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EC8394
TRANSMITTER
The Transmitter is a collection of one or more electronic devices and circuits which converts the original
source information in to a suitable form for propagating over the communication channel.
This conversion is achieved by modulating the carrier signal (high frequency signal), (that is, a high frequency
signal to carry a low frequency modulating signal’s characteristics) which may be an electromagnetic signal.This signal
is called as modulated signal.
CHANNEL
The transmission medium or the communication channel provides a mean of transporting of the signals from a
transmitter to a receiver. The channel may be wired or wireless. The channel can be simple as a pair of copper wires or
complex as Microwave, Satellite, or optical fiber communication systems. The channel is often referred to a specific
band of frequencies allocated to a particular service. A standard voice band channel occupies around 3KHz bandwidth
and is used for transmission of voice signals.AM broadcast channel occupies around a 10 KHz,30MHz or more
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bandwidth for High Frequency Radio channels.
In a communication system, addition of noise is unavoidable. Noise is nothing but the unwanted electrical
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signals that are interfering with the information signal. Distortion is the process of changing the shape of the signal
that may mislead the message signal instead of reaching destination. Usually the noise appearing in a communication
system is additive in nature. This means, If added at some point the same can be subtracted at some other point by
suitable technique.
RECEIVER
A Receiver is a collection of electronic devices and circuits that accepts the transmitted signals from the
transmission medium and then reproduces the message signal in electrical form. This process is known as
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demodulation or detection.
INFORMATION SINK/DESTINATION
Destination is the last stage, which is used to convert an electrical message signal into its original form.
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1.1.4 TYPES OF COMMUNICATION SYSTEM:

The transmitting system at source end, the receiving system at destination end and the channel between
these two systems through which the information is transferred comprises a communication system. Depending on
the type of information being transmitted the communication system can be of two types. They are
 Analog Communication system
 Digital Communication system
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If in the communication channel, the signals which could continuously vary with respect to time (analog
signals) are transmitted, then such a system is called Analog communication system.
If in the communication channel, the signals which are discontinuous and discrete in nature (Digital signals)
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are transmitted, then such a system is called Digital communication system. In practice, the analog information is
converted in to digital information using A/D convertors at Transmitting end and is sent through digital
communication channels. At the receiving end the received digital signals are converted back into analog form using
D/A convertors.
1.1.5 DIFFERENCE BETWEEN ANALOG AND DIGITAL COMMUNICATION SYSTEM
ANALOG COMMUNICATION SYSTEMS DIGITAL COMMUNICATION SYSTEMS
No A/D or D/A convertors required. Transmitted Requires A/D at Transmitter and a D/A at
as analog signals Receiver
Required Bandwidth & cost of components is In terms of QOS, less prone to noise and is
less. Most preferred for Short haul preferred for Long haul communication
communication
Regenerative repeaters are used at intermediate Regenerative repeaters are used at intermediate
stages to reduce signal attenuation stages to retransmit the binary information
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1.2 NOISE
In all communication process, electrical disturbance are appearing as interference which is called as NOISE.
Noise is an unwanted signal that corrupts a desired message signal. In general noise may be predictable or Random in
nature.
Predictable Noise
This kind of noise can be estimated and eliminated by proper engineering design. For Eg. Power supply hums,
Ignition radiation Pickup, spurious oscillations in feedback systems etc. The predictable noise is generally man-made
and it can be eliminated easily.
Unpredictable Noise
It varies randomly with time, and we have no control over this noise. In the Absence of noise, identification of
message signal at the receiver is perfect. Presence of Random noise complicated the communication system. The
amount of noise power present in the received signal decides the minimum power level of desired message signal at
the receiver. In general the term noise is used to represent Random Noise.
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1.2.1 SOURCES OF NOISE
The sources of noise are broadly classified as follows:
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Noise
External Noise Internal Noise
Erratic Natural Man-Made Shot Noise Thermal Noise
Disturbances Noise
1.2.2EXTERNAL NOISE
The external noise is the one which is created outside the communication circuit. This happens because of
natural disturbances or they are man-made.
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Erratic Natural Disturbances
This kind of noise occurs randomly. It is caused by lightening, electrical storms, and other atmospheric
disturbances. This noise is unpredictable in nature. This is also known as atmospheric noise (or) Static noise.
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This kind of noise is less severe above 30 MHz .They are also called Extra terrestrial Noise.
Man-Made Noise
This noise occurs due to undesired pickups from electrical appliances, such as motors, switch gears,
automobile and aircraft ignitions etc. This kind of noise is under human control and can be eliminated by removing the
sources of noise. This noise has an effective range of 1MHz to 500MHz.
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1.2.3 INTERNAL NOISE

Internal noise is created by active and passive components within the communication system. This noise is
also known as Fluctuation noise. It is caused by spontaneous fluctuations in the physical system. The two important
Fluctuation noise are shot noise and Thermal noise.
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Shot Noise
Shot noise is present in active devices due to the random behavior of charge carriers. In electron tubes, shot
noise are generated due to the random emission of electrons from cathodes. In semiconductor, shot noise occurs due
to the random diffusion of minority carriers (or) random generation and recombination of electron-hole pairs.
In electron devices, current flows in the form of discrete pulses, every time a charge carrier moves from one
place to another. Hence although the current flows in continuous manner, it is still be regarded as a discrete
phenomenon.
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The nature of current variation with time is shown below.
The current fluctuates about a mean value I0. The current in(t) which wiggles around the mean value is called
Shot noise. The total current i(t) is expressed as,
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i(t) =I0+in(t)
Where, I0= Constant
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In(t) = fluctuating current
The current In(t) is random in nature. It is specified by its power spectral density. The number of electrons
contributing to the random stationary current is large. By assuming there is no interaction among moving electrons,
and by applying central limit theorem, the electron movement process follows a Gaussian distribution.
The power spectral density of statistically independent non interacting random noise current In(t) is given as,
si(ω) =qI0Where, q= electronic charge
R I0= Mean value of current in Amperes
The power spectral density function is frequency independent. This property is valid for a certain range of
frequencies, which is decided by the transit time of an electron to move from anode to cathode.
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For all practical purposes, the si(ω) is considered as the frequency independent below 100MHz.
The mean square value for the randomly fluctuating noise current will be
in2 = 2qI0(Δf)
Where, 2Δf is the bandwidth of measuring system. This formula is known as Schottky formula.
If the electron movement is considered with interaction among moving electrons, in such cases the power spectral
density is given by
si(ω) = α qI0
Where, α= smoothing constant (0.01< α<1)
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THERMAL NOISE
Thermal Noise is an internal noise which occurs due to the thermal agitation of electrons inside electron
devices.This type of noise arises due to random motion of electrons in a conducting medium such as a resistor, and
this motion in turn is randomized through collisions caused by imperfection in the structure of conductors. The net
effect of motion of all electrons constitutes an electronic current flowing through the resistor, causing the noise. This
noise is also known as Resistor Noise and Johnson noise.
The Power spectral density of thermal noise is given by
si(ω) = 2KTG / *1 + (ω/α)2]
Where
T = Ambient temperature (Kelvin)
G = conductance of Resistive elements in mhos.
K = Boltzmann constant
α = average number of collisions per sec per electron.
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Since |ω/α| is always very small.
Therefore,
si(ω) = 2KTG
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EQUIVALENT CIRCUIT OF A NOISY RESISTOR

A Noisy Resistor R can be represented by noiseless conductance G in parallel with a noisy current source of in(t)as
shown below. The Thevenin equivalent of the same is also shown.
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Now, According to Ohms Law,

Vn(t) =In(t) x R
In terms of Power Spectral Density
Sv(ω) = R2 Si(ω)
Since the power spectral density is a function of square of voltage or the current, the relation between power spectral
density Si(ω) of current source is written in terms of power spectral density of voltage as given above
Hence Sv(ω) = R2 (2KTG)
= R2 (2KT(1/R))
Sv(ω) = 2KTR
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The noise power Pnis given by
Pn = 1/2π∫
Pn = 1/2π∫
= 2KTR/2π ** ω]-
=
This integral becomes infinite over an infinite bandwidth.
For a finite bandwidth of 2Δf. The noise power is given by

Pn=Sv(ω)2Δf
Substituting the value ofSv(ω),
Pn=2KTR2Δf
=4KTRΔf
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Since the power of a signal is same as its mean square value,
Pn=Vn2
Therefore
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Vn=√
FOR SERIAL AND PARALLEL COMBINATION OF SEVERAL RESISTORS

If Resistors are in series,
RS=R1+R2+R3+……..
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Vn=√
If Resistors are in Parallel,
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RP=* +
Vn=√
1.2.4PARTITION NOISE
Partition Noise is generated whenever a current has to divide between two or more branches in a circuit and
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results from random fluctuation in the division. It would be expected therefore that a diode would be less noisy than a
BJT, if the third terminal draws current.
For this Reason, the input stage of Microwave receiver is often a diode circuit rather than a transistor, for low
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frequency amplification. Partition noise has a flat spectrum.

1.2.5 FLICKER NOISE
Flicker Noise Occurs due to the imperfections present in cathode tubes and on the surface of semiconductors,
especially around the junctions. This kind of noise is inversely proportional to operating frequency.
1.2.6 BURST NOISE
It is observed in low frequency ranges, particularly in bipolar Transistors. This happens when the transistor is
operated in a non-linear region for a linear operation. Due to this noise, popping sounds are introduced and hence it is
also known as Popcorn noise.
1.2.7 AVALANCHE NOISE
This noise occurs when the diode is operated in reverse biased mode. Due to the extensive increase of reverse
current for a slight variation in the magnitude of reverse voltage, In depletion region holes and electrons are ionized
by collision, resulting in a large spikes.
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1.3 WHITE NOISE
White Noise is one type of noise, but it is not a source of noise. The noise which has Gaussian distribution and
has a flat power spectral density over wide range of frequencies is called White noise.
When the probability of occurrence of a white noise level is specified by a Gaussian distribution function, it is
known as Gaussian noise or White Gaussian Noise.
The White Gaussian noise contains all frequency components in equal amount. The Power Spectral Density of
WGN is given by,
Sw(ω) =
Where, N0=K Te
Te= Equivalent noise temperature
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1.4 NOISE CALCULATIONS
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The following are the expressions used to calculate various noises involved in a communication channel.
1. Shot noise
Output current of any semiconductor diode : i(t) =I0+in(t)
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Mean square shot noise current of the diode junction (squared) : Is2 = 2 IdcqBn
Bn= Noise equivalent of Bandwidth
Idc= DC current in Amperes
2. Thermal Noise Power
Pn=4KTRΔf =2KTR(Bn) Since 2Δf=Bn
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Volt equivalent of Thermal Noise: Vn=√ Volts

Thermal Noise Power Density: No= kT
Thermal Noise power of a noisy transistor: Pn= KTBn Watts
Thermal Noise power of a noisy transistor in dbW: Pn(dbW) = 10 log(k) +10 log T+ 10 log Bn dbW
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3. Superposition of Noises:
“For a system involves multiple independent sources of noise, the power density spectrum is equal to the sum
of individual power density spectrum evaluated individual sources at a time.”
Sy1+y2+y3+….(ω) = Sy1(ω) + Sy2(ω) + Sy3(ω) +…..
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1.5 MODULATION
Modulation is defined as the process by which some parameters (Amplitude,Frequency and Phase) of a high
frequency carrier signal is varied in accordance with the amplitude of message signal.
Any signal can be represented by its three parameters namely Amplitude, Frequency and Phase. If any of
these parameters of a signal are changed with respect to the amplitude of a relatively lower frequency signal, one
such a process is called modulation.
The circuit which performs this process is called Transmitter. The original information signal with which the
other signal’s parameters are changed is known as modulating signal or message signal. Message signal always assume
a lower frequency than carrier.
The high frequency signal whose parameters are changed is known as Carrier. The resulting signal of
modulation process is called modulated signal or modulated carrier.
In a communication system, if the message signal is transmitted to the channel without performing a
modulation, such a transmission is called Baseband transmission. In the other hand, if the transmission of a signal is
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followed after a modulation process, such system are said to be Broadband transmission.
Demodulation is the reverse process of modulation. If the original, Low frequency message signal is recovered
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from the modulated carrier signal, then it is called Demodulation/Detection. It is a technique followed at receiver by a
circuit called Demodulator/Detector.
1.5.1 NEED FOR MODULATION
If the Low frequency, Low amplitude message signals are transmitted for a long distance through a noisy
transmission medium it will not reach the destination properly. Due to Noise and Interference present, the signal may
fade or getting disturbed.
In order to avail successful long haul communication and faithful reception, Modulation by a high frequency
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carrier at transmitter and Demodulation at receiver is required.
The other needs for modulation are listed below.
1. Radiation antenna height – If modulated, antenna with lesser heights is enough
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2. Long haul communication – distance between Transmitter and Receiver can be increased relatively if
modulated
3. Ease of Radiation- After Modulation, the message signals are translated to a higher frequency, hence it is
easy to construct radiation circuits such as amplifiers and matching networks.
4. Multiplexing- Transmission of two or more signals over the same channel is called multiplexing. Due to
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modulation, the number of channels required is reduced.

5. Improve the quality of Reception-Noise is highly limited by performing modulation, which in turn
improves the quality of Reception
6. Avoids mixing of signals-If more than one message signal has to be transmitted, each low frequency
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message signal can be modulated with different carrier frequencies and hence they can be allotted to
various slots in frequency domain, which essentially avoids mixing of closely spaced message signals.
1.5.2 TYPES OF MODULATION
Based on the type of carrier and the type of message signal used, the classification of Modulation is done.
In General Modulation is classified in to two categories. They are
1. Analog Modulation
2. Digital modulation
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The broad classificationof Modulation techniques is given below.
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NOTE:
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In Continuous wave modulation such as AM, FM and PM, Carrier : Analog signal (High Frequency)
Message : Analog signal (Low frequency)
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In Pulse Analog modulation such as PAM, PWM and PPM, Carrier : Digital Pulse signal (High Frequency)
Message : Analog signal (Low frequency)
In Pulse Transmission methods such as PCM,DPCM, DM and ADM,

Carrier : Digital Pulse signal (High Frequency)
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Message : Digital code signal (Low frequency)
In Shift Keying modulation such as ASK,FSK and PSK Carrier : Analog signal (High Frequency)
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Message : Digital Pulse signal (Low frequency)
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1.6 AMPLITUDE MODULATION
Amplitude modulation is the process by which amplitude of the high frequency analog carrier signal is varied
in accordance with the instantaneous amplitude values of the low frequency analog message signal.AM is a relatively
inexpensive, low quality form of modulation that is used for commercial broadcasting of both audio and video signals.
1.6.1 MATHEMATICAL REPRESENTATION OF AN AM WAVE
Let the modulating signal Vm(t) = VmSinωmt
Carrier signal Vc(t) =VC Sinωct
Where, Vm Instantaneous value of message signal
VC  Instantaneous value of message signal
The Instantaneous amplitude of the carrier signal is changed after modulation
VAM =VC +Vm(t)
= VC +VmSinωmt
= VC[1+(Vm/VC)Sinωmt]
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VAM = VC[1+ ma Sinωct] ma=Modulation index or Depth of modulation
Hence , the expression for representing AM equation is
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VAM(t) = VC[1+ maSinωmt]Sinωct
or
VAM(t) = VC[1+ ma Sin2πfct]Sin2πfct
This expression represents the time domain representation of an AM wave.
1.6.2 ENVELOPE OF AN AM WAVE
AM is simply called as a Double Side Band Full Carrier or Conventional AM or Simply AM.
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The shape of modulated wave (AM) is called AM envelope which contains all the frequencies and is used to
transfer the information through the system. It is called envelope, because it contains all frequency components
including Carrier term, USB and LSB frequencies.
o If the instantaneous amplitude of message signal increases, the amplitude of carrier is also increased.
o If the instantaneous amplitude of message signal decreases, the amplitude of carrier is also decreased.
Without Modulating signal, the AM output is simple the carrier signal. The repetition rate of the envelope is
equal to the frequency of modulating message signal. The shape of the envelope is similar to that of modulating signal.
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Hence after modulation, the resulting wave form contains the amplitude information from the message signal
and possessing the frequency as that of the carrier.
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1.6.3 AM FREQUENCY SPECTRUM AND BANDWIDTH
An AM modulator is a non-linear device. Therefore, non- linear mixing occurs and the output envelope is a
complex wave made up of a dc voltage, the carrier frequency and the sum (fc+ fm) and difference (fc- fm) frequencies.
Frequency Spectrum of AM:
The AM wave is given by
VAM(t) = VC [1+ maSinωmt]Sinωct
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= VCSinωct+VC maSinωmtSinωct
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Since ,SinωmtSinωct=
VAM(t) = VCSinωct+maVc
VAM(t) = VCSinωct+ -
This equation contains 3 radicals,
VCSinωct Carrier
Lower Side Band ; (-)  Upper Side Band
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 The negative sign included with USB term represents a phase shift of 180°.
 The first term is nothing else but the unmodulated carrier signal.
 The second term represents sinusoidal signal at frequency (fc-fm).It is also called LSB. Its amplitude is .
 The third term represents sinusoidal signal at frequency (fc+fm).It is also called USB. Its amplitude is .
1.6.4 SIDEBANDS OF AM
Whenever a carrier is modulated by an information signal, new signals at different frequencies are generated
as part of processes. These new frequencies are called side frequencies or side bands. The sidebands are occurs in the
frequency spectrum directly above and below the carrier frequency. Assuming a carrier frequency and a modulating
frequency of fm, the upper side band fUSB and Lower side band f LSB are determined as follows.
fUSB= fc+fm
f LSB= fc-fm
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1.6.5 FREQUENCY DOMAIN REPRESENTATION OF AM
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1.6.6 BANDWIDTH OF AM
The bandwidth of AM signal is given by the subtraction of highest and lowest frequency component in the
entire frequency spectrum.
BW = fUSB-f LSB
=fc+fm–( fc-fm)
BW =2fm
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From this it is evident that for AM-DSBFC system, the Bandwidth is twice that of the maximum frequency component
present in the message signal.
1.6.7 PHASOR REPRESENTATION OF AM
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The amplitude variation in an AM system can be explained with the help of a phasor diagram. The Phasor of
USB is leading the resultant (in anti-clockwise) by an angle ωm. The Phasor of LSB is lagging the resultant (in clockwise)
by an angle ωm. The resulting amplitude of the modulated wave at any instant is the vector sum of the two side band
phasor. Vc is carrier wave phasor, taken as reference phasor and the resulting phasor is VAM(t). The phasor for the
carrier and the upper and lower side frequencies combine, sometimes in phase and sometimes out of phase.
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1.6.8 MODULATION INDEX (or)DEGREE OF MODULATION
Modulation index or Degree of modulation or co-efficient of modulation is a parameter which represents the
extent of amplitude variation in AM about an unmodulated carrier amplitude.It is defined as the ratio of maximum
amplitude of modulating signal to the maximum amplitude of carrier signal.
Modulation Index (ma) =
For Distortion, not to occur in an AM modulation system, Value of Vm should always lesser than Vc. Hence the
maximum value of modulation index should be 1. The modulation index should be a number between 0 and 1.
THREE DEGREES OF MODULATION:
Case(i) Vm<Vc
If Vm<Vc, ma<1, such type of modulation is called Under Modulation. Here the envelope of amplitude modulated
signals does not reach the zero amplitude axis. Hence the message signal is fully preserved in the envelope of AM
wave. An AM envelope detector can recover the message signal without any distortion.
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Case(ii) Vm= Vc
If Vm= Vc, ma=1, such a type of modulation is called Critical Modulation. Here the envelope of modulated signal just
reaches the zero amplitude axis. The message signal remains preserved. An AM envelope detector can recover the
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message signal without any distortion.
Case(iii) Vm>Vc
If Vm>Vc, ma>1, such a type of modulation is called Over Modulation. Here the envelope of modulated signal crosses
the zero amplitude axis and the positive and negative extents of message signal are clipped out. The message signal is
not preserved inside the envelope. An AM envelope detector cannot recover the message signal without any
distortion.
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50% MODULATION UNDER MODULATION 100% MODULATION CRITICAL MODULATION 150% MODULATIONOVER MODULATION
PERCENTAGE MODULATION
When modulation index is expressed as percentage, it is known as percentage modulation. It gives the
percentage change in the amplitude of Modulated carrier, when the carrier is acted on by a modulating message
signal.
% Modulation = x 100%
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MODULATION INDEX AFTER MODULATION
There is one more method for calculating Modulation index. After modulation, if the envelope is observed, the
swing of the envelope is between Vmax and Vmin.
Where
Vmax =Vc+ Vm
Vmin= Vc- Vm
From the Envelope diagram,
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From this,
Then modulation index is given by,

m a=
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%ma= x 100%
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This expressions can be used for calculating the modulation index, when the Vmax and Vmin values are known after
modulation.
1.6.9 AM VOLTAGE DISTRIBUTION
We know that, the AM envelope equation is
VAM(t) = VCSinωct+
R -
From this the voltage distribution in modulated wave can be expressed.
In an AM envelope, there are three segments Viz.
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1. No modulation region or Carrier
2. USB
3. LSB
Voltage distribution of these three segments can be tabulated as follows
ENVELOPE SEGMENT VOLTAGE
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No modulation region VC
USB
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LSB
1.6.10 AM POWER DISTRIBUTION

An AM wave consists of carrier and two side bands. The carrier component of the modulated wave has the
same amplitude as that of the unmodulated carrier. The modulated wave contains extra power in two side band
components.
The amplitude of the sidebands depends on the modulation index. Hence the total power in the AM depends
on ma.
The total power in AM is given by
Pt= Pc+PUSB+PLSB
=
Where all three voltages are in rms values and R is the antenna Resistance.
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Carrier Power Pc:
Pc=
( )
√
=
Pc =
Power in Side bands:
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√
PUSB=PLSB=
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=
= R
But Pc
PUSB=PLSB=
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TOTAL POWER IN AM:

Pt= Pc+PUSB+PLSB
= + +
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=Pc* +
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Pt = Pc * +
This is called the Transmission power of AM. The AM power spectrum is shown below.
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1.6.11 TRANSMISSION EFFICIENCY OF AM:
The amount of useful information present in AM after modulation is expressed as transmission
efficiency. It is the ratio of the transmitted power which contains the information to the total transmitted
power.
Transmission efficiency (η) % = x 100%
= x 100%
= x 100%
= X100%
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(η) % = X100%
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From this, it is evident, if ma= 1, only 33% efficient, which means only one third of the total power is
confined in side bands and remaining two third of power is wasted in carrier. Hence AM is recommended
only for low cost, low quality reception systems.
1.7 OTHER FORMS OF AM
The AM scheme discussed before is a simple AM technique, in which the useful message signal is
modulated on to a carrier possessing 3 spectral components namely Carrier, USB and LSB.
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But it is evident from the analysis that only USB and LSB contain useful information and carrier
component contributes for wasteful of power.
Hence to improve the efficiency, some other forms of AM are developed they are
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1. DSB-SC AM Double Side Band –Suppressed Carrier AM
2. SSB-SC AM Single Side Band –Suppressed Carrier AM
3. VSB-AM Vestigial Side Band AM
1.7.1 DSB-SC AM
In this method it contains only LSB and USB components and carrier term is suppressed, resulting in
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Transmission bandwidth becomes twice that of the frequency of message signal.

Let,
Vm(t) = Vm sin ωmt&Vc(t) = Vc sin ωct
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When multiplying these signals, it results in DSB-SC AM.
VDSB-SC(t) =Vm(t) .Vc(t)
= Vm sin ωmt .Vc sin ωct
= VmVc [sin ωmt .sin ωct]
= VmVc[ ]
= [ ]
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Observing the above equation, it is trivial that there is No Carrier [No modulation term] term in the envelope
and possesses only the Side band terms [USB and LSB]. Hence this modulation is called Double Side Band –Suppressed
Carrier Amplitude Modulation [DSB-SC AM]. Since there is no Wasteful carrier present inside the Modulated envelope,
the transmission power of DSB-SC AM is lesser than that of DSB-FC AM.
FREQUENCY SPECTRUM OF DSB-SC AM
The Frequency domain spectral representation of DSB-SC AM is given below.
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BANDWIDTH OF DSB-SC AM
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BANDWIDTH = USB –LSB
=
BW =  which is similar to the bandwidth of DSB-FC.
PHASOR REPRESENTATION
The Phasor representation of the DSB-SC AM is given below
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1.7.2 SSB-SC AM
In AM DSB-FC, Both Transmission power and Transmission Bandwidth are wasted. To avoid this, DSB-SC
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scheme is introduced.
But in DSB-SC also only the transmission power is reduced, but not the transmission bandwidth. To increase
the efficiency and to further increase the saving of power, one more sideband is eliminated along with the carrier, and
such a technique is called SSB-SC technique.
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Elimination of one side band will not result in loss of message signal information, since LSB and USB are
uniquely symmetry with respect to carrier frequency.Usually USB term is suppressed along with carrier in SSB
techniques, as the LSB [Low frequency components] contains much of the message information than USB.
1.7.3 METHOD OF GENERATING SSB-SC AM
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The above schematic represents the method of generating SSB-AM with suppressed carrier. In this
the message signal [Vm(t)] split in to quadrature [V1(t)] and in-phase signals [V2(t)].
In-phase signal  0° phase shift with Actual message signal
Quadrature signal  90° phase shift with Actual message signal
Likewise the carrier signal [Vc( t)] is also split in to quadrature and in-phase signals. Thereon the in-
phase message signal is amplitude modulated using the in-phase carrier signal to produce [VAM1(t)]. And the
90°out of phase message signal is amplitude modulated using the quadrature carrier to produce[VAM2(t)].
Both the modulated AM waves are summed up linearly to produce a Single side band AM wave with
suppressed carrier [VSSB(t)].
During modulation, the SSB-SC AM envelope is formed as follows
[VAM1(t)] = Vm sin (ωmt+90°) .Vc sin (ωct+90°)
= Vmcos (ωmt) .Vccos (ωct)  Quadrature Carrier with Quadrature Message signal
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&
[VAM2(t)] = Vm sin (ωmt) .Vc sin (ωct)  In-phase Carrier with In-phase Message signal
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Now,
[VSSB(t)] = [VAM1(t)] +[VAM2(t)]
= Vmcos (ωmt) .Vccos (ωct) +Vm sin (ωmt) .Vc sin (ωct)
= VmVc [cos (ωmt) .cos (ωct) + sin (ωmt) .sin (ωct) ]

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Since cos = [cos A . cos B + sin A . sin B ]
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[VSSB(t)] = cos t
1.7.4 FREQUENCY SPECTRUM OF SSB-SC AM
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1.7.5 PHASOR REPRESENTATION OF SSB-SC AM
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1.7.6 ADVANTAGES ,DISADVANTAGES AND APPLICATIONS OF SSB-SC AM
ADVANTAGES:
 BW of SSB-SC AM = ½ BW of DSB-SC AM.
Therefore the twice No.of channels can be accommodated at a given frequency spectrum.
 Carrier is suppressed. Therefore the interference among several carriers is avoided.
 Improved SNR (9- 12 dB larger than the SNR of DSB-AM).
 Less Fading
 Cheaper than DSB-AM
DISADVANTAGES:
 Transmission and Reception becomes more complex. So not used for commercial broadcasting.
 To perform demodulation, Local generation of the same carrier is required (or) Pilot carrier method is used.
Demodulation is complex.
APPLICATIONS:
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 Police Wireless Systems
 SSB-Telegraphy
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 VHF-UHF radio communication
 Point to point radio links
1.7.7 VESTIGAL SIDE BAND AM (VSB AM)
Suppose if the message signal has a very large bandwidth, for Eg. Picture signal transmission in TV
communication and having very low frequency components, these very low frequency components give rise to
sidebands, very close to carrier frequency in DSB-AM or SSB-AM.
That is, if message signal frequency is very low, then ( or ( ≈
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Therefore it is difficult to suppress the carrier (or) one side band completely, if is very low. If suppressed
then phase distortion will occur at these frequencies. Hence a part of LSB is suppressed. Therefore the Transmitted
signal contains a full USB, a carrier component and Vestige of LSB (partially suppressed LSB).
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 Vestige Part of interest

VSB-AM is useful in TV signal transmission.
1.8 EVOLUTION AND DESCRIPTION OF SSB SYSTEMS
Single-sideband modulation (SSB) or single-sideband suppressed-carrier (SSB-SC) is a refinement of amplitude
modulation that more efficiently uses transmitter power and bandwidth. Amplitude modulation produces an output
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signal that has twice the bandwidth of the original baseband signal.
Single-sideband modulation avoids this bandwidth doubling, and the power wasted on a carrier, at the cost of
increased device complexity and more difficult tuning at the receiver. One method of producing an SSB signal is to
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remove one of the sidebands via filtering, leaving only either the upper sideband (USB), the sideband with the higher
frequency, or less commonly the lower sideband (LSB), the sideband with the lower frequency.
Most often, the carrier is reduced or removed entirely (suppressed), being referred to in full as single
sideband suppressed carrier (SSBSC). Assuming both sidebands are symmetric, which is the case for a normal AM
signal, no information is lost in the process.
Since the modulation is concentrated in a single sideband, the effective power output is greater than in
normal AM (the carrier and redundant sideband account for well over half of the power output of an AM transmitter).
Though SSB uses substantially less bandwidth and power, it cannot be demodulated by a simple envelope detector like
standard AM.
In conventional (double sideband) DSBFC-AM the carrier signal is considered to represent a waste of
transmission power and therefore SSB transmission generally attempts to reduce the (amplitude) level of the carrier
signal to as close as possible to zero. This is known as suppressed carrier SSB (SSB-SC).
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However to ensure that the transmitted AM signal is resolved at the correct pitch it is essential that the
receiver is tuned to exactly the same frequency as the transmitter is operating on and it is often difficult to achieve the
necessary level of precision.
As a result, The AM signals transmitted by SSB suffer from poor clarity. To overcome this problem a small
amount of carrier signal is transmitted in order that receivers with the necessary circuitry can synchronize with the
transmitted signal. This transmission mode is known as single sideband reduced (or pilot) carrier.
1.9 ADVANTAGES DISADVANTAGES AND APPLICATIONS OF AM
ADVANTAGES:
 Simple and Robust of all modulation systems
 Cheap and inexpensive
 Am wave can travel over long distances.
DISADVANTAGES:
 Poor Noise performance [Much affected by Noise. No methods for noise rejection]
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 Low quality form modulation
 Low efficiency
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 Requires Bandwidth
APPLICATIONS:
 Picture signal transmission in TV communication
 Commercial broadcasting of Audio and video signals
 2-Way mobile radio communication systems such as CB-Radio
 Aircraft communications @ VHF range  30 to 300 MHz
1.10 EXPONENTIAL MODULATION OR ANGLE MODULATION
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It is the process by which the angle [frequency or phase] of a carrier signal is changed in accordance with the
instantaneous amplitude of the modulating message signal. Amplitude of the carrier remains unchanged in these
modulations.
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FEATURES OF ANGLE MODULATION SYSTEMS

 Noise Reduction
 Improved system fidelity
 More efficient use of Power
TYPES
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1. Frequency Modulation - FM
2. Phase Modulation - PM
Direct Frequency Modulation [FM]
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The process of varying the frequency of a carrier, in accordance to the instantaneous amplitude of message
signal at a rate equal to the frequency of the message signal. Amplitude of the carrier signal remains unaltered after
modulation.
Indirect Frequency Modulation or Direct Phase Modulation[PM]
The process of varying the phase angle of a carrier, in accordance to the instantaneous amplitude of message
signal at a rate equal to the frequency of the message signal. Amplitude of the carrier signal remains unaltered after
modulation.
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Note:
In general, a signal is represented as
a(t) = A sin (ωt+θ)
where, A  Instantaneous amplitude
ω  Frequency
θ  Phase
Let
Vx(t) = Vc sin (ωct+θ(t))
= Vc sin (φ(t))
Where,
φ(t) =ωct+θ(t)  Angle of the signal
Vx(t)  Angle modulated wave
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Vc(t)  Maximum amplitude of carrier
ωc  Angular frequency of carrier
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θ(t)  Phase of the signal
INSTANTANEOUS PHASE ANGLE (φ)
Phase of the carrier at any instant of time.
INSTANTANEOUS PHASE DEVIATION(θ(t))
The Change in phase of carrier at any instant of time with respect to its reference phase.
INSTANTANEOUS FREQUENCY ωi(t)
The frequency of the carrier at any instant of time.
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ωi(t) =
( )
=
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=ωc+θ’(t)
INSTANTANEOUS FREQUENCY DEVIATION θ’(t)
The change in frequency of the carrier.First derivative with respect to time of the instantaneous phase
deviation.
DEVIATION SENSITIVITY [K]
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The relationship between the output parameter change and the input parameter change is given by Deviation
sensitivity. It expresses the amount of change in output frequency or output phase for a small change in input
amplitude.
Hence For FM:
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KFM =
KFMVm =Δω
For PM:
KPM =
KPMVm =Δθ
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1.11.EXPRESSION FOR FREQUENCY MODULATION
Let message signal be
Vm(t) = Vmcosωmt
& Let Carrier Signal be
Vc(t) = Vc sin [ωct +θ]
Assume that, φ(t) = ωct+θ(t)

Then ,
Vc(t) = Vc sin [φ(t)]
Differentiating,
(Vc(t) ) = [ Vc sin [φ(t)]]
= [Vc sin [ωct +θ]]
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Since it is known that,
(φ(t)) = φ’(t)
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During FM, the frequency of carrier signal is changed in accordance to the instantaneous value of amplitude of
message signal.
That is, θ changes with respect to the changes in Vm
Therefore the frequency of carrier after modulation is given by
ωi = ωc+ KFMVm(t)
ωi = ωc+ KFMVmcosωmt
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To find the instantaneous phase angle of modulated signal, integrate the above equation.
φi=∫
=∫
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φi = + +Constant
This constant involved with integration can be neglected since it has no role to play in modulation.
According to the definition of FM,
VFM(t) = Vc sinφi
= Vc sin [ + ]
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VFM(t) = Vc sin [ + ]
Where m f=  Modulation index of FM
It is known that,
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ωi = ωc+ KFMVmcosωmt
MAXIMUM AND MINIMUM FREQUENCIES IN FM MODULATED WAVE
The maximum and minimum values of any cosine function is
Max(ωi) = ωc+ KFMVm{ if cosωmt =+1}
Min(ωi) = ωc-KFMVm { if cosωmt =+1}

1.12 EXPRESSION FOR PHASE MODULATION
Let message signal be
Vm(t) = Vmcosωmt
& Let Carrier Signal be
Vc(t) = Vc sin [ωct +θ]
Assume that, φ(t) = ωct+θ(t)
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Then, According to the definition of PM, the phase angle of the carrier changes with respect to the changes in
Amplitude of message signal.
 That is, θ changes with respect to the changes in Vm
After Phase Modulation,

VPM(t) = Vc sin [ + ]
Where, Modulation index of Phase Modulation.
1.13 FREQUENCY DEVIATION AND PHASE DEVIATION
In case of FM,
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m f=
Or mf  frequency deviation
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 If is +ve In FM wave frequency isgreater than the carrier wave
 If is -ve In FM wave frequency is lesser than the carrier wave
In case of PM,
mp=
Phase deviation
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 If is +ve Phase shift of PM wave is greater than that of the carrier wave
 If is -ve  Phase shift of PM wave is lesser than that of the carrier wave
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1.14 WAVE FORMS OF FM & PM
FM WAVEFORMS:
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PM WAVFORMS:
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FM WAVEFORM
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PM WAVEFORM
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1.15 SPECTRUM OF ANGLE MODULATED WAVES
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Frequency modulation can be classified as narrowband if the change in the carrier frequency is about the
same as the signal frequency, or as wideband if the change in the carrier frequency is much higher (modulation index
>1) than the signal frequency.
If is less for a FM system, such FM is called Narrowband FM
If is large for a FM system, such FM is called Wideband FM
For the case of a carrier, frequency modulated by a single sine wave, the resulting frequency spectrum can be
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calculated using Bessel functions of the first kind, as a function of the sideband number and the modulation index. The
carrier and sideband amplitudes are illustrated for different modulation indices of FM signals. For particular values of
the modulation index, the carrier amplitude becomes zero and all the signal power is in the sidebands.
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It is established that,
VFM(t) = Vc sin [ + ]
=Vc [sin . cos + cos . sin
Since,sin (A+B) = sin A cos B +cos A sin B
The above equation can be simplified using Bessel function of first kind.
VFM(t) = Vc{Jo(mf)sin }
+ VcJ1(mf)[sin }
+ VC{J2(mf)[sin
+ VC{J3(mf)[sin + …………..and so on.
Where, J is the Bessel function coefficient.
is the modulation index
( ) ( )
Jn(mf) =( ) [ ]
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From the Bessel function table, with several values of modulation index, if m=0 produces no side frequencies,
if m>value, more set of side frequencies and if m<1, higher order side frequencies become insignificant.All side bands
are symmetric around the carrier. As m increases the number of significant sidebands also increases. Hence it is
evident that the bandwidth of angle modulated signals is a function of modulation index.
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1.16 BANDWIDTH REQUIREMENTS OF ANGLE MODULATED WAVES
Theoretically the Bandwidth of FM &PM waves are infinite. But in practice the bandwidth is calculated by the
number of sub bands having significant amplitude. Bandwidth of FM and PM depends on their modulation index.
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If m<1 low index angle modulation
If m=1 medium index angle modulation
If m>1 high index angle modulation
In case of FM,
(1) Low index corresponds to Narrowband FM, whose bandwidth can be calculated as
BW= =
R As =
BW=2
(2) High index corresponds to Wideband FM, whose bandwidth can be calculated as
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BW=2
(3) The maximum possible Bandwidth for a FM system is determined by
BW=2n
n Number of sidebands having significant measurable amplitude
(4) The Approximate Transmission Bandwidth necessary for faithful transmission of angle modulated waves
can be found using Carson’s rule (Thumb rule).“The Necessary Bandwidth required for the transmission of angle
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modulated waves is twice the sum of peak frequency deviation and highest frequency component in the message
signal.”
BW=2
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=2
=2 ( )
BW = 2 Hz
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1.17 PREEMPHASIS AND DE-EMPHASIS OF EXPONENTIAL MODULATED WAVES
In case of AM, the modulation itself takes care of the improvement of power level of the transmission signal.
That is by merely performing the modulation, the signal power can be improved in AM systems. But in angle
modulation systems there is no guarantee for the power level improvement in transmission signal by modulation.
Hence in angle modulated systems, to improve the power level of the transmission signal, the actual message
signal is pre-emphasized before modulation and at receiver it is de-emphasized after demodulation.
PRE-EMPHASIS
It refers to a system process designed to increase (within a frequency band) the magnitude of some higher
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frequency with respect to the magnitude of some lower frequency in order to improve overall SNR (power) by
minimizing attenuation.
DE-EMPHASIS
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It refers to a system process designed to decrease (within a frequency band) the magnitude of some higher
frequency with respect to the magnitude of some lower frequency in order to improve overall SNR (power) by
minimizing attenuation.
1.18 SUPERHETERODYNE RECEIVER
To recover the message signal from the FM signal, frequency demodulation must be performed.
The most basic frequency demodulator consists of a frequency discriminator, which is a differentiator followed by an
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envelope detector, as shown in Figure.
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The Frequency discriminator detects the slope of the message signal from FM waveform and forms a relative
AM signal, from which the message signal can be detected using an envelope detector. This method can be employed
in Narrowband systems.
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A super-heterodyne receiver is a universal receiver which is capable of detecting AM, FM and PM. It is widely
used in Wireless Radio systems. A Super-heterodyne Receiver has three blocks. They
1. RF Amplifier
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2. IF section
3. Baseband processor
A super heterodyne receiver, uses frequency mixing to convert a received signal to a fixed intermediate
frequency (IF), which can be more conveniently processed than the original radio carrier frequency. At the cost of an
extra frequency converter stage, the super-heterodyne receiver provides superior selectivity and sensitivity compared
with simpler designs.
The principle of operation of the super-heterodyne receiver depends on the use of heterodyning or frequency
mixing. The signal from the antenna is filtered sufficiently at least to reject the image frequency and possibly
amplified. A local oscillator in the receiver produces a sine wave, which mixes with that signal, shifting it to a
specific intermediate frequency (IF), usually a lower frequency. The IF signals is itself filtered and amplified and
possibly processed in additional ways. The demodulator uses the IF signals rather than the original radio frequency to
recreate a copy of the original information.
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The diagram shows the minimum requirements for a single-conversion super-heterodyne receiver design. The
following essential elements are common to all super-heterodyne circuits.
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 A receiving antenna
 A tuned stage, which may optionally contain amplification (RF amplifier)
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 A variable frequency local oscillator
 A frequency mixer
 A band pass filter and intermediate frequency (IF) amplifier and
 A demodulator plus additional circuitry to amplify or process the original audio signal
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1.19 INDRECT METHOD OF GENERATING FM AND PM
Angle modulation such as FM and PM are mutually alterable. That is if Fm is available, PM can be easily obtained and
Vice versa.
PM from FM:
Phase modulated wave can be obtained from Fm by simply integrating it.
FM from PM:
Frequency modulated wave can be obtained from PM by differentiating it.
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AP
1.20 COMPARISON OF ANALOG MODULATION SYSTEMS
PARAMETER
FOR AM FM PM
COMPARISON
DEFINITION A modulation technique in which the A modulation technique in which the A modulation technique in which the phase
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amplitude of the carrier is varied in frequency of the carrier is varied in of the carrier is varied in accordance with
accordance with the change in amplitude of accordance with the change in amplitude of the change in amplitude of message signal.
message signal. message signal.
MESSAGE
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SIGNAL
CARRIER SIGNAL
MODULATED
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SIGNAL
MODULATION AM signal ‘s instantaneous amplitude FM signal ‘s instantaneous frequency directly PM signal ‘s instantaneous phase directly
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directly proportional to Amplitude of proportional to Amplitude of Message signal proportional to Amplitude of Message signal
RELATION
Message signal
MATHEMATICAL VAM (t) =VC Sinωct +
VFM(t)= Vc sin [ + ] VPM(t) = Vc sin [ + ]
EXPRESSION
-
BANDWIDTH 2fm 2(Δf + fm) 2(Δf + fm)

REQUIRED
FREQUENCY
SPECTRUM
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UNIT-2 DIGITAL COMMUNICATION

Amplitude Shift Keying (ASK) – Frequency Shift Keying (FSK) Minimum Shift Keying (MSK) –Phase Shift Keying (PSK) – BPSK – QPSK
– 8 PSK – 16 PSK - Quadrature Amplitude Modulation (QAM) – 8 QAM – 16 QAM – Bandwidth Efficiency– Comparison of various
Digital Communication System (ASK – FSK – PSK – QAM).
2.1 INTRODUCTION TO DIGITAL COMMUNICATION
Digital communication refers to the process of transmission and reception of information by means of digital
signals. There are two methods in Digital communication. They are
1. Digital Baseband Communication
2. Digital Modulation
In Digital Baseband communication, Transmission and Reception of digital data directly to a noiseless channel.
In this method there is no major modulation performed. In Digital Modulation techniques, either modulating a digital
carrier by an analog message signal (Pulse modulation) or modulating an analog carrier by a digital message signal
(Shift keying technique) is carried out. In general Digital modulation refers to Shift keying techniques. It is a method in
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which a relatively high frequency carrier is modulated by low frequency digital information.
ADVANTAGES OF DIGITAL COMMUNICATION
AP
 Better Noise immunity
 More flexible
 Easy for multiplexing and Demultiplexing
 Easy and effective error correction and decoding
APPLICATIONS OF DIGITAL MODULATION
 Best suitable for Cabled and Wireless Communication
 Relatively low speed ,voice-band data communication MODEMS
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 High speed data transmission systems such as Broadband Digital Subscriber Links (DSL)
 Digital Microwave and Satellite Communications
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 Cellular phones and Personal computer systems
Digital signals are more tolerant to noise, but digital signals can be completely corrupted in the presence of excess
noise. In digital signals, noise could cause a 1 to be interpreted as a 0 and vice versa, which makes the received data
different than the original data. Luckily, there are systems in place to prevent this sort of scenario, such as checksums
and CRCs, which tell the receiver when a bit has been corrupted and ask the transmitter to resend the data. The
primary benefit of digital signals is that they can be handled by simple, standardized receivers and transmitters, and
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the signal can be then dealt with in software.

2.2 BLOCK DIAGRAM OF DIGITAL COMMUNICATION SYSTEM
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The block diagram of Digital communication system is shown above. The digital communication system can be
divided into three sections, namely
1. Transmitter Section
2. Channel
3. Receiver Section
Transmitter section consists of the following blocks.
 Information source and Input transducer
 Source Encoder
 Channel Encoder
 Digital Modulator
A communication channel is present in between the transmitter and receiver sections. Receiver section consists of the
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following blocks.
 Digital Demodulator
 Channel Decoder
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 Source Decoder
 Output transducer
INFORMATION SOURCE AND INPUT TRANSDUCER
Information source is nothing but the Source of information. In shift keying techniques the message signal
happens to be a digital bit sequence. Input Transducer along with an ADC can convert the acquired real time analog
signals in to digital bit streams.
SOURCE ENCODER
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Source coding, also known as quantization or signal formatting, is a way of processing data in order to reduce
redundancy or prepare it for later processing. Usually, the Binary information is converted into NRZ codes in this block.
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[Binary informationit’s electrical representation in a specified format+
CHANNEL ENCODER
To combat the effects noise and channel corruption, the channel decoding block performs operations such as
convolutional coding and decoding techniques to implement error detection and correction. For simple error
detection with no inherent correction, a cyclic redundancy check capability is also available in this block.
DIGITAL MODULATOR
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A Digital modulator is a block which modulates an analog carrier by a digital information sequence.
CHANNEL
The Communication Channel may be a wired channel or a wireless open air interface. The Digitally modulated
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carrier is transmitted over this channel and is received and detected at the receiver section.
DIGITAL DEMODULATOR
A Digital modulator is a block which recovers the digital information sequence from the modulated wave. This
may be a coherent process or a non-coherent process.
CHANNEL DECODER
In this block, the detected wave is decoded and is represented as an Electrical equivalent of the binary
information.
SOURCE DECODER
Source decoder performs the conversion of electrical representation of the information into some Data
format. This is usually a thresholding device or a decision device.
OUTPUT TRANSDUCER
This block performs the D to A operation of the binary information.
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2.3 SHIFT KEYING TECHNIQUES
The process of changing the parameters (Amplitude, Frequency (period), Phase) of a high frequency Analog
Carrier by a digital modulating signals (bits or Binary) is called shift Keying.
(i.e) In Shift keying techniques,
Message signal  Digital Signal
Carrier signal  Analog Signal
o If Amplitude of Analog carrier is modulated by Digital Message, it is Amplitude Shift Keying or ASK
o If Frequency of Analog carrier is modulated by Digital Message, it is Frequency Shift Keying or FSK
o If Phase of Analog carrier is modulated by Digital Message, it is Phase Shift Keying or PSK
Let us consider,
Vc(t) = Vc sin (ωct+θ)
Vc(t) =Vc sin (2πfct+θ) be the high frequency carrier
Where,
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VcAmplitude of the carrier (volts)
fc Frequency of the carrier (Hz)
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θ Phase of the carrier (radians)
And consider Vm (t) be the digital message signal, possessing a value from the sample space of {0,1}
Then,
If Vc is modulated by Vm (t), it is called ASK Process similar to AM
If fc is modulated by Vm (t), it is called FSKProcess similar to FM
If θ is modulated by Vm (t), it is called PSKProcess similar to PM
2.4 AMPLITUDE SHIFT KEYING or DIGITAL AMPLITUDE MODULATION or ON-OFF KEYING
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2.4.1PRINCIPLE
Binary information signal directly modulates the amplitude of high frequency analog carrier. It is also called
Digital AM.
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Mathematically ASK is represented as,

VASK(t)= * +
VASK(t) = * + * +
Where,
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* + Unmodulated Carrier term & Digitally Amplitude Modulated Carrier

VASK(t) ASK Wave
ωc (or)2πfc  frequency of analog carrier
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A/2 Unmodulated Carrier amplitude

In the above equation, is the normalized digital data where binary bits {0,1} are considered.
Let ={ }Bipolar NRZ encoding logic
Hence for Logic 1 as input, the equation becomes
VASK(t) = * +
VASK(t) = * +
VASK(t) = Volts
Hence for Logic 0 as input, the equation becomes
VASK(t) = * +
VASK(t) = * +
VASK(t) = 0 Volts
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2.4.2 ASK MODULATION SAMPLE WAVEFORM
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ADVANTAGE DISADVANTAGE
 Simple to Implement  Applicable only to low bit
rate applications
[Up to 100 bps]
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NOTES:
In the ASK modulated wave, the analog carrier term is repeated (ON) if message contains Logic 1 and it is
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absent (OFF) for Logic 0 in the message. So ASK is sometimes called ON-OFF Keying [OOK]
 The time for transmitting one bit [either 0 or 1] is called symbol duration or bit duration [tb]
 The time taken for transmitting one DAM carrier unit is called signaling time [ts]
 Therefore if the binary input is high, with constant amplitude, constant frequency signal, the carrier is ON at
the output and if the binary output is Low, carrier is OFF at the output.
2.4.3 BLOCK DIAGRAM OF ASK MODULATION
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The binary data in NRZ format is obtained by source encoding. The NRZ encoded signal is modulated by a high
frequency analog carrier through a product modulator. The Modulated signal is band restricted using a Band Pass
Filter. The resulting signal is the ASK waveform.
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2.4.4 BIT RATE AND BAUD RATE
 tb(bit duratrion)= [bits per second]-1or [tb]-1 = bits per second  (data rate or bit rate)
 [ts] -1 = (signaling duration) -1= [Baud or Symbol rate ]
 If in an ASK scheme, Baud rate = Bit rate (Nyquist Rate), then it is called Nyquist rate ASK.
 = =[tb]-1 ,N No. of Bits Encoded in to each signaling element.
BAUD: Baud refers to the rate of change of signal on a transmission medium after encoding and modulation process.
It is a unit of transmission rate variation or Symbol rate.It is the reciprocal of time of one output signaling element.
Symbol rate or transmission rate is measured in (Baud / Sec).
Usually, Baud rate< Bit rate
 In low level digital systems such as BFSK,BPSK: Baud=bps
 In high level digital systems such as QPSK,8PSK: Baud<bps
NYQUIST BANDWIDTH: According to Nyquist, the minimum Bandwidth required for distortion less transmission of
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digitally modulated carrier is given, when the transmission rate (symbol rate) is twice that of the Ideal Bandwidth.
 [ts] -1=2 [tb]-1 =2 fb = BWi
 Minimum Required Bandwidth for distortion less transmission of digitally
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modulated carrier
Where, fb bitrate
BWi Ideal Nyquist Bandwidth
2.5 BINARY AND M-ARY ENCODING SCHEMES FOR DIGITAL MESSAGE SIGNAL
The digital message signal used for Shift keying modulation may be of any kind, either a binary encoded signal
or an m-ary [m=2n] encoded signal. These Different kinds of encoding schemes are used in complex shift keying
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methods like FSK and PSK, but not in simple ASK method.
Binary signal
 Symbol’s Sample space contains one bit group for a symbol
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(i.e.) one symbol is formed by one digit grouping (n=1), therefore m=21=2.
Sample space contains 2 possible symbols. {0,1}
If Modulation is done with above sample space, it is called Binary Scheme
m- Ary signal
 Symbol’s Sample space may vary from 2 bit group combination to n-bit group combination.
(i.e.) one symbol is formed by n-digit grouping and 2n=m.
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If n=2, therefore m=22=4  Sample space contains 4 possible symbols

{00, 01, 10, 11}
If Modulation is done with above sample space, it is called 4-ary Scheme or Quaternary Scheme
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{000,001, 010, 011, 100, 101, 110, 111}
If Modulation is done with above sample space, it is called 8-ary Scheme
{0000, 0001, 0010, 0011, 0100, 0101, 0110, 0111, 1000, 1001, 1010, 1011, 1100, 1101, 1110, 1111}
If Modulation is done with above sample space, it is called 16-ary Scheme
And so on…
In General,
If there are ‘n’ digits in a group to form a symbol, the number of possible symbols can be generated is m=2n.
And the scheme is called m-ary scheme.
If Modulation is done with above sample space, it is called m-ary Scheme
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No.of Bits used to form No.of Possible Symbols Used{sample space} Scheme name
one symbol (n) symbols m=(2^n)
“Symbol space”
S0=0
1 21=2 BINARY
S1=1
2 22=4 S0=00 S1=01 QUARTERNARY

S2=10 S3=11
3 3
2 =8 S0=000 S1=001 S4=100 S5=101 8-ARY
S2=010 S3=011 S6=110 S7=111
S0=0000 S1=0001 S4=0100 S5=0101
4 24=16 S2=0010 S3=0011 S6=0110 S7=0111 16-ARY
S8=1000 S9=1001 S10=1100 S11=1101
S12=1010 S13=1011 S14=1110 S15=1111
Notes:
 If the Digital Message signal used for FSK is encoded with Binary Scheme, then the resulting FSK is called BFSK.
 If the Digital Message signal used for FSK is encoded with m-ary Scheme, then the resulting FSK is called m-ary
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FSK.
2.6 FREQUENCY SHIFT KEYING [FSK]
2.6.1 PRINCIPLE
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FSK is similar to FM, but the only difference in FSK is the message signal happens to be a digital signal. Hence,
if a Digital Message signal is used to frequency modulate a high frequency analog carrier it is called Frequency Shift
Keying or FSK.
Vm= +1 or 5V Logic 1
Let the modulating digital message signal, Vm(t) = Vm= -1 or -5V Logic 0
And let,
Carrier signal
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Vc(t) = VC Cosωct
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Assume Frequency deviation for the carrier shift during modulation be ,Δ .
Frequency Deviation is defined as the difference between either Mark or Space frequency with the center carrier
frequency fc. It is also the half of the difference between Mark and Space frequencies.
Then according to FSK modulation, Carrier signal Vc(t) is modulated by the digital message signal Vm(t) as follows.
 If Logic 1 present as the instantaneous value of the message signal, the carrier is modulated with a positive
frequency deviation and
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 If Logic 0 present as the instantaneous value of the message signal, the carrier is modulated with a negative
frequency deviation.
ST
(i.e.)
Logic 1 at Vm Δ
for
Logic 0 at Vm Δ
Where Fm Mark frequency Maximum Frequency component in FSK signal

Fs Space frequency Minimum Frequency component in FSK signal
VFSK(t) = VC Cos t
Hence
If Logic 1 @ I/p:
VFSK(t) = VC Cos t
If Logic 0 @ I/p:
VFSK(t) = VC Cos t
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2.6.2 FSK MODULATION SAMPLE WAVEFORMS
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2.6.3 BLOCK DIAGRAM OF FSK MODULATION
AP
A Voltage Controlled Oscillator can act as a FSK modulator. According to the principle of FSK, the binary
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message is to be source encoded as a two level logic signal for frequency modulating a high frequency analog carrier
signal. Hence an NRZ Bipolar encoder is used before the VCO modulator.
The NRZ encoder converts the Binary signal in to a two level bipolar logic signal such that for Logic 1 the
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message signal constitutes a Positive voltage level and for Logic 0, the message signal constitutes a negative voltage
level. The Voltage controlled Oscillator is a circuit, producing sustained oscillations with a frequency which is governed
by the input voltage level. The VCO initially runs with a frequency called the free running frequency, which is assumed
to be the carrier frequency fc. Now with the Input Binary logic, the carrier frequency range is increased or decreased
 If Logic 1 present as the instantaneous value of the message signal, the VCO output frequency becomes (fc+fm)
 If Logic 0 present as the instantaneous value of the message signal, the VCO output frequency becomes (fc-
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fm)
The Output frequency of VCO is shifted between two preset frequency values (fc+fm) Mark and (fc-fm) Space.
A VCO-FSK Modulator is operated in sweep mode, for better performance where the peak frequency deviation is
ST
Where,K1 Deviation sensitivity of the sweep [Hz/V].

In BFSK, The amplitude of input signal can only be two values, one for Logic 1 and another for Logic 0.
So, and is the plus or minus the peak voltage of binary signal times the deviation sensitivity
of VCO. Since the magnitude of peak voltage is same for Logic 1 and Logic 0 in bipolar signal, the Magnitude of
Frequency deviation will also be same for Both Logic values.
2.6.4 FSK DEMODULATION SCHEMES
To do FSK demodulation and to recover the original binary information from the signaling unit there are three
demodulation schemes available. They are
1. Non- coherent FSK Demodulation
2. Coherent FSK Demodulation
3. PLL-FSK Demodulation
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NON-COHERENT FSK DEMODULATION
In this method there is no local carrier generation for the detection of message signal from the FSK signal.
Hence it is called Non-coherent method. The FSK signal is split and filtered by BPF. After splitting it passes Mark and
Space frequency components to envelope detector. The detector detects the power from the pass bands, Power from
each detector is compared and using a thresholding decision device the binary output data is formed.
P
AP
COHERENT FSK DEMODULATION
In this method a local carrier is generated and multiplied with the FSK signal to extract the actual message
signal. Hence this method is called coherent detection method. The Incoming FSK signal is multiplied by a locally
generated carrier at receiver (having the same frequency and phase as that of the transmitted carrier).
The Two Transmission frequencies Mark and Space are not continuous and they are discrete. They are
identified and are split and are sent to two distinct channels and those split signals are multiplied by local carrier
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frequencies in order to remove the carrier components from FSK signals.
Then the signal is filtered using a LPF of suitable cutoff frequency, to extract the binary message signal. The
filtered signals from each channel are given to comparator as input. After comparing the power from pass bands they
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are assigned with Binary values using a thresholding decision device.

U
ST
PLL- FSK DEMODULATOR

It is the most common Detector used for FSK demodulation. FSK signal shifts between Mark (for Logic 1) and
Space (for Logic 0). The DC error signal Produced after the comparison at mixer (phase comparator) follows the
frequency shift. Since there are only two input frequencies (Fm and Fs) hence there are only two possible error signals.
Among these two error values one corresponds for Logic 1 at message and the other corresponds for Logic
0.Therefore the output of PLL will be a Binary data. The free running frequency of PLL is made equal to the
transmission carrier frequency, so that the Local carrier can be generated at PLL itself. The changes in the DC error
signal follow the changes in the analog input frequency and are symmetrical around 0V.
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2.6.5 FSK DEMODULATION SAMPLE WAVEFORM
2.6.6 BANDWIDTH OF FSK

The minimum Bandwidth required for transmitting a FSK signal is given by
BWfsk = |(Fs+ fb)-(Fm –fb)|
= |Fs+ fb- Fm +fb)|
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= |Fs- Fm +2fb)|
= |Fm- Fs|+2fb
But it is known that,
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| |
= | |
And hence BWfsk =
BWfsk =
Where, BWfsk Minimum transmission bandwidth for FSK
 Frequency Deviation
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 Binary input bit rate * +
2.6.7 PROS AND CONS OF FSK
PROS CONS
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 Less error when compared to ASK and PSK  Not useful in low performance, low cost
asynchronous circuits like DATA MODEMS and
Voice phones etc.
 Can be Used in high performance Digital radio  Non-Linear Distortions will be more which causes
applications fickle reception
2.7 MINIMUM SHIFT KEYING [MSK]
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In FSK Non-linear distortions may occur during the detection due to the non-linear thresholding decision
device. This creates much jarring effects when FSK is used in asynchronous voice MODEMS and in Telecommunication
circuits. To avoid this problem a variant of FSK is developed and it is called MSK. In digital modulation, Minimum Shift
ST
Keying (MSK) is a type of continuous-phase frequency-shift keying . MSK is encoded with bits alternating between
quadrature components, with the Q component delayed by half the symbol period.
MSK encodes each bit as a half sinusoid. This results in a constant-modulus signal, which reduces problems
caused by non-linear distortion. In addition to being viewed as related to OQPSK, MSK can also be viewed as a
continuous phase frequency shift keyed (CPFSK) signal with a frequency separation of one-half the bit rate.In MSK the
difference between the higher and lower frequency is identical to half the bit rate. Consequently, the waveforms used
to represent a 0 and a 1 bit differs by exactly half a carrier period. Thus, the maximum frequency deviation is =
0.25 fm where fm is the maximum modulating frequency.
As a result, the modulation index m is 0.5. This is the smallest FSK modulation index that can be chosen such
that the waveforms for 0 and 1 are orthogonal. A variant of MSK called GMSK is used in the GSM mobile
phone standard.
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The name Minimum Shift Keying (MSK) implies the minimum frequency separation that allows orthogonal
detection as two FSK signals VIN-PHASE(t) & VQUADRATURE(t).MSK is a spectrally efficient modulation scheme and is
particularly attractive for use in mobile communication systems because of its possesses properties such as : Constant
envelope, Spectral efficiency, Good BER performance, Self-synchronizing capability.
2.7.1 MATHEMATICAL REPRESENTATION Vm= +1 or 5V Logic 1
Let the modulating digital message signal, Vm(t) = Vm= -1 or -5V Logic 0
And let,
Carrier signal Vc(t) = VC Cosωct
Assume Frequency deviation for the carrier shift during modulation be, Δ .
For MSK Δ
The resulting signal is represented by the formula
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( ) ( )
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Where and encode the even and odd information respectively with a sequence of square pulses of
duration 2Tand is delayed by T with respect to .The carrier frequency is .Using the trigonometric
identity, this can be rewritten in a form where the phase and frequency modulation are more obvious,
( ) (k)
Where ,Bk(t) = +1 when = and
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-1 when , are of opposite signs
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And (k) =0 , if is 1, and
= , otherwise.
Therefore, the signal is modulated in frequency and phase, and the phase changes continuously and linearly.
2.7.2 MSK WAVEFORMS
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ST
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P
AP
2.7.3 GAUSSIAN MSK
In digital communication, Gaussian minimum shift keying or GMSK is a continuous-phase frequency-shift
keying modulation scheme. It is similar to standard minimum-shift keying (MSK), however the digital data stream is
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first shaped with a Gaussian filter before being applied to a frequency modulator. This has the advantage of
reducing sideband power, which in turn reduces out-of-band interference between signal carriers in adjacent
frequency channels. However, the Gaussian filter increases the modulation memory in the system and causes inter-
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symbol interference, making it more difficult to differentiate between different transmitted data values and requiring
more complex channel equalization algorithms such as an adaptive equalizer at the receiver. GMSK has high spectral
efficiency, but it needs a higher power level than QPSK, for instance, in order to reliably transmit the same amount
of data. GMSK is most notably used in the Global System for Mobile Communications (GSM) and the Automatic
Identification System (AIS) for maritime navigation.
2.8. PHASE SHIFT KEYING
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The Process of varying the phase of the carrier (Analog) with the accordance to the level of digital message
signal is called Phase Shift Keying [PSK].
In BPSK,
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Message : Binary digital signal

Carrier : High frequency analog signal
BPSK O/P : A high frequency analog signal with limited number of phases (2-different phases) at Output.
The Binary Input information is encoded in to group of bits before modulating the carrier. This is called Source coding
or Pre-coding. The input binary sequence has so many number of bits. During pre-coding, the number of bits grouped
to represent (or) encode one symbol is called “symbol space”.
For example, if 4 bits are grouped, combined and encoded together as a symbol, then it is called 16-ary (24 )
encoding. In general if n- nits are grouped together and are represented as 1 symbol. Then it is called 2n-ary scheme. If
2n=M, then it is called M –ary encoding.
 If encoding is done with n=1 and M=2, and the resulting 2 symbols are used to phase modulate an analog
carrier, then it is called BPSK. Hence in output there will be 2 different phases.
 If encoding is done with n=2 and M=4, and the resulting 4 symbols are used to phase modulate an analog
carrier, then it is called QPSK. Hence in output there will be 4 different phases.
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2.8.1 BINARY PHASE SHIFT KEYING:
With Binary PSK, n=1 and M =21=2, so with BPSK there are possibly 2-Phases at the Output. Among the two
phases at the output 1-phase represents Logic 1, and the other 1-phase represents Logic 0.As the Input signal changes
from 0 to 1 and from 1 to 0, the phase of Output carrier shifts between 2 angles that are separated by 180°s [π-
Radians]
(i.e.)BPSK, If Logic 1 at input Output signal is 0° phase with respect to carrier [In-phase]
If Logic 0 at input Output signal is 180° phase with respect to carrier [Out of phase]
The other names of BPSK are
1. Phase reversal keying
2. Bi-phase Modulation
2.8.2 BPSK TRANSMITTER
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AP
Balanced Modulator can act as a Phase reversing switch. Hence it is widely used as BPSK
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modulator/Transmitter. Depending upon the logic condition of binary input (Logic 0 or Logic 1), the carrier is
transferred to output either in phase or with 180° Out of phase with reference to the Reference Carrier.
The Balanced ring modulator has 2 inputs. Namely,
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1. Reference Carrier (Analog Carrier)
2. Binary Input [Digital message for keying, Either 0 or 1]
For the Proper operation of a balanced modulator, binary message signal voltage level should be greater than
the carrier signal’s amplitude. There are four diodes present in the balanced modulator circuit forming a loop as
shown in the figure. Diodes are linear semi-conducting devices, which can pass a signal applied at its anode to
cathode only when it is in forward biased state.
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If the anode of the diode is connected to a positive voltage terminal and cathode connected to a negative
terminal and also the supply voltage is greater than 0.6V, then the diode is said to be forward biased. If the anode of
the diode is connected to a negative voltage terminal and cathode connected to positive terminal OR if the supply
ST
voltage is lesser than 0.6V, then the diode is said to be reverse biased.
During forward bias condition diodes acts as a short circuit, providing a path for the signal to pass through.
When they are reverse biased they act as Open circuit and no signal can pass through that.
There are two signal transformers available which transfers the signals from one part of the circuit to other
without loss in energy through a flux linkage. Each transformer provides 180° phase shift at their secondary, for the
inputs given to their primary.
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AP
R
CO
The Binary input sequence switches the diodes between ON and OFF states. The operation is explained below.
Case (i) If Logic 1 applied at input :
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If Logic 1 (positive voltage in bipolar format) is applied at digital input terminal, diodes D1 and D2 enters to forward
bias state. Hence they are ON and are acting as short circuit, providing the path for the analog carrier signal from the
secondary. The carrier is transferred to the output transformer without any phase shift. Hence the output is Inphase
ST
with respect to the Reference carrier. In this case diodes D3 and D4 are reverse biased and they are in open circuit
condition.
Case (ii) If Logic 0 applied at input :
If Logic 0 (negative voltage in bipolar format) is applied at digital input terminal, diodes D3 and D4 enters to forward
bias state. Hence they are ON and are acting as short circuit, providing the path for the analog carrier signal from the
secondary. The carrier is transferred to the output transformer with 180° phase shift due to the inverted path for
carrier. Hence the output is 180° out of phase with respect to the Reference carrier. In this case diodes D1 and D2 are
reverse biased and they are in open circuit condition.
TRUTH TABLE:
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Binary input Carrier Output signal Output phase
Logic 0 VcSinωct VcSin [ωct+π+ 180°
Logic 1 VcSinωct VcSinωct 0°
PHASOR DIAGRAM:
CONSTELLATION DIAGRAM:
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Constellation diagram represents the relative positions of peak points of carriers. It is also a signal state space
diagram, same like a phasor diagram.
AP
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2.8.3 BANDWIDTH OF BPSK
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In BPSK, the output signaling unit contains either - VcSinωct (out of phase carrier) or VcSinωct (in-phase carrier).
Hence the output rate of change (baud) is equal to the input rate of change (bit rate). Hence to calculate the Required
bandwidth of BPSK scheme , It is wise to calculate when the message signal is an alternating bit sequence.
(i.e.) if message is like, *1 0 1 0 1 0 1 0 …….+ maximum rate of change.
The duration of input message signal is chosen Such that the fundamental frequency (fa) of an alternating bit
sequence at input if equal to inverse of twice the bit rate.
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If tb time of one bit transmission duration, then it is related to fundamental frequency as ,
fa=1/2tb
fa=fb/2
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where ,fb=1/tb
Assuming the Carrier’s frequency to be fc, finding the bandwidth of BPSK signal as follows.
BPSK signal = [ x[
π
=
Now the minimum required Nyquist Bandwidth is
Band width (BPSK) = Maximum frequency component – Minimum frequency component
=
Band width (BPSK) =
Or Band width (BPSK) = b /2 =fb
But it is known that fb is inversely proportional to bit rate tb. fb=1/tb .Hence Bandwidth is also inversely proportional
to bit rate.
2.8.4 BPSK MODULATION WAVEFORM
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AP
2.8.5 BPSK RECEIVER
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The Input to BPSK receiver is a signal having Sinωct. The coherent carrier recovery circuit detects and
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regenerates a carrier that is similar to the original carrier in all aspects (frequency and phase). The balanced modulator
(product modulator) acts as a product detector here to demodulate the signal. The output is the product of 2 inputs.
(i.e.)Recovered Message signal = BPSK modulated signal * Recovered Carrier. The LPF separates recovered binary
ST
data from the demodulated signal and then decoded.

Mathematically, the Demodulation process can be explained as follows.
Using Carrier recovery circuit, a local coherent carrier is generated. And it is Sinωct. The Received BSPK signal
contains either +Sinωct (in-phase) or -Sinωct (out of phase) components.
Case i :
If the Received BPSK signal contains in-phase component Sinωct, Then at the Product detector, +Sin2ωct arises.
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It is known that,
Channel Logic Symbol Channel value
+Sin2ωct = [1 – Cos2ωct]/2
= (+ve)½ - (½ Cos2ωct)
If this is fed to LPF, the filtered output is (+ve) 1/2 .with Positive sign, the thresholding device will decode this as Logic
1.
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Case ii :
If the Received BPSK signal contains in-phase component Sinωct, Then at the Product detector, -Sin2ωct.
AP
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It is known that,
-Sin2ωct = -[1 – Cos2ωct]/2
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= (-ve)½ + (½ Cos2ωct)
If this is fed to LPF, the filtered output is (+ve) 1/2 .with negative sign, the thresholding device will decode this as Logic
0.
2.8.6 BPSK DEMODULATION WAVEFORM
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value Value
S4 I Logic 1 +1 I Sinωct (+1) Sinωct +45°
11
Q Logic 1 +1 Q Cosωct (+1) Cosωct
S3 I Logic 1 +1 I Sinωct (+1) Sinωct +135°
10
Q Logic 0 -1 Q Cosωct (-1) Cosωct
S2 I Logic 0 -1 I Sinωct (-1)Sinωct -45°
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01 [+315°]
Q Logic 1 +1 Q Cosωct (+1)Cosωct
AP
S1 I Logic 0 -1 I Sinωct (-1)Sinωct -135°
00 [+225°]
Q Logic 0 -1 Q Cosωct (-1)Cosωct
2.9 QUATERNARY PHASE SHIFT KEYING [QPSK]

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In quaternary scheme, instead of encoding one bit alone, 2 bits are combined together and are encoded as a single
symbol. Therefore n=2 and hence M= 2n =22=4. If 2 bits per symbol are used, there are 4 possible combinations
available for grouping. They are
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Symbol 1 Symbol 0 Di-bit encoded Symbol
0 0 S4
0 1 S3
1 0 S2
1 1 S1
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With 4 PSK, 2 bits are processed to produce a single phase change. In this case each symbol consists of 2 bits, which
are referred to as a di-bit. The actual phases that are produced by a 4 PSK modulator are shown in the table below.
Di-bits Di-bit encoded Symbol Output Phase
00 S4 -135°[+225°]
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01 S3 -45°[+315°]
10 S2 +135°
11 S1 +45°
Note: Between each distinct symbol the relative phase shift is 90°
The binary input sequence is divided in odd and even bit by bit splitter with a scaled clocking circuit(scale factor=2).
The odd and even bits are transmitted to In-phase channel [I-channel] and quadrature channel [Q-channel]
respectively. The signal from I-channel is Phase modulated [BPSK] by a normal carrier and the signal from Q-channel is
Phase modulated [BPSK] by a quadrature carrier [Carrier with 90°Phase difference]. These Phase Modulation
operations are carried out by two balanced modulator circuits. After PSK modulations the signals are band restricted
using BPFs of suitable pass band. Both the I-channel and Q-channel signals are linearly summed up using an LIC
amplifier to form the QPSK signal. The QPSK signal is again filtered by a BPF to preserve Bandwidth.
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The Output of linear summing amplifier has four possible output phases. There are four possible combinations
that can arise in 4-ary encoding [Di-bit encoding] viz. S4,S3,S2, and S1.Each symbol corresponds to a particular phase
with respect to the reference carrier. The truth tables for 4-PSK modulation are given below.
NOTE: For QPSK, Baud Rate = Bitrate/2
2.9.1QPSK MODULATION WAVEFORMS
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AP
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2.9.2 QPSK DEMODULATION

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ST
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In QPSK Demodulation, the received signal power is split in to two halves and is demodulated separately with
in-phase and quadrature carriers in order to recover the original binary message signal. Initially the QPSK signal is band
restricted using a Band pass filter. Then it is Split into two halves using a power splitter circuit. The two halves of the
received circuit are sent to balanced detectors. The in-phase and quadrature Carriers required for demodulating the
QPSK signal are locally generated. After Demodulating the Signal at balanced product demodulator the higher order
harmonics produced as a result are filtered using a LPF and DC terms are considered for thresholding. Depending on
the polarity of the Filtered DC term the decision is taken and binary data is formed. This is carried out for I& Q
channels.
Phase Difference phase At Product detector after Power splitting AFTER LPF At Decision
in Received QPSK component Device
signal w.r.t carrier
I-CHANNEL =(+Sin2ωct) +( Sinωct Cosωct) +1/2 Logic 1
(+Sinωct + Cosωc t )*(Sinωct) = +1/2 +HIGHER ORDER HARMONICS
45° +Sinωct + Cosωc t
Q-CHANNEL =( Sinωct Cosωct)+ (cos2ωct) +1/2 LOGIC 1
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(+Sinωct + Cosωc t )* (Cosωct) = +1/2 +HIGHER ORDER HARMONICS
I-CHANNEL =(+Sin2ωct) -( Sinωct Cosωct) +1/2 LOGIC 1
(+Sinωct - Cosωc t )*(Sinωct) = +1/2 +HIGHER ORDER HARMONICS
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135° +Sinωct - Cosωc t
Q-CHANNEL =(-Cos2ωct) +( Sinωct Cosωct) -1/2 LOGIC 0
(+Sinωct - Cosωc t )* (Cosωct) = -1/2 +HIGHER ORDER HARMONICS
I-CHANNEL = (-Sin2ωct)+( Sinωct Cosωct) -1/2 LOGIC 0
(-Sinωct + Cosωc t )*(Sinωct) = -1/2 +HIGHER ORDER HARMONICS
-45° -Sinωct + Cosωc t
Q-CHANNEL (-Sinωct Cosωct)+(Cos2ωct) +1/2 LOGIC 1
(-Sinωct + Cosωc t )* (Cosωct) = +1/2 +HARMONICS
I-CHANNEL (-Sin2ωct)-( Sinωct Cosωct) -1/2 LOGIC 0
= -1/2 +HIGHER ORDER HARMONICS
-135° -Sinωct - Cosωc t
R (-Sinωct - Cosωc t )*(Sinωct)
Q-CHANNEL (-Cos2ωct)-( Sinωct Cosωct) -1/2 LOGIC 0
(-Sinωct - Cosωc t )* (Cosωct) = -1/2 +HIGHER ORDER HARMONICS
2.10 8-ARY PHASE SHIFT KEYING
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When Three bits are brought together to form one symbol, such an encoding is called 8-ary encoding. If PSK is
performed with 8-ary encoded digital symbols, the resulting PSK is called 8-ary PSK or simply 8-PSK.
(i.e.) if n=3, M= 2n =23= 8
Therefore in 8-PSK scheme, there are 8 different phases possible at the output signal.The Truth table and signal
constellation for 8-ary PSK is given below.
8-ary Encoded Symbol Output
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binary message Name Phase

signal
ST
0 0 0 S0 0°
0 0 1 S1 45°
0 1 0 S2 90°
0 1 1 S3 135°
1 0 0 S4 180°
1 0 1 S5 225°
1 1 0 S6 270°
1 1 1 S7 315°
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2.11 16 -ARY PHASE SHIFT KEYING
When Four bits are brought together to form one symbol, such an encoding is called 16-ary encoding. If PSK is
(i.e.) if n=4, M= 2n=24 = 16
Therefore in 16-PSK scheme, there are 16 different phases possible at the output signal.
The Truth table and signal constellation for 16-ary PSK is given below.
16-ary Encoded Symbol Output Phase
binary message signal Name
0 0 0 0 S0 0°
0 0 0 1 S1 22.5°
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0 0 1 0 S2 45°
0
0 0 1 1 S3 67.5°
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0 1 0 0 S4 90°
0 1 0 1 S5 112.5°
0 1 1 0 S6 135°
0 1 1 1 S7 157.5°
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1 0 0 0 S8 180°
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1 0 0 1 S9 202.5°
1 0 1 0 S10 225°
0
1 0
0 1 1 S11 247.5°
1 1 0 0 S12 270°
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1 1 0 1 S13 292.5°
1 1 1 0 S14 315°
ST
1 1 1 1 S15 337.5°
2.12 DPSK Modulation and Demodulation

It is an alternate form of Digital Modulation, where binary input information is contained in the difference
between 2 successive signaling elements rather than the absolute phase. There are two operations involved in DPSK.
They are
1. Differential Encoding
2. Phase shift Keying.
To perform DPSK the input data bit is supposed to Logical EX-NOR operation with its one bit duration delayed
signal and then it is phase modulated in a balanced product modulator. By doing so, there is no need of coherent
carrier during the demodulation.
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2.12.1 DPSK MODULATION AND DEMODULATION PROCESS
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Let Initial Reference bit be Logic 0 and Reference Phase be π.To Send Symbol 0 phase advance the by π radians
and to send symbol 1send the actual carrier as such. The differential encoding process starts with an initial reference
assumption which is usually Logic 0 and then proceeded by the Logical EX-NOR operation between the second bit and
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the first output.
It is represented as
First Bit Reference bit Output (by XNOR) Transmitted Phase

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0 0 1 0
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0 1 0
1 0 0
1 1 1 0
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2.13 QUADRATURE AMPLITUDE MODULATION
In all PSK Methods, One symbol is distinguished with other symbol by phase and amplitude remains constant
for all symbols. Noise immunity will improve if the signal vectors differ not only in phase but also in amplitude. Hence
direct modulation of carriers with quadrature phase angles is included in PSK and such a system is called Quadrature
Amplitude Modulation [QAM].
TYPES:
METHOD Bits/Symbol No. Of symbols used
4-QAM 2 4
8-QAM 3 8
16-QAM 4 16
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32-QAM 5 32
64-QAM 6 64
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On moving to higher order QAM systems, the mode and High data rates can be achieved with the help of high
power RF carriers and they can be employed in Microwave communication systems. This QAM scheme spreads data in
to more number of phases into the spectrum. Hence Noise immunity is increased when compared to other systems.
4-QAM 8-QAM 16-QAM
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4-symbols 8-symbols 16 Symbols

1 amplitude 4 different amplitudes 4 different amplitudes
4 different pahses 8 different phases 12 different phases
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BLOCK DIAGRAM OF 8 –QAM GENERATOR
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BLOCK DIAGRAM OF 16 –QAM GENERATOR
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2.14 BANDWIDTH EFFICIENCY

The three factors that influence the choice of digital modulation system are
• Bandwidth efficiency: The number of bits per second that can be transmitted per Hertz of channel
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bandwidth
• Error performance: The probability of making a bit error at the receiver, as a function of the signal-to-noise
ratio
• Equipment complexity: A parameter which effectively corresponds to the cost of the system.
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Bandwidth efficiency or Spectral efficiency or spectrum efficiency refers to the information rate that can be
transmitted over a given bandwidth in a specific communication system. It is a measure of how efficiently a limited
frequency spectrum is utilized by the Communication system. Bandwidth efficiency describes how efficiently the
allocated bandwidth is utilized or the ability of a modulation scheme to accommodate data, within a limited
bandwidth. This table shows the theoretical bandwidth efficiency limits for the main modulation types. The Spectral
Efficiency (measured in b/s/Hz) of a modulation scheme with transmission rate R and bandwidth B is defined as
ρ = R/B = ⁄
%ρ = R/B *100%
Bandwidth Efficiency is generally normalized to a 1-Hz Bandwidth and, thus it indicates the number of bits that can be
transmitted through a medium for every 1 Hz. Usually Bandwidth efficiency is represented as % efficiency.
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Consider an M-ary modulation scheme with baseband pulse s(t) with duration T and null-to-null bandwidth 2/T, then R
= (1/T) log2(M)
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2.15 COMPARISON OF VARIOUS DIGITAL MODULATION SCHEMES(ASK, FSK, PSK, QAM)

PARAMATERS ASK FSK PSK QAM
BPSK QPSK DPSK
MESSAGE M:10110100 M:10110100
DI-BIT ENCODING:
10 –S2 -> 135° ASSUMING 4-QAM
11-S1 ->+45° DI-BIT ENCODING:
01-S3 ->-45° 10 –S2 -> 135°
00-S4 ->-135° 11-S1 -> +45°
01-S3 -> -45°
MSG:1010 00-S4 ->-135
DIFFERENTIAL ENCODING
0 1 1 0 WITH REF=0
TX: π-0-0- π
CARRIER
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MODULATED
WAVEFORM
LOGIC 1 = Sinωct LOGIC 1 =Sin(fc+Δf)t LOGIC 1 = Sinωct 00 -Sinωct -Cosωc t AFTER DIFFERENTIAL 00 -Sinωct -Cosωc t
TRUTH TABLE ENCODING
LOGIC 0 = 0
R LOGIC 0 = Sin(fc-Δf)t LOGIC 0 = -Sinωct
LOGIC 1 = Sinωct+ π
01 -Sinωct + Cosωc t LOGIC 0 = Sinωct 01 -Sinωct + Cosωc t
10 +Sinωct - Cosωc t 10 +Sinωct - Cosωc t

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11 +Sinωct + Cosωc t 11 +Sinωct + Cosωc t
ENCODING n=1; M= 2n=2 n=1; M= 2n=2 n=1; M= 2n=2 n=2; M= 2n=4 Differential Encoding by
XNOR LOGIC
DIBIT ENCODING FOR 4-
QAM
SCHEME n=2; M= 2n=4
(n,M)
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NYQUIST fb fb fb fb/2 <fb/2 fb/2

BANDWIDTH
Coherent: MODERATE MAJOR MAJOR MAJOR MODERATE MAJOR
COMPLEXITY
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Non-coherent: MINOR
APPLICATION Device Control Coherent: Most Used for High Used for High speed Widely used in For Long haul
(RARELEY USED) widely used for speed data data transmission lesser complex Transmission of
domestic transmission systems, wireless picture signals.
communication systems, asynchronous communication
systems asynchronous modems, systems
Non-Coherent: Slow modems DSL links
speed data modems
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WORKSHEET UNIT-2
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WORKSHEET UNIT-2
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WORKSHEET UNIT-2
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WORKSHEET UNIT-2
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WORKSHEET UNIT-2
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WORKSHEET UNIT-2
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UNIT-3 DATA AND PULSE COMMUNICATION

Data Communication: History of Data Communication - Standards Organizations for Data Communication- Data
Communication Circuits - Data Communication Codes - Error Detection and Correction Techniques - Data communication Hardware
- serial and parallel interfaces.
Pulse Communication: Pulse Amplitude Modulation (PAM) – Pulse Time Modulation (PTM) – Pulse code Modulation (PCM) -
Comparison of various Pulse Communication System (PAM – PTM – PCM).
3.1 INTRODUCTION
Data  generally defined as information stored in digital form
3.1.1 HISTORY OF DATA COMMUNICATION
It is the process if transfer of digital information (transmission, reception and processing of digital information)
between two or more points. The analog source information is in analog form (human voice or music) or in digital
form (binary coded numbers or alpha numeric codes)
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3.1.2 NETWORK
Set of devices (called as nodes or stations) interconnected by media links.
3.1.3 DATA COMMUNICATION NETWORK
These are systems of inter connected computers and computer equipment and can be a simple personal
computer connected to a printer or two or more computers connected together through the public telephone
network.
3.1.4 COMPLEX DATA COMMUNICATION NETWORK
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 One or more main frame computers
 Hundreds, thousands or even millions of remote terminals, personal computers, work stations
Note: There is no limit to the capacity size of a data communication network
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3.1.5 DATA COMMUNICATIONS
 Exchange of data between two devices via some form of transmission medium (wire cable)
 For data communication to occur, the communicating devices must be a communication system made up of
combination of hardware and software
3.1.6 FOUR FUNDAMENTAL CHARACTERISTICS OF DATA COMMUNICATION NETWORKS
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1. Delivery
2. Accuracy
3. Timeliness
4. Jitter
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3.1.7 COMPONENTS OF DATA COMMUNICATIONS

There are five components
1. Message (text, number, pictures, audio, and video)
2. Sender (computer, work station, telephone handset, video camera)
3. Receiver (computer, work station, telephone handset, video camera)
4. Transmission medium (twisted pair cable, co-axial cable, fiber optic cable, radio waves)
5. Protocols  set of rules that govern data communication
An agreement bits two communicating device without protocol, two devices may be connected but not
communicating.
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3.1.8 DATA FLOW
Communication between two devices can be
1. Simplex
2. Half- duplex
3. Full duplex
Simplex: Unidirectional, only one of the two devices transmit on a link, the other can only receive. (eg: one way
street) Example: key board, monitor
Half duplex: Two way communication, but not at same time. (eg: traffic of walkie-talkies, CB Radios)
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Full duplex: Transmit and receive simultaneously, example: telephone
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Note :
1. The entire capacity of the channel is fully utilized in simplex and in half duplex
2. In full duplex, capacity of the channel is divided into two directions
Link must have two physically separate transmission path, one for sending and receiving a capacity of the
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channel is divided between signals travelling in both directions
3.2 STANDARD ORGANIZATION FOR DATA COMMUNICATION
Computer network communicate using protocols which defines the procedures that the system involved in the
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communication process will use. Numerous protocols are used today to provide networking capabilities such as how
much data can be sent, how it will be sent, how it will be addressed, and what procedure will be used to ensure that
are no undetected errors.
International Standard Organization (ISO):
 It creates the set of rules and standards for graphics, document exchange and related technologies
(multinational body)
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 ISO is responsible for coordinating the work of other standard organizations

Consultative Committee for International Telephony & Telegraphy (CCITT):
 The membership of the CCITT consists of government authorities and representatives from many countries
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 It is the standard organization for united nation

 It develops the recommended sets of rules and standards for telephone & telegraph communications
 It has developed 3 sets of specifications
1. V series for modem interfacing
2. X series for data communication
3. I and Q series for Integrated Services digital network
American National Standards Institutions (ANSI):
 ANSI official standards agency for the united states is the U.S. voting representative for ISO
 It is a private, nonprofit cooperation not affiliated with U.S. federal government
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International Telecommunication Union Telecommunication standards sector (ITU-T):
 By early 1970’s, a number of countries were defining national standards for telecommunication. The united
nations responded by forming a part of its International Telecommunication union (ITU) a committee, the
consultative committee for international telephony and telegraphy (CCITT).
 Devoted to the research & establishment of standards for telecommunications, in general and for phone and
data systems
 On March 1, 1993, the name of this committee was changed to International Telecommunication Union
Telecommunication standards sector (ITU-T)
Institute of Electrical and Electronics Engineering (IEEE):
 Largest professional engineering society in the world
 It aims to advance theory, creativity and product quality in the fields of electrical engineering, electronics,
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radio as well as all branches in engineering
Electronics Industries Association (EIA):
 Aligned with ANSI, EIA is a nonprofit organization devoted to the promotion of electronics manufacturing
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concerns
 Activity  public awareness education and lobbying efforts in standards development
 In the field of information technology the EIA has made significant contributions by defining physical
connection interfaces and electronic signaling specification for data
Standards Council of Canada (SCC):
 SCC is the official standard agency for Canada with similar responsibilities like ANSI.
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3.3 DATA COMMUNICATION CIRCUITS
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 The block diagram consists of a source of digital information (primary station), a transmission medium
(facility) and a destination (secondary station).
 Primary or host location, secondary or remote station, secondary are the users of the network
 Depending on the system and the applications
 How many secondary stations are used?
 How they are interconnected with each other?
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There are many types of transmission media,
1. Free space radio transmission (terrestrial &) satellite microwave)
2. Metallic cable facilities (both digital & analog systems)
3. Optical fiber cable (light wave propagation)
3.3.1 DTE:
Data Terminal Equipment is a general term that describes the interface equipment used at stations to adapt
digital signals from the computer and terminals to a form more suitable for transmission
 UART  Universal Asynchronous Receiver Transmitter
 USART  Universal Synchronous Receiver Transmitter
3.3.2 DCE:
Data Communication Equipment, the equipment that converts digital signals to analog signals.
DCE  MODEM  modulator / demodulator and interfaces the data terminal equipment to the analog transmission
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medium.
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3.3.3 TRANSMISSION MODES:
Transmission of data from one device to another is in means of wiring, suppose wiring is data stream, how the bits
are sent, either one bit at a time or either group of bits at a time. Depending on number of bits transmitted, they are
classified as
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Parallel transmission:
 Binary data consists of 0’s and 1’s are organized into group of n-bits each
 So by using n wires, to send n bits at a time, so each bit has its own wire, so all the n bits of one group can be
transmitted with each clock pulse from one device to another
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 All the four bits can be transmitted simultaneously during the time of a single clock pulse (T). this type of
transmission called parallel by bit (or) serial by characters
 Limited to short distances, is high speed device
Serial transmission:
 One bit follows another, so only one communication channel is used instead of n wires
 Due to only one wire, cost is also reduced, since communication within devices is parallel, so conversion are
used
 To transmit the bits 0110, It requires 4 clock pulses to transmit entire words. This type is called as serial – by –
bit used for long distance communications
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Asynchronous transmission:
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Asynchronous at the byte levels, bits are still synchronized. The receiver does not know when the bit is coming, to
alert the receiver an extra bit is added to the beginning of each byte. Example: connection of keyboard to the
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computer is asynchronous transmission. In each byte an extra bit ‘0’is added called start bit and the extra bit ‘1’ is
added at the end of each byte to know the receiver that the byte is finished called as stop bit.
Example:
8 bit  I byte word
4 Bit  nibble
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The gap is in between each byte, this gap is filled by additional stop bits are by an idle channel.
In asynchronous mode, Speed is less.
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Synchronous transmission:
 Bit stream is combined into longer frames which contain multiple bytes
 Each byte is introduced on to the transmission medium without a gap between and next one
 It is left to the receiver to separate bit streams into bytes for decoding purpose
Example: transmission of data from one computer to another computer
 Accuracy depends on only the receiver device
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3.3.4 TRANSMISSION CONFIGURATIONS:

 A network is a two (or) more devices connected through links.
 A link is a communication path that transfers from one device to another.
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 For simplicity any link is a line drawn between two points

 So for communication to occur two device must be connected in some way to the same link
Types of connections:
 Point to point connection (two point)
 Multipoint connection
Point to point:
 This connection provides a dedicated link between two devices
 Connection between only two locations or stations
 Transfer of information between a mainframe computer and a remote computer terminal
Example: television  remote control

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Multipoint connection: (Multi drop connection)
 It involves three or more stations
 Multipoint  interconnects a single mainframe computer ( host) to many remote computer terminals
 The link is shared
3.3.5 TOPOLOGY:
The topology or architecture of a data communication circuit identifies how various locations within the network
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connected. Two or more devices connected to a link (or) two or more links from topology.
Categories of topology:
1. Mesh
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2. Star
3. Ring (or) loop
4. Bus or Multidrop
Mesh (point to point):
 Every device has a dedicated point to point link to every other device
 The link carries data only between the two devices it connects
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 Each node must be connected to every other node

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 Example: Node 1  n-1 node

Node 2  n-1 node etc
Node 3  n-1 node
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So n(n-1) physical links are used, if to find how many physical links are used. n(n-1)/2  for duplex mode.
Star topology : (point to point)
S1, S2, S3, S4  stations

 Each device has a dedicated point to point link only to a central controller called hub
 Devices are not directly linked to the others
 Controller acts as a exchange
 Less expensive
 Information from one device to another is not direct
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Bus topology: (multi point)
 One long cable acts as a backbone to link all the devices in a network
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 Nodes are connected to the cable by tap and drop lines.
Ring topology:
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 Each device has a dedicated point to point connection with only 2 devices on either sides of it
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 A signal is passed along the ring in a direction, from device to device till it reaches destination
 Each device in the ring is incorporated with repeater. The repeater passes the bits to another device
 Easy to install and reconfigure
 Each device is linked to only its immediate neighbors (either physically or logically)
3.3.6 TYPES OF PROTOCOLS:
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Connection oriented:
 Logical connection is established between the end points prior to the transmission of data
 Example: standard telephone call where sequence of operations are done  setting up the call, establishing
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the connection and then disconnecting. That is it involves dial tone, touch tone, signaling, ringing, ring back
signals, busy signals
 In this a set of procedures for establishing the connection, transferring the data, acknowledging the data,
then cleaning the connection
 In this each packet is assigned a unique sequence number and an associated acknowledgement number to
track the data as they travel through the network
 If data are lost or damaged, the destination station requests to resend
 Provide high degree reliability
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Connectionless protocols:
 In this data are exchanged in a unplanned manner, without coordination between end points
 Advantage  transmission speed than connection oriented
 Disadvantage  do not provide high degree of reliability like connection oriented
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 Postal service information is formatted, placed in an envelope with source and destination addresses and
mailed
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Connection oriented Connection less
Hand shake signals occur between two stations Sends data without a source and destination
before any data are actually transmitted address without a handshake to ensure that
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data is ready to receive the data
Provides some means of error control (detection Do not support error control
and correction)
3.3.7 TYPES OF STANDARDS:
1) Proprietary (closed) system
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2) Open system
Proprietary system:
 Generally manufactured and controlled by one company
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 Other companies are not allowed to manufacture equipment or write software using this standard
Advantages
 Tighter control
 easier consensus
 monopoly
Disadvantages
 Lack of choice for the customers
 Higher financial investment
 Overpricing
Open system
 Any company can produce compatible equipment or software but royalty should be made to the original
company.
 Example: IBM’s PC
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Advantage
 Customer choice, compatibility between vendors and competition by small company
Disadvantage
 Less product control increased difficulty acquiring agreements between vendors for changes or updates
Extra standards:
1) Telecommunication Industry Association (TIA)
2) Internet Architecture Board
3) Internet Engineering Task Force (IETF)
4) Internet Research Task Force (IRTF)
3.4 DATA COMMUNICATION CODES
These are prescribed bit sequences used for encoding characters and symbols, in simple used to represent
character and symbol. So data communication codes are called as character codes, character sets, symbol codes or
character languages.
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Codes
1) Baud of code
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2) ASCII code
3) EBCDIC code
4) BAR code
Baud of code : Telex code
 It is called as Telex code.
 It was the first fixed length character code developed for machines than people
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 French postal engineer Thomas Murray developed in 1875
 Baud of code  Baw dough is a fixed length source or fixed length block code
 With fixed length all characters are represented in binary and hence the same number of bits (symbols)
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 It is a five bit character code used for low speed teletype equipment such as TWX/Telex system and radio teletype
(RTTY)
 Latest version recommended by the CCITT as international alphabet number (2).
Example:
Letter Figure Bit : 4 3 2 1 0
A - 1 1 0 0 0
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B ? 1 0 0 1 1
Figure shift, Letter shift, Space, Line feed (LF) , blank (null)
ASCII code:
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 1963, to standardize the data communication codes, the U.S. adopted Bell system model, 33 teletype code as the
United States of American Standard Code for Information Interchange (USASCII) known as better ASCII-63
 Versions 1965, 1967, 1977 are being recommended by ITU as alphabet number 5
 ASCII  standard character set for source coding the alpha numeric character set that humans understand but
computers do not
 7 bit character set
 Bits are referred by the order than by their positions b0, b1  zero order, first order…
 By using serial transmission, b0 is transmitted first.
 ASCII probably used code in data communication network recently
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EBCDIC code:
 Extended Binary Coded Decimal Interchange Code (EBCDIC)
 8 bit fixed length character developed in 1962 by IBM (International Business Machines Corporation )
 Used almost with IBM mainframe computers and peripheral equipment
 With 8 bits, 28 = 256 codes are possible out of 256, only 139 are assigned characters
 Binary coded decimal  because the second hexadecimal character for all letter and digit codes contain only the
hexadecimal values for 0 to 9 which have same binary sequence HEX : 00 to 09, 0A to 0F, 10 to 19, 1A to 1F.
BAR codes:
 Black and white stripped stickers that seem to appear on every customer item
 Developed in 1960’s and used in the mid of 1980’s
 BAR code is a series of vertical black bars separated by vertical white bars (called spaces)
 The width of the bar and spaces represents 1’s and 0’s and combinations of bits represents to identify specific
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items
 BAR codes may contain information regarding
 Cost
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 Inventory management control
 Security access
 Production counting
 Automatic billing
 There are several BAR code formats
 The format selection depends on how the data are stored, what type of data are stored, system performance
Classification :
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1)Discrete
2)Continuous
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3)Two dimensional (2 D)
Discrete: A discrete bar code has spaces or gap between characters, so each character within the bar code is
independent of every other character. Example: code 39 (code 3 of 9 or code 9 of 3)
Continuous: It does not include spaces between characters. Example: Universal Product Code (UPC)
2 D code: stores data in 2 dimensional in contrast with conventional bar code (stores data along only one axis). They
have a larger storage capacity than one dimensional (1 kilobyte or more data symbol)
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3.5 ERROR CONTROL

Errors are unpredictable changes occurs because of interference.
Classification:
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1) Single bit
2) Multiple bit error
3) Burst error
Single bit error:
 Only one bit at a given data unit (byte, character, packet) is changed from 1 to 0 (or) from 0 to 1.
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Multiple bit error:
 When two or more nonconsecutive bits within a given data string are error
 It will affect one or more characters within the message
Burst error:
When two or more consecutive bits within a given data string are in error. In this corrupted bits are measured
from the first corrupted bits to the last computed bit.
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Note:
 Error performance is in the rate in which error occurs, which is expected or empirical value
 The theoretical (mathematical) expectation of the rate of error which occur is called probability of error (Pe)
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 Where the actual record of system error performance is called “Bit Error Rate” (BER)
Categories of error control:
 Error detection
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 Error correction
3.5.1 ERROR DETECTION:
 Process of monitoring data transmission and determining when errors have occurred
 Error detection neither correct errors nor identify which bits are in error, they indicate only when a error has
occurred.
 Most common error detection technique is redundancy checking
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Redundancy checking:
 Some extra bits are added with the data. These redundant bits are added by sender and removed by the receiver.
Their presence allows the receiver to detect or correct computer bits
 Extra redundant bits are added
 So costly and message will be long
Vertical Redundancy Check (VRC):
 Vertical Redundancy Check
 Simple way of error detecting technique called as character parity or parity
 Character parity  each character has its own error detection bits called redundant bits
 Error detection bits ∝ length of the message
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 To each character single parity bit is added to force the number of logic 1’s in the character with parity bit is odd
or even number
 Example: ASCII code for letter C is 43 hex or P1000011
Parity bit
P  0  odd parity
P  1  even parity
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Merits:
 Simplicity
Demerits:
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 Only 50% of errors are detected

 When even number of bits is received in error, the parity checker will not detect them because when the logic
condition of an even number of bits is changed, the parity of the character remains the same.
Types:
1) Marking parity  parity bit is always 1
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2) No parity  parity bits not sent or check

3) Ignored parity  parity bit is always 0
Checksum:
 Simple form of redundancy error checking where each character has a numerical value assigned to it
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 Suppose the data to be sent is a list of 4 bit numbers (example: 7, 11, 12, 0, 6) so in addition by sending this bits,
the sum of the bits is also added at the end of the bits called checksum
Example: (7, 11, 12, 0, 6)  to be sent
(7, 11, 12, 0, 6, 36)  will be transmitted
Checksum (or) error checking character
Receiver adds the bits (5 numbers), if it matches with the sum, no error, accepts 5 numbers and discards the sum.
If not matching, error is present and data will not be accepted. Internet has been using a 16 bit checksum.
Longitudinal redundancy checking:
 It uses a parity to determine if a transmission error has occurred within a message and also called as message
parity.
 With LRC, each bit position is having a parity bit
 So from each character in the message is X-OR ed with b0 from all other character in the message
 Similarly b1, b2, …b7 are also X-OR ed.
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 So LRC X-OR ing the character codes that make up the message, so even parity is generally used LRC bits are
computed in the transmitter while data are being sent and then appended to the end of the message as a
redundant character.
At Receiver  LRC is recomputed from the data and the recomputed LRC is compared to the LRC appended to the
data.
If 2 LRC characters are same  no error
If 2 LRC characters are different  error have occurred
Example: consider a message “THE CAT”
LRC = 2F (hex)  character / in ASCII after LRC character is appended to the message if read “THE CAT /”
Group of characters that is a message called a block or frame of data. So bit sequence of LRC is called as block check
sequence (BCS) or frame check sequence (FCS)
Cycle redundancy check:
 Most reliable redundancy checking technique for error detection is cyclic redundancy checking
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 With CRC, approximately 99.999% of all transmission errors are detected
 In US, most common CRC is CRC-16. With CRC-16, 16 bits are used for the block check sequence
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 With CRC, the entire data stream is treated as long continuous binary number
 Cyclic block codes are written as (n,k) cyclic codes
N= bit length of transmission
K= bit length of message
So BCC= n-k
It detects single bit, double bit, all odd number of bit errors, all burst error of 16 bit or less and burst error greater than
16 bits.
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CRC-16 generating circuit:
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 BCC- remainder of the binary division process

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 A data message polynomial G(x) is divided by a unique generator polynomial function p(x)
 The quotient is discarded, the remainder is truncated to 16 bits and appended to the message as BCS (Block
Check Sequence)
 The generator polynomial should be a prime number
 CRC generation, modulo-2 division is used, where the remainder is derived from an XOR operation
 In the receiver, the data stream including CRC code is divided by the same generating function P(x)
 No transmission errors, remainder is ‘0’
 In the receiver the message and CRC characters pass through a block check register. After the entire message
has passed through the register contents should be zero, so received message has no error.
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Where, G(x) message polynomial
P(x) generator polynomial
Q(x) quotient
R(x)  remainder
Pbm 1: Determine BCS and CRC generating polynomial
Data G(x) = x7+x5+x4+x2+x1+x0
= 10110111
CRC P(x) = x5+x4+ x1+x0
= 110011
CRC  generally used with 8 bit codes such as EBCDIC or seven bit codes when parity is not used.
With CRC-16, 16 bits are used for the BCS.
Generator polynomial for CRC-16 is x16+x12+ x5+x4
Pbm 2: Determine the BCS for the following data and CRC generating polynomial
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Data G(x) = x7+x5+x4+x2+x1+x0
= 10110111
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CRC P(x) = x5+x4+ x1+x0
= 110011
So G(x) is multiplied by the number of bits in the CRC code (the number of bits in the CRC code is equal
to the highest component of the generator polynomial)
 In the above example:
= x5 (x7+x5+x4+x2+x1+x0)
= x12+x10+x9+x6+x1+x5
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= 1011011100000
 Divide by P(x)
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CRC is appended to the data to give the following transmitted data stream
G(x) CRC
10110111 01001  Remainder of division process
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3.5.2 ERROR CORRECTION
There are two basic types of errors
1. Lost message  A message that never arrives at the destination or arrive that is
damages and unrecognizable
2. Damaged message  Recognized at the destination but contain one or more transmission
errors
This enables the receiver to determine when an error has occurred and which bit is in error.
Types of error correcting:
1. Symbol substitution
2. Retransmission
3. Forward error
Symbol substitution:
 Designed to be used in a human environment, where there is a human being in the receiver terminal to
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analyze the received data and make decisions
 With this, if a character is received in error, instead of displaying incorrect character, a unique character
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that is undefined by the character code, such as reverse question mark (¿) is substituted for bad character
 It is a form of selective retransmission
 Example: NAME
If error displayed as “Game”, an operator can discard the correct message by inspection and
retransmission is not necessary
 Example : “ $5,000,00” received, an operator cannot determine correct character and asks for
retransmission
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Retransmission:
 When a receiver station requests the transmit station to resend a message (or a portion of the message),
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when the message receive in error
 The receiver terminal automatically calls for retransmission of the entire message
 Retransmission is called as “ARQ”  Automatic Repeat Request (or) Automatic Retransmission Request
 Most reliable method for retransmission
 If short message are sent the occurrence during transmission is small
 But short messages require more acknowledgements and line turn around than long messages
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Acknowledgement:
 When the recipients of data sends a short message back to the sender acknowledging receipt of the last
transmission
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 Acknowledgement may be positive (successive transmission) or negative (unsuccessful transmission)

acknowledgement
Line turn arounds:
 Receiver will be a transmit station such as when acknowledgements are sent or when retransmission are
sent in response to a negative acknowledgements
 For long message, error occur is high but less turnaround time is needed
Note: Message size between 256 and 512 characters long are the optimum size for ARQ error correction
TYPES OF ARQ:
1. Discrete ARQ
2. Continuous ARQ
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1. Discrete ARQ:
 Uses acknowledgements to indicate the successful or unsuccessful reception of data (Positive or negative
acknowledgement)
 If sending station does not receive any acknowledgement after a predetermined length of time (“time
out”) it retransmits the message called “retransmission after time out”.
2. Continuous ARQ:
 This is used when messages are divided into smaller block or frames that are sequentially numbered and
transmitted, without waiting for acknowledgements within blocks.
 Continuous ARQ allows the destination station to asynchronously request the retransmission of a specific
data and receiver will reconstruct the entire message once all frames have been transmitted also called as
“selective ARQ”
 Since it can be used to all, for a retransmission of an entire message or only a portion of message
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Forward error correction:
 Only error correction schemes that actually corrects and detects transmission error when they are
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received without requiring a retransmission
 Redundant bits are added to the message before transmission, when an error has occurred, redundant
bits will determine which bit is in error
 Correcting is complementing the bits
 FEC limited to one, two or three bit error
 FEC ideally suited for data communication
 When acknowledgements are not possible, used for simple transmission
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 FEC is used to eliminate the time wasted for retransmission
Popular method – “Hamming Code”
 Named after a mathematician Richard W. Hamming, working in Bell Telephone laboratories
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 Correct only single bit errors, cannot correct multiple error or burst errors and it cannot identify the error
that occur in Hamming bits itself
 Hamming code as with all FEC codes requires the addition of overhead in the messages
(acknowledgement and time turn around) increasing the length of transmission
 Hamming bits (called error bits) are inserted into a character at random locations
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 Combination of data bits and Hamming bits are called “Hamming code”
 To calculate that number of redundant Hamming bits necessary for a given character length is given as
m + n  total number of bits in one data units
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m  character bits
n  Hamming bits
 Since Hamming bits are able to identify which bit is in error, n- Hamming bits are able to indicate at least
m+n+1 codes differently
 Out of m+n codes, one code indicates no error and remaining one code indicates which bit is in error
 Since n bits can produce 2n different codes, 2n must be greater than or equal to m+n+1. So number of
Hamming bits is determined by
2n ≥ m+n+1
m number of bits in each character data
n  number of Hamming bits
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Data unit comprised of m character and n Hamming bits:
Pbm 1: For 12 bit data string 101100010010, determine the number of Hamming bits arbitrarily place the
Hamming bits into data string, logic condition of each Hamming bit, single bit transmission error.
Solution
m = 12
2n ≥ m+n+1
So, n=5
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25 = 32 ≥ 12+5+1 =18
Hamming bits n=5
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So total number of bits 12 + 5 = 17
Data bits Hamming bits
Arbitrarily placing Hamming bits

17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1
H 1 0
R 1 H 1 0 0 H H 0 1 0 H 0 1 0
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Logic condition of Hamming bits, express all bit positions that contain logic 1 as a five bit binary number and
XOR them
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Hamming bit = 10110

b17 =1, b13 =0, b9 =1, b8 =1, b4 =1,
so 17 bit hamming code is
H H H H H
1 1 0 1 0 1 0 0 1 1 0 1 0 0 0 1 0
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To detect errors:
1. Assume a single bit error is occurred
2. Assume a data is received with error in bit 14
3. To determine the bit patter in error, extract the hamming bits and XOR them with the binary code for logic 1
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3.6 DATA COMMUNICATION HARDWARES
Data communications, or telecommunications, is the electronic collection and distribution of information
between two points. Data communications hardware is used to transmit digital information between terminals and
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computers, or between computers and other computers.
Examples:
 Host computer Concentrators
 Front-end processor Hubs
 Multiplexer Network Interface Cards
 Router Modem
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3.6.1 STANDARD SERIAL INTERFACES

 A standard serial interface is used to interconnect DTE and DCE for an orderly flow of data.
 It coordinates,
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1. Flow of data
2. Control signals, and
3. Timing information between DTE and DCE.
 Every company that manufactured data communication equipment used a different interface configuration,
before serial interfaces were standardized.
 A serial interface standards should provide the following,
1. A specific range of voltages for transmit and receive signals.
2. Limitations for the electrical parameters of the transmission line including
source and load impedance, cable capacitance.
3. Standard cable and cable connector.
4. Functional description of each signal in the interface.
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Here, we are going to discuss in depth about.
1. RS-232 serial interface standard,
a. RS-232 electrical equivalent circuit
b. RS-232 functional description
c. RS-232 signals, and
d. RS-232 Asynchronous data transmission
1. RS-232 SERIAL INTERFACE STANDARD
 The RS-232 interfaces mechanical specification specifies a cable with two connectors.
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RS-232 serial interface connector
 From the fig, RS-232 cable is a sheath containing 25 wires with a DB25P-Compatible male connector as
plug on one end which connects to the DTE and a DB 25S compatible female connector as receptacle on
the other end which connects to the DCE.
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EIA-561 modular connector

 The eight pin EIA561 modular connector is designed exclusively for dial up telephone connections, which is
shown in fig.
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Limitations:
 Even though Rs-232is a cable with 2 connectors, the standards specifies limitations on the voltage levels
that the DTE and DCE can output onto or receive from the cable.
 To overcome this voltage leveling circuits are used, which converts the internal voltage levels from the
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DTE and DCE to RS-232 values.

Voltage leveler:
 It is called a driver (or) transmitter if it output signals onto the cable, and
 A terminator (or) receiver if it accepts signals from the cable.
RS-232 ELECTRICAL EQUIVALENT CIRCUIT
 The following figure shows the RS-232 electrical equivalent circuit with driver and the terminator.
 The maximum length of RS-232 cable is approximately 50 feet, with a bit rate of 20kbps.
RS-232 FUNCTIONAL DESCRIPTION
The RS-232 interface cable pins are categorized as five groups namely,
A – Ground (signal and chassis)
B – Data (transmit and receive)
C –Control (handshaking and diagnostic)
D –Timing (clocking signals), and
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E – Secondary channel.
PINS SPECIFICATIONS
 The pins 9,10,11,18 and 25 are unassigned.
 Pins 1 and 7 are grounds.
 Pins 2,3,14 and 16 are data pins.
 Pins 15,17 and 24 are timing pins, and
 All other pins are used for control or handshaking signals.
USES OF PINS:
 PINS 1 through 8 are used with both asynchronous and synchronous modems.
 Pins 5, 17 and 24 are used only with synchronous modems.
 Pins 12,13,14,16 and 19 are used only when the DCE is equipped with a secondary data channel.
 Pins 20 and 22 are used exclusively when interfacing a DTE to a modem.
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CHANNELS OF RS-232:
Two types of channels are,
o Primary data channel, and
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o Secondary data channel.
The following table shows the EIA signal designations with direction of propagation plus its functions.
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PINS DESIGNATIONS AND DIRECTION OF PROPOGATION
PIN PIN NAME DIRECTION OF FUNCTIONS
NO. PROPOGATION
1 Protective ground (frame ground or None Protection against accidental electrical shock
chassis ground)
2 Transmit data (send data) DTE or DCE Transmitting the data from the DTE to DCE as well as
receiving the data from DCE to DTE.
3 Receive data DCE to DTE Receive data for DTE and transmit data for DCE.
4 Request to send DTE to DCE Request permission from the DCE to transmit data on
the primary data channel.
5 Clear to send DCE to DTE Handshaking from DCE to DTE in response to an active
condition on RTS.
6 Data set ready (modem ready) DCE to DTE Used by DCE to indicate DTE about the availability of
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the communication channel.
7 Signal ground (reference ground) None Used as a signal reference for all data, control and
timing signals.
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8 Receive line signal detect (carrier DCE to DTE Used by DCE to signal the DTE about the reception of
detect or data carrier detect) valid analog carrier.
9 Unassigned None Often used for testing +12V dc.
10 Unassigned None Often used for testing -12V dc.
11 Unassigned None Used by modem to signal the DTE when the modem is
self adjusting its internal equalizers because error
performance is assumed to be poor.
12
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Secondary receive line signal detect DCE to DTE Same as pin8 (RLDs)expect for secondary data channel.
13 Secondary cleat to send DCE to DTE Send from DCE to DTE as a response to the secondary
request to send signal(pin 19)
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14 Secondary transmit data (secondary DTE to DCE Diagnostic data transmission from DTE to DCE, which is
send data) enabled by active conditions on SCTS.
15 Transmit signal element timing- DCE to DTE The transmit clocking signal is sent from DCE to DTE on
DCE(serial clock transmit-DCE) this pin.
16 Secondary receive data DCE to DTE Diagnostic data transmission from DCE to DTE, which is
enabled by an active conditions on secondary receive
line signal detect (srlsd).
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17 Receive signal element timing (serial DCE to DTE The check information recovered by DCE is sent to the
clock receive) DTE on this pin, when the synchronous modems are
used.
18 Unassigned None Local loopback(LL) signal
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19 Secondary request to send DCE to DTE Used by DTE to bid for the secondary data channel
from the DCE.
20 Data terminal ready DCE to DTE Used by the DTE to signal DCE about the availability of
DTE.
21 Signal quality detect DCE to DTE Used by the DCE to signal DTE about the quality of the
received analog carrier.
22 Ring indicator DCE to DTE Used by DCE to inform DTE about the incoming call
ready.
23 Data signal rate selector DCE to DTE Used by DTE to select one of the two transmission bit
rates when the DCE is equipped to offer two rates.
24 Transmit signal element timing- DTE to DCE Used to transmit clock signal from the DTE to DCE
DTE(serial clock transmit-DTE) when synchronous modems are used.
25 Unassigned None Used to indicate the local loopback mode of DCE to
DTE.
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RS-232 SIGNALS
 The following diagram shows the timing diagram for the transmission of one asynchronous data character
over the RS-232 interface.
 The character is comprised of one start bit, one stop bit, seven ASCII character bits, and one even parity
bit.
 The transmission rate is 100 bps and a time of one bit is 1ms, and the total time to transmit one ASCII
character is 10ms.
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RS-232 data timing diagram
RS-232 ASYNCHRONOUS DATA TRANSMISSION

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The functional block diagram for the drivers and terminators needed for the transmission of asynchronous
data over the RS-232 interface between a DTE and a DCE is shown below,
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Functional block diagram for the driver and terminators for transmission of asynchronous data over the RS-
232 interface between a DTE and a CDE(modem)- transmit circuits
Functional block diagram for the driver and terminators for transmission of asynchronous data over the RS-
232 interface between a DTE and a CDE(modem)- receive circuits
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Only first eight pins of the interface are required. They are,
 Signal ground and chassis ground,
 Transmit data receive data,
 Request to send,
 Clear to send,
 Data set ready, and
 Receive line signal detect.
Timing diagram
Transmitter
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Typical timing diagram for control and data signals for a asynchronous data transmission over the RS-232 interface
between a DTE and a DCE(modem)- transmit timing diagram
The transmission of asynchronous data over an RS-232 interface is possible with the following parameters as,
 Modern RTS-CTS delay = 50 ms.
 DTE primary data message length=100ms
 Modem training sequence = 50 ms
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 Propagation time=10 ms,

 Modern RLSD turn-off delay =5ms.
The transmitter timing diagram for control and data signals for a typical asynchronous data transmission is
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given in fig.
Procedure:
 DTE enables request to send (RTS)(t=0), when it is ready to transmit data on the primary channel, then CTS
goes active.
 The modem outputs an analog carrier, which is modulated by a training sequence during 50 ms RTS/CTS
delay.
 The analog carrier is used to initialize the communication channel and distinct receive modem.
 The transmit data line (TD) is enabled, after the RTS/CTS delay; the DTE begins transmitting user data.
 After the transmission is complete (t=150ms), RTS goes low, which turns off the modems analog carrier.
 The modem acknowledges the inactive condition of RTS with an inactive condition on CTS.
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Receiver:
 The receive modem receives a valid analog carrier after a propagation delay (Pd) of 10ms and enables
RLSD.
 The DCE sends an active RLSD signal across the RSS-232 interface cable to the DT, which enables the
receive data line (RD).
 The first 50ms of the received data is the training sequence of logic 1’s which is ignored by the DTE.
 At t=60ms,the DTE identifies the beginning of the user data by recognizing the transition from high to low.
 The DCE holds RLSD active for a predetermined RLSD-turn off delay time (10ms), to ensure that all the
data received have been demodulated and outputted onto RS-232 interface, at the end of the message.
 The timing diagram is shown in figure as,
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Typical timing diagram for control and data signals for asynchronous data transmission over the RS-232
interface between a DTE and DCE(modem)- receive timing diagram
3.6.2 PARALLEL INTERFACES
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Definition :
Parallel interface allows the user to transfer data between two devices with 8 or more bits at a same time (or)
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simultaneously. It is also called as serial by world transmission.
Advantages:
 The speed of data transfer is increased when compared to serial transmission.
 Most computer terminals and peripheral equipments process data parallel.
 It will avoid the conversion from serial to parallel.
Disadvantage:
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 Number of wires and number of connection gets increases.

 It results in increase complexity.
Here, we are going to discuss about,
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 Centronics parallel interface, and

 IEEE 488 bus.
Centronics parallel interface
The Centronics parallel interface was originally designed to be used for transferring data between
microcomputer and a printer. It is shown as,
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Control data and status lines for Centronics parallel interface
Data lines:
 There are eight parallel data lines d0 to d7.
 All these are unidirectional lines taking data from computer to printer.
 Characters are transmitted from computer to printer in the form of seven-bit ASCII.
Control lines:
 There are four unidirectional control lines to send the control information from computer to printer.
 The control limes are,
 Strobe(STB),
 Auto feed(AF),
 Prime(PRIME), and
 Select(SLCTIN)
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Strobe(STB) line:
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 This is a negative edge triggered signal used by the computer to direct the printer to accept data.
Auto feed (AF) signal:
 It is an active low signal. It indicates whether the printer automatically performs a line feed function after
receiving carriage return character from the computer,
 The (AF) is low(active) then the printer responds to the carriage return character by performing a carriage
return character by performing a carriage return and a line feed.
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Prime(PRIME) control signal:
 This line is also called as initialize line. It is an active low line used by the computer to clear computer to
clear printers memory which includes print programming and print buffer.
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 When printer detects an active low condition on this line, it returns to its original position.
Select(SLCTIN):
 This line is not used very frequently. When used, a low signal on this should be seen by the before
accepting the data from the computer. Many printers connect this line permanently to ground.
Status line:
 There are unidirectional lines used for conveying the information from printer to computer.
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 Via these lines, the printer tells computer what the printer is doing.
 The status lines are, (ACK),BUSY,PO,SLCT and (ERROR)
Acknowledge (ACK)
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 The printer makes this line low after it receives an active signal from the computer. It tells the computer
that the printer is ready to receive another character from the printer.
Busy:
 It is an active high line which becomes 1 when the printer is busy and cannot accept data from the
computer. Following are the conditions that make a printer busy.
 Printer is inputting data from data lines or printer’s data buffer is full.
 The printer is printing or processing the data.
 The printer is off.
 The printer’s (ERROR) line is low.
Paper out (PO):
 It becomes active when the printer is out of paper, when PO line is activated to high the (ERROR) line also
is activated to low.
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Select(SLCT):
 This is an active high line which indicates whether the printer is selected or not. the printer activates this
line when it is on line.
Error (ERROR):
 This status line is an active low line. It is used to indicate printer problem. This line becomes active under
the following operating conditions.
 The printer is offline.
 The printer is out of paper.
 Some other problem.
Miscellaneous lines:
 These include the return path for (STB),(ACK), BUSY and(PRIME) lines. One more is the frame ground (FG)
which electrically connects the cassis of the computer and the printer to reduce the possibility shocks.
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 One more line is used for providing +5V ac to the printer from the computer
3.7 INTRODUCTION TO PULSE COMMUNICATION
Transmission of digital information through a physical medium is called digital transmission. To perform digital
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transmission, the message has to undergo either shift keying or pulse modulation. To transmit the modulated signal in
a wireless medium, shift keying methods are preferred. In the other hand, if the modulated signal has to be
transmitted using physical links such as optical fibers, cables and wires, pulse modulation is carried out.
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The encoded information is transmitted over a long distance through a physical medium.
Message Carrier
Shift keying Digital analog Modulation: transmission of digital data over a long
distance in wireless medium
Pulse modulation Analog Digital Encoding: encoded information is transmitted over
a long distance through a physical medium
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IN DIGITAL TRANSMISSION
PULSE MODULATION: If analog signals converted DIGITAL ENCODING: If digital signals are encoded to
to digital form prior to transmission and at some other digital format and transmitted.
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reception, it is converted back to analog form. Example: computer to computer communication in

computer networks
DISADVANTAGE:
 Digitally encoded analog signals require more bandwidth than simply transmitting the original analog
source.
 Since analog signals are converted to digital form during transmission and again converted back to analog
form at receiver, encoder and decoder circuits are required.
PULSE MODULATION
Message  analog (low frequency)
Carrier  digital (high frequency)
It is the process of varying the amplitude (or) width (or) position of a digital pulse waveform with respect to the
instantaneous amplitude of the analog message signal.
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There are three types of pulse modulation. They are
 PAM
 PWM (or) PDM
 PPM
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AP
3.8 PULSE CODE MODULATION
PCM is an digital encoding technique, which involves three important steps to convert an analog signal directly
into a digital code. The following are the steps involved in PCM process. Pulse-code modulation (PCM) is a method
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used to digitally represent sampled analog signals, which was invented by Alec Reeves in 1937. It is the standard form
for digital audio in computers and various Blue-ray, Compact Disc and DVD formats, as well as other uses such as
digital telephone systems. A PCM stream is a digital representation of an analog signal, in which the magnitude of the
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analogue signal is sampled regularly at uniform intervals, with each sample being quantized to the nearest value
within a range of digital steps.
PCM streams have two basic properties that determine their fidelity to the original analog signal: the sampling
rate, which is the number of times per second that samples are taken; and the bit depth, which determines the
number of possible digital values that each sample can take.
3.8.1 SAMPLING
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Picking up of sample (instantaneous) values from analog signal at regular time intervals
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3.8.2 QUANTIZING:
Assigning discrete values to the sampled signal amplitude.
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3.8.3 ENCODING:
Converting the quantized values to binary bit streams.
0  000 -A  100
A  001 -2A 101
2A 010 -3A 110
3A 011
In signal coding the message signal is sampled and the amplitude of each sample is rounded off to nearest one of
a finite set of discrete levels such that both time and amplitude can be represented in discrete form. (discretizing the
amplitude levels is called quantization).
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In quantization, infinite levels are converted into finite levels.
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3.9 PCM SAMPLING

SAMPLING THEOREM:
A band limited signal of finite energy which has no frequency components higher than ‘w’ Hertz can be completely
recovered from its samples, if it is sampled at a rate of seconds (or if sampling frequency is 2w)
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(or)
If sampling frequency fs > 2fmax, the signal can be recovered completely from its samples.
fs ≥ 2fmax
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SAMPLING PROCESS:
The incoming analog message signal is sampled with a train of narrow rectangular pulses, so as to closely
approximate the actual signal from its samples (perfect reconstruction from samples). For perfect reconstruction,
fs > 2fmax
fs  frequency of rectangular pulses
fmax  maximum frequency present in analog message signal
TYPES OF SAMPLING:
 Natural sampling  sampler
 Flat top sampling  sample and hold circuit
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NATURAL SAMPLING FLAT TOP SAMPLING
Sampling pulse has a finite width T. it is also called as The hold circuit is also used. Therefore, it stores
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chopper sampling. Therefore, the sampled waveform the samples value for a duration of one pulse
appears to be clipped off version of original message period. Hence the top of sampled waveform
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version appears flat.
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FET  switch
Z1, z2  voltage follower amplifiers
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C  storage capacitor
SPECTRUM OF SAMPLED SIGNAL:
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ERRORS IN SAMPLING:
1. Round off error  when various samples are rounded off
2. Truncation error if the sampling is done for finite time
3. Aliasing  if sampling rate is not enough high (i.e.) fs < 2fmax
APERTURE ERROR:
The sampling process alters the frequency spectrum and introduces an error called aperture error. When the
amplitude of the sampled signal changes during the same pulse time, this error will occur. The magnitude of error
depends on how much the analog signal voltage changes while the sampling is being taken and width of sampling
pulse.
QUANTIZATION:
It is the process by which transforming the analog sample of original message signal into a discrete amplitude from
the finite set of possible amplitude.
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 Process of assigning values to sampled signal (discrete levels)
 Process of rounding off amplitudes of flat top samples to a manageable number of levels
3.10 SIMPLEX PCM SYSTEM
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The analog signal is sampled and converted then into a n-bit binary sequence called code. Code is used for
transmission. The basic operation of PCM involves, the PAM signals are quantized and then coded (assigning a binary
value), thus PCM is obtained. In PCM, both time and amplitude are represented as discrete values. PCM system
involves three processes. They are
 Sampling
 Quantizing
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 Encoding A to D conversion
 In PCM each code has same number of bits and same run length for transmission.
 It is the only digitally encoded modulation technique commonly used for transmission.
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 PCM is not actually a modulation technique, but it is an encoding technique through which analog signals are
assigned with fixed length (size) digital values.
 PCM is a binary system in which a pulse within a prescribed time period represents logic 1 and a lack of pulse
within a time period represents logic 0 (i.e.) for particular time interval,
 If pulse arrives  logic 1
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Band Pass Filter:
To limit the analog input frequency up to standard voice band (300Hz to 3000Hz). It is used to band restrict the
input signal.
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Sample And Hold Circuit:
It converts continuous time analog signal into discrete time signal.
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A/D:
Converts discrete time signal into digital bit stream (parallel).
Parallel To Serial Converter:
Converts n-bit parallel binary sequence into n-bit binary frame (serial). Serial n-bit digital frame  serial PCM code
Transmission path:
Physical medium such as wires, twisted pair, co-axial cables etc. in the transmission path, regenerative repeaters will
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be there.
Regenerative repeaters:
They receive, amplify or boost the signals and retransmit them. They are used for the purpose of reducing signal
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attenuation.
Serial To Parallel Converter:
Converts received serial PCM code frame into n-bit parallel stream.
D/A:
Digital to analog conversion by successive approximation method. It results in distant time pulses.
Hold Circuit:
Reconstruction of analog signal at output.
LPF:
Filters transmission noise and quantization noise and gives error free output
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3.11COMPARISON OF PAM, PTM and PCM:
PAM PTM/PWM/PDM PCM

The amplitude of digital pulse The spacing between each ON time In PCM, the analog message is directly
carrier is varied in accordance duration is varied with respect to the encoded in to n-bit serial binary code,
with the relative amplitude of amplitude variations of analog by sampling and quantization.
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analog message signal message signal.
Applications: Ethernet Applications: Local signaling ,DC Applications: Voice transmission in
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signaling, Driver applications motor speed control, Driver telephony, Data compression algorithms
applications
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UNIT-4 SOURCE AND ERROR CONTROL CODING

Entropy, Source encoding theorem, Shannon-Fano coding, Huffman coding, mutual information, channel capacity, channel coding
theorem, Error Control Coding, linear block codes, cyclic codes, convolution codes, Viterbi decoding algorithm.
4.1 INFORMATION THEORY
Information theory is needed to enable the communication system to carry information (signals) from sender
to receiver over a communication channel.
• It deals with mathematical modelling and analysis of a communication system
• Its major task is to answer to the questions of signal compression and data transfer rate.
Those answers can be found and solved by entropy and channel capacity.
Information is a measure of uncertainty. The less is the probability of occurrence of a certain message, the higher is
the information. Since the information is closely associated with the uncertainty of the occurrence of a particular
symbol, when the symbol occurs the information associated with its occurrence is defined as,
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1
I k  log ( )  - log(Pk )
Pk
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where Pk is the p robability of occurrence of sy mbol ' k'
and I k is the informatio n carried by sy mbol ' k'.
4.2 ENTROPY
Entropy is defined in terms of probabilistic behaviour of a source of information. In information theory the
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source output are discrete random variables that have a certain fixed finite alphabet with certain probabilities.Entropy
is an average information content for the given source symbol. (bits/message)
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K 1
1
H   pk log 2 ( )
k 0 pk
4.2.1 RATE OF INFORMATION:
If a source generates at a rate of ‘r’ messages per second, the rate of information ‘R’ is defined as the average number
of bits of information per second. ‘H’is the average number of bits of information per message. Hence
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R = rH bits/sec
4.3 SOURCE CODING
Source coding (a.k.a lossless data compression) means that we will remove redundant information from the
signal prior the transmission. Basically this is achieved by assigning short descriptions to the most frequent outcomes
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of the source output and vice versa.The common source-coding schemes are prefix coding, huffman coding, lempel-ziv
coding.
4.3.1 SOURCE CODING THEOREM
Source coding theorem states that the output of any information source having entropy H units per symbol
can be encoded into an alphabet having N symbols in such a way that the source symbols are represented by code
words having a weighted average length not less than H/log N.
Hence source coding theorem says that encoding of messages from a source with entropy H can be done,
bounded by the fundamental information theoretic limitation that the Minimum average number of symbols/message
isH/log N
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4.3.2 SHANNON-FANO CODING
In Shannon–Fano coding, the symbols are arranged in order from most probable to least probable, and then
divided into two sets whose total probabilities are as close as possible to being equal. All symbols then have the first
digits of their codes assigned; symbols in the first set receive "0" and symbols in the second set receive "1".
As long as any sets with more than one member remain, the same process is repeated on those sets, to determine
successive digits of their codes. When a set has been reduced to one symbol, of course, this means the symbol's code
is complete and will not form the prefix of any other symbol's code.
4.3.3 HUFFMAN CODING
Create a list for the symbols, in decreasing order of probability. The symbols with the lowest probability are
assigned a ‘0’ and a ‘1’. These two symbols are combined into a new symbol with the probability equal to the sum of
their individual probabilities. The new symbol is placed in the list as per its probability value. The procedure is
repeated until we are left with 2 symbols only for which 0 and 1 are assigned. Huffman code is the bit sequence
obtained by working backwards and tracking sequence of 0’s and 1’s assigned to that symbol and its successors.
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4.3.4 LEMPEL-ZIV CODING
A drawback of Huffman code is that knowledge of probability model of source is needed. Lempel-Ziv coding is
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used to overcome this drawback. While Huffman’s algorithm encodes blocks of fixed size into binary sequences of
variable length, Lempel-Ziv encodes blocks of varying length into blocks of fixed size. Lempel-Ziv coding is performed
by parsing the source data into segments that are the shortest subsequences not encountered before.
4.4 MUTUAL INFORMATION
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 Consider a communication system with a source of entropy H(X). The entropy on the receiver side be H(Y).
 H (X|Y) and H (Y|X) are the conditional entropies, and H(X,Y) is the joint entropy of X and Y.
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 Then the Mutual information between the source X and the receiver Y can be expressed as:
I(X,Y) = H(X) - H(X|Y)
 H(X) is the uncertainty of source X and H(X/Y) is the uncertainty of X given Y.
 Hence the quantity H(X) - H(X|Y) represents the reduction in uncertainty of X given the knowledge of Y. Hence
I(X, Y) is termed mutual information.
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4.5 CHANNEL CAPACITY

 Capacity in the channel is defined as a intrinsic ability of a channel to convey information.
 Using mutual information the channel capacity of a discrete memoryless channel is the maximum average
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mutual information in any single use of channel over all possible probability distributions.
 Thus Channel capacity C=max (I(X, Y)).
4.6 SHANNON’S CHANNEL CODING THEOREM
 The Shannon theorem states that given a noisy channel with channel capacity C and information transmitted
at a rate R, then if R < C there exist codes that allow the probability of error at the receiver to be made
arbitrarily small. This means that theoretically, it is possible to transmit information nearly without error at
any rate below a limiting rate, C.
 The converse is also important. If R>C, an arbitrarily small probability of error is not achievable. All codes will
have a probability of error greater than a certain positive minimal level, and this level increases as the rate
increases. So, information cannot be guaranteed to be transmitted reliably across a channel at rates beyond
the channel capacity.
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4.7 SHANNON-HARTLEY THEOREM OR INFORMATION CAPACITY THEOREM
 An application of the channel capacity concept to an additive white Gaussian noise channel with B Hz
bandwidth and signal-to-noise ratio S/N is the Information Capacity Theorem.
 It states that for a band-limited Gaussian channel operating in the presence of additive Gaussian noise, the
channel capacity is given by
C = B log2(1 + S/N)
Where C is the capacity in bits per second, B is the bandwidth of the channel in Hertz, and S/N is the signal-to-
noise ratio.
4.7.1 RATE DISTORTION FUNCTION
 The functions that relate the rate and distortion are found as the solution of the following minimization
problem.
| |
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∫ ∫ ∫ ∫ |
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 In the above equation, I(X,Y) is the Mutual information.
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A Plot of the Rate distortion function for Gaussian source

4.8 LOSSY SOURCE CODING
 Lossy source coding is the representation of the source in digital form with as few bits as possible while
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maintaining an acceptable loss of information.

 In lossy source coding, the source output is encoded at a rate less than the source entropy.
 Hence there is reduction in the information content of the source.
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Eg: It is not possible to digitally encode an analog signal with a finite number of bits without producing some
distortion
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4.9 ERROR CONTROL CODING
In all communication systems, errors are possible during transmission due to channel insufficiency and noise. That
errors caused by noise during transmission can be corrected at the receiver. The process of encoding for protection
against channel errors is called error-control coding. Error-control codes are used in a variety of applications, including
satellite communication, deep-space communication, mobile radio communication, and computer networking.
Forward Error Correction (FEC)
– Coding designed so that errors can be corrected at the receiver
– Appropriate for delay sensitive and one-way transmission (e.g., broadcast TV) of data
– Two main types, namely block codes and convolutional codes. We will only look at block codes
Block Codes
• Only binary data is considered
• Data is grouped into blocks of length k bits (dataword)
• Each dataword is coded into blocks of length n bits (codeword), where in general n>k
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• This is known as an (n,k) block code
• A vector notation is used for the datawords and codewords,
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• Dataword d = (d1d2….dk)
• Codeword c = (c1 c2……..cn)
• The redundancy introduced by the code is quantified by the code rate,
• Code rate = k/n
• i.e., the higher the redundancy, the lower the code rate
Hamming Distance:
• Error control capability is determined by the Hamming distance
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• The Hamming distance between two codewords is equal to the number of differences between them,
e.g.,
10011011 and 11010010 have a Hamming distance = 3
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Alternatively, can compute by adding codewords (mod 2)

=01001001 (now count up the ones)
• The maximum number of detectable errors is d 1
min
d min  1
• That is the maximum number of correctable errors is given by, t  
 2 
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Where dmin is the minimum Hamming distance between 2 code words and means the smallest integer
Linear Block Codes:
• As seen from the second Parity Code example, it is possible to use a table to hold all the codewords for a code
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and to look-up the appropriate codeword based on the supplied dataword

• Alternatively, it is possible to create codewords by addition of other codewords. This has the advantage that
there is now no longer the need to held every possible codeword in the table.
• If there are k data bits, all that is required is to hold k linearly independent codewords.
i.e., a set of k codewords none of which can be produced by linear combinations of 2 or more codewords in
the set.
• The easiest way to find k linearly independent codewords is to choose those which have ‘1’ in just one of the
first k positions and ‘0’ in the other k-1 of the first k positions.
• For example for a (7,4) code, only four codewords are required, e.g.,
1 0 0 0 1 1 0
0 1 0 0 1 0 1
0 0 1 0 0 1 1
0 0 0 1 1 1 1
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• So, to obtain the codeword for dataword 1011, the first, third and fourth codewords in the list are added
together, giving 1011010
• This process will now be described in more detail
• A (n,k) block code has code vectors
d=(d1d2….dk) and c=(c1 c2……..cn)
• The block coding process can be written as c=dG
where G is the Generator Matrix  a11 a12 ... a1n   a1 
a a22 ... a2 n  a 2 
G  21

 . . ... .   . 
   
ak1 ak 2 ... akn  a k 
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• Thus,
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k
c d a
i 1
i i
• ai must be linearly independent, i.e. Since codewords are given by summations of the ai vectors, then to avoid
2 datawords having the same codeword the ai vectors must be linearly independent.
• Sum (mod 2) of any 2 codewords is also a codeword, i.e. Since for datawords d1 and d2 we have;
So, d 3  d1  d 2
k k
R k k
c3   d 3i a i   (d1i  d 2i )a i  d1i a i   d 2i a i
c3  c1  c 2
i 1 i 1 i 1 i 1
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Error Correcting Power of LBC:

• The Hamming distance of a linear block code (LBC) is simply the minimum Hamming weight (number of 1’s or
equivalently the distance from the all 0 codeword) of the non-zero codewords
• Note d(c1,c2) = w(c1+ c2) as shown previously
• For an LBC, c1+ c2=c3
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• So min (d(c1,c2)) = min (w(c1+ c2)) = min (w(c3))

• Therefore to find min Hamming distance just need to search among the 2kcodewords to find the min
Hamming weight – far simpler than doing a pair wise check for all possible codewords.
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Linear Block Codes – example 1:

• For example a (4,2) code, suppose;
1 0 1 1
G  
0 1 0 1

a1 = [1011]
a2 = [0101] 1 0 1 1
• For d = [1 1], then;
 0 1 0 1
c 
_ _ _ _
 1 1 1 0
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Linear Block Codes – example 2:
A (6,5) code with
1 0 0 0 0 1
0 1 0 0 0 1

G  0 0 1 0 0 1 Is an even single parity code
 
0 0 0 1 0 1
0 0 0 0 1 1
Systematic Codes:
• For a systematic block code the data word appears unaltered in the codeword – usually at the start
• The generator matrix has the structure,
1 p1R 
P
0 .. 0 p1 1 p1 2 ..
0 1 .. 0 p2 1 p2 2 .. p2 R 
G   I | P 
.. .. .. .. .. .. .. .. 
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 
0 0 .. 1 pk 1 pk 2 .. pkR 
R=n–k
• P is often referred to as parity bits
• I is k*k identity matrix. Ensures data word appears as beginning of codeword P is k*R matrix.
Decoding Linear Codes:
• In this case received codeword is used as an address
• Example – Even single parity check code;
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AddressData
000000 0
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000001 1
000010 1
000011 0
……… .
• Data output is the error flag, i.e., 0 – codeword ok,
• If no error, dataword is first k bits of codeword
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• For an error correcting code the ROM can also store datawords
• Another possibility is algebraic decoding, i.e., the error flag is computed from the received codeword (as in the
case of simple parity codes)
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Parity Check Matrix:

• A linear block code is a linear subspace Ssub of all length n vectors (Space S)
• Consider the subset Snull of all length n vectors in space S that are orthogonal to all length n vectors in Ssub
• It can be shown that the dimensionality of Snull is n-k, where n is the dimensionality of S and k is the
dimensionality of Ssub .It can also be shown thatSnull is a valid subspace of S and consequently Ssub is also the
null space of Snull. Snull can be represented by its basis vectors. In this case the generator basis vectors (or
‘generator matrix’ H) denote the generator matrix for Snull - of dimension n-k = R
• This matrix is called the parity check matrix of the code defined by G, where G is obviously the generator
matrix for Ssub - of dimension k
• Note that the number of vectors in the basis defines the dimension of the subspace
• So the dimension of H is n-k (= R) and all vectors in the null space are orthogonal to all the vectors of the code
• Since the rows of H, namely the vectors bi are members of the null space they are orthogonal to any code
vector.So a vector y is a codeword only if yHT=0. Note that a linear block code can be specified by either G or
H
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Parity Check Matrix:
 b11 b12 ... b1n   b1 
b b22 ... b2 n   b 2 
H   21 
 . . ... .   . 
   
bR1 bR 2 ... bRn  b R 
R=n-k
• So H is used to check if a codeword is valid,
• The rows of H, namely, bi, are chosen to be orthogonal to rows of G, namely ai
• Consequently the dot product of any valid codeword with any bi is zero
This is so since,
k
and so, c   di a i
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b j .c  b j . d i a i   d i (a i .b j )  0
i 1 i 1
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• This means that a codeword is valid (but not necessarily correct) only if cHT = 0. To ensure this it is required
that the rows of H are independent and are orthogonal to the rows of G
• That is the bi span the remaining R (= n - k) dimensions of the code space
• For example consider a (3,2) code. In this case G has 2 rows, a1 and a2
• Consequently all valid codewords sit in the subspace (in this case a plane) spanned by a1 and a2
• In this example the H matrix has only one row, namely b1. This vector is orthogonal to the plane containing
the rows of the G matrix, i.e., a1 and a2
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• Any received codeword which is not in the plane containing a1 and a2 (i.e., an invalid codeword) will thus have
a component in the direction of b1 yielding a nonzero dot product between itself and b1.
Error Syndrome:
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• For error correcting codes we need a method to compute the required correction
• To do this we use the Error Syndrome, s of a received codeword, cr
s = crHT
• If cr is corrupted by the addition of an error vector, e, then
cr = c + e and
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s = (c + e) HT = cHT + eHT
s = 0 + eHT Syndrome depends only on the error
• That is, we can add the same error pattern to different codewords and get the same syndrome.
– There are 2(n - k) syndromes but 2n error patterns
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– For example for a (3,2) code there are 2 syndromes and 8 error patterns
– Clearly no error correction possible in this case
– Another example. A (7,4) code has 8 syndromes and 128 error patterns.
– With 8 syndromes we can provide a different value to indicate single errors in any of the 7 bit
positions as well as the zero value to indicate no errors
• Now need to determine which error pattern caused the syndrome
• For systematic linear block codes, H is constructed as follows,
G = [ I | P] and so H = [-PT | I]
where I is the k*k identity for G and the R*R identity for H
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Example, (7,4) code, dmin= 3
0 1 1 1 1 0 0 
1 1
0
0
1
0
0
0
0
0
1
1
0 1
 
H  - P T | I  1 0 1 1 0 1 0
G  I | P    1 1 0 1 0 0 1
0 0 1 0 1 1 0
 
0 0 0 1 1 1 1
Error Syndrome:– Example:
For a correct received codeword cr = [1101001]
In this case,
0 1
P
1
1 0 1
 
1 1 0
 
s  c r H T  1 1 0 1 0 0 11 1 1   0 0 0
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1 0 0
 
0 1 0
0 1
 0 
Hamming Codes:
• We will consider a special class of SEC codes (i.e., Hamming distance = 3) where,
– Number of parity bits R = n – k and n = 2R – 1
– Syndrome has R bits
– 0 value implies zero errors
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– 2R – 1 other syndrome values, i.e., one for each bit that might need to be corrected
– This is achieved if each column of H is a different binary word – remember s = eHT
• Systematic form of (7,4) Hamming code is,
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1 0 0 0 0 1 1 0 1 1 1 1 0 0 
0 1
G  I | P   
1 0 0 1 0  T

H  - P | I  1 0 1 1 0 1 0
0 0 1 0 1 1 0
  1 1 0 1 0 0 1
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0 0 0 1 1 1 1
1 1 1 0 0 0 0
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1 0 0 0 1 1 1 1
0
H  0 1 1 0 0 1 1
0 0 1 1 0
G 
0 1 0 1 0 1 0
  1 0 1 0 1 0 1
1 1 0 1 0 0 1
The original form is non-systematic,
• Compared with the systematic code, the column orders of both G and H are swapped so that the columns of H
are a binary count
• The column order is now 7, 6, 1, 5, 2, 3, 4, i.e., col. 1 in the non-systematic H is col. 7 in the systematic H.
Convolutional Code Introduction:
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• Convolutional codes map information to code bits sequentially by convolving a sequence of information bits
with “generator” sequences
• A convolutional encoder encodes K information bits to N>K code bits at one time step
• Convolutional codes can be regarded as block codes for which the encoder has a certain structure such that
we can express the encoding operation as convolution
• Convolutional codes are applied in applications that require good performance with low implementation cost.
They operate on code streams (not in blocks)
• Convolution codes have memory that utilizes previous bits to encode or decode following bits (block codes are
memory less)
• Convolutional codes achieve good performance by expanding their memory depth
• Convolutional codes are denoted by (n,k,L), where L is code (or encoder) Memory depth (number of register
stages)
• Constraint lengthC=n(L+1) is defined as the number of encoded bits a message bit can influence to the
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• Convolutional encoder, k = 1, n = 2, L=2
– Convolutional encoder is a finite state machine (FSM) processing information bits in a serial manner
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– Thus the generated code is a function of input and the state of the FSM
– In this (n,k,L) = (2,1,2) encoder each message bit influences a span of C=n(L+1)=6 successive output
bits = constraint length C
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– Thus, for generation of n-bit output, we require n shift registers in k = 1 convolutional encoders
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x ' j  m j3  m j2  m j

x '' j  m j3  m j1  m j
x ''' j  m j2  m j
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Here each message bit influences a span of C = n(L+1)=3(1+1)=6 successive output bits
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Representing convolutional codes- Code tree:
 x ' j  m j 2  m j 1  m j

 x '' j  m j 2  m j
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xout  x '1 x ''1 x '2 x ''2 x '3 x ''3 ...
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(n,k,L) = (2,1,2) encoder

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REPRESENTING CONVOLUTIONAL CODES COMPACTLY: CODE TRELLIS AND STATE DIAGRAM:

STATE DIAGRAM
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INSPECTING STATE DIAGRAM: STRUCTURAL PROPERTIES OF CONVOLUTIONAL CODES:

• Each new block of k input bits causes a transition into new state
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• Hence there are 2k branches leaving each state
• Assuming encoder zero initial state, encoded word for any input of k bits can thus be obtained. For instance,
below for u=(1 1 1 0 1), encoded word v=(1 1, 1 0, 0 1, 0 1, 1 1, 1 0, 1 1, 1 1) is produced
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- encoder state diagram for (n,k,L)=(2,1,2) code
- note that the number of states is 2L+1 = 8
Distance for some convolutional codes:
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THE VITERBI ALGORITHM:

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• Problem of optimum decoding is to find the minimum distance path from the initial state back to initial state
(below from S0 to S1). The minimum distance is the sum of all path metrics that is maximized by the correct
path.
ln p(y, x m )   j 0 ln p( y j | xmj )
• Exhaustive maximum likelihood method must search all the paths in phase trellis (2k paths emerging/
entering from 2 L+1 states for an (n,k,L) code)
• The Viterbi algorithm gets its efficiency via concentrating into survivor paths of the trellis
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THE SURVIVOR PATH:
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• Assume for simplicity a convolutional code with k=1, and up to 2k = 2 branches can enter each state in trellis
diagram
• Assume optimal path passes S. Metric comparison is done by adding the metric of S into S1 and S2. At the
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survivor path the accumulated metric is naturally smaller (otherwise it could not be the optimum path)
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• For this reason the non-survived path can be discarded -> all path alternatives need not to be considered
• Note that in principle whole transmitted sequence must be received before decision. However, in practice
storing of states for input length of 5L is quite adequate
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The maximum likelihood path:

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The decoded ML code sequence is 11 10 10 11 00 00 00 whose Hamming distance to the received sequence is 4 and
the respective decoded sequence is 1 1 0 0 0 0 0 (why?). Note that this is the minimum distance path.
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(Black circles denote the deleted branches, dashed lines: '1' was applied)
• In the previous example it was assumed that the register was finally filled with zeros thus finding the minimum
distance path.
• In practice with long code words zeroing requires feeding of long sequence of zeros to the end of the message
bits: this wastes channel capacity & introduces delay
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UNIT-5 MULTI-USER RADIO COMMUNICATION
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Advanced Mobile Phone System (AMPS) - Global System for Mobile Communications (GSM) - Code division multiple access
(CDMA) – Cellular Concept and Frequency Reuse - Channel Assignment and Handoff - Overview of Multiple Access Schemes -
Satellite Communication - Bluetooth.
5.1 INTRODUCTION TO MULTI USER RADIO COMMUNICATION
Multiuser communications refer to the simultaneous use of a communication Channel by a number of users.
Multiple access techniques are used to allow a large number of mobile users to share the allocated spectrum in the
most efficient manner. As the spectrum is limited, so the sharing is required to increase the capacity of cell or over a
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geographical area by allowing the available bandwidth to be used at the same time by different users. And this must
be done in a way such that the quality of service doesn’t degrade within the existing users.
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The main aim in Multi user communication system design is to be able to increase the capacity of the Channel in a
given bandwidth with a sufficient level of quality of service. There are several different ways to allow access to the
Channel. These include mainly the following:
1) Frequency division multiple-access (FDMA)
2) Time division multiple-access (TDMA)
3) Code division multiple-access (CDMA)
4) Space Division multiple access (SDMA)
FDMA, TDMA and CDMA are the three major multiple access techniques that are used to share the available
bandwidth in a wireless communication system. Depending on how the available bandwidth is allocated to the users
these techniques can be classified as narrowband and wideband systems.
5.1.1 NARROWBAND SYSTEMS
The term narrowband is used to relate the bandwidth of the single Channel to the expected coherence
bandwidth of the Channel. The available spectrum is divided in to a large number of narrowband Channels. The
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Channels are operated using FDD.
Channel bandwidth < Coherence bandwidth
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Coherence bandwidth:
It is the approximate maximum bandwidth or frequency interval over which two frequencies of a signal are
likely to experience comparable or correlated amplitude fading. In narrow band FDMA, a user is assigned a particular
Channel which is not shared by other users in the vicinity and if FDD is used then the system is called FDMA/FDD.
Narrow band TDMA allows users to use the same Channel but allocated a unique time slot to each user on the
Channel, thus separating a small number of users in time on a single Channel. For narrow band TDMA, there generally
are a large number of Channels allocated using FDD or TDD, each Channel is shared using TDMA. Such systems are
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called TDMA/FDD and TDMA/TDD access systems.
5.1.2 WIDEBAND SYSTEMS
In wideband systems, the transmission bandwidth of a single Channel is much larger than the coherence
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bandwidth of the Channel. Thus, multipath fading does not greatly affect the received signal within a wideband
Channel, and frequency selective fades occur only in a small fraction of the signal bandwidth.
Channel bandwidth > Coherence bandwidth
5.2 FREQUENCY DIVISION MULTIPLE ACCESS [FDMA]
This was the initial multiple-access technique for cellular systems in which each individual user is assigned a
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pair of frequencies while making or receiving a call as shown in Figure. One frequency is used for downlink and one
pair for uplink.
This is called frequency division duplexing (FDD). That allocated frequency pair is not used in the same cell or
adjacent cells during the call so as to reduce the co Channel interference. Even though the user may not be talking, the
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spectrum cannot be reassigned as long as a call is in place. Different users can use the same frequency in the same cell
except that they must transmit at different times.
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5.2.1 FEATURES OF FDMA
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The features of FDMA are as follows:
 The FDMA Channel can be used by only one user at a time. If an FDMA Channel is not in use, then it sits idle
and it cannot be used by other users to increase share capacity.
 After the assignment of the Channel to a particular User, The user is allowed to transmit and receive
simultaneously and continuously through the duplexed Channel.
 The bandwidths of FDMA systems are generally narrow.
 The symbol time is large compared to the average delay spread.
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 The complexity of the FDMA mobile systems is lower than that of TDMA mobile systems.
 FDMA requires tight filtering to minimize the adjacent Channel interference.
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A Guard band is an unused part of the radio spectrum between radio bands, for the purpose of preventing
interference. It is a narrow frequency range used to separate two wider frequency ranges to ensure that both can
transmit simultaneously without interfering with each other. A Guard band with BW fg always present in between two
FDMA Channels, for the purpose of preventing Interferences.
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5.2.2 FDMA/FDD IN AMPS
The first U.S. analog cellular system, AMPS (Advanced Mobile Phone System) is based on FDMA/FDD. A single
user occupies a single Channel while the call is in progress, and the single Channel is actually two simplex Channels
which are frequency duplexed with a 45 MHz split.
When a call is completed or when a handoff occurs the Channel is vacated so that another mobile subscriber
may use it. Multiple or simultaneous users are accommodated in AMPS by giving each user a unique signal. Voice
signals are sent on the forward Channel from the base station to the mobile unit, and on the reverse Channel from the
mobile unit to the base station. In AMPS, analog narrowband frequency modulation (NBFM) is used to modulate the
carrier.
5.2.3 FDMA/TDD IN CT2
Using FDMA, CT2 system splits the available bandwidth into radio Channels in the assigned frequency domain.
In the initial call setup, the handset scans the available Channels and locks on to an unoccupied Channel for the
duration of the call. Using TDD (Time Division Duplexing), the call is split into time blocks that alternate between
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transmitting and receiving.
5.2.4 FDMA AND NEAR-FAR PROBLEM
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The near-far problem is one of detecting or filtering out a weaker signal amongst stronger signals. The near-far
problem is particularly difficult in CDMA systems where transmitters share transmission frequencies and transmission
time. In contrast, FDMA and TDMA systems are less vulnerable. FDMA systems offer different kinds of solutions to
near-far challenge. Here, the worst case to consider is recovery of a weak signal in a frequency slot next to strong
signal. Since both signals are present simultaneously as a composite at the input of a gain stage, the gain is set
according to the level of the stronger signal; the weak signal could be lost in the noise floor. Even if subsequent stages
have a low enough noise floor to provide.
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5.3 TIME DIVISION MULTIPLE ACCESS [TDMA]
In digital systems, continuous transmission is not required because users do not use the allotted bandwidth all
the time. In such cases, TDMA is a complimentary access technique to FDMA. Global Systems for Mobile
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communications (GSM) uses the TDMA technique.

In TDMA, the entire bandwidth is available to the user but only for a finite period of time. In most cases the
available bandwidth is divided into fewer Channels compared to FDMA and the users are allotted time slots during
which they have the entire Channel bandwidth at their disposal, as shown in Figure.
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TDMA requires careful time synchronization since users share the bandwidth in the frequency domain. The
number of Channels is less, inter Channel interference is almost negligible. TDMA uses different time slots for
transmission and reception.
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This type of duplexing is referred to as Time division duplexing (TDD).The features of TDMA include the following:
1. TDMA shares a single carrier frequency with several users where each user makes use of non-overlapping time
slots.
2. The number of time slots per frame depends on several factors such as modulation technique, available
bandwidth etc.
3. Data transmission in TDMA is not continuous but occurs in bursts. This results in low battery consumption since
the subscriber transmitter can be turned OFF when not in use.
4. Because of a discontinuous transmission in TDMA the handoff process is much simpler for a subscriber unit,
since it is able to listen to other base stations during idle time slots.
5. TDMA uses different time slots for transmission and reception thus duplexers are not required.
6. TDMA has an advantage that is possible to allocate different numbers of time slots per frame to different users.
Thus bandwidth can be supplied on demand to different users by concatenating or reassigning time slot based
on priority.
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5.3.1 TDMA FRAME STRUCTURE

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In TDMA, the preamble contains the address and synchronization information that both the base station and
the mobiles use to identify each other. Trail bits signify the end of TDMA frame. Synchronization with receiver clock is
ensured by synchronization bits.
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Since there are significant delays between users, each user receives the reference burst with a different phase,
and its traffic burst is transmitted with a correspondingly different phase within the time slot. There is a guard
interval at the end of each time slot. As the transmission moves into the guard period, the mobile network adjusts the
timing advance to synchronize the transmission.
There is therefore a need for guard times to take account of this uncertainty. Each Time Slot is therefore
longer than the period needed for the actual traffic burst, thereby avoiding the overlap of traffic burst even in the
presence of these propagation delays.
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5.3.2TDMA/FDD IN GSM
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As discussed earlier, GSM is widely used in Europe and other parts of the world.GSM uses a variation of TDMA
along with FDD. GSM digitizes and compresses data, then sends it down a Channel with two other streams of user
data, each in its own time slot.
It operates at either the 900 MHz or 1800 MHz frequency band. Since many GSM network operators have
roaming agreements with foreign operators, users can often continue to use their mobile phones when they travel to
other countries.
5.3.3 TDMA/TDD IN DECT
DECT is a pan European standard for the digitally enhanced cordless telephony using TDMA/TDD. DECT
provides 10 FDM Channels in the band 1880-1990 MHz each Channel supports 12 users through TDMA for a total
system load of 120 users.
DECT supports Hand-Over. Users can roam over from cell to cell as long as they remain within the range of the
system. DECT antenna can be equipped with optional spatial diversity to deal with multipath fading.
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5.4 SPREAD SPECTRUM MULTIPLE ACCESS [SSMA]
Spread spectrum multiple access (SSMA) uses signals which have a transmission bandwidth whose magnitude
is greater than the minimum required RF bandwidth. A pseudo noise (PN) sequence converts a narrowband signal to a
wideband noise like signal before transmission. SSMA is not very bandwidth efficient when used by a single user.
However since many users can share the same spread spectrum bandwidth without interfering with one another,
spread spectrum systems become bandwidth efficient in a multiple user environment.
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There are two main types of spread spectrum multiple access techniques:
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Frequency Hopped Multiple Access (FHMA)
2. Direct sequence multiple access (DSMA) or Code division multiple access (CDMA).
5.4.1 FREQUENCY HOPPED MULTIPLE ACCESS (FHMA)
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This is a digital multiple access system in which the carrier frequencies of the individual users are varied in a
pseudo random fashion within a wideband Channel. The digital data is broken into uniform sized bursts which are then
transmitted on different carrier frequencies.
5.4.2 CODE DIVISION MULTIPLE ACCESS
In CDMA, the same bandwidth is occupied by all the users, however they are all assigned separate codes,
which differentiates them from each other (shown in Figure). CDMA utilize a spread spectrum technique in which a
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spreading signal (which is uncorrelated to the signal and has a large bandwidth) is used to spread the narrowband
message signal.
Direct Sequence Spread Spectrum (DS-SS)
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This is the most commonly used technology for CDMA. In DS-SS, the message signal is multiplied by a Pseudo
Random Noise Code.
Each user is given his own code-word which is orthogonal to the codes of other users and in order to detect
the user, the receiver must know the codeword used by the transmitter. There are, however, two problems in such
systems which are discussed below.
5.4.3 CDMA/FDD IN IS-95
In this standard, the frequency range is: 869-894 MHz (for Rx) and 824-849 MHz (for Tx). In such a system,
there are a total of 20 Channels and 798 users per Channel. For each Channel, the bit rate is 1.2288 Mbps. For
orthogonality, it usually combines64 Walsh-Hadamard codes and an m-sequence.
5.4.3 SELF-INTERFERENCE PROBLEM OR SELF JAMMING
In CDMA, self-interference arises from the presence of delayed replicas of signal due to multipath. The delays
cause the spreading sequences of the different users to lose their orthogonality, as by design they are orthogonal only
at zero phase offset.
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Hence in dispreading a given user’s waveform, nonzero contributions to that user’s signal arise from the
transmissions of the other users in the network. This is distinct from both TDMA and FDMA, wherein for reasonable
time or frequency guard-bands, respectively, orthogonality of the received signals can be preserved.
5.4.4 NEAR-FAR PROBLEM
The near-far problem is a serious one in CDMA. This problem arises from the fact that signals closer to the
receiver of interest are received with smaller attenuation than are signals located further away. Therefore the strong
signal from the nearby transmitter will mask the weak signal from the remote transmitter.
In TDMA and FDMA, this is not a problem since mutual interference can be filtered. In CDMA, however, the
near-far effect combined with imperfect orthogonality between codes(e.g. due to different time sifts), leads to
substantial interference. Accurate and fast power control appears essential to ensure reliable operation of multiuser
DS-CDMA systems.
5.5 SPACE DIVISION MULTIPLE ACCESS [SDMA]
SDMA utilizes the spatial separation of the users in order to optimize the use of the frequency spectrum. A
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primitive form of SDMA is when the same frequency is reused in different cells in a cellular wireless network.
The radiated power of each user is controlled by Space division multiple access. SDMA serves different users
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by using spot beam antenna.
These areas may be served by the same frequency or different frequencies. However for limited co-Channel
interference it is required that the cells are sufficiently separated. This limits the number of cells a region can be
divided into and hence limits the frequency re-use factor.
A more advanced approach can further increase the capacity of the network. This technique would enable
frequency re-use within the cell. In a practical cellular environment it is improbable to have just one transmitter fall
within the receiver beam width.
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Therefore it becomes imperative to use other multiple access techniques in conjunction with SDMA. When
different areas are covered by the antenna beam, frequency can be re-used, in which case. TDMA or CDMA is
employed, for different frequencies FDMA can be used.
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5.6 COMPARISON OF VARIOUS MULTIPLE DIVISION TECHNIQUES

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5.7 ADVANCED MOBILE PHONE STANDARD [AMPS]
Advanced Mobile Phone System (AMPS) was an analog mobile phone system standard developed by Bell Labs,
and officially introduced in the Americas in 1983.It was the primary analog mobile phone system in North
America (and other locales) through the 1980s and into the 2000s. Later this technology is succeeded by Digital AMPS.
While Motorola was developing a cellular phone, from 1968-1983 Bell Labs worked out a system
called Advanced Mobile Phone System (AMPS), which became the first cellular network standard in the U.S. Motorola
and others designed and built the cellular phones for this and other cellular systems. AMPS is a first-
generation cellular technology that uses separate frequencies, or "Channels", for each conversation (Frequency-
division multiple access (FDMA)). It therefore required considerable bandwidth for a large number of users.
In general terms, AMPS was very similar to the older "0G" Improved Mobile Telephone Service, but used
considerably more computing power in order to select frequencies, hand off conversations to PSTN lines, and handle
billing and call setup.
"Back end" call setup functionality is a special feature of AMPS when compared to older generation systems.
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In AMPS, the cell centers could flexibly assign Channels to handsets based on signal strength, allowing the same
frequency to be re-used in various locations without interference. This allowed a larger number of phones to be
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supported over a geographical area. AMPS pioneers coined the term "cellular" because of its use of small hexagonal
"cells" within a system.
AMPS suffered from many weaknesses compared to digital technologies. As an analog standard, it was
susceptible to static and noise, and there was no protection from 'eavesdropping' using a scanner. AMPS cellular
service operated in the 850 MHz Cellular band. D-AMPS was a digital, 2G standard used mainly by AT&T
Mobility and U.S. Cellular in the United States, succeeded and replaced the analog AMPS.
Parameters Specifications
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Uplink frequencies 824-849 MHz
Downlink frequencies 869- 894 MHz
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Channel spacing 30 MHz
Modulation Frequency modulation
No .of Channels 832
Multiple Access FDMA
5.8 GSM – GLOBAL SYSTEM FOR MOBILE COMMUNICATIONS
5.8.1 Historical Overview
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The Global System for Mobile communications (GSM) is by far the most successful mobile Communication
system worldwide. Its development started in 1982. The European Conference of Postal and Telecommunications
Administrations (CEPT), predecessor of the European Telecommunication. Standards Institute (ETSI), founded the
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Groupe Speciale Mobile, with the mandate to develop proposals for a pan-European digital mobile communication
system. Two goals were supposed to be achieved:
 First, a better and more efficient technical solution for wireless communications – it had become
evident at that time that digital systems would be superior in respect to user capacity, ease of use,
and number of possible additional services compared with the then-prevalent analog systems.
 Second, a single standard was to be realized all over Europe, enabling roaming across borders.
This was not possible before, as incompatible analog systems were employed in different
countries.
In the following years, several companies developed proposals for such a system. These proposals covered
almost all possible technical approaches in different technical areas. For multiple access, Time Division Multiple Access
(TDMA), Frequency Division Multiple Access (FDMA), and Code Division Multiple Access (CDMA) were suggested. The
proposed modulation techniques were Gaussian Minimum Shift Keying (GMSK), 4-Frequency Shift Keying (4-FSK),
Quadrature Amplitude Modulation (QAM), and Adaptive Differential Pulse Modulation (ADPM). Transmission rates
varied from 20 kbit/s to 8Mbit/s.
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In the early 1990s, it was realized that GSM should have functionalities that had not been included in the
original standard. Therefore, the so-called phase-2 specifications, which included these functions, were developed
until 1995. Further enhancements, which include packet radio - General Packet Radio Service (GPRS), and the more
efficient modulation of Enhanced Data rates for GSM Evolution (EDGE), have been introduced since then. Because of
these extensions GSM is often referred to as the 2.5th generation system, as its functionalities are beyond those of a
second-generation system, but do not enable all third-generation functionalities.
The success of GSM exceeded all expectations. Though it was originally developed as a European system, it
has spread all over the world in the meantime. Australia was the first non-European country that signed the basic
agreement (Memorandum of Understanding (MoU)). Since then, GSM has become the worldwide mobile
communication standard,1 with a number of subscribers that approached 3.5 billon in 2009. A few exceptions remain
in Japan and Korea, where GSM was never implemented. In the U.S.A., GSM was competing with the CDMA-based
Interim Standard-95 (IS-95) system. In contrast to most countries where spectral licenses were provided on condition
that the network operator would use GSM, the licenses in the U.S.A. were sold without requiring companies to
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implement a specific system. In 2009, there were two major operators offering GSM based services, while another two
were using rival technologies.
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There are three versions of GSM, each using different carrier frequencies. The original GSM system uses
carrier frequencies around 900 MHz. GSM1800, which is also called Digital Cellular System at the 1800-MHz band
(DCS1800), was added later to support the increasing numbers of subscribers. Its carrier frequencies are around 1,800
MHz, the total available bandwidth is roughly three times larger than the one around 900 MHz, and the maximal
transmission power of MSs is reduced. Apart from this, GSM1800 is identical to the original GSM. Thus, signal
processing, switching technology, etc. can be reused without changes. The higher carrier frequency, which implies a
smaller path gain, and reduced transmission power reduce the sizes of the cells significantly.
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5.8.2 SYSTEM OVERVIEW
A GSM system consists essentially of three parts – namely, the Base Station Subsystem (BSS), the Network and
Switching Subsystem (NSS), and the Operation Support System (OSS).
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5.8.2.1 BASE STATION SUBSYSTEM

The BSS consists of Base Transceiver Stations (BTSs) and the Base Station Controllers (BSCs). The BTS
establishes and maintains the connection to the MSs within its cell. The interface between the MS and the BTS is the
air interface, called the Um-interface in the GSM context. The BTS hosts, at a minimum, the antennas and the Radio
Frequency (RF) hardware of a BS, as well as the software for multiple access. Several – or, rarely, one – BTSs are
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connected to one BSC; they are either collocated, or connected via landline, directional microwave radio links, or
similar connections. The BSC has control functionality. It is, among other things, responsible for HandOver HO)
between two BTSs that are connected to the same BSC. The interface between BTS and BSC is called the Abis-
interface. In contrast to the other interfaces, this interface is not completely specified in the standard.2 Distribution of
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the functionalities between BTS and BSC may differ depending on the manufacturer. In most cases, one BSC is
connected to several BTSs. Therefore, it is possible to increase the efficiency of implementation by shifting
as much functionality as possible to the BSC. However, this implies increased signaling traffic on the link between the
BTS and the BSC, which might be undesirable (remember that these links are often rented landline connections). In
general, the BSS covers a large set of functionalities. It is responsible for Channel assignment, maintenance of link
quality and handover, power control, coding, and encryption.
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5.8.2.2 NETWORK AND SWITCHING SUBSYSTEM
The main component of the NSS is the Mobile-services Switching Center (MSC), which controls the traffic
between different BSCs. One function of the MSC is mobility management, which comprises all the functions that are
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necessary to enable true mobility for subscribers. To give but one example, one function of the MSC is the
management of HOs that occur when an MS is leaving the area of one BSC and moving into the area covered by
another BSC. Other functions are the so-called paging and location update. All interactions with other networks –
especially the landline Public Switched Telephone Network (PSTN) – are also performed by the MSC.
The NSS includes some databases, too. The Home Location Register (HLR) contains all the numbers of the
mobile subscribers associated with one MSC and information about the location of each of these subscribers. In the
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event of an incoming call, the location of the desired subscriber is looked up in the HLR and the call is forwarded to
this location. Therefore, we can conclude that from time to time a traveling MS has to send updates of its location to
its HLR. The Visitor Location Register (VLR) of one MSC contains all the information about mobile subscribers from
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other HLRs that are in the area of this MSC and are allowed to roam in the network of this MSC. Furthermore, a
temporary number will be assigned to the MS to enable the “host” MSC to establish a connection to the visiting MS.
The Authentication Center (AUC) verifies the identity of each MS requesting a connection. The Equipment Identity
Register (EIR) contains centralized information about stolen or misused devices.
5.8.2.3 OPERATING SUPPORT SYSTEM
The OSS is responsible for organization of the network and operational maintenance. More specifically, the
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OSS mainly covers the following functions:

1. Accounting: how much does a specific call cost for a certain subscriber? There are also plenty of different services
and features, from which each subscriber may choose an individual selection included in a specific plan. While this rich
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choice of services and prices is vital in the marketplace, the administrative support of this individualism is rather
complicated.
2. Maintenance: the full functionality of each component of the GSM network has to be maintained all the time.
Malfunctions may either occur in the hardware or in the software components of the system. Hardware malfunctions
are more costly, as they require a technician to drive to the location of the malfunction. In contrast, software is
nowadays administrated from a central location. For example, new versions of switching software can be installed in
the complete BSS from a central location, and activated all over the network at a specific time. Revision and
maintenance software often constitutes a considerable part of the overall complexity of GSM control software.
3. MS management: even though all MSs have to pass a type approval, it may happen that “bad apple” devices, which
cause system wide interference, are operating in the network. These devices have to be identified and their further
activities have to be blocked.
4. Data collection: the OSS collects data about the amount of traffic, as well as the quality of the links.
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5.8.3 THE AIR INTERFACE
GSM employs a combined FDMA/TDMA approach which further combines with Frequency Domain Duplexing
(FDD)
FDD
In the first GSM version, frequencies from 890 to 915MHz and from 935 to 960MHz were available. The lower
band is used for the uplink (connection from the MS to the BS). The upper band is used for the downlink. The
frequency spacing between the uplink and downlink for any given connection is 45MHz. Therefore, relatively cheap
duplex filters are sufficient for achieving very good separation between the uplink and downlink. For GSM1800, the
frequency ranges are 1,710–1,785MHz for the uplink, and 1,805–1,880MHz for the downlink. In North America, 1,850–
1,910MHz is used for the uplink and 1,930–1,990MHz for the downlink.
FDMA
Both uplink and downlink frequency bands are partitioned into a 200-kHz grid. The outer 100 kHz of each 25-
MHz band are not used, as they are guard bands to limit interference in the adjoined spectrum, which is used by other
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systems. The remaining 124 200-kHz subbands are numbered consecutively by the so-called Absolute Radio Frequency
Channel Numbers (ARFCNs).
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TDMA
Due to the very-bandwidth-efficient modulation technique (GMSK, see below), each 200-kHz subband
supports a data rate of 271 kbit/s. Each subband is shared by eight users. The time axis is partitioned into timeslots,
which are periodically available to each of the possible eight users. Each timeslot is 576.92 μs long, which is equivalent
to 156.25 bits. A set of eight timeslots is called a frame; it has duration of 4.615 ms. Within each frame, the timeslots
are numbered from 0 to 7. Each subscriber periodically accesses one specific timeslot in every frame on one frequency
subband. The combination of timeslot number and frequency band is called the physical Channel. The kind of data
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5.8.4 LOGICAL AND PHYSICAL CHANNELS
In addition to the actual payload data, GSM also needs to transmit a large amount of signaling information.
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These different types of data are transmitted via several logical Channels. The name stems from the fact that each of
the data types is transmitted on specific timeslots that are parts of physical Channels. The first part of this section
discusses the kind of data that is transmitted via logical Channels. The second part describes the mapping of logical
Channels to physical Channels.
Logical Channels
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Traffic Channels (TCHs)

Payload data are transmitted via the TCHs. The payload might consist of encoded voice data or “pure” data.
There is a certain flexibility regarding the data rate: Full-rate Traffic Channels (TCH/F) and Half-rate Traffic Channels
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(TCH/H). Two half-rate Channels are mapped to the same timeslot, but in alternating frames.
Full-Rate Traffic Channels
• Full-rate voice Channels: the output data rate of the voice encoder is 13 kbit/s.Channel coding increases the
effective transmission rate to 22.8 kbit/s.
• Full-rate data Channels: the payload data with data rates of 9.6, 4.8, or 2.4 kbit/s are encoded with Forward Error
Correction (FEC) codes and transmitted with an effective data rate of 22.8 kbit/s.
Half-Rate Traffic Channels
• Half-rate voice Channels: voice encoding with a data rate as low as 6.5 kbit/s is feasible. Channel coding increases
the transmitted data rate to 11.4 kbit/s.
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Broadcast Channels (BCHs)
BCHs are only found in the downlink. They serve as beacon signals. They provide the MS with the initial
information that is necessary to start the establishment of any kind of connection. The MS uses signals from these
Channels to establish a synchronization in both time and frequency. Furthermore, these Channels contain data
regarding, e.g., cell identity. As the BSs are not synchronized with respect to each other, the MS has to track these
Channels not only before a connection is established, but all the time, in order to provide information about possible
HOs.
Frequency Correction Channels (FCCHs)
The carrier frequencies of the BSs are usually very precise and do not vary in time, as they are based on
rubidium clocks. However, dimension considerations and price considerations make it impossible to implement such
good frequency generators in MSs. Therefore, the BS provides the MS with a frequency reference (an unmodulated
carrier with a fixed offset from the nominal carrier frequency) via the FCCH. The MS tunes its carrier frequency to this
reference; this ensures that both the MS and the BS use the same carrier frequency.
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Synchronization Channel (SCH)
In order to transmit and receive bursts appropriately, an MS not only has to be aware of the carrier
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frequencies used by the BS but also of its frame timing on the selected carrier. This is achieved with the SCH, which
informs the MS about the frame number and the Base Station Identity Code (BSIC). Decoding of the BSIC ensures that
the MS only joins admissible GSM cells and does not attempt to synchronize to signals emitted by other systems in
the same band.
Broadcast Control Channel (BCCH)
Cell-specific information is transmitted via the BCCH. This includes, e.g., Location Area Identity (LAI),7
maximum permitted signal power of the MS, actual available TCH, frequencies of the BCCH of neighboring BSs that are
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permanently observed by the MS to prepare for a handover, etc.
Common Control Channels (CCCHs)
Before a BS can establish a connection to a certain MS, it has to send some signaling information to all MSs in
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an area, even though only one MS is the desired receiver. This is necessary because in the initial setup stage, there is
no dedicated Channel established between the BS and a MS. CCCHs are intended for transmission of information to all
MSs.
Paging Channel (PCH)
When a request – e.g., from a landline – arrives at the BS to establish a connection to a specific MS, the BSs
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within a location area send a signal to all MSs within their range. This signal contains either the permanent
International Mobile Subscriber Identity (IMSI) or the Temporary Mobile Subscriber Identity (TMSI) of the desired MS.
The desired MS continues the process of establishing the connection by requesting (via a Random Access Channel
(RACH)) a TCH. The PCH may also be used to broadcast local messages like street traffic information or commercials to
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all subscribers within a cell. Evidently, the PCH is only found in the downlink.
Random Access Channel (RACH)
A mobile subscriber requests a connection. This might have two reasons. Either the subscriber wants to
initiate a connection, or the MS was informed about an incoming connection request via the PCH. The RACH can only
be found in the uplink.
Access Grant Channel (AGCH)
Upon the arrival of a connection request via the RACH, the first thing that is established is a Dedicated Control
Channel (DCCH) for this connection. This Channel is called the Standalone Dedicated Control Channel (SDCCH). This
Channel is assigned to the MS via the AGCH, which can only be found in the downlink.
Dedicated Control Channels (DCCHs)
Similar to the TCHs, the DCCHs are bidirectional – i.e., they can be found in the uplink and downlink. They
transmit the signaling information that is necessary during a connection. As the name implies, DCCHs are dedicated to
one specific connection.
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Standalone Dedicated Control Channel (SDCCH)
After acceptance of a connection request, the SDCCH is responsible for further establishing this connection.
The SDCCH ensures that the MS and the BS stay connected during the authentication process. After this process has
been finished, a TCH is finally assigned for this connection via the SDCCH.
Slow Associated Control Channel (SACCH)
Information regarding the properties of the radio link is transmitted via the SACCH. This information need not
be transmitted very often, and therefore the Channel is called slow. The MS informs the BS about the strength and
quality of the signal received from serving BSs and neighboring BSs. The BS sends data about the power control
and runtime of the signal from the MS to the BS.
Fast Associated Control Channel (FACCH)
The FACCH is used for HOs that are necessary for a short period of time; therefore, the Channel has to be able
to transmit at a higher rate than the SACCH. Transmitted information is similar to that sent by the SDCCH. The SACCH
is associated with either a TCH or a SDCCH; the FACCH is associated with a TCH.
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5.9 CELLULAR CONCEPTS
Radio telephone system should be structured to achieve high capacity with limited radio spectrum while at the
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same time covering very large areas.
Older System:
• Achieve a large coverage area by using a simple, high powered transmitter.
• Put BS on top of mountains or tall towers, so that it could provide coverage for a large area.
• The next BS was so far away that interference was not an issue.
• Severely limit the number of users that could communicate simultaneously.
• Noise-limited system with few users.
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• The Bell mobile system in New York City in the 1970s could only support a maximum of twelve
simultaneous calls over a thousand square miles.
The number of simultaneous calls a mobile wireless system can accommodate is essentially determined by the
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total spectral allocation for that system and the bandwidth required for transmitting signals used in handling a call.
Example- Using a typical analog system, each channel needs to have a bandwidth of around 25 kHz to enable sufficient
audio quality to be carried, as well as allowing for a guard band between adjacent signals to ensure there are no
undue levels of interference. Using this concept, it is possible to accommodate only forty users in a frequency band 1-
MHz wide. Even if 100 MHz were allocated to the system, this would enable only 4000 users to have access to the
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system. Today cellular systems have millions of subscribers, and therefore a far more efficient method of using the
available spectrum is needed.
 Cellular systems accommodate a large number of users over a large geographic area, within a
limited frequency spectrum.
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 High capacity is achieved by limiting the coverage of each base station transmitter to a small
geographic area called a cell so that the same radio channels may be reused by another base
station located some distance away.
 The coverage area is divided into many cells.
 Replace a single, high power transmitter (large cell) with many low power transmitters (small
cells) each providing coverage to only one cell area (a small portion of the service area).
 A sophisticated switching technique called a handoff enables a call to proceed un-interrupted
when the user moves from one cell to another.
The concept of cells was first proposed as early as 1947 by Bell Laboratories in the US, with a detailed proposal
for a High-Capacity Mobile Telephone System incorporating the cellular concept submitted by Bell Laboratories to the
FCC in 1971. The first AMPS system was deployed in Chicago in 1983.
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Basic cellular system
o Consist of mobile stations, base stations, and a mobile switching center (MSC).
o Mobile switching center (MSC)
o Sometimes called a mobile telephone switching office (MTSO)
o Coordinates the activities of all of the base stations
o Connect the entire cellular system to the PSTN.
o Accommodates all billing and system maintenance functions
o Each mobile communicates via radio with one of the base stations and may be handed-off to any
number of base stations throughout the duration of a call.
o Mobile station
o Contains a transceiver, an antenna, and control circuitry.
o Base stations
o Serve as a bridge between all mobile users in the cell and connects the simultaneous mobile calls via
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telephone lines or microwave links to the MSC.
o Consist of several transmitters and receivers which simultaneously handle full duplex communications
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o Generally have towers which support several transmitting and receiving antennas.
Communication between the base station and the mobiles is defined by a standard common air interface (CAl) that
specifies four different channels.
(a) Forward voice channels (FVC): voice transmission from the base station to mobiles
(b) Reverse voice channels (RVC): voice transmission from mobiles to the base station
(c) Forward control channels (FCC) and
(d)Reverse control channels (RCC)
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o Often called setup channels
o Involve in setting up a call and moving it to an unused voice channel.
o Initiate mobile calls
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o Transmit and receive data messages that carry call initiation and service requests, and are monitored
by mobiles when they do not have a call in progress.
o FCCs also serve as beacons which continually broadcast all of the traffic requests for all mobiles in the
system.
o Supervisory and data messages are sent in a number of ways to facilitate automatic channel changes
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and handoff instructions for the mobiles before and during a call.
5.10 FREQUENCY REUSE
Frequency reuse refers to the use of radio channels on the same carrier frequency to cover different areas
which are separated from one another by sufficient distances so that co-channel interference is not objectionable.
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Frequency reuse is employed not only in mobile-telephone service but also in entertainment broadcasting and most
other radio services. Most modern wireless systems are organized into geographic cells, each controlled by a base
station. Exceptions: Small-area systems such as local-area wireless networks and personal-area networks, Ad hoc and
wireless sensor networks.
Cellular radio systems rely on an intelligent allocation and reuse of channels throughout a coverage region.
Each cellular base station is allocated a group of radio channels to be used within a small geographic area called a cell.
Base stations in adjacent cells are assigned channel groups which contain completely different channels than
neighboring cells. The base station antennas are designed to achieve the desired coverage within the particular cell. By
limiting the coverage area within the boundaries of a cell, the same group of channel may be used to cover different
cells that are separated from one another by distances large enough to keep interference levels within tolerable limits.
The distance between two cells that an use the same frequency channels is called the reuse distance.
This reuse distance can be computed from link budgets. Why can't we use each frequency in each cell?
Same reason as why we separate the BSs of the older system far away from one another.
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Suppose user A is at the boundary of its assigned cell, so that distances from the useful BS and from a
neighboring BS are the same. If the neighboring BS transmits in the same frequency channel (in order to communicate
with user B in its own cell), then the signal-to-interference ratio (SIR) seen by user A is 0 dB.
So, reuse a frequency not in every cell, but only in cells that have a certain minimum distance from each other.
Using different allocated frequency bands, adjacent cells can overlap without causing interference.
We use hexagonal cell shape as a simplistic model of the radio coverage for each base station. Universally
adopted since the hexagon permits easy and manageable analysis of a cellular system. The actual radio coverage of a
cell is known as the footprint and is determined from field measurements or propagation prediction models.
In reality, it is not possible to define exactly the edge of a cell. The signal strength gradually reduces, and
towards the edge of the cell performance falls. As the mobiles themselves also have different levels of sensitivity, this
adds a further greying of the edge of the cell. It is therefore impossible to have a sharp cut-off between cells. In some
areas they may overlap, whereas in others there will be a hole in coverage. Although the real footprint is amorphous
in nature, a regular cell shape is needed for systematic system design and adaptation for future growth.
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Why hexagon? Adjacent circles cannot be overlaid upon a map without leaving gaps or creating overlapping regions.
When considering geometric shapes which cover an entire region without overlap and with equal area, there are three
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sensible choices: a square, an equilateral triangle, and a hexagon. A cell must be designed to serve the weakest
mobiles within the footprint, and these are typically located at the edge of the cell. For a given distance between the
center of a polygon and its farthest perimeter points, the hexagon has the largest area of the three. By using the
hexagon geometry, the fewest number of cells can cover a geographic region. Closely approximate a circular radiation
pattern which would occur for an omni directional base station antenna and free space propagation. They Permit easy
and manageable analysis of a cellular system.
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The above figure illustrates the concept of cellular frequency reuse, where cells labeled with the same letter use the
same group of channels. The frequency reuse plan is overlaid upon a map to indicate where different frequency
channels are used.
Cluster
 The total coverage area is divided into clusters.
 There can be no co-channel interference within a cluster.
 The number of cells in a cluster is called the cluster size. This number is denoted by N.
 The N cells collectively use the complete set of available frequencies.
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Capacity
Let S = the total number of available duplex radio channels for the system
k = the number of channels allocated to each cell (k < S)
N = cluster size
(a) If the S channels are divided among N cells into unique and disjoint channel groups which each have the same
number of channels,
S = kN
(b) If a cluster is replicated M times within the system, the total number of duplex channels, C, is given by
C = MS = MkN
(c) For a fixed total coverage area Atotal and the coverage area Acell of each cell, the number of cells in the system is
In which case,
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C x
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5.11 CHANNEL ASSIGNMENT
5.11.1 FIXED CHANNEL ASSIGNMENT

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5.11.2 DYNAMIC CHANNEL ASSIGNMENT

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5.12 HANDOFF
When a mobile moves into a different cell while a conversation is in progress, the MSC automatically transfers
the call to a new channel belonging to the new base station. This handoff operation not only involves identifying a new
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base station, but also requires that the voice and control signals be allocated to channels associated with the new base
station.
Processing handoffs is an important task in any cellular radio system. Many handoff strategies prioritize
handoff requests over call initiation requests when allocating unused channels in a cell site. Handoffs must be
performed successfully and as infrequently as possible, and be imperceptible to the users. In order to meet these
requirements, system designers must specify an optimum signal level at which to initiate a handoff. Once a particular
signal level is specified as the minimum usable signal for acceptable voice quality at the base station receiver (normally
taken as between –90 dBm and –100 dBm), a slightly stronger signal level is used as a threshold at which a handoff is
made. This margin, given by Δ = Pr handoff – Pr minimum usable, cannot be too large or too small. If Δ is too large,
unnecessary handoffs which burden the MSC may occur, and if Δ is too small, there may be insufficient time to
complete a handoff before a call is lost due to weak signal conditions. Therefore, Δ is chosen carefully to meet these
conflicting requirements.
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Dropped call event can happen when there is an excessive delay by the MSC in assigning a handoff or when
the threshold Δ is set too small for the handoff time in the system. Excessive delays may occur during high traffic
conditions due to computational loading at the MSC or due to the fact that no channels are available on any of the
nearby base stations (thus forcing the MSC to wait until a channel in a nearby cell becomes free).
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5.13 BLUETOOTH
Bluetooth is a standard used in links of radio of short scope, destined to replace wired connections between
electronic devices like cellular telephones, Personal Digital Assistants (PDA), computers, and many other devices.
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Bluetooth technology can be used at home, in the office, in the car, etc. This technology allows to the users
instantaneous connections of voice and information between several devices in real time. The way of transmission
used assures protection against interferences and safety in the sending of information. Between the principal
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characteristics, must be named the hardiness, low complexity, low consume and low cost. The Bluetooth is a small
microchip that operates in a band of available frequency throughout the world. Communications can realize point to
point and point multipoint.
How it works?
Every device will have to be equipped with a microchip (transceiver) that transmits and receives in the
frequency of 2.4 GHz that is available in the whole world (with some variations of bandwidth in different countries).
Besides the information, there are three channels of voice available.
The information can be exchanged to speeds of up to 1 megabit for second (2 megabits for second in the
Second Generation of this Technology). A scheme of “frequency hop” (jumps of frequency) allows to the devices to
communicate inclusive in areas where a great electromagnetic interference exists. Besides that is provided with
schemes of encryptation and check.
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Frequency Bands
The standard Bluetooth operates in the band of 2,4 GHz. Though worldwide, this band is available, the width
of the band can differ in different countries. This is the frequency of band of the scientific and medical industries 2.45
GHz (ISM*). The ranges of the bandwidth in The United States and Europe are between 2.400 to 2.483,5 MHz and it
covers part of France and Spain. The ranges of the bandwidth in Japan are between 2.471 to 2.497 MHz. So the system
can be used worldwide due to that the transmitters of radio covers 2.400 and 2.500 MHz and it is possible to select
the appropriate frequency. This ISM is opened for any system of radio and must take care of the interferences of
monitors for baby, the controls for doors of garages, the wireless telephones and the microwave ovens (the source
with higher interference).
Power
The equipments of transmission are qualified in 3 groups according to the level of power of emission, as we
can see below. The recipient equipment must possess a sensibility of at least 70 dBm, and the rate of admissible
mistake must be a minor or equal to 0,1 %.
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The chip is going to be incorporated in portable devices and powered by batteries, that´s why it must has a
very limited consumption of power (up to 97 % less than a mobile telephone). If the Bluetooth devices do not
exchange information, then they establish the way of "wait" to save energy, staying to the scout of messages. The
power of transmission that is used as specification is of 1 mW for a scope of 10 m, 100mW for a scope of up to 100 m.
Protocols
Different applications can operate under different sets of protocols; nevertheless, all of them have a link of
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information and a physical cap common Bluetooth. The figure below shows the set of protocols:
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MAIN APPLICATIONS
Bluetooth's applications are very varied and allow changing radically the form that the users interact with the
mobile telephones and other devices. Inside the field of the technology, the application is immediate because it allows
an easy, instantaneous communication, in any place and low cost. We cannot forget the impact in the way of
realizing the processes, on having replaced the conventional means and having made new business and applications
possible.
More prevalent applications of Bluetooth:
 Wireless control of and communication between a mobile phone and a hands-free headset. This was one of the
earliest applications to become popular.
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 Wireless networking between PCs in a confined space and where little bandwidth is required.
 Wireless communications with PC input and output devices, the most common being the mouse, keyboard and
printer.
 Transfer of files between devices with OBEX (a kind of communications protocol).
 Replacement of traditional wired serial communications in test equipment, GPS receivers, medical equipment,
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bar code scanners, and traffic control devices.
 For controls where infrared was traditionally used.
 Sending small advertisements from Bluetooth enabled advertising hoardings to other, discoverable, Bluetooth
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devices.
 Two seventh-generation game consoles, Nintendo's Wii and Sony's PlayStation 3 use
 Bluetooth for their respective wireless controllers.
 Dial-up internet access on personal computer or PDA using a data-capable mobile
 phone as a modem.
5.14 CODE DIVISION MULTIPLE ACCESS
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Spread-spectrum telecommunications is a technique in which a telecommunication signal is transmitted on

a bandwidth considerably larger than the frequency content of the original information. Frequency hopping is a basic
modulation technique used in spread spectrum signal transmission.
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In telecommunication and radio communication, spread-spectrum techniques are methods by which

a signal (e.g. an electrical, electromagnetic, or acoustic signal) generated with a particular bandwidth is deliberately
spread in the frequency domain, resulting in a signal with a wider bandwidth. These techniques are used for a variety
of reasons, including the establishment of secure communications, increasing resistance to
natural interference, noise and jamming, to prevent detection, and to limit power flux density (e.g.
in satellite downlinks).
Spread-spectrum telecommunications is a signal structuring technique that employs direct
sequence, frequency hopping, or a hybrid of these, which can be used for multiple access and/or multiple functions.
This technique decreases the potential interference to other receivers while achieving privacy. Spread spectrum
generally makes use of a sequential noise-like signal structure to spread the normally narrowband information signal
over a relatively wideband (radio) band of frequencies. The receiver correlates the received signals to retrieve the
original information signal. Originally there were two motivations: either to resist enemy efforts to jam the
communications (anti-jam, or AJ), or to hide the fact that communication was even taking place, sometimes called low
probability of intercept (LPI).
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Frequency-hopping spread spectrum (FHSS), direct-sequence spread spectrum (DSSS), time-hopping spread
spectrum (THSS), chirp spread spectrum (CSS), and combinations of these techniques are forms of spread spectrum.
Each of these techniques employs pseudorandom number sequences — created using pseudorandom number
generators— to determine and control the spreading pattern of the signal across the allocated bandwidth.
Frequency-hopping spread spectrum (FHSS) is a method of transmitting radio signals by rapidly switching
a carrier among many frequency channels, using a pseudorandom sequence known to both transmitter and receiver.
It is used as a multiple access method in the frequency-hopping code division multiple access (FH-CDMA) scheme.
In telecommunications, direct-sequence spread spectrum (DSSS) is a modulation technique. As with
other spread spectrum technologies, the transmitted signal takes up more bandwidth than the information signal that
modulates the carrier or broadcast frequency. The name 'spread spectrum' comes from the fact that the carrier signals
occur over the full bandwidth (spectrum) of a device's transmitting frequency. Certain IEEE 802.11 standards use DSSS
signaling.
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5.14.1 DIRECT SEQUENCE SPREAD SPECTRUM

Using CDMA the spread spectrum signal is obtained by modulating initial signal s(t) using specially formed
sequence of positive and negative pulses— pseudo-noise (PN). Such method is called direct spreading.
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5.14.2 FREQUENCY HOPING SPREAD SPECTRUM
For a long time spread spectrum signals were obtained by modulating radio oscillation frequency using saw
form signal. Using such modulation frequency of modulated signal is periodically changed. Spread spectrum signals
also can be obtained by discretely modulating one frequency signal. It is frequency hopping method.
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5.15 SATELLITE COMMUNICATION

A satellite is an artificial object which has been intentionally placed into orbit. Such objects are sometimes
called artificial satellites to distinguish them from natural satellites such as the Moon. Satellites are used for a large
number of purposes. Common types include military and civilian Earth observation satellites, communications
satellites, navigation satellites, weather satellites, and research satellites. Satellites that are used for transmitting or
broadcasting useful information for long ranges are known as communication satellites. A communications
satellite or comsat is an artificial satellite sent to space for the purpose of telecommunications.
For fixed (point-to-point) services, communications satellites provide a microwave radio relay technology
complementary to that of communication cables. They are also used for mobile applications such as communications
to ships, vehicles, planes and hand-held terminals, and for TV and radio broadcasting.
Satellites are usually semi-independent computer-controlled systems. Satellite subsystems attend many tasks,
such as power generation, thermal control, telemetry, attitude control and orbit control.
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5.15.1 SATELLITE ORBITS
Satellite orbits vary greatly, depending on the purpose of the satellite, and are classified in a number of ways.
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Well-known (overlapping) classes include
 Low Earth orbit
 polar orbit, and
 geostationary orbit.
Modern communications satellites use a variety of orbits including geostationary orbits, Molniya orbits, elliptical orbits
and low (polar and non-polar) Earth orbits.
5.15.2 GEOSTATIONARY ORBITS
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A geosynchronous orbit (sometimes abbreviated GSO) is an orbit around the Earth with an orbital period of
one side real day (approximately 23 hours 56 minutes and 4 seconds), matching the Earth's sidereal rotation
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period. The synchronization of rotation and orbital period means that, for an observer on the surface of the Earth, an
object in geosynchronous orbit returns to exactly the same position in the sky after a period of one sidereal day.
A special case of geosynchronous orbit is the geostationary orbit, which is a circular geosynchronous orbit at
zero inclination (that is, directly above the equator). A satellite in a geostationary orbit appears stationary, always at
the same point in the sky, to ground observers. Popularly or loosely, the term "geosynchronous" may be used to mean
geostationary. Specifically, geosynchronous Earth orbit (GEO) may be a synonym for geosynchronous equatorial orbit,
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or geostationary earth orbit. Communications satellites are often given geostationary orbits, or close to geostationary,
so that the satellite antennas that communicate with them do not have to move, but can be pointed permanently at
the fixed location in the sky where the satellite appears.
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A geostationary orbit (GEO) is a circular geosynchronous orbit in the plane of the Earth's equator with a radius
of approximately 42,164 km (26,199 mi). A satellite in such an orbit is at an altitude of approximately 35,786 km
(22,236 mi) above mean sea level. It maintains the same position relative to the Earth's surface.
To an observer on the earth, a satellite in a geostationary orbit appears motionless, in a fixed position in the
sky. This is because it revolves around the earth at the earth's own angular velocity. A geostationary orbit is useful for
communications because ground antennas can be aimed at the satellite without their having to track the satellite's
motion. This is relatively inexpensive.
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5.15.3 LOW EARTH ORBIT
A low Earth orbit (LEO) is an orbit around Earth with an altitude between 160 kilometers (99 mi), with
an orbital period of about 88 minutes, and 2,000 kilometers (1,200 mi), with an orbital period of about 127 minutes.
Objects below approximately 160 kilometers (99 mi) will experience very rapid orbital decay and altitude loss.
With the exception of the manned lunar flights of the Apollo program, all human spaceflights have taken place
in LEO (or were suborbital).
Objects in LEO encounter atmospheric drag in the form of gases in the thermosphere (approximately 80–
500 km up) or exosphere (approximately 500 km and up), depending on orbit height. Objects in LEO orbit Earth
between the atmosphere and below the inner Van Allen radiation belt. The altitude is usually not less than 300 km for
satellites, as that would be impractical due to atmospheric drag.
The orbital velocity needed to maintain a stable low earth orbit is about 7.8 km/s, but reduces with increased
orbital altitude. Although the Earth's pull due to gravity in LEO is not much less than on the surface of the Earth,
people and objects in orbit experience weightlessness because they are in free fall. A low earth orbit is simplest and
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most cost effective for a satellite placement and provides high bandwidth and low communication latency.
5.15.4 APSIS- APOGEE AND PERIGEE
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Apogee and Perigee refer to the moon and its relation to the earth. The moon's orbit is not circular, but
elliptical and when the moon is at its closest point to earth, it is said to be at its perigee and when it's at its furthest
point it is said to be at its apogee.
Apogee and perigee are terms which indicate the distance between the earth and the moon. Apogee is when
the moon is furthest from the earth, and perigee is when it is the closest.
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5.15.5 KEPLER'S LAWS OF PLANETARY MOTION

Johannes Kepler, developed three laws which described the motion of the planets across the sky.
 The Law of Orbits: All planets move in elliptical orbits, with the sun at one focus.
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 The Law of Areas: A line that connects a planet to the sun sweeps out equal areas in equal
times.
 The Law of Periods: The Square of the period of any planet is proportional to the cube of the
semi major axis of its orbit.
In astronomy, Kepler's laws of planetary motion are three scientific laws describing the motion of planets around the
Sun. Kepler's laws are now traditionally enumerated in this way:
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 The orbits are ellipses, with focal points ƒ1 and ƒ2for the first planet and ƒ1 and ƒ3 for the
second planet. The Sun is placed in focal point ƒ1.
 The two shaded sectors A1 and A2 have the same surface area and the time for planet 1 to
cover segment A1 is equal to the time to cover segment A2.
 The total orbit times for planet 1 and planet 2 have a ratio a13/2: a23/2.
FIRST LAW:
"The orbit of every planet is an ellipse with the Sun at one of the two foci."
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Kepler's first law placing the Sun at the focus of an elliptical orbit
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Heliocentric coordinate system (r, θ) for ellipse. Also shown are: semi-major axis a, semi-minor axis b and semi-latus
rectum p; center of ellipse and its two foci marked by large dots. For θ = 0°, r = rmin and for θ = 180°, r = rmax.
Mathematically, an ellipse can be represented by the formula:
where (r, θ) are polar coordinates, d is the focal parameter, and ε is the eccentricityof the ellipse.
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Note that 0 < ε < 1 for an ellipse; in the limiting case ε = 0, the orbit is a circle with the sun at the
centre
SECOND LAW:
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"A line joining a planet and the Sun sweeps out equal areas during equal intervals of time."
The same shaded area is swept out in a given time. The green arrow is velocity. The purple arrow directed
towards the Sun is the acceleration. The other two purple arrows are acceleration components parallel and
perpendicular to the velocity.
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This law relates radial distance and angular velocity in elliptical orbits. That is, in a perfectly circular orbit, the
orbital radius of the satellite would be constant and therefore so would be its observed angular velocity. In elliptical
orbits, the orbital radius of the satellite will vary and therefore so will its angular velocity. This is shown in the above
animation where the satellite travels "faster" (greater angular velocity) when closer to the parent object, then
"slower" (less angular velocity) at a more distant radius. The result is that the blue sectors are shorter but wider when
close to the body, then longer but narrower at a greater distance. Kepler's 2nd law states that for a given elliptical
orbit, any two sectors of equal time duration will have the same area. This implies that radial distance and angular
velocity have an inversely proportional relationship in a given orbit; angular velocity is minimum at apoapsis and
maximum at periapsis. The constant of proportionality is the rate at which area in the ellipse is covered.
In a small time the planet sweeps out a small triangle (or, more precisely, a sector) having base
line and height and area and so the constant areal velocity is
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The planet moves faster when it is closer to the Sun.
The area enclosed by the elliptical orbit is
so the period satisfies
and the mean motion of the planet around the Sun

R satisfies
THIRD LAW:
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The square of the orbital period of a planet is directly proportional to the cube of the semi-major axis of its
orbit. The third law captures the relationship between the distance of planets from the Sun, and their orbital periods.
Kepler’s enunciated this third law in a laborious attempt to determine what he viewed as the "music of the spheres"
according to precise laws, and express it in terms of musical notation. So it used to be known as the harmonic law.
Mathematically, the law says that the expression has the same value for all the planets in the solar
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system. The modern formulation with the constant evaluated reads:

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with
T the orbital period of the orbiting body,
M the mass of the star,
G the universal gravitational constant and
r the radius, the semi-major axis of the ellipse.
5.15.6 SATELLITE LINKS:
The communication going from a satellite to ground is called downlink, and when it is going from ground to a
satellite it is called uplink. When an uplink is being received by the spacecraft at the same time a downlink is being
received by Earth, the communication is called two-way. If there is only an uplink happening, this communication is
called upload. If there is only a downlink happening, the communication is called one-way. The communication
happening between two satellites is characterized by a Cross link.
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5.15.7 TRANSPONDER:
In telecommunication, a transponder is one of two types of devices. In air navigation or radio
frequency identification, a flight transponder is a device that emits an identifying signal in response to an interrogating
received signal. In a communications satellite, a transponder gathers signals over a range of uplink frequencies and re-
transmits them on a different set of downlink frequencies to receivers on Earth, often without changing the content of
the received signal or signals.
The term is a portmanteau for Transmitter-responder. It is variously abbreviated as XPDR, XPNDR, TPDR or TP.
A communications satellite’s channels are called transponders, because each is a separate transceiver or repeater.
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With digital video data compression and multiplexing, several video and audio channels may travel through a
single transponder on a single wideband carrier. Original analog video only has one channel per transponder,
with subcarriers for audio and automatic transmission identification service (ATIS). Non-multiplexed radio stations can
also travel in single channel per carrier (SCPC) mode, with multiple carriers (analog or digital) per transponder. This
allows each station to transmit directly to the satellite, rather than paying for a whole transponder, or
using landlines to send it to an earth station for multiplexing with other stations.
5.15.8 LINK BUDGET
A link budget is the accounting of all of the gains and losses from the transmitter, through the medium (free
space, cable, waveguide, fiber, etc.) to the receiver in a telecommunication system. It accounts for the attenuation of
the transmitted signal due to propagation, as well as the antenna gains, feed line and miscellaneous losses. Randomly
varying channel gains such as fading are taken into account by adding some margin depending on the anticipated
severity of its effects. The amount of margin required can be reduced by the use of mitigating techniques such
as antenna diversity or frequency hopping.
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A simple link budget equation looks like this:
Received Power (dBm) = Transmitted Power (dBm) + Gains (dB) − Losses (dB)
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A link budget equation including all these effects, expressed logarithmically, might look like this:
where:
= received power (dBm)
= transmitter output power (dBm)
= transmitter antenna gain (dBi)
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= transmitter losses (coax, connectors...) (dB)
= free space loss or path loss (dB)
= miscellaneous losses (fading margin, body loss, polarization mismatch, other losses...) (dB)
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= receiver antenna gain (dBi)

= receiver losses (coax, connectors...) (dB)
The loss due to propagation between the transmitting and receiving antennas, often called the path loss, can
be written in dimensionless form by normalizing the distance to the wavelength:
(dB) = 20×log*4×π×distance/wavelength+ (where distance and wavelength are in the same units)
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When substituted into the link budget equation above, the result is the logarithmic form of the Friis transmission
equation.In some cases it is convenient to consider the loss due to distance and wavelength separately, but in that
case it is important to keep track of which units are being used, as each choice involves a differing constant offset.
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Some examples are provided below.

(dB) = 32.45 dB + 20×log[frequency(MHz)] + 20×log[distance(km)]
(dB) = - 27.55 dB + 20×log[frequency(MHz)] + 20×log[distance(m)]
(dB) = 36.6 dB + 20×log[frequency(MHz)] + 20×log[distance(miles)]
These alternative forms can be derived by substituting wavelength with the ratio of propagation velocity (c,
approximately 3×10^8 m/s) divided by frequency, and by inserting the proper conversion factors between km or miles
and meters, and between MHz and (1/sec).
5.15.9 FOOT PRINTS OF A SATELLITE
The footprint of a communications satellite is the ground area that its transponders offer coverage, and
determines the satellite dish diameter required to receive each transponder's signal. There is usually a different map
for each transponder (or group of transponders) as each may be aimed to cover different areas of the ground.
Footprint maps usually show either the estimated minimum satellite dish diameter required or the signal
strength in each area measured in dBW.
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PART-A
UNIT-1 ANALOG COMMUNICATION
1 What do you mean by communication?
COMMUNICATION is the process of establishing a physical connection (or link) between two points for
information exchange. The information generated at source part may need to travel thousand/hundred miles
via the established link, called channel to reach the destination
2 What is an electronic communication system? Draw its block diagram.
An electronic communication system is a complete system, which exchanges information from one place to
another by electrical means. The transmitting system at source end, the receiving system at destination end
and the channel between these two systems through which the information is transferred comprises a
communication system.
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3 What are the types of communication systems based on the type of information being sent?
Depending on the type of information being transmitted, the communication system can be of two
types. They are
 Analog Communication system
 Digital Communication system
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If in the communication channel, the signals which could continuously vary with respect to time
(analog signals) are transmitted, then such a system is called Analog communication system.If in the
communication channel, the signals which are discontinuous and discrete in nature (Digital signals) are
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transmitted, then such a system is called Digital communication system.

4 Spot the differences between analog and digital communication systems
ANALOG COMMUNICATION SYSTEMS DIGITAL COMMUNICATION SYSTEMS
No A/D or D/A convertors required. Transmitted Requires A/D at Transmitter and a D/A at
as analog signals Receiver
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Required Bandwidth & cost of components is In terms of QOS, less prone to noise and
less. Most preferred for Short haul is preferred for Long haul
communication communication
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Regenerative repeaters are used at intermediate Regenerative repeaters are used at

stages to reduce signal attenuation intermediate stages to retransmit the
binary information
5 What is noise in communication systems?

In all communication process, electrical disturbance are appearing as interference which is called as NOISE.
Noise is an unwanted signal that corrupts a desired message signal. In general noise may be predictable or
Random in nature.
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6 Give the classification of noises based on their sources.

The sources of noise are broadly classified as follows:
Noise
External Noise Internal Noise
Erratic Natural Man-Made Shot Noise Thermal Noise
Disturbances Noise
7 What is Man-made noise?

This noise occurs due to undesired pickups from electrical appliances, such as motors, switch gears,
automobile and aircraft ignitions etc. This kind of noise is under human control and can be eliminated by
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removing the sources of noise. This noise has an effective range of 1MHz to 500MHz.
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8 When Thermal noise occurs in a system?
Thermal Noise is an internal noise which occurs due to the thermal agitation of electrons inside electron
devices.This type of noise arises due to random motion of electrons in a conducting medium such as a resistor,
and this motion in turn is randomized through collisions caused by imperfection in the structure of conductors.
The net effect of motion of all electrons constitutes an electronic current flowing through the resistor, causing
the noise. This noise is also known as Resistor Noise and Johnson noise.
The Power spectral density of thermal noise is given by
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si(ω) = 2KTG / *1 + (ω/α)2]
Where
T = Ambient temperature (Kelvin)
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G = conductance of Resistive elements in mhos.

α = average number of collisions per sec per electron.
9 Obtain the equivalent circuit of a noisy resistor?
A Noisy Resistor R can be represented by noiseless conductance G in parallel with a noisy current source of
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in(t)as shown below. The Thevenin equivalent of the same is also shown.
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If Resistors are in series,

RS=R1+R2+R3+……..
If Resistors are in Parallel,
RP=* +
10 What is partition noise? How it is reduced in a microwave receiver?
Partition Noise is generated whenever a current has to divide between two or more branches in a circuit and
results from random fluctuation in the division. It would be expected therefore that a diode would be less
noisy than a BJT, if the third terminal draws current.For this Reason, the input stage of Microwave receiver is
often a diode circuit rather than a transistor, for low frequency amplification. Partition noise has a flat
spectrum.
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11 How thermal noise power can be calculated?

Thermal Noise PowerPn=4KTRΔf =2KTR(Bn) Since 2Δf=Bn
Volt equivalent of Thermal Noise: Vn=√ Volts
Thermal Noise Power Density: No= kT
Thermal Noise power of a noisy transistor: Pn= KTBn Watts
Thermal Noise power of a noisy transistor in dbW: Pn(dbW) = 10 log(k) +10 log T+ 10 log Bn dbW
12 Explain –Superposition of noises.
“For a system involves multiple independent sources of noise, the power density spectrum is equal to
the sum of individual power density spectrum evaluated individual sources at a time.”
Sy1+y2+y3+….(ω) = Sy1(ω) + Sy2(ω) + Sy3(ω) +…..
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13 Define Modulation.
Modulation is defined as the process by which some parameters (Amplitude, Frequency and Phase) of a high
frequency carrier signal is varied in accordance with the amplitude of message signal.
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Any signal can be represented by its three parameters namely Amplitude, Frequency and Phase. If any of these
parameters of a signal are changed with respect to the amplitude of a relatively lower frequency signal, one
such a process is called modulation.
14 What is demodulation of a signal?
Demodulation is the reverse process of modulation. If the original, Low frequency message signal is recovered
from the modulated carrier signal, then it is called Demodulation/Detection. It is a technique followed at
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receiver by a circuit called Demodulator/Detector.
15 List the needs for modulation
If the Low frequency, Low amplitude message signals are transmitted for a long distance through a noisy
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transmission medium it will not reach the destination properly. Due to Noise and Interference present, the
signal may fade or getting disturbed. In order to avail successful long haul communication and faithful
reception, Modulation by a high frequency carrier at transmitter and Demodulation at receiver is required.
The other needs for modulation are listed below.
1. Radiation antenna height – If modulated, antenna with lesser heights is enough
2. Long haul communication – distance between Transmitter and Receiver can be increased
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relatively if modulated
3. Ease of Radiation- After Modulation, the message signals are translated to a higher frequency,
hence it is easy to construct radiation circuits such as amplifiers and matching networks.
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4. Multiplexing- Transmission of two or more signals over the same channel is called multiplexing.
Due to modulation, the number of channels required is reduced.
5. Improve the quality of Reception-Noise is highly limited by performing modulation, which in turn
improves the quality of Reception
6. Avoids mixing of signals-If more than one message signal has to be transmitted, each low
frequency message signal can be modulated with different carrier frequencies and hence they can
be allotted to various slots in frequency domain, which essentially avoids mixing of closely spaced
message signals.
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16 Give the Detailed classification of modulation.
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16 Define Amplitude modulation.Where it can be employed?
Amplitude modulation is the process by which amplitude of the high frequency analog carrier signal is varied in
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accordance with the instantaneous amplitude values of the low frequency analog message signal.AM is a
relatively inexpensive, low quality form of modulation that is used for commercial broadcasting of both audio
and video signals.
17 Envelope of AM- Explain briefly.
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The shape of modulated wave (AM) is called AM envelope which contains all the frequencies and is
used to transfer the information through the system. It is called envelope, because it contains all frequency
components including Carrier term, USB and LSB frequencies.
o If the instantaneous amplitude of message signal increases, the amplitude of carrier is also
increased.
o If the instantaneous amplitude of message signal decreases, the amplitude of carrier is also
decreased.
Without Modulating signal, the AM output is simple the carrier signal. The repetition rate of the
envelope is equal to the frequency of modulating message signal. The shape of the envelope is similar to that
of modulating signal.
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18 Draw the spectral representation of AM. Also write the expression to find the Bandwidth of AM.
The bandwidth of AM signal is given by the subtraction of highest and lowest frequency component in the
entire frequency spectrum.
BW = fUSB-f LSB
=fc+fm–( fc-fm)
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BW =2fm
19 Draw the Phasor of DSB-Full Carrier AM
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20 What is Modulation index.
Modulation index or Degree of modulation or co-efficient of modulation is a parameter which represents the
extent of amplitude variation in AM about an unmodulated carrier amplitude. It is defined as the ratio of
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maximum amplitude of modulating signal to the maximum amplitude of carrier signal.

Modulation Index (ma) =
For Distortion, not to occur in an AM modulation system, Value of Vm should always lesser than Vc. Hence the
maximum value of modulation index should be 1. The modulation index should be a number between 0 and 1.
21 What are the 3 degrees of amplitude modulation based on Modulation index.
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If Vm<Vc, ma<1, such type of modulation is called Under Modulation

If Vm= Vc, ma=1, such a type of modulation is called Critical Modulation
If Vm>Vc, ma>1, such a type of modulation is called Over Modulation
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22 How the percentage modulation is calculated after modulation?

There is one more method for calculating Modulation index. After modulation, if the envelope is
observed, the swing of the envelope is between Vmax and Vmin.
Where Vmax = Vc+ Vm & Vmin= Vc- Vm
From the Envelope of AM,
&
From this,
Then modulation index is given by,

m a= = x 100%
This expressions can be used for calculating the modulation index, when the Vmax and Vmin values are known
after modulation.
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23 Write the voltage distribution of AM-DSBFC
We know that, the AM envelope equation is
VAM(t) = VCSinωct+ -
From this the voltage distribution in modulated wave can be expressed.
In an AM envelope, there are three segments Viz.
1. No modulation region or Carrier
2. USB
3. LSB
Voltage distribution of these three segments can be tabulated as follows
ENVELOPE SEGMENT VOLTAGE
No modulation region VC
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USB
LSB
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24 Write an expression for Transmission power of AM
Transmission power of AM
Pt = P c * +
Pc– Power of Carrier
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ma- Modulation index
25 What is the efficiency of AM?
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The amount of useful information present in AM after modulation is expressed as transmission
efficiency. It is the ratio of the transmitted power which contains the information to the total
transmitted power.
Transmission efficiency (η) % = x 100%
(η) % = X100%
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26 Why AM-DSB-Full Carrier is in efficient? What are the other forms of AM?
In AM-DSB-FC, the useful message signal is modulated on to a carrier possessing 3 spectral
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components namely Carrier, USB and LSB. But it is evident from the analysis that only USB and LSB
contain useful information and carrier component contributes for wasteful of power. Only the
modulated signal contains 33% of useful power and the remaining are wasteful side band
components. Hence it is considered to be inefficient
Hence to improve the efficiency, some other forms of AM are developed they are
1. DSB-SC AM Double Side Band –Suppressed Carrier AM
2. SSB-SC AM Single Side Band –Suppressed Carrier AM
3. VSB-AM Vestigial Side Band AM
27 Draw the Spectrum and phasor of DSB-SC AM. Also write the expression for its Bandwidth.
VDSB-SC(t) = [ ]
Observing the above equation, it is trivial that there is No Carrier [No modulation term] term in the envelope
and possesses only the Side band terms [USB and LSB]. Hence this modulation is called Double Side Band –
Suppressed Carrier Amplitude Modulation [DSB-SC AM].
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FREQUENCY SPECTRUM OF DSB-SC AM
The Frequency domain spectral representation of DSB-SC AM is given below.
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PHASOR REPRESENTATION
The Phasor representation of the DSB-SC AM is given below.
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BANDWIDTH OF DSB-SC AM
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BANDWIDTH = USB –LSB
=
BW =  which is similar to the bandwidth of DSB-FC.
28 What is SSB-SC AM?
In AM DSB-FC, Both Transmission power and Transmission Bandwidth are wasted. To avoid this, DSB-SC
scheme is introduced. But in DSB-SC also only the transmission power is reduced, but not the transmission
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bandwidth. To increase the efficiency and to further increase the saving of power, one more sideband is
eliminated along with the carrier, and such a technique is called SSB-SC technique.
Elimination of one side band will not result in loss of message signal information, since LSB and USB are
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uniquely symmetry with respect to carrier frequency. Usually USB term is suppressed along with carrier in SSB
techniques, as the LSB [Low frequency components] contains much of the message information than USB.
29 Draw the Spectrum and phasor of SSB-SC AM. Also write the expression for its Bandwidth.
FREQUENCY SPECTRUM OF SSB-SC AM
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PHASOR REPRESENTATION OF SSB-SC AM
30 List out the Advantages, disadvantages and applications of AM

ADVANTAGES:
 Simple and Robust of all modulation systems
 Cheap and inexpensive
 Am wave can travel over long distances.
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DISADVANTAGES:
 Poor Noise performance [Much affected by Noise. No methods for noise rejection]
 Low quality form modulation
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 Low efficiency
 Requires Bandwidth
APPLICATIONS:
 Picture signal transmission in TV communication
 Commercial broadcasting of Audio and video signals
 2-Way mobile radio communication systems such as CB-Radio
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 Aircraft communications @ VHF range  30 to 300 MHz
31 What is angle modulation? List out its types & features.

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It is the process by which the angle [frequency or phase] of a carrier signal is changed in accordance with the
instantaneous amplitude of the modulating message signal. Amplitude of the carrier remains unchanged in
these modulations.
TYPES
1. Frequency Modulation – FM
2. Phase Modulation -PM
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FEATURES OF ANGLE MODULATION SYSTEMS

 Noise Reduction
 Improved system fidelity
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 More efficient use of Power

32 Define Deviation sensitivity.
The relationship between the output parameter change and the input parameter change is given by Deviation
sensitivity. It expresses the amount of change in output frequency or output phase for a small change in input
amplitude. Hence
For FM: KFM =
KFMVm =Δω
For PM:KPM =
KPMVm =Δθ
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33 Write expression for Frequency Deviation for FM and phase deviation for PM.
In case of FM, m f=
Or mf  frequency deviation
 If is +ve In FM wave frequency is greater than the carrier wave
 If is -ve  In FM wave frequency is lesser than the carrier wave
In case of PM, mp=
Phase deviation
 If is +ve Phase shift of PM wave is greater than that of the carrier wave
 If is -ve  Phase shift of PM wave is lesser than that of the carrier wave
34 Explain the two types of Fm systems based on their bandwidth.
Frequency modulation can be classified as narrowband if the change in the carrier frequency is about the same
as the signal frequency, or as wideband if the change in the carrier frequency is much higher (modulation index
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>1) than the signal frequency.
If is less for a FM system, such FM is called Narrowband FM
If is large for a FM system, such FM is called Wideband FM
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35 What is Pre-Emphasis and De-emphasis? Why it is required in angle modulated systems?
In case of AM, the modulation itself takes care of the improvement of power level of the transmission signal.
That is by merely performing the modulation, the signal power can be improved in AM systems. But in angle
modulation systems there is no guarantee for the power level improvement in transmission signal by
modulation.
Hence in angle modulated systems, to improve the power level of the transmission signal, the actual
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message signal is pre-emphasized before modulation and at receiver it is de-emphasized after demodulation.
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PRE-EMPHASIS
It refers to a system process designed to increase (within a frequency band) the magnitude of some
higher frequency with respect to the magnitude of some lower frequency in order to improve overall SNR
(power) by minimizing attenuation.
DE-EMPHASIS
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It refers to a system process designed to decrease (within a frequency band) the magnitude of some
higher frequency with respect to the magnitude of some lower frequency in order to improve overall SNR
(power) by minimizing attenuation.
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36 Write the expressions that are used for calculating bandwidth for various types of FM systems.
Theoretically the Bandwidth of FM &PM waves are infinite. But in practice the bandwidth is calculated by the
number of sub bands having significant amplitude. Bandwidth of FM and PM depends on their modulation
index.
If m<1 low index angle modulation
If m=1 medium index angle modulation
If m>1 high index angle modulation
In case of FM,
(1) Low index corresponds to Narrowband FM, whose bandwidth can be calculated as
BW=2
(2) High index corresponds to Wideband FM, whose bandwidth can be calculated as
BW=2
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(3) The maximum possible Bandwidth for a FM system is determined by
BW=2n n  Number of sidebands having significant measurable
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amplitude
(4) The Approximate Transmission Bandwidth necessary for faithful transmission of angle modulated
waves can be found using Carson’s rule (Thumb rule). “The Necessary Bandwidth required for the transmission
of angle modulated waves is twice the sum of peak frequency deviation and highest frequency component in
the message signal.”
BW=2 (or)BW = 2 Hz
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What is a super heterodyne receiver?
A super-heterodyne receiver is a universal receiver which is capable of detecting AM, FM and PM. It is widely
used in Wireless Radio systems. A Super-heterodyne Receiver has three blocks. They
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1. RF Amplifier
2. IF section
3. Baseband processor
A super heterodyne receiver, uses frequency mixing to convert a received signal to a fixed intermediate
frequency (IF), which can be more conveniently processed than the original radio carrier frequency. At the cost
of an extra frequency converter stage, the super-heterodyne receiver provides superior selectivity and
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sensitivity compared with simpler designs. The principle of operation of the super-heterodyne receiver
depends on the use of heterodyning or frequency mixing.
38 Draw the block diagram of a super heterodyne receiver.
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39 How FM can be generated from PM?
Frequency modulated wave can be obtained from PM by differentiating it.
40 How PM can be obtained from FM?

Phase modulated wave can be obtained from Fm by simply integrating it.
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PART-A
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UNIT-2 DIGITAL COMMUNICATION
1 What is digital Communication? What are the two types of digital communication?
Digital communication refers to the process of transmission and reception of information by means of digital
signals. There are two methods in Digital communication. They are
1. Digital Baseband Communication
2. Digital Modulation
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In Digital Baseband communication, Transmission and Reception of digital data directly to a noiseless channel.
In this method there is no major modulation performed.
In Digital Modulation techniques, either modulating a digital carrier by an analog message signal (Pulse
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modulation) or modulating an analog carrier by a digital message signal (Shift keying technique) is carried out.
In general Digital modulation refers to Shift keying techniques. It is a method in which a relatively high
frequency carrier is modulated by low frequency digital information
2 List out the advantages and applications of Digital communication.
ADVANTAGES OF DIGITAL COMMUNICATION
 Better Noise immunity
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 More flexible
 Easy for multiplexing and Demultiplexing
 Easy and effective error correction and decoding
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APPLICATIONS OF DIGITAL MODULATION

 Best suitable for Cabled and Wireless Communication
 Relatively low speed ,voice-band data communication MODEMS
 High speed data transmission systems such as Broadband Digital Subscriber Links (DSL)
 Digital Microwave and Satellite Communications
 Cellular phones and Personal computer systems
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3 Draw the block diagram of a Digital communication system.
4 What are shift keying methods? Enumerate its types.

The process of changing the parameters (Amplitude, Frequency (period), Phase) of a high frequency Analog
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Carrier by a digital modulating signals (bits or Binary) is called shift Keying.(i.e) In Shift keying techniques,
Message signal  Digital Signal &Carrier signal  Analog Signal
o If Amplitude of Analog carrier is modulated by Digital Message, it is Amplitude Shift Keying or
AP
ASK
o If Frequency of Analog carrier is modulated by Digital Message, it is Frequency Shift Keying or
FSK
o If Phase of Analog carrier is modulated by Digital Message, it is Phase Shift Keying or PSK
5 Describe the Logic of Direct digital AM (ASK).
In the ASK modulated wave, the analog carrier term is repeated (ON) if message contains Logic 1 and it is
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absent (OFF) for Logic 0 in the message. So ASK is sometimes called ON-OFF Keying [OOK] or Direct digital AM.
 The time for transmitting one bit [either 0 or 1] is called symbol duration or bit duration [tb]
 The time taken for transmitting one DAM carrier unit is called signaling time [ts]
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 Therefore if the binary input is high, with constant amplitude, constant frequency signal, the carrier is
ON at the output and if the binary output is Low, carrier is OFF at the output.
6 Draw a sample ASK modulated waveform for a digital binary alternating sequence.
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7 Define Bit rate, Baud and Baud rate.

 tb(bit duratrion)= [bits per second]-1 or [tb]-1 = bits per second  (data rate or bit rate)
-1 -1
 [ts] = (signaling duration) = [Baud or Symbol rate ]
 = =[tb]-1 ,N No.of Bits Encoded in to each signaling element.
BAUD: Baud refers to the rate of change of signal on a transmission medium after encoding and modulation
process. It is a unit of transmission rate variation or Symbol rate. It is the reciprocal of time of one output
signaling element. Symbol rate or transmission rate is measured in (Baud / Sec).Usually, Baud rate< Bit rate
 In low level digital systems such as BFSK,BPSK: Baud=bps
 In high level digital systems such as QPSK,8PSK: Baud<bps
8 What is m-ary scheme of Shift keying?
If there are ‘n’ digits in a group to form a symbol, the number of possible symbols can be generated is m=2n.
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And the scheme is called m-ary encoding. If Modulation is done with such a kind of encoded message, it is
called m-ary Scheme of shift keying.
9 Draw a Binary FSK Modulator.
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10 What are the different demodulation techniques for FSK.

To do FSK demodulation and to recover the original binary information from the signaling unit there are three
demodulation schemes available. They are
1. Non- coherent FSK Demodulation
2. Coherent FSK Demodulation
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3. PLL-FSK Demodulation
11 Draw the sample wave FSK modulation for a binary message sequence 1001011001010 2.
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12 Write the Pros and Cons of FSK modulation.
PROS CONS
 Less error when compared to ASK and PSK  Not useful in low performance, low cost
asynchronous circuits like DATA MODEMS and
Voice phones etc.
 Can be Used in high performance Digital radio  Non-Linear Distortions will be more which caus
applications fickle reception
13 Write the minimum required bandwidth for BFSK.
The minimum Bandwidth required for transmitting a FSK signal is given by
BWfsk = |(Fs+ fb)-(Fm –fb)|
= |Fs+ fb- Fm +fb)|
= |Fs- Fm +2fb)|
= |Fm- Fs |+2fb
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But it is known that,
| |
=
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| |
And hence BWfsk =
BWfsk =
Where, BWfsk Minimum transmission bandwidth for FSK
 Frequency Deviation
 Binary input bit rate * +
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14 How MSK differs from FSK?
In FSK Non-linear distortions may occur during the detection due to the non-linear thresholding decision
device. This creates much jarring effects when FSK is used in asynchronous voice MODEMS and in
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Telecommunication circuits. To avoid this problem a variant of FSK is developed and it is called MSK. In digital
modulation, Minimum Shift Keying (MSK) is a type of continuous-phase frequency-shift keying . MSK is
encoded with bits alternating between quadrature components, with the Q component delayed by half
the symbol period. MSK encodes each bit as a half sinusoid. This results in a constant-modulus signal, which
reduces problems caused by non-linear distortion. In addition to being viewed as related to OQPSK, MSK can
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also be viewed as a continuous phase frequency shift keyed (CPFSK) signal with a frequency separation of one-
half the bit rate. In MSK the difference between the higher and lower frequency is identical to half the bit rate.
Consequently, the waveforms used to represent a 0 and a 1 bit differs by exactly half a carrier period. Thus, the
maximum frequency deviation is = 0.25 fm where fm is the maximum modulating frequency. As a result, the
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modulation index m is 0.5. This is the smallest FSK modulation index that can be chosen such that the
waveforms for 0 and 1 are orthogonal.
15 What is PSK, BPSK?
The Process of varying the phase of the carrier (Analog) with the accordance to the level of digital
message signal is called Phase Shift Keying [PSK].
In BPSK,
Message : Binary digital signal
Carrier : High frequency analog signal
BPSK O/P : A high frequency analog signal with limited number of phases (2-different phases) at Output.
With Binary PSK, n=1 and M =21=2, so with BPSK there are possibly 2-Phases at the Output. Among the two
phases at the output 1-phase represents Logic 1, and the other 1-phase represents Logic 0.As the Input signal
changes from 0 to 1 and from 1 to 0, the phase of Output carrier shifts between 2 angles that are separated by
180°s [π-Radians]
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(i.e.) BPSK, If Logic 1 at input Output signal is 0° phase with respect to carrier [In-phase]
If Logic 0 at input Output signal is 180° phase with respect to carrier [Out of phase]
The other names of BPSK are
1. Phase reversal keying
2. Bi-phase Modulation
16 Draw the block diagram of BPSK Transmitter.
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17
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Write the Truth table of BPSK.Also draw its constellation.
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TRUTH TABLE
Binary input Carrier Output signalUnit Output
phase
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Logic 0 VcSinωct VcSin [ωct+π+or- VcSinωct 180°

Logic 1 VcSinωct VcSinωct 0°
CONSTELLATION
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18 What is the minimum required bandwidth of BPSK modulation.

The minimum required Nyquist Bandwidth is
Band width (BPSK) = Maximum frequency component – Minimum frequency component
=
Band width (BPSK) = or Band width (BPSK) = b /2 =fb
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19 Draw a QPSK Transmitter .
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AP
20 Write the Dibit encoding table for QPSK and also write the Truth table for QPSK modulation.
If there are ‘2’ digits in a group to form a symbol, the number of possible symbols can be generated is m=22=4.
And the scheme is called 4-ary encoding or Quaternary encoding or Dibit encoding. If PSK Modulation is done
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with such a kind of encoded message, it is called Quaternary Phase shift keying.
Symbol 1 Symbol 0 Di-bit encoded Symbol
0 0 S4
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0 1 S3
1 0 S2
1 1 S1
With 4 PSK, 2 bits are processed to produce a single phase change. In this case each symbol consists of 2 bits,
which are referred to as a di-bit. The actual phases that are produced by a 4 PSK modulator are shown in the
table below.
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Di-bits Di-bit encoded Symbol Output Phase

00 S4 -135°[+225°]
01 S3 -45°[+315°]
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10 S2 +135°
11 S1 +45°
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21 What is 8-PSK? Give its truth table and constellation?
When Three bits are brought together to form one symbol, such an encoding is called 8-ary encoding. If PSK is
(i.e.) if n=3, M= 2n =23= 8.Therefore in 8-PSK scheme, there are 8 different phases possible at the output signal.
The Truth table and signal constellation for 8-ary PSK is given below.
8-ary Symb Outp
Encoded ol ut
binary Name Phas
message
0 0 0 S0 e 0°
signal
0 0 1 S1 45°
0 1 0 S2 90°
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0
0 1 1 S3 135°
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1 0 0 S4 180°
1 0 1 S5 225°
1 1 0 S6 270°
1 1 1 S7 315°
22
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Draw the signal constellation of 16-ary encoded PSK.
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23 What is DPSK? Explain

It is an alternate form of Digital Modulation, where binary input information is contained in the difference
between 2 successive signaling elements rather than the absolute phase. There are two operations involved in
DPSK. They are
1. Differential Encoding
2. Phase shift Keying.
To perform DPSK the input data bit is supposed to Logical EX-NOR operation with its one bit duration delayed
signal and then it is phase modulated in a balanced product modulator. By doing so, there is no need of
coherent carrier during the demodulation.
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24 What is Quadrature Amplitude Modulation? What are their types?
In all PSK Methods, One symbol is distinguished with other symbol by phase and amplitude remains constant
for all symbols. Noise immunity will improve if the signal vectors differ not only in phase but also in amplitude.
Hence direct modulation of carriers with Quadrature phase angles is included in PSK and such a system is called
Quadrature Amplitude Modulation [QAM].
TYPES:
METHOD Bits/Symbol No. Of symbols used
4-QAM 2 4
8-QAM 3 8
16-QAM 4 16
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32-QAM 5 32
64-QAM 6 64
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25 How noise immunity can be improved by QAM?
On moving to higher order QAM systems, the mode and High data rates can be achieved with the help of high
power RF carriers and they can be employed in Microwave communication systems. This QAM scheme spreads
data in to more number of phases into the spectrum. Hence Noise immunity is increased when compared to
other digital modulation systems.
26 What are the three factors that influence the choice of digital modulation system?
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1. Bandwidth efficiency: The number of bits per second that can be transmitted per Hertz of channel
bandwidth
2. Error performance: The probability of making a bit error at the receiver, as a function of the signal-to-
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noise ratio
3. Equipment complexity: A parameter which effectively corresponds to the cost of the system.
27 Define Bandwidth Efficiency.
Bandwidth efficiency or Spectral efficiency or spectrum efficiency refers to the information rate that can be
transmitted over a given bandwidth in a specific communication system. It is a measure of how efficiently a
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limited frequency spectrum is utilized by the Communication system. Bandwidth efficiency describes how
efficiently the allocated bandwidth is utilized or the ability of a modulation scheme to accommodate data,
within a limited bandwidth. This table shows the theoretical bandwidth efficiency limits for the main
modulation types. The Spectral Efficiency (measured in b/s/Hz) of a modulation scheme with transmission rate
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R and bandwidth B is defined as
ρ = R/B = ⁄
%ρ = R/B *100%
Bandwidth Efficiency is generally normalized to a 1-Hz Bandwidth and, thus it indicates the number of bits that
can be transmitted through a medium for every 1 Hz. Usually Bandwidth efficiency is represented as %
efficiency.
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PART-A
UNIT-3 DATA AND PULSE COMMUNICATION
1 Define Data?
Data is generally defined as information stored in digital form.
2 What is network?
Set of devices (called as nodes or stations) interconnected by media links.
3 List the characteristics of data communication networks?
1. Delivery
2. Accuracy
3. Timeliness
4. Jitter
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4 Mention the components of Data communication network?
1. Message (text, number, pictures, audio, and video)
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2. Sender (computer, work station, telephone handset, video camera)
3. Receiver (computer, work station, telephone handset, video camera)
4. Transmission medium (twisted pair cable, co-axial cable, fiber optic cable, radio waves)
5. Protocols = set of rules that govern data communication
5 What are the different types of Data flow?
1. Simplex
2. Half- duplex
3. Full duplex
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6 What is simplex data flow?
Simplex: Unidirectional, only one of the two devices transmit on a link, the other can only receive.
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(eg: one way street) Example: key board, monitor
7 What is meant by half duplex data communication?

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Half duplex: Two way communication, but not at same time. (eg: traffic of walkie-talkies, CB
Radios)
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8 What is meant by full duplex data communication?

Full duplex: Transmit and receive simultaneously, example: telephone
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9 What are the different standards organizations available for data communication?
International Standard Organization (ISO),
Consultative Committee for International Telephony & Telegraphy (CCITT),
American National Standards Institutions (ANSI),
International Telecommunication Union Telecommunication standards sector (ITU-T),
Institute of Electrical and Electronics Engineering (IEEE),
Electronics Industries Association (EIA),
Standards Council of Canada (SCC)
10 Mention the types of transmission media?
1. Free space radio transmission (terrestrial &) satellite microwave)
2. Metallic cable facilities (both digital & analog systems)
3. Optical fiber cable (light wave propagation)
11 What is meant by DTE?
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Data Terminal Equipment is a general term that describes the interface equipment used at stations
to adapt digital signals from the computer and terminals to a form more suitable for transmission
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 UART  Universal Asynchronous Receiver Transmitter
 USART  Universal Synchronous Receiver Transmitter
12 What is DCE?
Data Communication Equipment, the equipment that converts digital signals to analog signals.
DCE  MODEM  modulator / demodulator and interfaces the data terminal equipment to the
analog transmission medium
13 What is meant by transmission mode?
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Transmission of data from one device to another is in means of wiring, suppose wiring is data
stream, how the bits are sent, either one bit at a time or either group of bits at a time. Depending
on number of bits transmitted, they are classified as
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14 What is meant by parallel transmission?

Binary data consists of 0’s and 1’s are organized into group of n-bits each. So by using n wires, to
send n bits at a time, so each bit has its own wire, so all the n bits of one group can be transmitted
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with each clock pulse from one device to another
All the four bits can be transmitted simultaneously during the time of a single clock pulse (T). this
type of transmission called parallel by bit (or) serial by characters
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15 What is meant by serial transmission?
One bit follows another, so only one communication channel is used instead of n wires. Due to only
one wire, cost is also reduced, since communication within devices is parallel, so conversion is
used. To transmit the bits 0110, It requires 4 clock pulses to transmit entire words. This type is
called as serial – by – bit used for long distance communications
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AP
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16 Mention the different types of transmission configurations?
• A network is a two (or) more devices connected through links.
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• A link is a communication path that transfers from one device to another.
• For simplicity any link is a line drawn between two points
• So for communication to occur two device must be connected in some way to the same link
17 Mention the different types of connections?
• Point to point connection (two point)
• Multipoint connection
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18 What is meant by topology?

The topology or architecture of a data communication circuit identifies how various locations
within the network connected. Two or more devices connected to a link (or) two or more links
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from topology
19 What are the different types of topology?
1. Mesh
2. Star
3. Ring (or) loop
4. Bus or Multidrop
20 Distinguish between connection oriented and connection less?
Connection oriented Connection less
Hand shake signals occur between two Sends data without a source and
stations before any data are actually destination address without a handshake
transmitted to ensure that data is ready to receive
the data
Provides some means of error control Do not support error control
(detection and correction)
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21 What is Proprietary (closed) system?
• Generally manufactured and controlled by one company
• Other companies are not allowed to manufacture equipment or write software using this
standard
22 What are the advantages and disadvantages of Proprietary (closed) system?
Advantages
 Tighter control
 easier consensus
 monopoly
Disadvantages
 Lack of choice for the customers
 Higher financial investment
 Overpricing
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23 What is open system?
• Any company can produce compatible equipment or software but royalty should be made to the
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original company.
• Example: IBM’s PC
24 What are the advantages and disadvantages of open system?
Advantage
 Customer choice, compatibility between vendors and competition by small company
Disadvantage
 Less product control increased difficulty acquiring agreements between vendors for
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changes or updates
25 What are data communication codes?
These are prescribed bit sequences used for encoding characters and symbols, in simple used to
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represent character and symbol. So data communication codes are called as character codes,
character sets, symbol codes or character languages.
26 What are the different types of data communication codes?
1) Baud of code
2) ASCII code
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3) EBCDIC code
4) BAR code
27 What is Baud of code?
• It is called as Telex code. It was the first fixed length character code developed for machines than
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people
• Baud of code => Baw dough is a fixed length source or fixed length block code
• With fixed length all characters are represented in binary and hence the same number of bits
(symbols)
• It is a five bit character code used for low speed teletype equipment such as TWX/Telex system
and radio teletype (RTTY)
28 What is ASCII code?
• 1963, to standardize the data communication codes, the U.S. adopted Bell system model, 33
teletype code as the United States of American Standard Code for Information Interchange
(USASCII) known as better ASCII-63
• Versions 1965, 1967, 1977 are being recommended by ITU as alphabet number 5
• ASCII => standard character set for source coding the alpha numeric character set that humans
understand but computers do not
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29 What is EBCDIC Code?
• Extended Binary Coded Decimal Interchange Code (EBCDIC)
• 8 bit fixed length character developed in 1962 by IBM (International Business Machines
Corporation )
• Used almost with IBM mainframe computers and peripheral equipment
• With 8 bits, 28 = 256 codes are possible out of 256, only 139 are assigned characters
• Binary coded decimal = because the second hexadecimal character for all letter and digit codes
contain only the hexadecimal values for 0 to 9 which have same binary sequence HEX : 00 to 09, 0A
to 0F, 10 to 19, 1A to 1F.
30 What are bar codes?
 Black and white stripped stickers that seem to appear on every customer item
 Developed in 1960’s and used in the mid of 1980’s
 BAR code is a series of vertical black bars separated by vertical white bars (called spaces)
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 The width of the bar and spaces represents 1’s and 0’s and combinations of bits represents to
identify specific items
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 BAR codes may contain information regarding
 Cost
 Inventory management control
 Security access
 Production counting
 Automatic billing
31
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How redundancy checking helps in detecting errors?
• Some extra bits are added with the data. These redundant bits are added by sender and removed
by the receiver. Their presence allows the receiver to detect or correct computer bits
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• Extra redundant bits are added
• So costly and message will be long
32 Mention the types of error correcting?
1. Symbol substitution
2. Retransmission
3. Forward error
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33 What is symbol substitution?

• Designed to be used in a human environment, where there is a human being in the receiver
terminal to analyze the received data and make decisions
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• With this, if a character is received in error, instead of displaying incorrect character, a unique
character that is undefined by the character code, such as reverse question mark (¿) is substituted
for bad character
• It is a form of selective retransmission
34 What is ARQ?
• When a receiver station requests the transmit station to resend a message (or a portion of the
message), when the message receive in error
• The receiver terminal automatically calls for retransmission of the entire message
• Retransmission is called as “ARQ” => Automatic Repeat Request (or) Automatic Retransmission
Request
• Most reliable method for retransmission
35 Mention the types of ARQ?
Discrete ARQ and Continuous ARQ
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36 Explain discrete ARQ?
• Uses acknowledgements to indicate the successful or unsuccessful reception of data (Positive or
negative acknowledgement)
• If sending station does not receive any acknowledgement after a predetermined length of time
(“time out”) it retransmits the message called “retransmission after time out”.
37 What is continuous ARQ?
• This is used when messages are divided into smaller block or frames that are sequentially
numbered and transmitted, without waiting for acknowledgements within blocks.
• Continuous ARQ allows the destination station to asynchronously request the retransmission of a
specific data and receiver will reconstruct the entire message once all frames have been
transmitted also called as “selective ARQ”
• Since it can be used to all, for a retransmission of an entire message or only a portion of message
38 What is serial standard interface?
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A serial interface standards should provide the following,
1. A specific range of voltages for transmit and receive signals.
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2. Limitations for the electrical parameters of the transmission line including source and load
impedance, cable capacitance.
3. Standard cable and cable connector.
4. Functional description of each signal in the interface
39 What are the limitations of RS-232 serial interface cable?
• Even though Rs-232is a cable with 2 connectors, the standards specifies limitations on the
voltage levels that the DTE and DCE can output onto or receive from the cable.
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• To overcome this voltage leveling circuits are used, which converts the internal voltage levels
from the DTE and DCE to RS-232 values
40 Explain the functional description of RS-232?
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RS-232 FUNCTIONAL DESCRIPTION

The RS-232 interface cable pins are categorized as five groups namely,
A – Ground (signal and chassis)
B – Data (transmit and receive)
C –Control (handshaking and diagnostic)
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D –Timing (clocking signals), and

E – Secondary channel.
41 Define parallel interface?
Parallel interface allows the user to transfer data between two devices with 8 or more bits at a
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same time (or) simultaneously. It is also called as serial by world transmission

42 State the advantages & disadvantages of parallel interfacing?
Advantages:
 The speed of data transfer is increased when compared to serial transmission.
 Most computer terminals and peripheral equipment process data parallel.
 It will avoid the conversion from serial to parallel.
Disadvantage:
 Number of wires and number of connection gets increases.
 It results in increase complexity.
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43 Distinguish between pulse modulation and digital encoding?
PULSE MODULATION: If analog signals DIGITAL ENCODING: If digital signals are
converted to digital form prior to encoded to some other digital format and
transmission and at reception, it is transmitted.
converted back to analog form. Example: computer to computer
44 List the types of sampling?
communication in computer networks
Natural sampling  sampler alone
Flat top sampling  sample and hold circuit
45 Define aperture error?
The sampling process alters the frequency spectrum and introduces an error called aperture error.
When the amplitude of the sampled signal changes during the same pulse time, this error will
occur. The magnitude of error depends on how much the analog signal voltage changes while the
sampling is being taken and width of sampling pulse
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46 What is quantization?
It is the process by which transforming the analog sample of original message signal into a discrete
amplitude from the finite set of possible amplitude.
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Process of assigning values to sampled signal (discrete levels)
Process of rounding off amplitudes of flat top samples to a manageable number of levels
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PART-A
UNIT-4 SOURCE AND ERROR CONTROL CODING
1 What is information theory?
Information theory is needed to enable the communication system to carry information (signals) from sender to
receiver over a communication channel.
2 Define entropy?
 Entropy is defined in terms of probabilistic behavior of a source of information
 In information theory the source output are discrete random variables that have a certain fixed finite
alphabet with certain probabilities
3 How entropy is measured?
Entropy is an average information content for the given source symbol. (bits/message)
K 1
1
H   pk log 2 (
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)
k 0 pk
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4 What is meant by rate of information?
If a source generates at a rate of ‘r’ messages per second, the rate of information ‘R’ is defined as the average
number of bits of information per second.
‘H’ is the average number of bits of information per message. Hence R = rH bits/sec
5 What is source coding?
Source coding means that we will remove redundant information from the signal prior the transmission.
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Basically this is achieved by assigning short descriptions to the most frequent outcomes of the source output
and vice versa
6 List various source coding techniques?
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The common source-coding schemes are prefix coding, huffman coding, lempel-ziv coding.
7 State source coding theorem?
Source coding theorem states that the output of any information source having entropy H units per symbol can
be encoded into an alphabet having N symbols in such a way that the source symbols are represented by code
words having a weighted average length not less than H/logN.
Hence source coding theorem says that encoding of messages from a source with entropy H can be done,
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bounded by the fundamental information theoretic limitation that the Minimum average number of
symbols/message is H/logN.
8 What is prefix coding?
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Prefix coding has an important feature that it is always uniquely decodable and it also satisfies Kraft-McMillan
inequality term. Prefix codes can also be referred to as instantaneous codes, meaning that the decoding process
is achieved immediately
9 What is Shannon Fano coding?
In Shannon–Fano coding, the symbols are arranged in order from most probable to least probable, and then
divided into two sets whose total probabilities are as close as possible to being equal. All symbols then have the
first digits of their codes assigned; symbols in the first set receive "0" and symbols in the second set receive "1".
10 What is procedure for Huffman coding?
Huffman Coding: Create a list for the symbols, in decreasing order of probability. The symbols with the lowest
probability are assigned a ‘0’ and a ‘1’. These two symbols are combined into a new symbol with the probability
equal to the sum of their individual probabilities. The new symbol is placed in the list as per its probability value.
The procedure is repeated until we are left with 2 symbols only for which 0 and 1 are assigned. Huffman code is
the bit sequence obtained by working backwards and tracking sequence of 0’s and 1’s assigned to that symbol
and its successors.
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11 State the drawbacks of Huffman coding?
A drawback of Huffman code is that knowledge of probability model of source is needed.
Huffman’s algorithm encodes blocks of fixed size into binary sequences of variable length.
12 What is Lempel-ziv coding?
Lempel-Ziv coding is performed by parsing the source data into segments that are the shortest subsequences
not encountered before.
13 How is mutual information expressed?
Mutual information between the source X and the receiver Y can be expressed as:
I(X,Y) = H(X) - H(X|Y)
H(X) is the uncertainty of source X and H(X/Y) is the uncertainty of X given Y. Hence the quantity H(X) - H(X|Y)
represents the reduction in uncertainty of X given the knowledge of Y. Hence I(X,Y) is termed mutual
information
14 What is channel capacity?
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Capacity in the channel is defined as a intrinsic ability of a channel to convey information
15 How channel capacity is measured using mutual information?
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Using mutual information the channel capacity of a discrete memory less channel is the maximum average
mutual information in any single use of channel over all possible probability distributions.
Thus Channel capacity C=max( I(X,Y) ).
16 State Shannon’s channel coding theorem?
The Shannon theorem states that given a noisy channel with channel capacity C and information transmitted at
a rate R, then if R < C there exist codes that allow the probability of error at the receiver to be made arbitrarily
small. This means that theoretically, it is possible to transmit information nearly without error at any rate below
R
a limiting rate, C.
17 State converse of Shannon’s channel coding?
The converse is also important. If R > C, an arbitrarily small probability of error is not achievable. All codes will
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have a probability of error greater than a certain positive minimal level, and this level increases as the rate
increases. So, information cannot be guaranteed to be transmitted reliably across a channel at rates beyond the
channel capacity.
18 State Shannon Hartley theorem?
An application of the channel capacity concept to an additive white Gaussian noise channel with B Hz bandwidth
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and signal-to-noise ratio S/N is the Information Capacity Theorem.

It states that for a band-limited Gaussian channel operating in the presence of additive Gaussian noise, the
channel capacity is given by C = B log2(1 + S/N) where C is the capacity in bits per second, B is the bandwidth of
the channel in Hertz, and S/N is the signal-to-noise ratio.
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19 State Information capacity theorem?

An application of the channel capacity concept to an additive white Gaussian noise channel with B Hz bandwidth
and signal-to-noise ratio S/N is the Information Capacity Theorem.
It states that for a band-limited Gaussian channel operating in the presence of additive Gaussian noise, the
channel capacity is given by C = B log2(1 + S/N) where C is the capacity in bits per second, B is the bandwidth of
the channel in Hertz, and S/N is the signal-to-noise ratio.
20 What is the tradeoff between SNR and bandwidth?
As the bandwidth of the channel increases, it is possible to make faster changes in the information signal,
thereby increasing the information rate.
However, as B is infinite, the channel capacity does not become infinite since, with an increase in bandwidth,
the noise power also increases.
As S/N increases, one can increase the information rate while still preventing errors due to noise.
For no noise, S/N is infinite and an infinite information rate is possible irrespective of bandwidth.
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21 Draw the bandwidth efficiency diagram?
22 State rate distortion theory?

Rate distortion theory is the branch of information theory addressing the problem of determining the minimal
amount of entropy or information that should be communicated over a channel such that the source can be
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reconstructed at the receiver with a given distortion.
23 When rate distortion theory is applicable?
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Rate distortion theory can be used for the given below situations:
1. Source coding in which the coding alphabet cannot exactly represent the source information.
2. when the information is to be transmitted at a rate greater than channel capacity
24 What is rate distortion function?
The functions that relate the rate and distortion are found as the solution of the following minimization
problem.
25 Give the expression for rate distortion function?
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26 Mention the rate distortion function for Gaussian memory less source?
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If Px(X) is Gaussian, variance is

independent, we find the following analytical expression for the rate distortion function.
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27 What is lossy source coding?

Lossy source coding is the representation of the source in digital form with as few bits as possible while
maintaining an acceptable loss of information
28 How lossy source coding helps in reduction of information at source?
In lossy source coding, the source output is encoded at a rate less than the source entropy. Hence there is
reduction in the information content of the source
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PART-A UNIT-5 MULTI USER RADIO COMMUNICATION

1 What does multi-user communication refers to?
Multiuser communications refer to the simultaneous use of a communication Channel by a number of
users. Multiple access techniques are used to allow a large number of mobile users to share the allocated
spectrum in the most efficient manner
2 List the different multiple access techniques?
1) Frequency division multiple-access (FDMA)
2) Time division multiple-access (TDMA)
3) Code division multiple-access (CDMA)
4) Space Division multiple access (SDMA)
3 What is meant by narrowband systems?
The term narrowband is used to relate the bandwidth of the single Channel to the expected coherence
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bandwidth of the Channel. The available spectrum is divided in to a large number of narrowband
Channels. The Channels are operated using FDD.
4 What is meant by wideband systems?
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In wideband systems, the transmission bandwidth of a single Channel is much larger than the coherence
bandwidth of the Channel. Thus, multipath fading does not greatly affect the received signal within a
wideband Channel, and frequency selective fades occur only in a small fraction of the signal bandwidth
5 Mention the salient features of FDMA?
•The FDMA Channel can be used by only one user at a time. If an FDMA Channel is not in use, then it sits
idle and it cannot be used by other users to increase share capacity.
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•After the assignment of the Channel to a particular User, The user is allowed to transmit and receive
simultaneously and continuously through the duplexed Channel.
•The bandwidths of FDMA systems are generally narrow.
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•The symbol time is large compared to the average delay spread
6 What is meant by Guard band?
 A Guard band is an unused part of the radio spectrum between radio bands, for the purpose of
preventing interference. It is a narrow frequency range used to separate two wider frequency
ranges to ensure that both can transmit simultaneously without interfering with each other.
 A Guard band with BW fg always present in between two FDMA Channels, for the purpose of
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preventing Interferences.
7 What is TDMA?
In TDMA, the entire bandwidth is available to the user but only for a finite period of time. In most cases
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the available bandwidth is divided into fewer Channels compared to FDMA and the users are allotted time
slots during which they have the entire Channel bandwidth at their disposal.
8 State the features of TDMA?
1. TDMA shares a single carrier frequency with several users where each user makes use of non-
overlapping time slots.
2. The number of time slots per frame depends on several factors such as modulation technique, available
bandwidth etc.
3. Data transmission in TDMA is not continuous but occurs in bursts. This results in low battery
consumption since the subscriber transmitter can be turned OFF when not in use.
4. Because of a discontinuous transmission in TDMA the handoff process is much simpler for a subscriber
unit, since it is able to listen to other base stations during idle time slots.
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9 What is SSMA?
Spread spectrum multiple access (SSMA) uses signals which have a transmission bandwidth whose
magnitude is greater than the minimum required RF bandwidth. A pseudo noise (PN) sequence converts a
narrowband signal to a wideband noise like signal before transmission.
SSMA is not very bandwidth efficient when used by a single user.
10 What are the types of spread spectrum multiple technique?
There are two main types of spread spectrum multiple access techniques:
1. Frequency Hopped Multiple Access (FHMA)
2. Direct sequence multiple access (DSMA) or Code division multiple access (CDMA).
11 What is meant by CDMA?
In CDMA, the same bandwidth is occupied by all the users, however they are all assigned separate codes,
which differentiates them from each other (shown in Figure). CDMA utilize a spread spectrum technique
in which a spreading signal (which is uncorrelated to the signal and has a large bandwidth) is used to
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spread the narrowband message signal.
12 List the essential parts of GSM?
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A GSM system consists essentially of three parts – namely, the Base Station Subsystem (BSS), the
Network and Switching Subsystem (NSS), and the Operation Support System (OSS).
13 What is need for traffic channels?
Payload data are transmitted via the TCHs. The payload might consist of encoded voice data or “pure”
data.
14 What are Broadcast channels?
BCHs are only found in the downlink. They serve as beacon signals. They provide the MS with the initial
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information that is necessary to start the establishment of any kind of connection. The MS uses signals
from these Channels to establish synchronization in both time and frequency. Furthermore, these
Channels contain data regarding, e.g., cell identity.
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15 When Paging channel is used?

When a request – e.g., from a landline – arrives at the BS to establish a connection to a specific MS, the
BSs within a location area send a signal to all MSs within their range. This signal contains either the
permanent International Mobile Subscriber Identity (IMSI) or the Temporary Mobile Subscriber Identity
(TMSI) of the desired MS. The desired MS continues the process of establishing the connection by
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requesting (via a Random Access Channel (RACH)) a TCH.

16 What is access grant channel?
Upon the arrival of a connection request via the RACH, the first thing that is established is a Dedicated
Control Channel (DCCH) for this connection. This Channel is called the Standalone Dedicated Control
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Channel (SDCCH). This Channel is assigned to the MS via the AGCH, which can only be found in the
downlink
17 What are dedicated control channels?
Similar to the TCHs, the DCCHs are bidirectional – i.e., they can be found in the uplink and downlink. They
transmit the signaling information that is necessary during a connection. As the name implies, DCCHs are
dedicated to one specific connection
18 What is meant by the term frequency reuse?
 Frequency reuse refers to the use of radio channels on the same carrier frequency to cover
different areas which are separated from one another by sufficient distances so that co-channel
interference is not objectionable.
 Frequency reuse is employed not only in mobile-telephone service but also in entertainment
broadcasting and most other radio services
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19 Why the cell shape is chosen hexagonal?
Universally adopted since the hexagon permits easy and manageable analysis of a cellular system.
20 What is channel assignment?
21 What are the different types of channel assignment?

1. Fixed channel assignment 2. Dynamic channel assignment
22 What is meant by fixed channel assignment?
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22 What is meant by dynamic channel assignment?
23 Explain handoff?
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When a mobile moves into a different cell while a conversation is in progress, the MSC automatically
transfers the call to a new channel belonging to the new base station. This handoff operation not only
involves identifying a new base station, but also requires that the voice and control signals be allocated to
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channels associated with the new base station
24 What are the different types of handoff?
1. Soft Handoff
2. Hard Handoff
25 What is hard handoff?
When the user moves to a new cell, he will be assigned with a new set of channels.
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26 What is soft handoff?

When the user moves to a new cell, the channel itself will be switched to the new base station. CDMA
uses soft handoff.
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27 What is Bluetooth?
Bluetooth is a standard used in links of radio of short scope, destined to replace wired connections
between electronic devices like cellular telephones, Personal Digital Assistants (PDA), computers, and
many other devices. Bluetooth technology can be used at home, in the office, in the car, etc. This
technology allows to the users instantaneous connections of voice and information between several
devices in real time.
28 Mention few applications of Bluetooth?
Wireless networking between PCs in a confined space and where little bandwidth is required.
Wireless communications with PC input and output devices, the most common being the mouse,
keyboard and printer.
Transfer of files between devices with OBEX (a kind of communications protocol).
Replacement of traditional wired serial communications in test equipment, GPS receivers, medical
equipment, bar code scanners, and traffic control devices.
For controls where infrared was traditionally used.
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29 What is a satellite?
A satellite is an artificial object which has been intentionally placed into orbit. Such objects are sometimes
called artificial satellites to distinguish them from natural satellites such as the Moon. Satellites are used
for a large number of purposes. Common types include military and civilian Earth observation satellites,
communications satellites, navigation satellites, weather satellites, and research satellites
30 What are the uses of satellites?
Satellites that are used for transmitting or broadcasting useful information for long ranges are known as
communication satellites. A communications satellite or comsat is an artificial satellite sent to space for
the purpose of telecommunications
31 List the classification of satellite orbits?
•Low Earth orbit
•Polar orbit
•Geostationary orbit
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32 What are geo-synchronous orbits?
A geosynchronous orbit (sometimes abbreviated GSO) is an orbit around the Earth with an orbital period
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of one side real day (approximately 23 hours 56 minutes and 4 seconds), matching the Earth's sidereal
rotation period.
The synchronization of rotation and orbital period means that, for an observer on the surface of the Earth,
an object in geosynchronous orbit returns to exactly the same position in the sky after a period of one
sidereal day
33 What is geo-stationary Earth orbits (GEO)?
A special case of geosynchronous orbit is the geostationary earth orbit, which is a circular geosynchronous
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orbit at zero inclination (that is, directly above the equator). A satellite in a geostationary orbit appears
stationary, always at the same point in the sky, to ground observers
34 What is meant by LEO (Low Earth Orbits)?
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A low Earth orbit (LEO) is an orbit around Earth with an altitude between 160 kilometers (99 mi), with an
orbital period of about 88 minutes, and 2,000 kilometers (1,200 mi), with an orbital period of about 127
minutes. Objects below approximately 160 kilometers (99 mi) will experience very rapid orbital decay and
altitude loss
35 Define apogee and Perigee?
 Apogee and Perigee refer to the moon and its relation to the earth.
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 The moon's orbit is not circular, but elliptical and when the moon is at its closest point to earth, it
is said to be at its perigee and when it's at its furthest point it is said to be at its apogee.
36 State Kepler’s first law of planetary motion?
ST
The orbit of every planet is an ellipse with the Sun at one of the two foci.
37 State Kepler’s second law of planetary motion?

A line joining a planet and the Sun sweeps out equal areas during equal intervals of time.
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38 State Kepler’s third law of planetary motion?
The square of the orbital period of a planet is directly proportional to the cube of the semi-major axis of
its orbit. The third law captures the relationship between the distance of planets from the Sun, and their
orbital periods.
39 What are satellite links?
The communication going from a satellite to ground is called downlink, and when it is going from ground
to a satellite it is called uplink
40 What is meant by two-way communication?
When an uplink is being received by the spacecraft at the same time a downlink is being received by
Earth, the communication is called two-way.
41 What is meant by one-way communication?
If there is only an uplink happening, this communication is called upload. If there is only a downlink
happening, the communication is called one-way
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42 What is meant by cross-link?
The communication happening between two satellites is characterized by a Cross link.
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HINTS TO PART-B QUESTIONS
PART B
UNIT-1
1. What is modulation? Explain
i)Modulation
ii)Need for modulation
iii)Types of modulation
2. Distinguish between FM and PM by giving its mathematical analysis.
The maximum and minimum values of any cosine function is
Max(ωi) = ωc+ KFMVm{ if cosωmt =+1}
Min(ωi) = ωc-KFMVm { if cosωmt =+1}

i)Expression For Phase Modulation
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ii)Frequency Deviation And Phase Deviation
iii)Wave Forms of FM & PM
3. Comparison Of Analog Modulation Systems
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PARAMETER
FOR AM FM PM
COMPARISON
A modulation technique in which the A modulation technique in A modulation technique in
amplitude of the carrier is varied in which the frequency of the which the phase of the carrier is
DEFINITION accordance with the change in carrier is varied in accordance varied in accordance with the
amplitude of message signal. with the change in amplitude of change in amplitude of message
message signal. signal.
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MESSAGE
SIGNAL
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CARRIER
SIGNAL
MODULATED
SIGNAL
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AM signal ‘s instantaneous amplitude FM signal ‘s instantaneous PM signal ‘s instantaneous

MODULATION directly proportional to Amplitude of frequency directly proportional phase directly proportional to
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RELATION Message signal to Amplitude of Message signal Amplitude of Message signal
ω ω
VAM (t) =VC Sinωct + VFM(t)= Vc sin [ VPM(t) = Vc sin [
MATHEMATICAL ω
ω ω + ] + ]
EXPRESSION - ω
BANDWIDTH 2fm 2(Δf + fm) 2(Δf + fm)

REQUIRED
FREQUENCY
SPECTRUM
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4. Discuss about the sets of side bands produced when a carrier is frequency modulated by a single frequency
sinusoid.
i)Spectrum Of Angle Modulated Waves
ii)Bessel Functions
iii)Bandwidth Requirements of Angle Modulated Waves
5. Explain noise function

In all communication process, electrical disturbance are appearing as interference which is called as
NOISE.
i) Sources of noise ii) External noise iii) Thermal noise iv) Partition noise v) Internal noise
vi) Flicker noise burst noise vii) Avalanche noise viii) Noise Calculations
6. Explain Amplitude Modulation.
i) Mathematical Representation of an Am Wave
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ii) Envelope of an AM Wave
iii) Frequency Spectrum and Bandwidth
iv) Sidebands of AM
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v) Frequency Domain Representation of AM Bandwidth of Am
vi) Representation of AM Voltage Distribution
vii) Power Distribution other Forms of AM
viii) DSB-SC AM
UNIT II
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1. Explain Amplitude shift keying.
i)Principle ii)Ask modulation sample waveform
iii)Diagram of ask modulationiv)Bit rate and baud rate
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2. Explain FSK.
i)Principl ii)FSK modulation sample waveforms block diagram of FSK modulation
iii)FSK demodulation schemes iv)FSK demodulation sample waveform
v)Bandwidth of FSK vi)Pros and cons of FSK
3. Explain MSK.
i)Mathematical representation ii)MSK waveforms iii)Gaussian MSK
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4. Explain PSK.
i)Phase shift keying ii)BPSK transmitter iii)Bandwidth of BPSK
iv)BPSK modulation waveform v)BPSK receiver vi)BPSK demodulation waveform
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5. Explain QPSK.
i)QPSK modulation waveforms ii)QPSK demodulation
6. Explain 8-Ary PSk and 16-Ary PSK with its truth table.
Wite the scheme logic of 8-Ary PSK and 16- ary PSK, with truth tables,constellation diagrams
7. Explain QAM in detail.
i)Types ii)Block diagram of 8 –QAM generator iii)Block diagram of 16 –QAM generator
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8. Explain the comparison of various digital modulation techniques.
PARAMATERS ASK FSK PSK QAM
BPSK QPSK DPSK
MESSAGE M:10110100 M:10110100
DI-BIT
ENCODING: ASSUMING
10 –S2 -> 135° 4-QAM
11-S1 ->+45° DI-BIT
01-S3 ->-45° ENCODING:
00-S4 ->-135° MSG:1010 10 –S2 ->
DIFFERENTIAL 135°
ENCODING 11-S1 -> +45°
0 1 1 0 WITH 01-S3 -> -45°
REF=0 00-S4 ->-135
TX: π-0-0- π
CARRIER
P
AP
MODULATED
WAVEFORM
TRUTH TABLE LOGIC 1 = Sinωct LOGIC 1 LOGIC 1 = Sinωct 00 -Sinωct -Cosωc AFTER 0 -Sinωct -
LOGIC 0 = 0 =Sin(fc+Δf)t LOGIC 0 = -Sinωct t DIFFERENTIAL 0 Cosωc t
LOGIC 0 = Sin(fc- ENCODING
RΔf)t 01 -Sinωct +
Cosωc t
LOGIC 1 = Sinωct+ 0
1
-Sinωct +
Cosωc t
π
10 +Sinωct - LOGIC 0 = Sinωct 1 +Sinωct -
Cosωc t 0 Cosωc t
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11 +Sinωct + 1 +Sinωct
Cosωc t 1 + Cosωc
t
ENCODING n=1; M= 2n=2 n=1; M= 2n=2 n=1; M= 2n=2 n=2; M= 2n=4 DifferentialEncodin DIBIT
SCHEME g by XNOR LOGIC ENCODING
(n,M) FOR 4-QAM
n=2; M= 2n=4
NYQUIST fb fb fb fb/2 <fb/2 fb/2
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BANDWIDTH
COMPLEXITY Coherent: MAJOR MAJOR MAJOR MODERATE MAJOR
MODERATE
Non-coherent:
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MINOR
APPLICATION Device Control Coherent: Most Used for High Used for High speed Widely used in For Long haul
(RARELEY USED) widely used for speed data data transmission lesser complex Transmission
domestic transmission systems, wireless of picture
communication systems, asynchronous communication signals.
systems asynchronous modems, systems
Non-Coherent: Slow modems DSL links
speed data modems
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UNIT III
1. What are all the standard organization for data communication? Explain.
Write shortly about the following standard organizations and the specifications given by those organizations
i)International Standard Organization (ISO)
ii)Consultative Committee for International Telephony & Telegraphy (CCITT)
iii)American National Standards Institutions (ANSI)
iv)International Telecommunication Union Telecommunication standards sector (ITU-T)
v)Institute of Electrical and Electronics Engineering (IEEE)
\vi)Standards Council of Canada (SCC)
2. Explain data communication with its circuit diagram.
i)DTE ii)DCE iii)TRANSMISSION MODES iv)TRANSMISSION CONFIGURATIONS
3. What is topology and explain its type?
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Categories of topology:
i)Mesh ii)Star iii)Ring (or) loop iv)Bus or Multidrop
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4. What is Protocols and explain its type?
i)Connection oriented ii)Connectionless oriented
5. What are the types of standards?
i)Proprietary (closed) system ii)Open system
6. Explain data communication codes and its types?
i)Baud of code ii)ASCII code iii)EBCDIC code iv)BAR code
7. What is error control and explain its classification?
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i)Single bit ii)Multiple bit error iii)Burst error
8. What is redundancy check and explain its types?
i)Vertical Redundancy Check (VRC) ii)Longitudinal redundancy checking
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iii)Cycle redundancy check iv)Checksum
9. What is error correction technique and explain its types
i)Symbol substitution ii)Retransmission iii)Forward error
10. What is a parallel interface. Explain its advantages and disadvantages.
Write about standard parallel interfaces and about IEEE 488 connectors
11. Explain Pulse Communication.
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UNIT IV
1. Explain Shannon-Hartley Theorem Or Information Capacity Theorem.
Statement, Expressions, Inference from the theorem, Conclusion.
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2. What is error control code and explain.

i)Forward Error Correction (FEC) ii) Block Codes iii) Hamming Distance
iv)Linear Block Codes v)Error Correcting Power of LBC vi)Systematic Codes
vii)Decoding Linear Codes viii)Error Syndrome
3. Explain convolution codes.
Write about convolutinal encoder and the Steps involved in convolutinal coding
4. What is Entropy and explain.
i)Rate Of Information
5. Explain Source coding.
i)SOURCE CODING THEOREM ii) SHANNON-FANO CODING iii) HUFFMAN CODING
iv)LEMPEL-ZIV CODING
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UNIT V
1. What is FDMA and explain its features?
i)Features of FDMA ii)FDMA/FDD in AMPS
iii)FDMA/TDD in CT2 iv)FDMA and NEAR-FAR PROBLEM
2. Explain TDMA with its Structure.
3. Explain Spread Spectrum Multiple Access.
i)FREQUENCY HOPPED MULTIPLE ACCESS (FHMA)
ii)CODE DIVISION MULTIPLE ACCESS CDMA/FDD IN IS-95
iii)NEAR-FAR PROBLEM
4. Explain Global system for mobile communication.
i)Base station subsystem
ii)Network and switching subsystem
iii)Operating support system
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5. Explain the cellular concept.
Principles of cellular networks, cell sectoring, cell assignment, frequency reuse, methods, Handoff strategies
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6. Explain in detail about Bluetooth.
Bluetooth Architecture, protocol stack, frame structures, Specifications, Applications
7. Explain CDMA.
i)Direct sequence spread spectrum ii)Frequency hopping spread spectrum
8. Explain Satellite communication.
i) Satellite orbits ii)Geostationary orbits iii)Low earth orbit
iv)Apsis- apogee and perigee v)Kepler's laws of planetary motion
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vi)Satellite links vii)Foot prints of a satellite
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ALL THE BEST
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1.1.4 TYPES OF COMMUNICATION SYSTEM:

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1.2.3 INTERNAL NOISE

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EQUIVALENT CIRCUIT OF A NOISY RESISTOR

Now, According to Ohms Law,

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For a finite bandwidth of 2Δf. The noise power is given by

FOR SERIAL AND PARALLEL COMBINATION OF SEVERAL RESISTORS

frequency amplification. Partition noise has a flat spectrum.

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Volt equivalent of Thermal Noise: Vn=√ Volts

Sy1+y2+y3+….(ω) = Sy1(ω) + Sy2(ω) + Sy3(ω) +…..

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modulation, the number of channels required is reduced.

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In Pulse Transmission methods such as PCM,DPCM, DM and ADM,

Message : Digital code signal (Low frequency)

Message : Digital Pulse signal (Low frequency)

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This equation contains 3 radicals,

Lower Side Band ; (-)  Upper Side Band

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Then modulation index is given by,

1.6.10 AM POWER DISTRIBUTION

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Power in Side bands:

TOTAL POWER IN AM:

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Transmission bandwidth becomes twice that of the frequency of message signal.

When multiplying these signals, it results in DSB-SC AM.

VDSB-SC(t) =Vm(t) .Vc(t)

= Vm sin ωmt .Vc sin ωct

= VmVc [sin ωmt .sin ωct]

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= VmVc [cos (ωmt) .cos (ωct) + sin (ωmt) .sin (ωct) ]

1.7.5 PHASOR REPRESENTATION OF SSB-SC AM

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 Vestige Part of interest

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FEATURES OF ANGLE MODULATION SYSTEMS

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Assume that, φ(t) = ωct+θ(t)

Min(ωi) = ωc-KFMVm { if cosωmt =+1}

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After Phase Modulation,

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BANDWIDTH 2fm 2(Δf + fm) 2(Δf + fm)

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