, a

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receiver has to be able to cope with the signal distortion arising from echoes in the
channel as well as the rapid changes in the nature of this distortion. Such characteristics of
the mobile radio channel are described by the power delay profiles and the Doppler
spectra which are obtained from wideband channel sounding measurements.

Delay Spread
As an electromagnetic wave can travel from the transmitter to the receiver via
multiple paths, the signal can reach the receiver with interference from its own
echoes. Delay spread measures the effect of the time dispersion in multipath
channels. Thus, the total power received in a multipath wireless channel occurs
over a spread of time referred to as the delay spread.
Figure shows how a transmitted pulse its received at the receiver with different
signal strength as it travels through a multipath channel with different propagation
delays (τ, τ1, τ2).

Figure: Power Delay Profile Example 1

Figure: Power Delay Profile Example 2

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Detection of the first arrival peak is not an easy task to do when generating a
power delay profile. Generally, one might think that the first peak corresponds to the
maximum peak of the signal, but this is not necessarily the case. It may be that the peak of
maximum energy is not the first peak, as this can suffer greater attenuation than other
arrivals in Non-Line of Sight (NLOS) situations.
Power Delay Profile is usually supplied as a table of values obtained from
empirical data and it serves as a guidance to system design. Nevertheless, it is not an
accurate representation of the real environment in which the mobile is destined to operate
at.
Power delay profile is a good representation of the average “geometry” of the
transmitter, the receiver, and the reflectors. To quantify “how spread-out” the arriving
signals are, we use time dispersion parameters:
Excess delay: the delay with respect to the first arriving signal (𝜏)
Maximum excess delay: the excess delay of the latest arriving Multi Path
Component (MPC)
Mean excess delay: the “mean” excess delay of all arriving MPC
RMS delay spread: the “standard deviation” of the excess delay of all arriving
MPC
In a nutshell, the RMS delay spread indicates the capability of the communication
channel of supporting high data rate communications by implying the probability
of performance degradation which may occur due to the ISI because of multipath
signal propagation.
Even though a same signal is transmitted from single transmission antenna, the
signal may go through various different path. Each of the different path may cause
different travel distance if the signal get reflected by one or more obstacles (like
buildings) and in some case different path may has different physical property of
propagation media, so it is highly likely that the signal travelling through different
path would arrive at the receiver antenna at different timing. So if you send a
signal from a transmitter antenna and measure the arrival time at the receiver
antenna which is a certain distance away from the transmitter antenna, you would
get multiple different arrival timing.
If you plot those arrival timing on the axis of time, you would see a certain
variation (spread) of those values. This spread is called 'Delay Spread'.
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Power delay profile

Figure: (1) Discrete PDP (2) Continuous/exponential PDP

For continuous PDP, the RMS delay spread (τrms) can be calculated as

Knowledge of the delay spread is essential in system design for determining the
trade-off between the symbol rate of the system and the complexity of the equalizers at
the receiver. The ratio of RMS delay spread (τrms ) and symbol time duration (Tsym )
quantifies the strength of inter symbol interference (ISI). Typically, when the symbol time
period is greater than 10 times the RMS delay spread, no ISI equalizer is needed in the
receiver. The RMS delay spread obtained from the PDP must be compared with the
symbol duration to arrive at this conclusion.
With the power delay profile, one can classify a multipath channel into frequency
selective or frequency non-selective category. The derived parameter, namely, the

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maximum excess delay together with the symbol time of each transmitted symbol, can be
used to classify the channel into frequency selective or non-selective channel.
PDP can be used to estimate the average power of a multipath channel, measured
from the first signal that strikes the receiver to the last signal whose power level is above
certain threshold. This threshold is chosen based on receiver design specification and is
dependent on receiver sensitivity and noise floor at the receiver.
Maximum excess delay, also called maximum delay spread, denoted as (Tm), is
the relative time difference between the first signal component arriving at the receiver to
the last component whose power level is above some threshold. Maximum delay spread
(Tm) and the symbol time period (Tsym) can be used to classify a channel into frequency
selective or non-selective category. This classification can also be done using coherence
bandwidth (a derived parameter from spaced frequency correlation function which in turn
is the frequency domain representation of power delay profile).
A channel is classified as frequency selective, if the maximum excess delay is
greater than the symbol time period, i.e, Tm >Tsym. This introduces inter-symbol
interference into the signal that is being transmitted, thereby distorting it. This occurs
since the signal components (whose powers are above either a threshold or the maximum
excess delay), due to multipath, extend beyond the symbol time. Inter-symbol interference
can be mitigated at the receiver by an equalizer.On the other hand, if the maximum excess
delay is less than the symbol time period, i.e, Tm <Tsym , the channel is classified as
frequency non-selective or zero-mean channel. Here, all the scattered signal components
(whose powers are above either a specified threshold or the maximum excess delay) due
to the multipath, arrive at the receiver within the symbol time. This will not introduce any
ISI, but the received signal is distorted due to inherent channel effects like SNR condition.
Equalizers in the receiver are not needed.
Formulae

∫ 𝜙(𝜏) 𝜏 𝑑𝜏
𝑀𝑒𝑎𝑛 𝑑𝑒𝑙𝑎𝑦, 𝜏 = −∞
∫−∞ 𝜙(𝜏) 𝑑𝜏


∫ (𝜏)(𝜏 − 𝜏)2𝑑𝜏
𝑅𝑀𝑆 𝑑𝑒𝑙𝑎𝑦, 𝜎𝑅𝑀𝑆 = √ −∞

∫−∞ 𝜙(𝜏)𝑑𝜏
𝜏 ∞

1
𝑚𝑎𝑥𝑖𝑚𝑢𝑚 𝑠𝑦𝑚𝑏𝑜𝑙 𝑟𝑎𝑡𝑒 𝑡𝑜 𝑎𝑣𝑜𝑖𝑑 𝐼𝑆𝐼 =
10 × 𝜎𝑅𝑀𝑆

42
ATC,CHAS,AHMEDNAGAR

1
𝐹𝑜𝑟 0.9 𝑐𝑜𝑟𝑟𝑒𝑙𝑎𝑡𝑖𝑜𝑛, 𝐶𝑜ℎ𝑒𝑟𝑎𝑛𝑐𝑒 𝐵𝑊 =
50 × 𝜎𝑅𝑀𝑆
Conclusion:
Hence we have calculated the RMS delay spread for a given Power profile and plot
the graph of Power VS Delay.

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