Wireless Communication

Anna University
Solved Question Papers
B.E./B.Tech. 7th Semester

Electronics and Communication
Engineering
ECE_Semester-V_FM.indd iii 7/19/2012 10:32:36 AM

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ACKNOWLEDGEMENT
Mr K. Murali Babu
Department of Electronics and Communication,
GIT, Vellore
Semester-VII
Wireless
Communication
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EEE_Sem-VI_Chennai_FM.indd iv 12/7/2012 6:40:43 PM

B.E./B.Tech. DEGREE EXAMINATION,
NOV/DEC 2013
Seventh Semester
Electronics and Communication Engineering
Wireless Communication
Time: Three hours Maximum: 100 marks
Answer ALL questions
PART A (10 2 = 20 marks)
1. What are three most important effects of small-scale multipath propagation?
2. What is a multiple access technique?
3. State the difference between Narrowband and wideband systems.
4. Find the far-field distance for an antenna with maximum dimension of

1 m and operating frequency of 900 MHz.
5. Give the expression for bit error probability of Gaussian Minimum shift
Keying modulation.
6. What is fading and Doppler spread?
7. What is Diversity?
8. What is Equalization?
9. What is a PN sequence? Give its significance in spread spectrum

modulation technique.
10. What is DECT?
PART B (5 16 = 80 marks)
11. (a) Discuss the types of services, requirements, spectrum limitations
and noise considerations of wireless communications.
Or

2.4 B.E./B.Tech. Question Papers
(b) Explain the principle of Cellular Networks and various types of

Handoff techniques.
1 2. (a) (i) Briefly explain the factors that influence small-scale fading.
(ii) If a transmitter produces 50 W of power, express the transmit
power in units of dBM and dBW. If 50 W is applied to a unity
gain antenna with a 900 MHz carrier frequency, find the
received power in dBM at a free space distance of 100 m from
the antenna. What is Pr (10 km)? Assume unity gain for the
receiver antenna.
Or
(b) (i) Briefly explain the three basic propagation mechanisms which
impact propagation in a mobile communication system.
(ii) What is Brewster angle? Calculate the Brewster angle for a
wave impinging on ground having a permittivity of er = 4.
1 3. (a) (i) Explain the Nyquist criterion for ISI cancellation.

(ii) With transfer function, explain the raised cosine roll off filter.
Or
(b) (i) Explain the QPSK transmission and detection techniques.
(ii) Explain the performance of Digital modulation in slow flat-
fading channels.
1 4. (a) Explain in detail about:

(i) Linear Equalizers.
(ii) Non Linear Equalizers.
Or
(b) (i)
With block diagram, explain the operation of a RAKE
receiver.
(ii) Briefly explain the frequency domain coding of speech signals.
1 5. (a) Explain in detail about:

(i) Direct sequence spread spectrum technique.
(ii) Frequency hopped spread spectrum technique.
Or
(b) Discuss in detail about second generation (2G) and third generation
(3G) wireless networks and standards.

Solutions
PART A
1. The three most important effects of small-scale multi-path propagation
are
(i)Rapid changes in signal strength over a small travel distance or time
interval
(ii)Random frequency modulation due to varying Doppler shifts on
different multi-path signals and
(iii)Time dispersion (echoes) caused by multi-path propagation delays.
2. Multiple access technique are used to allow many mobile users to share
simultaneously a finite amount of radio spectrum.
3. Narrowband systems Here the available radio spectrum is divided

into large number of narrowband channels. The bandwidth of a single
channel is nothing but coherence bandwidth of the channel.
Wideband systems The transmission bandwidth of a single channel
is much larger than the coherence bandwidth of the channel.
4. Operating frequency, f = 900 MHz

l = c/f = 3 108 m/s/900 106 Hz = 0.33 m
Fraunhofer distance, df = 2D2/l = 2(1)2/0.33 = 6 m
Path loss PL(dB) = -10 log [(l2)/(4p)2d2] = -10 log [(0.33)2/(4 3.14)2
36] = 47dB
5. The error probability for GMSK is given by

2g Eb
Pe = Q
N O
Where g is a constant related to bandwidth-bit duration product (BT) of
GMSK filter.
0.68 for GMSK with BT = 0.25
g
0.85 for simple MSK ( BT = )
6. Fading is used to describe the rapid fluctuations of the amplitudes,
phases, or multi-path delays of a radio signal over a short period of time
or travel distance.

Doppler spread is a measure of the spectral broadening caused by the

time rate of change of the mobile radio channel.
7. Diversity is technique used to compensate for fading channel impairments

and is usually implemented by using two or more receiving antennas.
8. Equalization compensates for inter symbol interference (ISI) created by

multipath within time dispersive channels.
9. The pseudo-noise (PN) sequence is the spreading codes which is a binary

sequence that appears random but can be reproduced in a deterministic
manner by intended receivers. Each user is assigned a unique PN code
which is approximately orthogonal to the codes of other users, the
receiver can separate each user based on their codes, even though they
occupy the same spectrum at all times.
10. The Digital European Cordless Telephone (DECT) is a universal cordless

telephone standard developed by the European Telecommunications
Standards Institute (ETSI). It is the first pan-European standard for
cordless telephones and was finalized in July 1992.
PART B
11. (a) Paging Systems
Paging systems are communication systems that send brief messages
to a subscriber. Depending on the type of service, the message may
be either a numeric message, an alphanumeric message, or a voice
message. Paging systems are typically used to notify a subscriber of
the need to call a particular telephone number or travel to a known
location to receive further instructions. In modern paging systems, news
headlines, stock quotations, and faxes may be sent. A message is sent to
a paging subscriber via the paging system access number (usually a toll-
free telephone number) with a telephone keypad or modem. The issued
message is called a page. The paging system then transmits the page
throughout the service area using base stations which broadcast the page
on a radio carrier.
Paging systems vary widely in their complexity and coverage
area. While simple paging systems may cover a limited range of 2 to
5 km, or may even be confined to within individual buildings, wide
area paging systems can provide worldwide coverage. Though paging
receivers are simple and inexpensive, the transmission system required

Wireless Communication (Nov/Dec 2013) 2.7
is quite sophisticated. Wide area paging systems consist of a network of

telephone lines, many base station transmitters, and large radio towers
that simultaneously broadcast a page from each base station (this is
called simulcasting). Simulcast transmitters may be located within the
same service area or in different cities or countries. Paging systems are
designed to provide reliable communication to subscribers wherever
they are; whether inside a building, driving on a highway, or flying in
Figure 1 A wide area paging system. The paging control center dispatches
pages received from the PSTN throughout several cities at the same time.
an airplane. This necessitates large transmitter powers (on the order of

kilowatts) and low data rates (a couple of thousand bits per second) for
maximum coverage from each base station. Figure shows a diagram of a
wide area paging system.
Cordless Telephone Systems
Cordless telephone systems are full duplex communication systems
that use radio to connect a portable handset to a dedicated base station,
which is then connected to a dedicated telephone line with a specific
telephone number on the public switched telephone network (PSTN).

In first generation cordless telephone systems (manufactured in the

1980s), the portable unit communicates only to the dedicated base unit
Figure 2 A cordless telephone system.
and only over distances of a few tens of meters. Early cordless telephones
operate solely as extension telephones to a transceiver connected to a
subscriber line on the PSTN and are primarily for in-home use.
Second generation cordless telephones have recently been introduced
which allow subscribers to use their handsets at many outdoor
locations within urban centers such as London or Hong Kong. Modern
cordless telephones are sometimes combined with paging receivers
so that a subscriber may first be paged and then respond to the page
using the cordless telephone. Cordless telephone systems provide the
user with limited range and mobility, as it is usually not possible to
maintain a call if the user travels outside the range of the base station.
Typical second generation base stations provide coverage ranges
up to a few hundred meters. Figure 2 illustrates a cordless telephone
system.
Cellular Telephone Systems

A cellular telephone system provides a wireless connection to the PSTN
for any user location within the radio range of the system. Cellular
systems accommodate a large number of users over a large geographic
area, within a limited frequency spectrum. Cellular radio systems provide
high quality service that is often comparable to that of the landline
telephone systems. 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. A sophisticated switching technique called

a handoff enables a call to proceed uninterrupted when the user moves

from one cell to another.
Figure 3 shows a basic cellular system which consists of mobile
stations, base stations and a mobile switching center (MSC). The mobile
switching center is sometimes called a mobile telephone switching
office (MTSO), since it is responsible for connecting all mobiles to the
PSTN in a cellular system. 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. The mobile station contains a
transceiver, an antenna, and control circuitry, and may be mounted in a
vehicle or used as a portable hand-held unit. The base stations consist
of several transmitters and receivers which simultaneously handle full
duplex communications and generally have towers which support several
transmitting and receiving antennas. The base station serves as a bridge
Figure 3 A cellular system. The towers represent base stations which

provide radio access between mobile users and the mobile switching center
(MSC).
between all mobile users in the cell and connects the simultaneous
mobile calls via telephone lines or microwave links to the MSC. The
MSC coordinates the activities of all of the base stations and connects
the entire cellular system to the PSTN. A typical MSC handles 100,000
cellular subscribers and 5,000 simultaneous conversations at a time, and
accommodates all billing and system maintenance functions, as well. In
large cities, several MSCs are used by a single carrier.

11. (b) Cellular Networks

Cellular radio systems rely on an intelligent allocation and reuse of
channels throughout a coverage region [Oet83]. 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 to within the boundaries of a cell, the same group of channels 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 design process of selecting and allocating channel groups for all of
the cellular base stations within a system is called or frequency reuse or
frequency planning [Mac79].
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. The hexagonal cell shape is conceptual and is a simplistic model
of the radio coverage for each base station, but it has been 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
foot print and is determined from field measurements or propagation
prediction models. Although the real footprint is amorphous in nature, a
regular cell shape is needed for systematic system design and adaptation
for future growth. While it might seem natural to choose a circle to
represent the coverage area of a base station, adjacent circles cannot
be overlaid upon a map without leaving gaps or creating overlapping
regions. Thus, when considering geometric shapes which cover an entire
region without overlap and with equal area, there are three sensible
choicesa 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. Thus, by using the hexagon geometry, the
fewest number of cells can cover a geographic region, and the hexagon
closely approximates a circular radiation pattern which would occur for
an omnidirectional base station antenna and free space propagation. Of
course, the actual cellular footprint is determined by the contour in which
a given transmitter serves the mobiles successfully.
When using hexagons to model coverage areas, base station transmitters
are depicted as either being in the center of the cell (center-excited cells) or on

three of the six cell vertices (edge-excited cells). Normally, omnidirectional

antennas are used in center-excited cells and sectored directional antennas
are used in corner-excited cells. Practical considerations usually do not
allow base stations to be placed exactly as they appear in the hexagonal
layout. Most system designs permit a base station to be positioned up to
one-fourth the cell radius away from the ideal location.
To understand the frequency reuse concept, consider a cellular system
which has a total of S duplex channels available for use. If each cell
is allocated a group of k channels (k < S), and if the S channels are
divided among N cells into unique and disjoint channel groups which
each have the same number of channels, the total number of available
radio channels can be expressed as
S = kN (1)
The N cells which collectively use the complete set of available
frequencies is called a cluster. If a cluster is replicated M times within
the system, the total number of duplex channels, C can be used as a
measure of capacity and is given by
C = MkN = MS (2)
Handoff Technique
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 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 D = Pr handoff - Pr minimum usable, cannot be too large or too small. If
D is too large, unnecessary handoffs which burden the MSC may occur,
and if D is too small, there may be insufficient time to complete a handoff
before a call is lost due to weak signal conditions. Therefore, D is chosen
carefully to meet these conflicting requirements. Figure 1 illustrates a
handoff situation. Figure 1 demonstrates the case where a handoff is not

made and the signal drops below the minimum acceptable level to keep
the channel active. This dropped call event can happen when there is an
excessive delay by the MSC in assigning a handoff or when the threshold
D 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
Figure 1
nearby base stations (thus forcing the MSC to wait until a channel in a
nearby cell becomes free).
Types: the two types of Handoff techniques are:
Hard Handoff: With hard handoff, the link to the prior base station
is terminated before or as the user is transferred to the new cells base
station. That is to say that the mobile is linked to no more than one base
station at a given time.

Soft Handoff: CDMA usessoft handoff. Soft handoff is beneficial

because it reduces interference into other cells and improves performance
by usingmacro diversity.
12. (a) (i) Factors Influencing Small-Scale Fading

Many physical factors in the radio propagation channel influence small-
scale fading. These include the following:
Multipath propagation The presence of reflecting objects and
scatterers in the channel creates a constantly changing environment
that dissipates the signal energy in amplitude, phase, and time. These
effects result in multiple versions of the transmitted signal that arrive
at the receiving antenna, displaced with respect to one another in
time and spatial orientation. The random phase and amplitudes of
the different multipath components cause fluctuations in signal
strength, thereby inducing small-scale fading, signal distortion, or
both. Multipath propagation often lengthens the time required for the
baseband portion of the signal to reach the receiver which can cause
signal smearing due to intersymbol interference.
Speed of the mobile The relative motion between the base
station and the mobile results in random frequency modulation due
to different Doppler shifts on each of the multipath components.
Doppler shift will be positive or negative depending on whether the
mobile receiver is moving toward or away from the base station.
Speed of surrounding objects If objects in the radio channel are
in motion, they induce a time varying Doppler shift on multipath
components. If the surrounding objects move at a greater rate than
the mobile, then this effect dominates the small-scale fading.
Otherwise, motion of surrounding objects may be ignored, and only
the speed of the mobile need be considered. The coherence time
defines the staticness of the channel, and is directly impacted by
the Doppler shift.
The transmission bandwidth of the signal If the transmitted radio
signal bandwidth is greater than the bandwidth of the multipath
channel, the received signal will be distorted, but the received signal
strength will not fade much over a local area (i.e., the small scale signal
fading will not be significant). As will be shown, the bandwidth of the
channel can be quantified by the coherence bandwidth which is related to
the specific multipath structure of the channel. The coherence bandwidth
is a measure of the maximum frequency difference for which signals
are still strongly correlated in amplitude. If the transmitted signal has
a narrow bandwidth as compared to the channel, the amplitude of the

signal will change rapidly, but the signal will not be distorted in time.
Thus, the statistics of small-scale signal strength and the likelihood of
signal smearing appearing over small-scale distances are very much
related to the specific amplitudes and delays of the multipath channel,
as well as the bandwidth of the transmitted signal.
12. (a) (ii) Given:

Transmitter power, Pt = 50 W
Carrier frequency, fc = 900 MHz
Using Equation (4.9),
(a) Transmitter power,
Pt (dBm) = 10log [Pt (mW)/ (1 mW)]
= 10log [50 103] = 47.0 dBm.
(b) Transmitter power,
Pt (dBW) = 10log [Pt (mW)/(1 mW)]
= 10log [50] = 17.0 dBW.
The received power can be determined using Equation
t tGr l
2
PG 50(1)(1)(1/ 3) 2
Pr = = = (3.5 10 6 )W = 3.5 10 3 mW
( 4p ) 2 d 2 L ( 4p ) 2 (100) 2 (1)
Pr (dBm) = 10 log Pr ( mW) = 10 log(3.5 10 3 mW) = 24.5 dBm.

The received power at 10 km can be expressed in terms of dBm using
Equation, where d0 = 100 m and d = 10 km
100
Pr (10 km) = Pr (100) + 20 log = 24.5 dBm 40 dB
10000
= 64.5 dBm.
12. (b) (i) The Three Basic Propagation Mechanisms

Reflection, diffraction, and scattering are the three basic propagation
mechanisms which impact propagation in a mobile communication
system. These mechanisms are briefly explained in this section, and
propagation models which describe these mechanisms are discussed
subsequently in this chapter. Received power (or its reciprocal, path
loss) is generally the most important parameter predicted by large-scale
propagation models based on the physics of reflection, scattering, and
diffraction. Small-scale fading and multipath propagation (discussed in
Chapter 5) may also be described by the physics of these three basic
propagation mechanisms.

Reflection occurs when a propagating electromagnetic wave impinges

upon an object which has very large dimensions when compared to the
wavelength of the propagating wave. Reflections occur from the surface
of the earth and from buildings and walls.
Diffraction occurs when the radio path between the transmitter and
receiver is obstructed by a surface that has sharp irregularities (edges).
The secondary waves resulting from the obstructing surface are present
throughout the space and even behind the obstacle, giving rise to a
bending of waves around the obstacle, even when a line-of-sight path
does not exist between transmitter and receiver. At high frequencies,
diffraction, like reflection, depends on the geometry of the object, as
well as the amplitude, phase, and polarization of the incident wave at the
point of diffraction.
Scattering occurs when the medium through which the wave travels
consists of objects with dimensions that are small compared to the
wavelength, and where the number of obstacles per unit volume is large.
Scattered waves are produced by rough surfaces, small objects, or by
other irregularities in the channel. In practice, foliage, street signs, and
lamp posts induce scattering in a mobile communications system.
12. (b) (ii) Brewster Angle
The Brewster angle is the angle at which no reflection occurs in the
medium of origin. It occurs when the incident angle qB is such that the
reflection coefficient || is equal to zero (see Figure). The Brewster angle
is given by the value of qB which satisfies
e1
sin(q B ) = (1)
e1 + e 2
For the case when the first medium is free space and the second medium
has a relative permittivity, er Equation (2) can be expressed
er 1
sin(q B ) = (2)
e r2 1
Note that the Brewster angle occurs only for vertical (i.e. parallel)
polarization.
In accordance to Equation (2), the Brewster angle can be given by,
sin(q B ) = 1/ 5 or, q B = sin 1 (1/ 5) = 26.56
For er = 15,
Brewster angle, q B = sin 1 (14 / 224) = 14.47

13. (a) (i) Nyquist Criterion for ISI Cancellation

Nyquist was the first to solve the problem of overcoming intersymbol
interference while keeping the transmission bandwidth low [Nyq28]. He
observed that the effect of ISI could be completely nullified if the overall
response of the communication system (including transmitter, channel,
and receiver) is designed so that at every sampling instant at the receiver,
the response due to all symbols except the current symbol is equal to zero.
If heff (t) is the impulse response of the overall communication system,
this condition can be mathematically stated as
Figure 1 Power spectral density of (a) unipolar NRZ, (b) bipolar RZ, and (c)
Manchester NRZ line codes.

Figure 2Time waveforms of binary line codes: (a) unipolar NRZ;

(b) bipolar RZ; (c) Manchester NRZ.
K n = 0
heff ( nTs ) = (1)
0 n 0

where Ts is the symbol period, n is an integer, and K is a non-zero constant.

The effective transfer function of the system can be represented as
heff (t ) = d (t ) * p (t ) * hc (t ) * hr (t ) (2)
where p(t) is the pulse shape of a symbol, hc(t) is the channel impulse
response, hr(t) and is the receiver impulse response. Nyquist derived
transfer functions Heff (f) which satisfy the conditions of Equation (1)
[Nyq28].
There are two important considerations in selecting a transfer
function Heff(f) which satisfy Equation (1). First, Heff(t) should have
a fast decay with a small magnitude near the sample values for n 0.
Second, if the channel is ideal (hc(t) = d(t)), then it should be possible
to realize or closely approximate shaping filters at both the transmitter

and receiver to produce the desired Heff(f). Consider the impulse

response in (3)
sin(p t /Ts )
heff (t ) = (3)
(p t ) /Ts
Clearly, this impulse response satisfies the Nyquist condition for ISI
cancellation given in Equation (1) (see Figure 3). Therefore, if the
overall communication system can be modeled as a filter with the
impulse response of Equation (3), it is possible to completely eliminate
the effects of ISI. The transfer function of the filter can be obtained by
taking the Fourier transform of the impulse response, and is given by
1 f
heff ( f ) = rect (4)
fs fs
This transfer function corresponds to a rectangular brick-wall filter
with absolute bandwidth fs/2, where fs is the symbol rate. While this
transfer function satisfies the zero ISI criterion with a minimum of
bandwidth, there are practical difficulties in implementing it, since it
corresponds to a noncausal system (heff(t) exists for t < 0) and is thus
difficult to approximate. Also, the (sin t)/t pulse has a waveform slope
that is 1/t at each zero crossing, and is zero only at exact multiples of
Ts, thus any error in the sampling time of zero-crossings will cause
significant ISI due to overlapping from adjacent symbols. (A slope of
1/t2 or 1/t3 is more desirable to minimize the ISI due to timing jitter in
adjacent samples.)
Figure 3 Nyquist ideal pulse shape for zero inter symbol interference
Nyquist also proved that any filter with a transfer function having a
rectangular filter of bandwidth f0 1/2Ts, convolved with any arbitrary
even function Z(f) with zero magnitude outside the passband of the

rectangular filter, satisfies the zero ISI condition. Mathematically, the

transfer function of the filter which satisfies the zero ISI condition can
be expressed as
f
H eff ( f ) = rect Z ( f ) (5)
f0
where Z(f) = Z(-f), and Z(f) = 0 for f f 0 1 / 2Ts . Expressed in

terms of the impulse response, the Nyquist criterion states that any filter
with an impulse response
sin(p t /Ts )
heff (t ) = z (t ) (6)
pt
can achieve ISI cancellation. Filters which satisfy the Nyquist criterion
are called Nyquist filters (Figure 4).
Assuming that the distortions introduced in the channel can be
completely nullified by using an equalizer which has a transfer function
that is equal to the inverse of the channel response, then the overall
transfer function Heff(f) can be approximated as the product of the transfer
functions of the transmitter and receiver filters. An effective end-to-end
transfer function of Heff(f) is often achieved by using filters with transfer
functions H eff (f ) at both the transmitter and receiver. This has the
advantage of providing a matched filter response for the system, while at
the same time minimizing the bandwidth and inter symbol interference.
Figure 4 Transfer function of a Nyquist pulse-shaping filter at baseband.
13. (a) (ii) Raised Cosine Roll off Filter

The most popular pulse shaping filter used in mobile communications is
the raised cosine filter. A raised cosine filter belongs to the class of filters

which satisfy the Nyquist criterion. The transfer function of a raised

cosine filter is given by
(1 a )
1 0 | f |
2Ts

1 p (| f | 2Ts 1 + a ) (1 a ) (1 + a )
H RC (f ) 1 + cos | f | (2)

2 2a 2Ts 2Ts
(1 + a )
0 | f |>
2Ts
where a is the rolloff factor which ranges between 0 and 1. This transfer
function is plotted in Figure 1 for various values of a. When a = 0,
the raised cosine roll off filter corresponds to a rectangular filter of
minimum bandwidth. The corresponding impulse response of the filter
can be obtained by taking the inverse Fourier transform of the transfer
function, and is given by
pt pat
sin cos
Ts Ts
hRC (t ) (11)
pt 4a t
2
1
2T s
The impulse response of the cosine rolloff filter at baseband is plotted

in Figure 2 for various values of a. Notice that the impulse response
decays much faster at the zero-crossings (approximately as 1/t3 for t Ts)
when compared to the brick-wall filter a = 0). The rapid time rolloff
allows it to be truncated in time with little deviation in performance
from theory. As seen from Figure 1, as the roll off factor a increases, the
bandwidth of the filter also increases, and the time sidelobe levels decrease
in adjacent symbol slots. This implies that increasing a decreases the
sensitivity to timing jitter, but increases the occupied bandwidth.
The symbol rate Rs that can be passed through a baseband raised
cosine rolloff filter is given by
1 2B
Rs = = (3)
Ts 1 + a
where B is the absolute filter bandwidth. For RF systems, the RF
passband bandwidth doubles and
B
Rs = (4)
1+ a

Figure 1 Magnitude transfer function of a raised cosine filter at baseband.
Figure 2 Impulse response of a raised cosine rolloff filter at baseband.
13. (b) (i) QPSK Transmission and Detection Techniques

Figure 2 shows a block diagram of a typical QPSK transmitter.
The unipolar binary message stream has bit rate Rb and is first converted
into a bipolar non-return-to-zero (NRZ) sequence using a unipolar to
bipolar converter. The bit stream m(t) is then split into two bit streams
mI(t) and mQ(t) (in-phase and quadrature streams), each having a bit rate
of Rs = Rb/2. The bit stream mI(t) is called the even stream and mQ(t)
is called the odd stream. The two binary sequences are separately
modulated by two carriers f1(t) and f2(t), which are in quadrature.

The two modulated signals, each of which can be considered to be a

BPSK signal, are summed to produce a QPSK signal. The filter at the
output of the modulator confines the power spectrum of the QPSK signal
within the allocated band. This prevents spill-over of signal energy
into adjacent channels and also removes out-of-band spurious signals
generated during the modulation process. In most implementations,
pulse shaping is done at baseband to provide proper RF filtering at the
transmitter output.
Figure 1 Block diagram of a QPSK transmitter.
Figure 2 Block diagram of a QPSK receiver

Figure 2 shows a block diagram of a coherent QPSK receiver. The
frontend bandpass filter removes the out-of-band noise and adjacent
channel interference. The filtered output is split into two parts, and
each part is coherently demodulated using the in-phase and quadrature

carriers. The coherent carriers used for demodulation are recovered from
the received signal using carrier recovery circuits of the type described
in Figure. The outputs of the demodulators are passed through decision
circuits which generate the in-phase and quadrature binary streams. The
two components are then multiplexed to reproduce the original binary
sequence.
13. (b) (ii)Performance of Digital Modulation in Slow Flat-Fading

Channels
As discussed in Chapter, flat-fading channels cause a multiplicative
(gain) variation in the transmitted signal s(t). Since slow flat-fading
channels change much slower than the applied modulation, it can be
assumed that the attenuation and phase shift of the signal is constant over
at least one symbol interval. Therefore, the received signal r(t) may be
expressed as
r (t ) = a (t ) exp( jq (t ))s (t ) + n (t ) 0 t T (1)
where a(t) is the gain of the channel, q(t) is the phase shift of the
channel, and n(t) is additive Gaussian noise.
Depending on whether it is possible to make an accurate estimate of
the phase q(t), coherent or noncoherent matched filter detection may be
employed at the receiver.
To evaluate the probability of error of any digital modulation scheme
in a slow flat-fading channel, one must average the probability of error
of the particular modulation in AWGN channels over the possible ranges
of signal strength due to fading. In other words, the probability of
error in AWGN channels is viewed as a conditional error probability,
where the condition is that a is fixed. Hence, the probability of error
in slow flat-fading channels can be obtained by averaging the error
in AWGN channels over the fading probability density function. In
doing so, the probability of error in a slow flat-fading channel can be
evaluated as

Pe = Pe ( X ) p ( X )dX (2)
0
where Pe (X) is the probability of error for an arbitrary modulation at

a specific value of signal-to-noise ratio X, X = a2Eb/N0, and p(X) is the
probability density function of X due to the fading channel. Eb and N0
are constants that represent the average energy per bit and noise power
density in a non-fading AWGN channel, and the random variable a2

is used to represent instantaneous power values of the fading channel,

with respect to the non-fading Eb/N0. It is convenient to assume 2 is
one, for a unity gain fading channel. Then, p(X) can simply be viewed as
the distribution of the instantaneous value of Eb/N0 in a fading channel,
and Pe(X) can be seen to be the conditional probability of bit errors for a
given value of the random Eb/N0 due to fading.
For Rayleigh fading channels, the fading amplitude a has a Rayleigh
distribution, so the fading power a2 and consequently X have a chi-
square distribution with two degrees of freedom. Therefore,
1 X
p(X ) = exp X 0 (3)

Eb 2
where = a is the average value of the signal-to-noise ratio. For
N0
a 2 = 1, note that corresponds to the average Eb/N0 for the fading
channel.
By using Equation (3) and the probability of error of a particular
modulation scheme in AWGN, the probability of error in a slow flat-
fading channel can be evaluated. It can be shown that for coherent binary
PSK and coherent binary FSK, Equation (4) evaluates to [Ste87]
1
Pe , PSK = 1 (coherent binary PSK) (4)
2 1+
1
Pe , FSK = 1 (coherent binary FSK) (5)
2 2+
It can also be shown that the average error probability of DPSK and
orthogonal noncoherent FSK in a slow, flat, Rayleigh fading channel are
given by
1
Pe , DPSK = (differential binary PSK) (6)
2(1 + )
1
Pe , NCFSK = (noncoherent orthogonal binary FSK) (7)
2+
Figure 3 illustrates how the BER for various modulations changes as a

function of Eb/N0 in a Rayleigh flat-fading environment. The figure was
produced using simulation instead of analysis, but agrees closely with
Equations (6) to (9) [Rap91b].

Figure 3Bit error rate performance of binary modulation schemes in a

Rayleigh flat-fading channel as compared to a typical performance curve in
AWGN.
For large values of Eb/N0 (i.e., large values of X), the error probability
Equations may be simplified as
1
Pe ,PSK = (coherent binary PSK) (8)
4
1
Pe ,FSK = (coherent binary FSK) (9)
2
1
Pe ,DPSK = (differential PSK)(10)
2
1
Pe ,NCFSK = (noncoherent orthogonal binary FSK) (11)

For GMSK, the expression for BER in the AWGN channel is given
in Equation (6.112.a) which when evaluated in Equation (2) yields a
Rayleigh fading BER of
1 d 1
Pe ,GMSK = 1 (coherent GMSK) (12)
2 d + 1 4d
where
0.68 for BT = 0.25

d (13)
0.85 for BT =

As seen from Equations (22) to (26), for lower error rates all five
modulation techniques exhibit an inverse algebraic relation between error
rate and mean SNR. This is in contrast with the exponential relationship
between error rate and SNR in an AWGN channel. According to these
results, it is seen that operating at BERs of 10-3 to 10-6 requires roughly a
30 dB to 60 dB mean SNR. This is significantly larger than that required
when operating over a nonfading Gaussian noise channel (20 dB to
50 dB more link is required). However, it can easily be shown that the
poor error performance is due to the non-zero probability of very deep
fades, when the instantaneous BER can become as low as 0.5. Significant
improvement in BER can be achieved by using efficient techniques such
as diversity or error control coding to totally avoid the probability of
deep fades, as shown in Chapter 7.
Work by Yao [Yao92] demonstrates how the analytical technique of
Equation (2) may be applied to desired signals as well as interfering
signals which undergo Rayleigh, Ricean, or log-normal fading.
14. (a) (i) Linear Equalizers

As mentioned in Section 7.5, a linear equalizer can be implemented
as an FIR filter, otherwise known as the transversal filter. This type of
equalizer is the simplest type available. In such an equalizer, the current
and past values of the received signal are linearly weighted by the filter
coefficient and summed to produce the output, as shown in Figure 1.
If the delays and the tap gains are analog, the continuous output of the
equalizer is sampled at the symbol rate and the samples are applied to
the decision device. The implementation is, however, usually carried out
in the digital domain where the samples of the received signal are stored
in a shift register. The output of this transversal filter before a decision is
made (threshold detection) is [Kor85]
N2
d k =
n = N1
(cn *) yk n (1)
where cn* represents the complex filter coefficients or tap weights, d k is

the output at time index k, yi is the input received signal at time t0 + iT,
t0 is the equalizer starting time, and N = N1 + N2 is the number of taps.
The values N1 and N2 denote the number of taps used in the forward
and reverse portions of the equalizer, respectively. The minimum mean
squared error E[|e(n)|2] that a linear transversal equalizer can achieve is
[Pro89]

Figure 1 Structure of a linear transversal equalizer.
p /T
T N0
E[| e ( n ) |2 ] = jw T 2
2 p /T | F (e )| + N 0
dw (2)
where F(ejwT) is the frequency response of the channel, and N0 is the

noise power spectral density.
The linear equalizer can also be implemented as a lattice filter, whose
structure is shown in Figure 2. In a lattice filter, the input signal yk is
transformed into a set of N intermediate forward and backward error signals,
fn(k) and bn(k) respectively, which are used as inputs to the tap multipliers
and are used to calculate the updated coefficients. Each stage of the lattice
is then characterized by the following recursive equations [Bin88]:
f 1 ( k ) = b1 ( k ) = y( k ) (3)
n
f n ( k ) = y( k ) K i y( k i ) = f n 1 ( k ) + K n 1 ( k )bn 1 ( k 1) (4)
i =1
n
bn ( k ) = y( k n) K i y( k n + i ) (5)
i =1
= bn 1 ( k 1) + K n 1 ( k )f n 1 ( k )
where Kn(k) is the reflection coefficient for the nth stage of the lattice.
The backward error signals, bn, are then used as inputs to the tap weights,
and the output of the equalizer is given by
N
d k = cn ( k )bn ( k ) (6)
n =1

Figure 2 The structure of a lattice equalizer [from [Pro91] IEEE].

Two main advantages of the lattice equalizer is its numerical stability
and faster convergence. Also, the unique structure of the lattice filter
allows the dynamic assignment of the most effective length of the lattice
equalizer. Hence, if the channel is not very time dispersive, only a
fraction of the stages are used. When the channel becomes more time
dispersive, the length of the equalizer can be increased by the algorithm
without stopping the operation of the equalizer. The structure of a
lattice equalizer, however, is more complicated than a linear transversal
equalizer.
(ii) Nonlinear Equalization

Nonlinear equalizers are used in applications where the channel distortion
is too severe for a linear equalizer to handle, and are commonplace in
practical wireless systems. Linear equalizers do not perform well on
channels which have deep spectral nulls in the pass band. In an attempt
to compensate for the distortion, the linear equalizer places too much
gain in the vicinity of the spectral null, thereby enhancing the noise
present in those frequencies.
Three very effective nonlinear methods have been developed which
offer improvements over linear equalization techniques and are used in
most 2G and 3G systems. These are [Pro91]:
1. Decision Feedback Equalization (DFE)
2. Maximum Likelihood Symbol Detection
3. Maximum Likelihood Sequence Estimation (MLSE)

Figure 1 Decision feedback equalizer (DFE).

Decision Feedback Equalization (DFE)
The basic idea behind decision feedback equalization is that once an
information symbol has been detected and decided upon, the ISI that it
induces on future symbols can be estimated and subtracted out before
detection of subsequent symbols [Pro89]. The DFE can be realized in
either the direct transversal form or as a lattice filter. The direct form
is shown in Figure1. It consists of a feed forward filter (FFF) and a
feedback filter (FBF). The FBF is driven by decisions on the output of
the detector, and its coefficients can be adjusted to cancel the ISI on the
current symbol from past detected symbols. The equalizer has N1 + N2 + 1
taps in the feed forward filter and N3 taps in the feedback filter, and its
output can be expressed as:
N2 N3
dk =
n = N1
cn* yk n + Fi dk i (1)
i =1
*
where c and yn are tap gains and the inputs, respectively, to the forward
n
filter, Fi are tap gains forthe feedback filter, and di(i < k) is the previous
*
decision made on the detected signal. That is, once dk is obtained using
Equation 1, dk is decided from it. Then, dk along with previous decisions
dk-1, dk-2, are fed back into the equalizer, and dk +1 is obtained using
Equation (1).

The minimum mean squared error a DFE can achieve is [Pro89]
T /T N0
E[| e( n) |2 ]min = exp ln j T 2 d (2)
2 / T
| F (e )| + N 0
It can be shown that the minimum MSE for a DFE in Equation (2) is
always smaller than that of an LTE in Equation unless | F (e jw T )|2 is a
constant (i.e., when adaptive equalization is not needed) [Pro89]. If
there are nulls in | F (e jw T )|2, a DFE has significantly smaller minimum
MSE than an LTE. Therefore, an LTE is well behaved when the channel
spectrum is comparatively flat, but if the channel is severely distorted or
exhibits nulls in the spectrum, the performance of an LTE deteriorates
and the mean squared error of a DFE is much better than a LTE. Also, an
LTE has difficulty equalizing a nonminimum phase channel, where the
strongest energy arrives after the first arriving signal component. Thus, a
DFE is more appropriate for severely distorted wireless channels.
The lattice implementation of the DFE is equivalent to a transversal
DFE having a feed forward filter of length N1 and a feedback filter of
length N2, where N1 > N2.
Another form of DFE proposed by Belfiore and Park [Bel79] is called
a predictive DFE, and is shown in Figure 3. It also consists of a feed
forward filter (FFF) as in the conventional DFE. However, the feedback
filter (FBF) is driven by an input sequence formed by the difference of
the output of the detector and the output of the feed forward filter. Hence,
the FBF here is called a noise predictor because it predicts the noise and
Figure 3 Predictive decision feedback equalizer.

the residual ISI contained in the signal at the FFF output and subtracts
from it the detector output after some feedback delay. The predictive DFE
performs as well as the conventional DFE as the limit in the number of
taps in the FFF and the FBF approach infinity. The FBF in the predictive
DFE can also be realized as a lattice structure [Zho90]. The RLS lattice
algorithm (discussed in Section 7.8) can be used in this case to yield fast
convergence.
Maximum Likelihood Sequence Estimation (MLSE) Equalizer
The MSE-based linear equalizers described previously are optimum
with respect to the criterion of minimum probability of symbol error
when the channel does not introduce any amplitude distortion. Yet
this is precisely the condition in which an equalizer is needed for a
mobile communications link. This limitation on MSE-based equalizers
led researchers to investigate optimum or nearly optimum nonlinear
structures. These equalizers use various forms of the classical maximum
likelihood receiver structure. Using a channel impulse response simulator
within the algorithm, the MLSE tests all possible data sequences (rather
than decoding each received symbol by itself), and chooses the data
sequence with the maximum probability as the output. An MLSE
usually has a large computational requirement, especially when the
delay spread of the channel is large. Using the MLSE as an equalizer
was first proposed by Forney [For78] in which he set up a basic MLSE
estimator structure and implemented it with the Viterbi algorithm.
This algorithm, described in Section 7.15, was recognized to be a
maximum likelihood sequence estimator (MLSE) of the state sequences
of a finite state Markov process observed in memoryless noise. It has
recently been implemented successfully for equalizers in mobile radio
channels.
The MLSE can be viewed as a problem in estimating the state of a
discrete-time finite state machine, which in this case happens to be the
radio channel with coefficients fk, and with a channel state which at any
instant of time is estimated by the receiver based on the L most recent
input samples. Thus, the channel has ML states, where M is the size of
the symbol alphabet of the modulation. That is, an ML trellis is used by
the receiver to model the channel overtime. The Viterbi algorithm then
tracks the state of the channel by the paths through the trellis and gives
at stage k a rank ordering of the ML most probable sequences terminating
in the most recent L symbols.
The block diagram of a MLSE receiver based on the DFE is shown
in Figure 4. The MLSE is optimal in the sense that it minimizes the
probability of a sequence error. The MLSE requires knowledge of the

channel characteristics in order to compute the metrics for making

decisions. The MLSE also requires knowledge of the statistical
distribution of the noise corrupting the signal. Thus, the probability
distribution of the noise determines the form of the metric for optimum
demodulation of the received signal. Notice that the matched filter
operates on the continuous time signal, whereas the MLSE and channel
estimator rely on discretized (nonlinear) samples.
Figure 4 The structure of a maximum likelihood sequence estimator (MLSE)

with an adaptive matched filter.
14. (b) (i) RAKE Receiver

In CDMA spread spectrum systems (see Chapter), the chip rate is
typically much greater than the flat-fading bandwidth of the channel.
Whereas conventional modulation techniques require an equalizer to
undo the intersymbol interference between adjacent symbols, CDMA
spreading codes are designed to provide very low correlation between
successive chips. Thus, propagation delay spread in the radio channel
merely provides multiple versions of the transmitted signal at the
receiver. If these multipath components are delayed in time by more than
a chip duration, they appear like uncorrelated noise at a CDMA receiver,
and equalization is not required. The spread spectrum processing gain
makes uncorrelated noise negligible after despreading.
However, since there is useful information in the multipath components,
CDMA receivers may combine the time delayed versions of the original
signal transmission in order to improve the signal-to-noise ratio at the
receiver. A RAKE receiver does just thisit attempts to collect the time-
shifted versions of the original signal by providing a separate correlation
receiver for each of the multipath signals. Each correlation receiver may
be adjusted in time delay, so that a microprocessor controller can cause

different correlation receivers to search in different time windows for

significant multipath. The range of time delays that a particular correlator
can search is called a search window. The RAKE receiver, shown in
Figure 5, is essentially a diversity receiver designed specifically for
CDMA, where the diversity is provided by the fact that the multipath
components are practically uncorrelated from one another when their
relative propagation delays exceed a chip period.
A RAKE receiver utilizes multiple correlators to separately detect the
M strongest multipath components. The outputs of each correlator are
then weighted to provide a better estimate of the transmitted signal than
is provided by a single component. Demodulation and bit decisions are
then based on the weighted outputs of the M correlators.
The basic idea of a RAKE receiver was first proposed by Price and
Green [Pri58]. In outdoor environments, the delay between multipath
components is usually large and, if the chip rate is properly selected,
the low autocorrelation properties of a CDMA spreading sequence can
assure that multipath components will appear nearly uncorrelated with
each other. However, the RAKE receiver in IS-95 CDMA has been
found to perform poorly in indoor environments, which is to be expected
since the multipath delay spreads in indoor channels (100 ns) are much
smaller than an IS-95 chip duration (800 ns). In such cases, a RAKE
will not work since multipath is unresolveable, and Rayleigh flat-fading
typically occurs within a single chip period.
To explore the performance of a RAKE receiver, assume M correlators
are used in a CDMA receiver to capture the M strongest multipath
components. A weighting network is used to provide a linear combination
Figure 5An M-branch (M-finger) RAKE receiver implementation.

Each correlator detects a time shifted version of the original CDMA
transmission, and each finger of the RAKE correlates to a portion of the
signal which is delayed by at least one chip in time from the other fingers.

of the correlator output for bit detection. Correlator 1 is synchronized to

the strongest multipath m2. Multipath component m2 arrives t1 later than
component m1 where t2 t1 is assumed to be greater than a chip duration.
The second correlator is synchronized to m2. It correlates strongly with
m2, but has low correlation with m1. Note that if only a single correlator is
used in the receiver (see Figure), once the output of the single correlator
is corrupted by fading, the receiver cannot correct the value. Bit decisions
based on only a single correlation may produce a large bit error rate. In a
RAKE receiver, if the output from one correlator is corrupted by fading, the
others may not be, and the corrupted signal may be discounted through the
weighting process. Decisions based on the combination of the M separate
decision statistics offered by the RAKE provide a form of diversity which
can overcome fading and thereby improve CDMA reception.
The M decision statistics are weighted to form an overall decision
statistic as shown in Figure 5. The outputs of the M correlators are
denoted as Z1, Z2... and ZM. They are weighted by a1, a2, and aM,
respectively. The weighting coefficients are based on the power or the
SNR from each correlator output. If the power or SNR is small out of a
particular correlator, it will be assigned a small weighting factor. Just as
in the case of a maximal ratio combining diversity scheme, the overall
signal Z is given by
M
Z = a m Z m (1)
m =1
The weighting coefficients, am, are normalized to the output signal

power of the correlator in such a way that the coefficients sum to unity,
as shown in Equation (2)
Z2
a m = M m (2)
Z m2 m =1
As in the case of adaptive equalizers and diversity combining, there

are many ways to generate the weighting coefficients. However, due
to multiple access interference, RAKE fingers with strong multipath
amplitudes will not necessarily provide strong output after correlation.
Choosing weighting coefficients based on the actual outputs of the
correlators yields better RAKE performance.
14. (b) (ii) Frequency Domain Coding of Speech

Frequency domain coders [Tri79] are a class of speech coders which
take advantage of speech perception and generation models without

making the algorithm totally dependent on the models used. In this

class of coders, the speech signal is divided into a set of frequency
components which are quantized and encoded separately. In this way,
different frequency bands can be preferentially encoded according to
some perceptual criteria for each band, and hence the quantization noise
can be contained within bands and prevented from creating harmonic
distortions outside the band. These schemes have the advantage that
the number of bits used to encode each frequency component can be
dynamically varied and shared among the different bands.
Many frequency domain coding algorithms, ranging from simple to
complex are available. The most common types of frequency domain
coding include sub-band coding (SBC) and block transform coding.
While a sub-band coder divides the speech signal into many smaller
sub-bands and encodes each sub-band separately according to some
perceptual criterion, a transform coder codes the short-time transform of
a windowed sequence of samples and encodes them with number of bits
proportional to its perceptual significance.
Sub-band Coding
Sub-band coding can be thought of as a method of controlling and
distributing quantization noise across the signal spectrum. Quantization is
a nonlinear operation which produces distortion products that are typically
broad in spectrum. The human ear does not detect the quantization
distortion at all frequencies equally well. It is therefore possible to
achieve substantial improvement in quality by coding the signal in
narrower bands. In a sub-band coder, speech is typically divided into four
or eight sub-bands by a bank of filters, and each sub-band is sampled
at a bandpass Nyquist rate (which is lower than the original sampling
rate) and encoded with different accuracy in accordance to a perceptual
criteria. Band-splitting can be done in many ways. One approach could
be to divide the entire speech band into unequal sub-bands that contribute
equally to the articulation index. One partitioning of the speech band
according to this method as suggested by Crochiere et al. [Cro76] is given
below.
Sub-band Number Frequency Range

1 200700 Hz
2 7001310 Hz
3 13102020 Hz
4 20203200 Hz

Another way to split the speech band would be to divide it into equal
width sub-bands and assign to each sub-band number of bits proportional
to perceptual significance while encoding them. Instead of partitioning
into equal width bands, octave band splitting is often employed. As the
human ear has an exponentially decreasing sensitivity to frequency, this
kind of splitting is more in tune with the perception process.
There are various methods for processing the sub-band signals. One
obvious way is to make a low pass translation of the sub-band signal to
zero frequency by a modulation process equivalent to single sideband
modulation. This kind of translation facilitates sampling rate reduction
and possesses other benefits that accrue from coding low-pass signals.
Figure 6 shows a simple means of achieving this low pass translation.
The input signal is filtered with a bandpass filter of width wn for the
n th band. w1n is the lower edge of the band and w2n is the upper edge
of the band. The resulting signal sn(t) is modulated by a cosine wave
cos (w1n t), and filtered using a low pass filter hn (t) with bandwidth
(0 - wn). The resulting signal rn (t) corresponds to the low pass translated
version sn(t) of and can be expressed as
rn(t) = [sn(t) cos w1nt)] hn(t)(1)
where denotes a convolution operation. The signal rn(t) is sampled

at a rate 2wn. This signal is then digitally encoded and multiplexed
with encoded signals from other channels as shown in Figure 6. At the
receiver, the data is demultiplexed into separate channels, decoded, and
bandpass translated to give the estimate of rn(t) for the n th channel.
The low pass translation technique is straightforward and takes
advantage of a bank of nonoverlapping bandpass filters. Unfortunately,
unless we use sophisticated bandpass filters, this approach will lead to
perceptible aliasing effects. Estaban and Galand proposed [Est77] a
scheme which avoids this inconvenience even with quasiperfect, sub-
band splitting. Filter banks known as quadrature mirror filters (QMFs)
are used to achieve this. By designing a set of mirror filters which
satisfy certain symmetry conditions, it is possible to obtain perfect alias
cancellation. This facilitates the implementation of sub-band coding
without the use of very high order filters. This is particularly attractive
for real time implementation as a reduced filter order means a reduced
computational load and also a reduced latency.
Sub-band coding can be used for coding speech at bit rates in the range
9.6 kbps to 32 kbps. In this range, speech quality is roughly equivalent to
that of ADPCM at an equivalent bit rate. In addition, its complexity and
relative speech quality at low bit rates make it particularly advantageous

for coding below about 16 kbps. However, the increased complexity of

subband coding when compared to other higher bit rate techniques does
not warrant its use at bit rates greater than about 20 kbps. The CD-900
cellular telephone system uses sub-band coding for speech compression.
Figure 6 Block diagram of a sub-band coder and decoder.
15. (a) (i) Code Division Multiple Access (CDMA)

In code division multiple access (CDMA) systems, the narrowband
message signal is multiplied by a very large bandwidth signal called the
spreading signal. The spreading signal is a pseudonoise code sequence
that has a chip rate which is orders of magnitudes greater than the data rate
of the message. All users in a CDMA system, as seen from figure 7, use
the same carrier frequency and may transmit simultaneously. Each user
has its own pseudorandom codeword which is approximately orthogonal
to all other codewords. The receiver performs a time correlation operation
to detect only the specific desired codeword. All other codewords appear
as noise due to decorrelation. For detection of the message signal, the

receiver needs to know the codeword used by the transmitter. Each user
operates independently with no knowledge of the other users.
In CDMA, the power of multiple users at a receiver determines the
noise floor after decorrelation. If the power of each user within a cell
is not controlled such that they do not appear equal at the base station
receiver, then the near-far problem occurs.
The nearfar problem occurs when many mobile users share the same
channel. In general, the strongest received mobile signal will capture the
demodulator at a base station. In CDMA, stronger received signal levels
raise the noise floor at the base station demodulators for the weaker
signals, thereby decreasing the probability that weaker signals will be
received. To combat the nearfar problem, power control is used in most
CDMA implementations. Power control is provided by each base station
in a cellular system and assures that each mobile within the base station
coverage area provides the same signal level to the base station receiver.
This solves the problem of a nearby subscriber overpowering the base
station receiver and drowning out the signals of far away subscribers.
Power control is implemented at the base station by rapidly sampling
the radio signal strength indicator (RSSI) levels of each mobile and then
sending a power change command over the forward radio link. Despite
the use of power control within each cell, out-of-cell mobiles provide
interference which is not under the control of the receiving base station.
The features of CDMA including the following:
Many users of a CDMA system share the same frequency. Either
TDD or FDD may be used.
Unlike TDMA or FDMA, CDMA has a soft capacity limit. Increasing
the number of users in a CDMA system raises the noise floor in a
linear manner. Thus, there is no absolute limit on the number of users
in CDMA. Rather, the system performance gradually degrades for
all users as the number of users is increased, and improves as the
number of users is decreased.
Multipath fading may be substantially reduced because the signal is
spread over a large spectrum. If the spread spectrum bandwidth is
greater than the coherence bandwidth of the channel, the inherent
frequency diversity will mitigate the effects of small-scale fading.
Channel data rates are very high in CDMA systems. Consequently,
the symbol (chip) duration is very short and usually much less than the
channel delay spread. Since PN sequences have low autocorrelation,
multipath which is delayed by more than a chip will appear as noise.
A RAKE receiver can be used to improve reception by collecting
time delayed versions of the required signal.

Since CDMA uses co-channel cells, it can use macroscopic spatial

diversity to provide soft handoff. Soft handoff is performed by the
MSC, which can simultaneously monitor a particular user from two
or more base stations. The MSC may chose the best version of the
signal at any time without switching frequencies.
Self-jamming is a problem in CDMA system. Self-jamming arises
from the fact that the spreading sequences of different users are not
exactly orthogonal, hence in the despreading of a particular PN code,
non-zero contributions to the receiver decision statistic for a desired
user arise from the transmissions of other users in the system.
The nearfar problem occurs at a CDMA receiver if an undesired
user has a high detected power as compared to the desired user.
15. (a) (ii) Frequency Hopped Multiple Access (FHMA)
Frequency hopped multiple access (FHMA) is a digital multiple access
system in which the carrier frequencies of the individual users are
varied in a pseudorandom fashion within a wideband channel. Figure 7
illustrates how FHMA allows multiple users to simultaneously occupy
the same spectrum at the same time, where each user dwells at a specific
narrowband channel at a particular instance of time, based on the
Figure 7Spread spectrum multiple access in

which each channel is assigned a unique PN code
which is orthogonal or approximately orthogonal
to PN codes used by other users.

particular PN code of the user. The digital data of each user is broken
into uniform sized bursts which are transmitted on different channels
within the allocated spectrum band. The instantaneous bandwidth
of any one transmission burst is much smaller than the total spread
bandwidth. The pseudorandom change of the channel frequencies of the
user randomizes the occupancy of a specific channel at any given time,
thereby allowing for multiple access over a wide range of frequencies.
In the FH receiver, a locally generated PN code is used to synchronize
the receivers instantaneous frequency with that of the transmitter. At
any given point in time, a frequency hopped signal only occupies a
single, relatively narrow channel since narrowband FM or FSK is used.
The difference between FHMA and a traditional FDMA system is that
the frequency hopped signal changes channels at rapid intervals. If the
rate of change of the carrier frequency is greater than the symbol rate,
then the system is referred to as a fast frequency hopping system. If the
channel changes at a rate less than or equal to the symbol rate, it is
called slow frequency hopping. A fast frequency hopper may thus be
thought of as an FDMA system which employs frequency diversity.
FHMA systems often employ energy efficient constant envelope
modulation. Inexpensive receivers may be built to provide noncoherent
detection of FHMA. This implies that linearity is not an issue, and
the power of multiple users at the receiver does not degrade FHMA
performance.
A frequency hopped system provides a level of security, especially
when a large number of channels are used, since an unintended (or an
intercepting) receiver that does not know the pseudorandom sequence of
frequency slots must retune rapidly to search for the signal it wishes to
intercept. In addition, the FH signal is somewhat immune to fading, since
error control coding and interleaving can be used to protect the frequency
hopped signal against deep fades which may occasionally occur during
the hopping sequence. Error control coding and interleaving can also
be combined to guard against erasures which can occur when two or
more users transmit on the same channel at the same time. Bluetooth
and HomeRF wireless technologies have adopted FHMA for power
efficiency and low cost implementation.
15. (b) 2G network:

Second Generation (2G) Cellular Networks
Most of todays ubiquitous cellular networks use what is commonly
called second generation or 2G technologies which conform to the

second generation cellular standards. Unlike first generation cellular

systems that relied exclusively on FDMA/FDD and analog FM, second
generation standards use digital modulation formats and TDMA/FDD
and CDMA/FDD multiple access techniques. The most popular second
generation standards include three TDMA standards and one CDMA
standard: (a) Global System Mobile (GSM), which supports eight time
slotted users for each 200 kHz radio channel and has been deployed
widely by service providers in Europe, Asia, Australia, South America,
and some parts of the US (in the PCS spectrum band only) [Gar99];
(b) Interim Standard 136 (IS-136), also known as North American
Digital Cellular (NADC), which supports three time slotted users
for each 30 kHz radio channel and is a popular choice for carriers in
North America, South America, and Australia (in both the cellular and
PCS bands); (c) Pacific Digital Cellular (PDC), a Japanese TDMA
standard that is similar to IS-136 with more than 50million users; and
(d) the popular 2G CDMA standard Interim Standard 95 Code Division
Multiple Access (IS-95), also known as cdmaOne, which supports up
to 64 users that are orthogonally coded and simultaneously transmitted
on each 1.25 MHz channel. CDMA is widely deployed by carriers in
North America (in both cellular and PCS bands), as well as in Korea,
Japan, China, South America, and Australia [Lib99, ch. 1], [Kim00],
[Gar00].
The 2G standards mentioned above represent the first set of wireless
air interface standards to rely on digital modulation and sophisticated
digital signal processing in the handset and the base station. As discussed
in Chapter 11, second generation systems were first introduced in the
early1990s, and evolved from the first generation of analog mobile phone
systems (e.g., AMPS, ETACS, and JTACS). Today, many wireless service
providers use both first generation and second generation equipment in
major markets and often provide customers with subscriber units that can
support multiple frequency bands and multiple air interface standards.
For example, in many countries it is possible to purchase a single tri-
mode cellular handset phone that supports CDMA in the cellular and
PCS bands in addition to analog first generation technology in the
cellular band. Such tri-mode phones are able to automatically sense and
adapt to whichever standard is being used in a particular market. Figure 1
illustrates how the world subscriber base was divided between the major
1G and2G technologies as of late 2001. Table 1 highlights the key
technical specifications of the dominant GSM, CDMA, and IS-136/PDC
second generation standards.

Figure 1 Worldwide subscriber base as a function of cellular technology in

late 2001.
In many countries, 2G wireless networks are designed and deployed for

conventional mobile telephone service, as a high capacity replacement
for, or in competition with, existing older first generation cellular
telephone systems. Modern cellular systems are also being installed
to provide fixed (non-mobile) telephone service to residences and
businesses in developing nationsthis is particularly cost effective for
providing plain old telephone service (POTS) in countries that have
poor telecommunications infrastructure and are unable to afford the
installation of copper wire to all homes. Since all 2G technologies offer
at least a three-times increase in spectrum efficiency(and thus at least a
3X increase in overall system capacity) as compared to first generation
analog technologies, the need to meet a rapidly growing customer
base justifies the gradual, ongoing change out of analog to digital 2G
technologies in any growing wireless network.
In mid-2001, several major carriers such as AT&T Wireless and
Cingular in the US and NTT in Japan announced their decisions
to eventually abandon the IS-136 and PDC standards as long term
technology options in favor of emerging third generation standards
based on the GSM
TDMA platform. Simultaneously, international wireless carrier
Nextel announced its decision to upgrade its iDen air interface standard
to support up to five times the number of current users based on a data
compression methodology using Internet protocol (IP) packet data. Most
other carriers throughout the world had already committed to adopting

Table 1 Key Specifications of Leading 2G Technologies (adapted from [Lib99])
cdmaOne, IS-95, ANSI GSM, DCS-1900, ANSI NADC, IS-54/IS-136,
J-STD-008 J-STD-007 ANSI J-STD-011, PDC
Uplink Frequencies 824849 MHz (US 890915 MHz 800 MHz, 1500 MHz
Cellular)18501910 MHz (Europe)18501910 MHz (Japan)18501910 MHz
(US PCS) (US PCS) (US PCS)
Downlink Frequencies 869894 MHz (US 935960 MHz 869894 MHz (US
Cellular)19301990 MHz (Europe)19301990 MHz Cellular)19301990 MHz
(US PCS) (US PCS) (US PCS) 800 MHz, 1500
M02_ECE_7th-SEMESTER_CH01(Nov-Dec-2013).indd 43
MHz (Japan)
Duplexing FDD FDD FDD
Multiple Access Technology CDMA TDMA TDMA
Modulation BPSK with Quadrature GMSK with BT = 0.3 p/4 DQPSK
Spreading
Carrier Separation 1.25 MHz 200 kHz 30 kHz (IS-136) (25 kHz
for PDC)
Wireless Communication (Nov/Dec 2013)
Channel Data Rate 1.2288 Mchips/sec 270.833 kbps 48.6 kbps (IS-136) (42 kbps
for PDC)
Voice channels per carrier 64 8 3
Speech Coding Code Excited Linear Residual Pulse Excited Vector Sum Excited Linear
Prediction (CELP)@ 13kbps, Long Term Prediction Predictive Coder (VSELP)
Enhanced Variable Rate (RPE-LTP) @ 13 kbps @ 7.95 kbps
2.43
Codec (EVRC) @ 8 kbps
4/25/2014 8:17:30 PM
a 3G standard based on either GSM or CDMA prior to 2001. Decisions

like these have set the stage for the inevitability of two universal and
competing third generation (3G) cellular mobile radio technologies, one
based on the philosophy and backward compatibility of GSM, and the
other based on the philosophy and backward compatibility of CDMA.
3G network
Third Generation (3G) Wireless Networks
3G systems promise unparalleled wireless access in ways that have never
been possible before. Multi-megabit Internet access, communications
using Voice over Internet Protocol (VoIP), voice-activated calls,
unparalleled network capacity, and ubiquitous always-on access are
just some of the advantages being touted by 3G developers. Companies
developing 3G equipment envision users having the ability to receive
live music, conduct interactive web sessions, and have simultaneous
voice and data access with multiple parties at the same time using a
single mobile handset, whether driving, walking, or standing still in an
office setting.
As mentioned in Chapter 1, and described in detail in [Lib99], the
International Telecommunications Union (ITU) formulated a plan to
implement a global frequency band in the 2000MHz range that would
support a single, ubiquitous wireless communication standard for all
countries throughout the world. This plan, called International Mobile
Telephone 2000 (IMT-2000), has been successful in helping to cultivate
active debate and technical analysis for new high speed mobile telephone
solutions when compared to 2G. However, as can be seen in Figures, the
hope for a single worldwide standard has not materialized, as the world-
wide user community remains split between two camps: GSM/IS-136/
PDC and CDMA.
The eventual 3G evolution for CDMA systems leads to cdma2000.
Several variants of CDMA 2000 are currently being developed, but they
all are based on the fundamentals of IS-95and IS-95B technologies.
The eventual 3G evolution for GSM, IS-136, and PDC systems leads
to Wideband CDMA (W-CDMA), also called Universal Mobile
Telecommunications Service(UMTS). W-CDMA is based on the network
fundamentals of GSM, as well as the merged versions of GSM and IS-
136 through EDGE. It is fair to say that these two major 3G technology
camps, cdma2000 and W-CDMA, will remain popular throughout the
early part of the 21st century.
Table 2 illustrates the primary worldwide proposals that were sub
mitted for IMT-2000 in 1998. Since 1998, many standards proposals

Table 2 Leading IMT-2000 Candidate Standards as of 1998 (adapted from [Lib99])
Air Interface Mode of Operation Duplexing Key Features
Method
cdma2000 US Multi-Carrier and Direct FDD and TDD Backward compatibility with IS-95A and IS-
TIATR45.5 Spreading DS-CDMA a N = Modes 95B. Downlink can be implemented using either
1.2288 Mcps with N = 1, 3, Multi-Carrier or Direct Spreading. Uplink can
6, 9, 12 support a simultaneous combination of Multi-
Carrier or Direct Spreading.
Auxiliary carriers to help with downlink channel
estimation in forward link beam-forming.
UTRA (UMTS DS_CDMA at Rates of N FDD and TDD Wideband DS_CDMA System.
Terrestrial Radio Ac- 0.960 Mcps with N = 4, Modes Backward compatibility with GSM/DCS-1900.
cess)ETSI SMG2 8, 16 Up to 2.048 Mbps on Downlink in FDD Mode.
Minimum forward channel bandwidth of 5GHz.
The collection of proposed standards represented
here each exhibit unique features, but support a
common set of chip rates, 10 ms frame structure,
with 16 slots per frame.
Wireless Communication (Nov/Dec 2013)
Connection-dedicated pilot bits assist in down-

link beamforming.
W-CDMA/NA
(Wideband CDMA)
North America)USA
T1P1-ATIS
2.45
(Continued)
4/25/2014 8:17:30 PM
Table 2 (Continued)
2.46
W-CDMA/Japan
(Wideband CDMA)
Japan ARIB
CDMA II South
Korea TTA
WIMS/W-CDMA
USA
TIA TR46.1
CDMA I DS-CDMA at N 0.9216 FDD and TDD Up to 512 kbps per spreading code, code
South Korea TTA Mcps with N = 1, 4, 16 Modes aggregation up to 2.048 Mbps.
UWC-136 (Univer- TDMA - Up to722.2 kbps FDD (Out- Backward compatibility and upgrade path for
sal Wireless Com- (Out-door/Vehicular), Up to door/Ve- both IS-136 and GSM.
munications Consor- 5.2 Mbps(Indoor Office) hicular), TDD Fits into existing IS-136 and GSM.
tium) USA TIA TR (Indoor Office) Explicit plans to support adaptive antenna
B.E./B.Tech. Question Papers
45.3 technology.
TD-SCDMA DS-CDMA 1.1136Mcps TDD RF channel bit rate up to 2.227 Mbps.
China Academy of Use of smart antenna technology is fundamental
Telecommunication (but not strictly required) in TD-SCMA.
Technology (CATT)
DECT 1150-3456 kbps TDMA TDD Enhanced version of 2G DECT technology.
ETSI Project (EP)
DECT
4/25/2014 8:17:30 PM
have capitulated and joined with either thecdma2000 or UMTS

(W-CDMA) camps. Commercial grade 3G equipment is expected to
be available in the 20022003 time frame. Two good Internet sources
for up-to-the-minute 3G developments can be found at GSM World
(www.gsmworld.com) and the CDMA Developers Group (www.cdg.
org). The ITU IMT-2000 standards organizations are currently separated
into two major organizations reflecting the two 3G camps: 3GPP (3G
Partnership Project for Wideband CDMA standards based on backward
compatibility with GSM and IS-136/PDC) and 3GPP2 (3GPartnership
Project for cdma2000 standards based on backward compatibility with
IS-95).
Countries throughout the world are currently determining new radio
spectrum bands to accommodate the 3G networks that will likely
be deployed in the 20042005 time frame. ITUs2000 World Radio
Conference established the 25002690 MHz, 17101885 MHz, and 806
960MHZ bands as candidates for 3G. In the US, additional spectrum in
the upper UHF television bands near 700 MHz is also being considered
for 3G. Given the economic downturn of the telecommunications industry
during 2001, many governments throughout the world, including the US,
had postponed their 3G auctions and spectrum decisions as of late 2001.
Some European governments, however, auctioned off radio spectrum
for 3G well before the telecommunications industry depression of
2001. The sale price of the spectrum was astounding! Englands first
ever spectrum auction netted $35.5 Billion USD in April 2000 for five
nationwide 3G licenses. Germanys 3G auction generated $46 Billion
USD later the same year, for four competing nationwide licenses [Buc00,
pp. 32-36.].

MAY/JUNE 2013
Seventh Semester
WIRELESS COMMUNICATION
1. Define: Frequency reuse.
2. State the operating principle of adhoc networks.
3. State the differences between small-scale and large-scale fading.
4. Define: Snells law.
5. Mention any two criteria for choosing a modulation technique for a specific
wireless application.
6. Draw the Structure of generic optimum receiver.
7. Define: Hamming distance.
8. State the principle of diversity.
9. Define: Direct Sequence-Speed Spectrum.
10. State the goals of a standard IMT-2000.
11. (a) (i) Explain the methods for increasing the capacity of wireless
cellular networks.
(ii) Brief about the principle of Time Division Multiple Access
(TDMA).
Or
M02_ECE_7th-SEMESTER_CH02(May-Jun-2013).indd 1 4/25/2014 11:35:47 AM

(b) (i) Describe in detail about the effects of multipath propagation in

wireless environment.
(ii) A Communication system has the following parameters:
P1 = 5 W, Gt (dB) = 13 dB, Gr (dB) = 17 dB, d = 80 km,
f = 3 GHz,
Determine the value of the received power.
12. (a) (i) Explain the time-variant two-path model of a wireless prop
agation channel.
(ii) Brief about the properties of Rayleigh distribution.
Or
(b) (i) Explain the narrow band modeling methods for Short scale
fading and Long scale fading.
(ii) Brief about the properties of Nakagami distribution.
13. (a) (i) Explain the principle of p/4 Differential Quadrature-Phase

Shift Keying from a signal space diagram.
(ii) Derive the expression for probability of error in Flat-Fading
channels.
Or
(b) (i) Explain the principle of Minimum Shift Keying (MSK)
modulation and derive the expression for power spectral density.
(ii) Derive the expression for probability of error in Frequency-
Dispersive Fading Channels.
14. (a) (i) Explain any two diversity techniques to combat small-scale
fading.
(ii) Describe any two adaptation algorithms for Mean Square Error
Equalizers.
Or
(b) (i) Write short notes on Linear Predictive voCoder.
(ii) The generator matrix for a linear binary code is
0 0 1 1 1 0 1
G = 0 1 0 0 1 1 1
1 0 0 1 1 1 0

Wireless Communication (May/June 2013) 2.3
(1) Express G in systematic [I/P] form.

(2) Determine the Parity check matrix H for the code.
(3) Construct the table of syndromes for the code.
(4) Determine the minimum distance of the code.
15. (a) (i) Explain the principle of cellular code division multiple access
systems.
(ii) Brief about the properties of spreading codes used in CDMA
systems.
Or
(b) (i) Describe in detail about the operation of OFDM transceiver
structures.
(ii) Explain the physical layer features of WCDMA systems.

Solutions
PART A
1. Physical separation of two cells is sufficiently wide; the same subset of
frequencies can be used in both cells. This is the concept of frequency
reuse. The ability of frequency is to expand the total system capacity
without the need of high power transmitters. This is called frequency
reuse or frequency planning.
2. MANET - Mobile Ad-hoc Network
It is a self configuring network of mobile stations connected by wireless
links. It is defined as Independent Basic Service Set in IEEE 802.11 standard.
3. In large scale fading, it characterize the signal strength over large
transmitter and receiver separation distances (several hundreds or
thousands of meters).
In small scale fading, it characterize the rapid fluctuations of the
received signal strength over very short distances between transmitter
and receiver (a few meters) or short time duration (seconds).
4. Snells law states how light ray reacts when it meets the interface of two
media having difference indexes of refraction.
Refractive model for Snells law

Let the two media have refractive indexes n1 and n2, where n1 > n2 and
f1 and f2 be the angles of incidence and angle of refraction respectively.
Snells law stated mathematically is
n1 sin f1 = n2 sin f2

5. The criteria for choosing a modulation technique for a wireless appli

cations are
(i) Fading and multi-path conditions in the mobile radio channel
(ii) Data rate and
(iii) Low bit rate.
6.
An M-branch (M-finger) RAKE receiver implementation. Each

correlator detects a time shifted version of the original CDMA
transmission, and each finger of the RAKE correlates to a portion
of the signal which is delayed by at least one chip in time from the
otherfingers.
7. Hamming distance are nothing but the difference between two code
words.
Ex. 1100
1000.
Here hamming distance are 1.
8. Diversity is technique used to compensate for fading channel impairments

and is usually implemented by using two or more receiving antennas.
9. In direct sequence spread spectrum (DS-SS), the narrow band message

signal is multiplied by pseudo-random noise code sequence called the
spreading signal. Each user has its own pseudorandom code word which
is approximately orthogonal to all other code words.
10. International Mobile Telecommunication (IMT-2000) formerly called
future public land mobile telecommunication system (FPLMTS) tried
to establish a common world wide communication system that allowed
for terminal and user mobility, supporting the data of Universal Personal
Telecommunication (UPT).

PART B
11. (a) (i)As the demand for wireless service increases, the number of
channels assigned to a cell eventually becomes insufficient to support
the required number of users. At this point, cellular design techniques
are needed to provide more channels per unit coverage area. Techniques
such as cell splitting, sectoring, and coverage zone approaches are
used in practice to expand the capacity of cellular systems. Cell
splitting allows an orderly growth of the cellular system. Sectoring uses
directional antennas to further control the interference and frequency
reuse of channels. The zone microcell concept distributes the coverage
of a cell and extends the cell boundary to hard-to-reach places. While
cell splitting increases the number of base stations in order to increase
capacity, sectoring and zone microcells rely on base station antenna
placements to improve capacity by reducing co-channel interference.
Cell splitting and zone microcell techniques do not suffer the trunking
inefficiencies experienced by sectored cells, and enable the base station
to oversee all handoff chores related to the microcells, thus reducing the
computational load at the MSC.
11. (a) (ii)Time Division Multiple Access (TDMA) systems divide the
radio spectrum into time slots, and in each slot only one user is allowed
to either transmit or receive. It can be seen from Figure 1 that each user
occupies a cyclically repeating time slot, so a channel may be thought
of as a particular time slot that reoccurs every frame, where N time slots
comprise a frame. TDMA systems transmit data in a buffer-and-burst
Code
Channel N
ts
Channel 3
lo
eS
Channel 2
m
Ti
Channel 1
Frequency
Time
Figure 1 TDMA scheme where each channel occupies a cyclically

repeating time slot

One TDMA Frame
Preamble Information Message Trail Bits
Slot 1 Slot 2 Slot 3 Slot
Trail Bits Sync. bits Information Data Guard Bits
Figure 2 TDMA frame structure. The frame is cyclically repeated

over time
method, thus the transmission for any user is noncontinuous. This implies
that, unlike in FDMA systems which accommodate analog FM, digital
data and digital modulation must be used with TDMA. The transmission
from various users is interlaced into a repeating frame structure as shown
in Figure 2. It can be seen that a frame consists of a number of slots.
Each frame is made up of a preamble, an information message, and tail
bits. In TDMA/TDD, half of the time slots in the frame information
message would be used for the forward link channels and half would be
used for reverse link channels. In TDMA/FDD systems, an identical or
similar frame structure would be used solely for either forward or reverse
transmission, but the carrier frequencies would be different for the forward
and reverse links. In general, TDMA/FDD systems intentionally induce
several time slots of delay between the forward and reverse time slots for
a particular user, so that duplexers are not required in the subscriber unit.
In a TDMA frame, the preamble contains the address and synchro
nization information that both the base station and the subscribers use
to identify each other. Guard times are utilized to allow synchronization
of the receivers between different slots and frames. Different TDMA
wireless standards have different TDMA frame structures, and some are
described in Chapter 11. The features of TDMA include the following:
TDMA shares a single carrier frequency with several users, where
each user makes use of nonoverlapping time slots. The number of
time slots per frame depends on several factors, such as modulation
technique, available bandwidth, etc.
Data transmission for users of a TDMA system 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 (which is
most of the time).

Because of discontinuous transmissions in TDMA, the handoff

process is much simpler for a subscriber unit, since it is able to
listen for other base stations during idle time slots. An enhanced link
control, such as that provided by mobile assisted handoff (MAHO)
can be carried out by a subscriber by listening on an idle slot in the
TDMA frame.
TDMA uses different time slots for transmission and reception, thus
duplexers are not required. Even if FDD is used, a switch rather than
a duplexer inside the subscriber unit is all that is required to switch
between transmitter and receiver using TDMA.
Adaptive equalization is usually necessary in TDMA systems, since
the transmission rates are generally very high as compared to FDMA
channels.
In TDMA, the guard time should be minimized. If the transmitted
signal at the edges of a time slot are suppressed sharply in order to
shorten the guard time, the transmitted spectrum will expand and
cause interference to adjacent channels.
High synchronization overhead is required in TDMA systems
because of burst transmissions. TDMA transmissions are slotted, and
this requires the receivers to be synchronized for each data burst. In
addition, guard slots are necessary to separate users, and this results in
the TDMA systems having larger overheads as compared to FDMA.
TDMA has an advantage in that it 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 slots based on priority.
Efficiency of TDMA The efficiency of a TDMA system is a measure of
the percentage of transmitted data that contains information as opposed
to providing overhead for the access scheme. The frame efficiency h f ,
is the percentage of bits per frame which contain transmitted data. Note
that the transmitted data may include source and channel coding bits,
so the raw end user efficiency of a system is generally less than h f . The
frame efficiency can be found as follows.
The number of overhead bits per frame is [Zie92],
bOH = N r br + N t bp + N t bg + N r bg (1)
where Nr is the number of reference bursts per frame, Nt is the number of

traffic bursts per frame, br is the number of overhead bits per reference

burst, bp is the number of overhead bits per preamble in each slot, and
bg is the number of equivalent bits in each guard time interval. The total
number of bits per frame, bT, is
bT = Tf R(2)
where Tf is the frame duration, and R f is the channel bit rate. The frame
efficiency hf is thus given as
b
h f = 1 OH 100%(3)
bT
Number of channels in TDMA system The number of TDMA

channel slots that can be provided in a TDMA system is found by
multiplying the number of TDMA slots per channel by the number of
channels available and is given by
m( Btot 2 Bguard )
N= (4)
Bc
Where m is the maximum number of TDMA users supported on each

radio channel. Note that two guard bands, one at the low end of the
allocated frequency band and one at the high end, are required to ensure
that users at the edge of the band do not bleed over into an adjacent
radio service.
11. (b) (i) Refer Q. no. 12. (a) (i) from Nov/Dec 2013
11. (b) (ii) Given Data:
Transmitter power, Pt = 5 W
Frequency, f = 3 GHz
Gt = 13 dB
Gr = 17 dB
d = 80 km
usually, L = 1 km
To find:
Received power, Pr
Pr (10 km)
Solution:
c
l=
f

3 108
=
3 10 9
1
l= or 0.1
10
Received Power (Pr)
t t Gr l
2
PG
=
( 4p ) d L
2 2

5 13 17 (0.1) 2
( 4 3.14) 2 (80) 2 1
11.05
=
157.75 6400
11.05
= 10.94 W
1009664
1 2. (a) (i) Refer Q. no. 12. (a) (ii) from April/May 2012
12. (a) (ii) Rayleigh Fading Distribution
In mobile radio channels, the Rayleigh distribution is commonly used to
describe the statistical time varying nature of the received envelope of a
flat fading signal, or the envelope of an individual multi path component.
It is well known that the envelope of the sum of two quadrature
Gaussian noise signals obeys a Rayleigh distribution. Figure 1 shows a
Rayleigh distributed signal envelope as a function of time. The Rayleigh
distribution has a probability density function (pdf) given by
r r2
2 exp 2 (0 r )
p( r ) = s 2s (1)
( r < 0)
0
where s is the rms value of the received voltage signal before envelope
detection, and s2 is the time-average power of the received signal before
envelope detection. The probability that the envelope of the received
signal does not exceed a specified value R is given by the corresponding
cumulative distribution function (CDF)
R R2
P ( R) = Pr ( r R) = p( r ) dr = 1 exp 2 (2)
0 2s

Figure 1 A typical Rayleigh fading envelope at 900 MHz [from [Fun93]

IEEE]
The mean value rmean of the Rayleigh distribution is given by

p
rmean = E[r ] = rp( r )dr = s = 1.2533s (3)
0
2
and the variance of the Rayleigh distribution is given by s r2 which
represents the ac power in the signal envelope

s 2p
s 2r = E[r 2 ] E 2 [r ] = r 2 p ( r ) dr
0
2

p
= s 2 2 = 0.4292s 2 (4)
2
The rms value of the envlope is the square root of the mean square, or
2s , where s is the standard deviation of the original complex Gaussian
signal prior to envelope detection.
The median value of r is found by solving
r
11 rmedian
= 0 PP((rr )) dr
median
and is = dr (5)
22
0
rrmedian = 1.177s
median = 1.177s
(6)
Thus the mean and the median differ by only 0.55 dB in a Rayleigh
fading signal. Note that the median is often used in practice, since fading

data are usually measured in the field and a particular distribution cannot
be assumed. By using median values instead of mean values, it is easy to
compare different fading distributions which may have widely varying
means.
12. (b)(i) Propagation models have traditionally focused on predicting
the average received signal strength at a given distance from the
transmitter, as well as the variability of the signal strength in close
spatial proximity to a particular location. Propagation models that
predict the mean signal strength for an arbitrary transmitterreceiver
(TR) separation distance are useful in estimating the radio coverage
area of a transmitter and are called large-scale propagation models,
since they characterize signal strength over large TR separation
distances (several hundreds or thousands of meters). On the other
hand, propagation models that characterize the rapid fluctuations of
the received signal strength over very short travel distances (a few
wavelengths) or short time durations (on the order of seconds) are
called small-scale or fading models.
As a mobile moves over very small distances, the instantaneous
received signal strength may fluctuate rapidly giving rise to small-scale
fading. The reason for this is that the received signal is a sum of many
contributions coming from different directions, as described in Chapter 5.
Since the phases are random, the sum of the contributions varies widely;
for example, obeys a Rayleigh fading distribution. In small-scale fading,
the received signal power may vary by as much as three or four orders of
magnitude (30 or 40 dB) when the receiver is moved by only a fraction
of a wavelength. As the mobile moves away from the transmitter over
much larger distances, the local average received signal will gradually
decrease, and it is this local average signal level that is predicted by large-
scale propagation models. Typically, the local average received power is
computed by averaging signal measurements over a measurement track
of 5 l to 40 l. For cellular and PCS frequencies in the 1 GHz to 2 GHz
band, this corresponds to measuring the local average received power
over movements of 1 m to 10 m.
Figure 1 illustrates small-scale fading and the more gradual large-
scale variations for an indoor radio communication system. Notice in the
figure that the signal fades rapidly (small-scale fading) as the receiver
moves, but the local average signal changes much more gradually with
distance. This chapter covers large-scale propagation and presents a
number of common methods used to predict received power in mobile
communication systems.

Figure 1 Small-scale and large-scale fading
1 3. (a) (i) o/4 QPSK

p
The shifted QPSK modulation is a quadrature phase shift keying
4
technique which offers a compromise between OQPSK and QPSK in
terms of the allowed maximum phase transitions. It may be demodulated
p
in a coherent or noncoherent fashion. In QPSK, the maximum
4
phase change is limited to 135, as compared to 180 for QPSK and
p
90 for OQPSK. Hence, the bandlimited QPSK signal preserves
4
the constant envelope property better than bandlimited QPSK, but is
more susceptible to envelope variations than OQPSK. An extremely
p
attractive feature of QPSK is that it can be noncoherently detected,
4
which greatly simplifies receiver design. Further, it has been found that
p
in the presence of multipath spread and fading, QPSK performs better
p 4
than OQPSK [Liu89]. Very often, QPSK signals are differentially
4
encoded to facilitate easier implementation of differential detection or
coherent demodulation with phase ambiguity in the recovered carrier.
p p
When differentially encoded, QPSK is called DQPSK.
p 4 4
In a QPSK modulator, signaling points of the modulated signal
4 p
are selected from two QPSK constellations which are shifted by
4
with respect to each other. Figure 1 shows the two constellations along
with the combined constellation where the links between two signal
points indicate the possible phase transitions. Switching between two

constellations, every successive bit ensures that there is at least a phase

p
shift which is an integer multiple of radians between successive
4
symbols. This ensures that there is a phase transition for every symbol,
which enables a receiver to perform timing recovery and synchronization.
13. (a) (ii) Refer Q. no. 13. (b) (ii) from Nov/Dec 2013
13. (b) (i)Minimum Shift Keying (MSK)
Minimum shift keying (MSK) is a special type of continuous phase-
frequency shift keying (CPFSK) wherein the peak frequency deviation is
equal to 1/4 the bit rate. In other words, MSK is continuous phase FSK
with a modulation index of 0.5. The modulation index of an FSK signal
is similar to the FM modulation index, and is defined as KFSK = (2DF)/
Rb, where DF is the peak RF frequency deviation and Rb is the bit rate. A
modulation index of 0.5 corresponds to the minimum frequency spacing
that allows two FSK signals to be coherently orthogonal, and the name
minimum shift keying implies the minimum frequency separation (i.e.,
bandwidth) that allows orthogonal detection. Two FSK signals vH (t) and
vL (t) are said to be orthogonal if
T
v
0
H (t )vL (t ) dt = 0 (1)
MSK is sometimes referred to as fast FSK, as the frequency spacing

used is only half as much as that used in conventional noncoherent FSK
[Xio94].
MSK is a spectrally efficient modulation scheme and is particularly
attractive for use in mobile radio communication systems. It possesses
properties such as constant envelope, spectral efficiency, good BER
performance, and self-synchronizing capability.
Block diagram of noncoherent FSK receiver

An MSK signal can be thought of as a special form of OQPSK where

the baseband rectangular pulses are replaced with half-sinusoidal pulses
[Pas79]. These pulses have shapes like the St. Louis arch during a period
of 2Tb. Consider the OQPSK signal with the bit streams offset as shown
in Figure. If half-sinusoidal pulses are used instead of rectangular pulses,
the modified signal can be defined as MSK and for an Nbit stream is
given by
N 1
sMSK ( t ) = m1 (t ) p(t 2iTb ) cos 2p f c t + (2)
i=0
N 1
m
i=0
Q (t ) p(t 2iTb Tb ) sin 2p f c t
where
pt
sin 0 t 2Tb (3)
p(t ) = 2Tb

0 elsewhere
and where mI(t) and mQ(t) are the odd and even bits of the bipolar
data stream which have values of 1 and which feed the in-phase and
quadrature arms of the modulator at a rate of Rb/2. It should be noted
that there are a number of variations of MSK that exist in the literature
[Sun86]. For example, while one version of MSK uses only positive
half-sinusoids as the basic pulse shape, another version uses alternating
positive and negative half-sinusoids as the basic pulse shape. However,
all variations of MSK are continuous phase FSK employing different
techniques to achieve spectral efficiency [Sun86].
The MSK waveform can be seen as a special type of a continuous phase
FSK if Equation (2) is rewritten using trigonometric identities as
2 Eb pt
sMSK (t ) = cos 2p f c t mI (t )mQ (t ) + fk (4)
Tb 2Tb
where fk is 0 or p depending on whether mI(t) is 1 or -1. From

Equation (4), it can be deduced that MSK has a constant amplitude.
Phase continuity at the bit transition periods is ensured by choosing the
carrier frequency to be an integral multiple of one fourth the bit rate,
1/4T. Comparing Equation (4) with Equation (6.97), it can be concluded
that the MSK signal is an FSK signal with binary signaling frequencies
of fc +1/4T and fc -1/4T.

13. (b) (ii)Frequency Selective Fading

If the channel possesses a constant-gain and linear phase response over a
bandwidth that is smaller than the bandwidth of transmitted signal, then
the channel creates frequency selective on the received signal. Under such
conditions, the channel impulse response has a multipath delay spread
which is greater than the reciprocal bandwidth of the transmitted message
waveform. When this occurs, the received signal includes multiple
versions of the transmitted waveform which are attenuated (faded) and
delayed in time, and hence the received signal is distorted. Frequency
selective fading is due to time dispersion of the transmitted symbols
within the channel. Thus the channel induces intersymbol interface (ISI).
Viewed in the frequency domain, certain frequency components in the
received signal spectrum have greater gains than others.
Frequency selective fading channels are much more difficult to model
than flat fading channels since each multipath signal must be modeled
and the channel must be considered to be a linear filter. It is for this
reason that wideband multipath measurements are made, and models are
developed from these measurements. When analyzing mobile commu
nication systems, statistical impulse response models such as the two-ray
Rayleigh fading model (which considers the impulse response to be made
up of two delta functions which independently fade and have sufficient
time delay between them to induce frequency selective fading upon the
applied signal), or computer generated or measured impulse responses,
are generally used for analyzing frequency selective small-scale fading.
Figure 2 illustrates the characteristics of a frequency selective fading
channel.
Figure 1 Flat fading channel characteristics

Figure 2 Frequency selective fading channel characteristics.
For frequency selective fading, the spectrum S(f) of the transmitted signal
has a bandwidth which is greater than the coherence bandwidth BC of the
channel. Viewed in the frequency domain, the channel becomes frequency
selective, where the gain is different for different frequency components.
Frequency selective fading is caused by multipath delays which approach
or exceed the symbol period of the transmitted symbol. Frequency
selective fading channels are also known as wide band channels since the
bandwidth of the signal s(t) is wider than the bandwidth of the channel
impulse response. As time varies, the channel varies in gain and phase
across the spectrum of s(t), resulting in time varying distortion in the
received signal r(t). To summarize, a signal undergoes frequency selective
fading if
BS > BC (1)
and
TS < st(2)
A common rule of thumb is that a channel is flat fading if TS 10 st
and a channel is frequency TS < 10 st, although this is dependent on the
specific type of modulation used.
14. (a) (i) Refer Q. no. 14. (a) (i) from April/May 2012
14. (b) (i)Linear Predictive Coders
LPC Vocoders
Linear predictive coders (LPCs) [Sch85a] belong to the time domain class
of vocoders. This class of vocoders attempts to extract the significant
features of speech from the time waveform. Though LPC coders are

computationally intensive, they are by far the most popular among the
class of low bit rate vocoders. With LPC, it is possible to transmit good
quality voice at 4.8 kbps and poorer quality voice at even lower rates.
The linear predictive coding system models the vocal tract as an all
pole linear filter with a transfer function described by
G
H ( z) = M
(1)
1 + bk Z k
k =1
where G is a gain of the filter and z-1 represents a unit delay operation.
The excitation to this filter is either a pulse at the pitch frequency or
random white noise depending on whether the speech segment is voiced
or unvoiced. The coefficients of the all pole filter are obtained in the
time domain using linear prediction techniques [Mak75]. The prediction
principles used are similar to those in ADPCM coders. However, instead
of transmitting quantized values of the error signal representing the
difference between the predicted and actual waveform, the LPC system
transmits only selected characteristics of the error signal. The parameters
include the gain factor, pitch information, and the voiced/unvoiced
decision information, which allow approximation of the correct error
signal. At the receiver, the received information about the error signal is
used to determine the appropriate excitation for the synthesis filter. That
is, the error signal is the excitation to the decoder. The synthesis filter
is designed at the receiver using the received predictor coefficients. In
practice, many LPC coders transmit the filter coefficients which already
represent the error signal and can be directly synthesized by the receiver.
Figure 1 shows a block diagram of an LPC system [Jay86].
Figure 1 Block diagram of a LPC coding system.

Determination of Predictor Coefficients The linear predictive coder

uses a weighted sum of p past samples to estimate the present sample,
where p is typically in the range of 1015. Using this technique, the
current sample sn can be written as a linear sum of the immediately
preceding samples snk
p
sn = ak sn k + en (2)
k =1
where en is the prediction error (residual). The predictor coefficients

are calculated to minimize the average energy E in the error signal
that represents the difference between the predicted and actual speech
amplitude
2
N p N
E = e = ak sn k (3)
2
n
n =1 n =1 k = 0
where a0 = -1 Typically, the error is computed for a time window of

10 ms, which corresponds to a value of N = 80. To minimize E with
respect to am, it is required to set the partial derivatives equal to zero
E N p
= 2 sn m a s k nk = 0 (4)
am n = 1 k =0
p N
= sn m sn k ak = 0 (5)
k = 0 n =1
The inner summation can be recognized as the correlation coefficient Crm

and hence the above equation can be rewritten as
p
C
k =0
mk ak = 0 (6)
After determining the correlation coefficients Crm, Equation (6) can

be used to determine the predictor coefficients. Equation (6) is often
expressed in matrix notation and the predictor coefficients calculated
using matrix inversion. A number of algorithms have been developed to
speed up the calculation of predictor coefficients. Normally, the predictor
coefficients are not coded directly, as they would require 8 bits to 10 bits per
coefficient for accurate representation [Del93]. The accuracy requirements
are lessened by transmitting the reflection coefficients (a closely related
parameter), which have a smaller dynamic range. These reflection
coefficients can be adequately represented by 6 bits per coefficient. Thus,
for a 10th order predictor, the total number of bits assigned to the model

parameters per frame is 72, which includes 5 bits for a gain parameter
and 6 bits for the pitch period. If the parameters are estimated every
15 ms to 30 ms, the resulting bit rate is in the range of 2400 bps to 4800
bps. The coding of the reflection coefficient can be further improved by
performing a nonlinear transformation of the coefficients prior to coding.
This nonlinear transformation reduces the sensitivity of the reflection
coefficients to quantization errors. This is normally done through a log-
area ratio (LAR) transform which performs an inverse hyperbolic tangent
mapping of the reflection coefficients, Rn (k)
1 + R n (k )
LAR n ( k ) = tanh 1 ( R n ( k )) log10 (1)
1 R n ( k )
Various LPC schemes differ in the way they recreate the error signal
(excitation) at the receiver. Three alternatives are shown in Figure
[Luc89]. The first one shows the most popular means. It uses two sources
at the receiver, one of white noise and the other with a series of pulses at
the current pitch rate. The selection of either of these excitation methods
is based on the voiced/unvoiced decision made at the transmitter and
communicated to the receiver along with the other information. This
technique requires that the transmitter extract pitch frequency information
which is often very difficult. Moreover, the phase coherence between the
harmonic components of the excitation pulse tends to produce a buzzy
twang in the synthesized speech. These problems are mitigated in the
other two approaches: Multipulse excited LPC and stochastic or code
excited LPC.
Multipulse Excited LPC
Atal showed [Ata86] that, no matter how well the pulse is positioned,
excitation by a single pulse per pitch period produces audible distortion.
Therefore, he suggested using more than one pulse, typically eight per
period, and adjusting the individual pulse positions and amplitudes
sequentially to minimize a spectrally weighted mean square error. This
technique is called the multipulse excited LPC (MPE-LPC) and results
in better speech quality, not only because the prediction residual is better
approximated by several pulses per pitch period, but also because the
multipulse algorithm does not require pitch detection. The number of
pulses used can be reduced, in particular for high pitched voices, by
incorporating a linear filter with a pitch loop in the synthesizer.
15. (a) Refer Q. no. 15. (a) from April/May 2012.

NOV/DEC 2012
Seventh Semester
1. What is flat fading?
2. Define signal to self-interference ratio.
3. Distinguish between narrowband and wideband systems.
4. What is Link Budget calculation?
5. Find the 3-dB bandwidth for a Gaussian low pass filter used to produce 0.25
GMSK with a channel data rate of Rb = 270 kbps. What is the 90% power
bandwidth in the RF channel?
6. What is slotted frequency hopping?
7. Assume four branch diversity is used, where each branch receives an

independent Rayleigh fading signal. If the average SNR is 20 dB, determine
the probability that the SNR will drop below 10 dB. Compare this with the
case of a single receiver without diversity.
8. Define coding gain.
9. What is duplexing?
10. What is the speech code used in IS-95 system? Why?
11. (a) (i) Explain about the factors that influence small scale fading.

(ii) Find the average fade duration for threshold levels? = 0.01,
? = 0.1 and ? = 1, when the Doppler frequency is 200 Hz.
Or
(b) (i) Write a note on Noise and Interference Limited Systems.
(ii) Discuss the principles of Cellular Networks.
12. (a) (i) How the received signal strength is predicted using the free
space propagation model? Explain.
(ii) Find the far field distance for an antenna with maximum
dimension of 1 m and operating frequency of 900 MHz.
Or
(b) (i) With system theoretic description, explain the characteristics of
Time Dispersive channels.
(ii) Explain the three basic propagation mechanisms in a mobile
communication system.
13. (a) (i) Briefly explain the structure of a wireless communication

link.
(ii) With block diagram, explain the MSK transmitter and receiver.
Derive an expression for MSK and its power spectrum.
Or
(b) Derive an expression for
(i) M-ary phase shift keying and M-ray quadrature amplitude
modulation.
(ii) Also derive an expression for their bit error probability.
14. (a) Explain in detail about

(i) Polarization diversity
(ii) Time diversity
(iii) Frequency diversity
Or
(b) (i) Explain the basic idea about linear and decision feedback
equalisers and derive an expression for its minimum mean
square error.
(ii) With a suitable diagram, explain the channel coding and speech
coding techniques.

15. (a) (i) Discuss in detail about cellular code division multiple access
systems with neat diagrams.
(ii) Write short notes on transceiver implementation.
Or
(b) (i) Explain with neat diagram of orthogonal frequency division
multiplexing.
(ii) Write a note on second generation and third generation wireless
networks and standards.

Solutions
PART A
1. If the mobile radio channel has a constant gain and linear phase response
over a bandwidth which is greater than the bandwidth of the transmitted
signal, then the received signal will undergo flat fading.
The conditions for the flat fading are
(i) BW of the signal < BW of the channel
(ii) Delay spread < Symbol period
2. In general, the signal-to-interference ratio for cellular network can be

written as
pdesired
SR =
interference, i
P
i
Here Pdesired is the signal strength from the desired BS and

PInterference is the signal strength from the ith interfering BS.
3. Narrowband systems Here the available radio spectrum is divided into

large number of narrowband channels. The bandwidth of a single channel
is nothing but coherence bandwidth of the channel.
Wideband systems The transmission bandwidth of a single channel
is much larger than the coherence bandwidth of the channel.
4. Link Calculation: In designing a system for reliable communications,

we must perform a link budget calculation to ensure that sufficient
power is available at the receiver to close the link and meet the SNR
Requirement.
5.
Figure 1 Block diagram of a GMSK transmitter using direct FM

generation


Figure 2 Block diagram of a GMSK receiver
From the problem statement
1 1
T = = = 3.7s
R b 270 103
Solving for B, where BT = 0.25,
0.25 0.25
B= = = 67.567 kHz
T 3.7 10 6
Thus the 3-dB bandwidth is 67.567 kHz. To determine the 90% power
band-width, use Table 6.3 to find that 0.57 Rb is the desired value. Thus,
the occupied RF spectrum for a 90% power bandwidth is given by
RF BW = 0.57R b = 0.57 270 103 = 153.9 kHz
6. A form of spread spectrum in which the signal is broadcast over a
seemingly random series of radio frequencies, hopping from frequency
to frequency at fixed intervals.
7. Given: Five branches diversity, specific threshold, g = 10dB, = 20dB.
Hence / = 0.1
P1(10 dB) = (1 - e-0.1)1 = 0.095, when M = 1 (no diversity)
P5(10 dB) = (1 - e-0.1)5 = 0.0000078, when M = 5 i.e. five branches are used.
Mean SNR, g = (1 + 1/ 2 + 1/ 3 + 1/ 4 + 1/ 5) = 20( 2.28) = 45.6dB >>>
20 dB i.e. due to selection diversity average SNR improves multifold.
Also, without diversity, the SNR drops below the specific threshold
with the probability, i.e. four orders of magnitude greater than, if five
branch diversity is used.

Number of message bits ( k )

8. Coding gain or Efficiency h =
Number of bits in code word ( n)
9. Duplexing is the process of transmitting information in both directions

simultaneously between the sender and the receiver.
10. The speech coder used in the IS-95 system is the Qualcomm 9600bps
Code Excited Linear Predictive (QCELP) coder. The original imple
mentation of this vocoder detects voice activity, and reduces the data rate
to 1200 bps during silent periods.
PART B
1 1. (a) (i) Refer 12. (a) (i) from Nov/Dec 2013
11. (a) (ii)
11. (b) (i)
11. (b) (ii) Refer 11. (a) from Nov/Dec 2013
12. (a) (i) Refer Q. no. 12. (a) (i) from April/May 2012
12. (a) (ii) Operating frequency, f = 900 MHz
l = c/f = 3 108 m/s/900 106 Hz = 0.33m
Fraunhofer distance, df = 2D2/l = 2(1)2/0.33 = 6 m
Path loss PL(dB) = -10 log [(l2)/(4p)2 d2] = -10 log [(0.33)2 / (4 3.14)2
36] = 47dB
12. (b) (i) Time Dispersion Parameters
In order to compare different multipath channels and to develop some
general design guidelines for wireless systems, parameters which
grossly quantify the multipath channel are used. The mean excess
delay, rms delay spread, and excess delay spread (X dB) are multipath
channel parameters that can be determined from a power delay profile.
The time dispersive properties of wide band multipath channels are most
commonly quantified by their mean excess delay (t ) and rms delay
spread (st). The mean excess delay is the first moment of the power
delay profile and is defined to be
a t 2
k k P(t )tk k
t = k
= k
(1)
a k
2
k P(t )
k
k
The rms delay spread is the square root of the second central moment of
the power delay profile and is defined to be

s t = t 2 (t ) 2 (2)
where
a t 2 2
k k P(t )tk
2
k
t2 = k
= k
(3)
a k
2
k P(t )
k
k
These delays are measured relative to the first detectable signal arriving
at the receiver at t0 = 0. Equations (1)-(3) do not rely on the absolute
power level of P(t), but only the relative amplitudes of the multipath
components within P(t). Typical values of rms delay spread are on the
order of microseconds in outdoor mobile radio channels and on the
order of nanoseconds in indoor radio channels. Table 1 shows the typical
measured values of rms delay spread.
It is important to note that the rms delay spread and mean excess delay
are defined from a single power delay profile which is the temporal or
spatial average of consecutive impulse response measurements collected
and averaged over a local area. Typically, many measurements are made
at many local areas in order to determine a statistical range of multipath
channel parameters for a mobile communication system over a large-
scale area [Rap90].
The maximum excess delay (X dB) of the power delay profile is
defined to be the time delay during which multipath energy falls to X
dB below the maximum. In other words, the maximum excess delay
is defined as tx - t0 where t0 is the first arriving signal and tX is the
maximum delay at which a multipath component is within X dB of the
strongest arriving multipath signal (which does not necessarily arrive at
t0). Figure 1 illustrates the computation of the maximum excess delay
for multipath components within 10 dB of the maximum. The maximum
excess delay (X dB) defines the temporal extent of the multipath that
is above a particular threshold. The value of tx is sometimes called the
excess delay spread of a power delay profile, but in all cases must be
specified with a threshold that relates the multipath noise floor to the
maximum received multipath component.
12. (b) (ii) Refer Q. no. 11 (b) (i) from Nov/Dec 2013
13. (a) (i)
13. (a) (ii) Refer Q. no. 13 (b) (i) from April/May 2013

13. (b) (i) M-ary Phase Shift Keying (MPSK)

In M-ary PSK, the carrier phase takes on one of M possible values,
namely, qi = 2(i - 1) p/M, where i = 1, 2, M. The modulated waveform
can be expressed as
2 Es 2p
si (t ) = cos 2p f c t + (i 1) , 0 t Ts i = 1, 2,, M (1)
Ts M
Where Es = (log2 M) Eb is the energy per symbol and Ts = (log2 M) Tb is the

symbol period. The above equation can be rewritten in quadrature form as
2 Es 2p
si (t ) = cos (i 1) cos( 2p f c t ) i = 1, 2,, M (2)
Ts M
2E s 2p
sin (i 1) sin( 2p f c t )
Ts M
2
By choosing orthogonal basis signals f1 (t ) = cos( 2p f c t ), and
Ts
2
f2 (t ) = sin( 2p f c t ) defined over the interval 0 t Ts, the M-ary
Ts
PSK signal set can be expressed as
p p
s M PSK (t ) = Es cos (i 1) f1 (t ), E s sin (i 1) f 2 (t ) (3)
2 2
i = 1, 2,, M
Since there are only two basis signals, the constellation of M-ary PSK
is two dimensional. The M-ary message points are equally spaced on a
circle of radius Es centered at the origin.
M-ary Quadrature Amplitude Modulation (QAM)
In M-ary PSK modulation, the amplitude of the transmitted signal was
constrained to remain constant, thereby yielding a circular constellation.
By allowing the amplitude to also vary with the phase, a new modulation
scheme called quadrature amplitude modulation (QAM) is obtained.
Figure 1 shows the constellation diagram of 16-ary QAM. The
constellation consists of a square lattice of signal points. The general
form of an M-ary QAM signal can be defined as
2E min 2E min
s i (t ) = ai cos( 2p f c t ) + bi sin( 2p f c t )
Ts Ts
0 t Ti = 1, 2,, M

where Emin is the energy of the signal with the lowest amplitude, and ai
and bi are a pair of independent integers chosen according to the location
of the particular signal point. Note that M-ary QAM does not have constant
energy per symbol, nor does it have constant distance between possible
symbol states. It reasons that particular values of si(t) will be detected
with higher probability than others.
13. (b) (ii)
14. (a) Polarization Diversity

At the base station, space diversity is considerably less practical than
at the mobile because the narrow angle of incident fields requires large
antenna spacings [Vau90]. The comparatively high cost of using space
diversity at the base station prompts the consideration of using orthogonal
polarization to exploit polarization diversity. While this only provides two
diversity branches, it does allow the antenna elements to be co-located.
In the early days of cellular radio, all subscriber units were mounted
in vehicles and used vertical whip antennas. Today, however, over half of
the subscriber units are portable. This means that most subscribers are no
longer using vertical polarization due to hand-tilting when the portable
cellular phone is used. This recent phenomenon has sparked interest in
polarization diversity at the base station.
Theoretical Model for Polarization Diversity
It is assumed that the signal is transmitted from a mobile with vertical
(or horizontal) polarization. It is received at the base station by a
polarization diversity antenna with two branches. Figure 1 shows the
theoretical model and the system coordinates. As seen in the figure, a
polarization diversity antenna is composed of two antenna elements V1
and V2, which make a a angle (polarization angle) with the Y axis. A
mobile station is located in the direction of offset angle b from the main
beam direction of the diversity antenna as seen in Figure 1b.
Some of the vertically polarized signals transmitted are converted to
the horizontal polarized signal because of multipath propagation. The
signal arriving at the base station can be expressed as
x = r1 cos (wt + f1)(1)
y = r2 cos (wt + f2)(2)
where x and y are signal levels which are received when b = 0. It is

assumed that r1 and r2 have independent Rayleigh distributions, and f1
and f2 have independent uniform distributions.

V2 V1
(a)
X
Multipath
Mobile Main Beam

(b)
Figure 1 Theoretical model for base station polarization diversity based on

[Koz85]: (a) x-y plane; (b) xz plane
The received signal values at elements V1 and V2 can be written as

V1 = (ar1 cos f1 + r2b cos f2) cos wt - (ar1 sin f1 + r2b sin f2) sin wt (3)
V2 = (-ar1 cos f1 + r2b cos f2) cos wt - (-ar1 sin f1 + r2b sin f2) sin wt (4)
where a = sina cosb and b = cos a
The correlation coefficient r can be written as

2
tan 2 (a ) cos 2 ( b )
r= 2 (5)
tan (a ) cos 2 ( b ) +
where
R 22
X = (6)
R12
and
R1 = r12 a 2 + r22 b 2 + 2r1r2 ab cos(f1 + f2 ) (7)
R2 = r12 a 2 + r22 b 2 2r1r2 ab cos(f1 + f2 ) (8)

Here, X is the cross polarization discrimination of the propagation path

between a mobile and a base station.
Frequency Diversity
Frequency diversity is implemented by transmitting information on
more than one carrier frequency. The rationale behind this technique is
that frequencies separated by more than the coherence bandwidth of the
channel will be uncorrelated and will thus not experience the same fades
[Lem91]. Theoretically, if the channels are uncorrelated, the probability
of simultaneous fading will be the product of the individual fading
probabilities.
Frequency diversity is often employed in microwave line-of-sight
links which carry several channels in a frequency division multiplex
mode (FDM). Due to tropospheric propagation and resulting refraction,
deep fading sometimes occurs. In practice, 1:N protection switching is
provided by a radio licensee, wherein one frequency is nominally idle but
is available on a stand-by basis to provide frequency diversity switching
for any one of the N other carriers (frequencies) being used on the same
link, each carrying independent traffic. When diversity is needed, the
appropriate traffic is simply switched to the backup frequency. This
technique has the disadvantage that it not only requires spare bandwidth
but also requires that there be as many receivers as there are channels
used for the frequency diversity. However, for critical traffic, the expense
may be justified.
New OFDM modulation and access techniques exploit frequency
diversity by providing simultaneous modulation signals with error control
coding across a large bandwidth, so that if a particular frequency undergoes
a fade, the composite signal will still be demodulated.
Time Diversity
Time diversity repeatedly transmits information at time spacings that
exceed the coherence time of the channel, so that multiple repetitions of
the signal will be received with independent fading conditions, thereby
providing for diversity. One modern implementation of time diversity
involves the use of the RAKE receiver for spread spectrum CDMA,
where the multipath channel provides redundancy in the transmitted
message. By demodulating several replicas of the transmitted CDMA
signal, where each replica experiences a particular multipath delay, the
RAKE receiver is able to align the replicas in time so that a better estimate
of the original signal may be formed at the receiver.

1 4. (b) (i) Refer Q. no. 14 (a) (ii) from April/May 2012

14. (b) (ii) Refer Q. no. 14 (b) from April/May 2012
15. (a) (i) Refer Q. no. 15 (a) from April/May 2012
15. (a) (ii)
15. (b) (i)
15. (b) (ii) Refer Q. no. 15 (b) from Nov/Dec 2013

MAY/JUNE 2012
Seventh Semester
Wireless Communication
1. What are the different types of multiple Access schemes?
2. Mention the significance of frequency reuse in Cellular Networks.
3. List the different types of wireless channels.
4. What is frequency selective fading? How to avoid fading problem?
5. List the advantages of QPSK.
6. Differentiate between MSK and GMSK.
7. List the different types of Speech coding techniques.
8. State the significance of linear and decision feedback equalizer.
9. State effects of multipath propagation on CDMA.
10. List a few wireless network standards.
11. (a) (i)
Explain in detail Wide Area Data Services and Broadband
Wireless Access services offered to Wireless networks.
(ii) What are Paging Systems? Explain.
Or
M02_ECE_7th-SEMESTER_CH04(May-Jun-2012).indd 1 4/25/2014 9:38:04 PM

(b) (i)
With a neat block diagram, explain the Cellular Network
Architecture.
(ii) Explain any one type of Multiple Access scheme.
1 2. (a) (i) Explain the free space path loss and derive the gain expression.
(ii) Describe in detail Two Ray Model propagation mechanism.
Or
(b) (i) Define the following Auto-correlation, Cross correlation and
power spectral density for narrow band fading model.
(ii) What is the need for link calculation? Explain with suitable
example.
13. (a) Explain with neat signal diagrams, the modulation and demodulation
technique of QPSK.
Or
(b) (i) Describe with a block diagram Offset-Quadrature Phase Shift
Keying and its advantages.
(ii) Explain the concept of GMSK and mention its advantages.
1 4. (a) (i) With a neat block diagram, explain the principle of diversity.
(ii) Explain in detail Decision feedback equalizer.
Or
(b) (i) Explain any one method of channel coding.
(ii) What are the advantages of speech coding? Explain any one
technique of speech coding
15. Explain:
(a) Code Division Multiple Access (CDMA) and compare its perfor
mance with TDMA.
Or
(b) What is orthogonal frequency division multiplexing? Explain
OFDM technique and mention its merits, demerits and application.

Solutions
PART A
1. The different types of multiple access schemes are
(i) Frequency Division Multiple Access (FDMA)
(ii) Time Division Multiple Access (TDMA)
(iii) Code Division Multiple Access (CDMA) and
(iv) Space Division Multiple Access (FDMA)
2. Physical separation of two cells is sufficiently wide; the same subset of

frequencies can be used in both cells. This is the concept of frequency
reuse. The ability of frequency is to expand the total system capacity
without the need of high power transmitters. This is called frequency
reuse or frequency planning.
3. Different types of wireless channels are

(i) Flat frequency (or) time selective channels and
(ii) Frequency selective channels
4. A signal undergoes frequency selective fading if

(i) BW of signal (BS) > BW of channel (BC) and
(ii) Symbol period (TS ) < Delay period (s t )
5. Advantages of QPSK
(i) It has twice the bandwidth efficiency of BPSK, since two bits are
transmitted in a single modulation symbol.
(ii) Higher data rate.
6. MSK:
(i)
Minimum shift keying is a special type of continuous phase-
frequency shift keying (CPFSK) wherein the peak frequency
deviation is equal to the bit rate.
(ii)MSK is a spectrally efficient modulation scheme and is particularly
attractive for use in mobile radio communication systems.
(iii)Main lobe of MSK is wide. So, it is not suitable for multi-user
communications.

Where as, GMSK:

(i)
Gaussian refers to the shape of a filter. It is a simple binary
modulation scheme.
(ii)The side lobe levels of the spectrum are much reduced by passing
the modulating NRZ data waveform through a premodulation
Gaussian pulse-shaping filter.
(iii)GMSK has excellent power efficiency and spectral efficiency than
conventional FSK.
7. Different types of speech coding techniques are

(i) Linear Predictive Coders (LPCs)
(ii) Adaptive Predictive Coding (APC)
(iii) Adaptive Differential Pulse Code Modulation (ADPCM)
(iv) Sub-Band Coding (SBC) and
(v) Adaptive Transform Coding (ATC)
8. Linear equalizer is a simple one and use FIR filter structure. Decision
feedback equalizer is a nonlinear equalizer that are used in applications
where the channel distortions too severe for linear equalizer to handle.
The main advantage for this, once an information symbol has been
detected and decided upon, the ISI that it induces on future symbols can
be estimated.
9. Effects of multi-path propagation on CDMA:

(i)In CDMA, the power of multiple users at a receivers determines the
noise floor after decorrelation. If the power of each user within a
cell is not controlled such that they do not appear equal at the base
station receiver.
(ii)The near-far problem occurs when many users share the same
channel.
10. (i) Digital European Cordless Telephone (DECT)

(ii) Advanced Mobile Phone System (AMPS)
(iii) Global System for Mobile Communication (GSM) and
(iv) Cellular Digital Packet Data (CDPD)

PART B
1 1. (a) (i) Refer Q. no. 11 (a) from Nov/Dec 2013
11. (a) (ii) Refer Q. no. 11 (a) from Nov/Dec 2013 (only paging systems)
11. (b) (i) Refer Q. no. 11 (b) from Nov/Dec 2013
11. (b) (ii)Time Division Multiple Access (TDMA)
Time Division Multiple Access (TDMA) systems divide the radio spectrum
into time slots, and in each slot only one user is allowed to either transmit
or receive. It can be seen from Figure 1 that each user occupies a cyclically
repeating time slot, so a channel may be thought of as a particular time
slot that reoccurs every frame, where N time slots comprise a frame.
TDMA systems transmit data in a buffer-and-burst method, thus the
transmission for any user is noncontinuous. This implies that, unlike in
FDMA systems which accommodate analog FM, digital data and digital
modulation must be used with TDMA. The transmission from various
users is interlaced into a repeating frame structure as shown in Figure 2.
It can be seen that a frame consists of a number of slots. Each frame is
made up of a preamble, an information message, and tail bits. In TDMA/
TDD, half of the time slots in the frame information message would be
used for the forward link channels and half would be used for reverse link
channels. In TDMA/FDD systems, an identical or similar frame structure
would be used solely for either forward or reverse transmission, but the
carrier frequencies would be different for the forward and reverse links.
In general, TDMA/FDD systems intentionally induce several time slots
of delay between the forward and reverse time slots for a particular user,
so that duplexers are not required in the subscriber unit.
Code
Channel N
ts
Channel 3
lo
eS
Channel 2
m
Ti
Channel 1
Frequency
Time
Figure 1 TDMA scheme where each channel occupies a cyclically

repeating time slot

In a TDMA frame, the preamble contains the address and synchronization

information that both the base station and the subscribers use to identify
each other. Guard times are utilized to allow synchronization of the
receivers between different slots and frames. Different TDMA wireless
standards have different TDMA frame structures, and some are described
in Chapter 11. The features of TDMA include the following:
One TDMA Frame
Preamble Information Message Trail Bits
Slot 1 Slot 2 Slot 3 Slot
Trail Bits Sync. bits Information Data Guard Bits
Figure 2 TDMA frame structure. The frame is cyclically repeated over

time
TDMA shares a single carrier frequency with several users, where

each user makes use of nonoverlapping time slots. The number of
time slots per frame depends on several factors, such as modulation
technique, available bandwidth, etc.
Data transmission for users of a TDMA system 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 (which is
most of the time).
Because of discontinuous transmissions in TDMA, the handoff
process is much simpler for a subscriber unit, since it is able to
listen for other base stations during idle time slots. An enhanced link
control, such as that provided by mobile assisted handoff (MAHO)
can be carried out by a subscriber by listening on an idle slot in the
TDMA frame.
TDMA uses different time slots for transmission and reception, thus
duplexers are not required. Even if FDD is used, a switch rather than
a duplexer inside the subscriber unit is all that is required to switch
between transmitter and receiver using TDMA.
Adaptive equalization is usually necessary in TDMA systems, since
the transmission rates are generally very high as compared to FDMA
channels.

In TDMA, the guard time should be minimized. If the transmitted

signal at the edges of a time slot are suppressed sharply in order to
shorten the guard time, the transmitted spectrum will expand and
cause interference to adjacent channels.
High synchronization overhead is required in TDMA systems
because of burst transmissions. TDMA transmissions are slotted, and
this requires the receivers to be synchronized for each data burst.
In addition, guard slots are necessary to separate users, and this
results in the TDMA systems having larger overheads as compared
to FDMA.
TDMA has an advantage in that it 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 slots based on priority.
Efficiency of TDMA The efficiency of a TDMA system is a measure
of the percentage of transmitted data that contains information as opposed
to providing overhead for the access scheme. The frame efficiency, hf, is
the percentage of bits per frame which contain transmitted data. Note that
the transmitted data may include source and channel coding bits, so the
raw end-user efficiency of a system is generally less than hf. The frame
efficiency can be found as follows.
The number of overhead bits per frame is [Zie92],
bOH = N r b r + N t b P + N t b g + N r b g (1)
where Nr is the number of reference bursts per frame, Nt is the number

of traffic bursts per frame, br is the number of overhead bits per reference
burst, bP is the number of overhead bits per preamble in each slot, and
bg is the number of equivalent bits in each guard time interval. The total
number of bits per frame,bT, is
bT = Tf R (2)
where Tf is the frame duration, and R is the channel bit rate. The frame
efficiency hf, is thus given as
b
h f = 1 OH 100%(3)
bT
Number of channels in TDMA system The number of TDMA

channel slots that can be provided in a TDMA system is found by

multiplying the number of TDMA slots per channel by the number of

channels available and is given by
m( Btot 2 Bguard )
N= (4)
Bc
Where m is the maximum number of TDMA users supported on each
radio channel. Note that two guard bands, one at the low end of the
allocated frequency band and one at the high end, are required to ensure
that users at the edge of the band do not bleed over into an adjacent
radio service.
12. (a) (i) The free space propagation model is used to predict received
signal strength when the transmitter and receiver have a clear,
unobstructed line-of-sight path between them. Satellite communication
systems and microwave line-of-sight radio links typically undergo free
space propagation. As with most large-scale radio wave propagation
models, the free space model predicts that received power decays as
a function of the TR separation distance raised to some power (i.e.
a power law function). The free space power received by a receiver
antenna which is separated from a radiating transmitter antenna by a
distance d, is given by the Friis free space equation,
t tGr l
2
PG
Pr (d ) = (1)
( 4p )2 d 2 L
Where Pt is the transmitted power, Pr(d) is the received power which is
a function of the TR separation, Gt is the transmitter antenna gain, Gr
is the receiver antenna gain, d is the TR separation distance in meters,
L is the system loss factor not related to propagation (L 1) and l is the
wavelength in meters. The gain of an, antenna is related to its effective
aperture Ae, by
4p A e
G= (2)
l2
The effective aperture Ae is related to the physical size of the antenna,
and l is related to the carrier frequency by
c 2l c
l= = (3)
f wc
where f is the carrier frequency in Hertz, wc is the carrier frequency in
radians per second, and c is the speed of light given in meters/s. The
values for Pt or Pr must be expressed in the small units, and Gt and Gr are
dimensionless quantities. The miscellaneous losses L (L 1) are usually
due to transmission line attenuation, filter losses, and antenna losses in

the communication system. A value of L = 1 indicates no loss in the

system hardware.
The Friis free space equation of (1) shows that the received power
falls off as the square of the TR separation distance. This implies that
the received power decays with distance at a rate of 20 dB/decade.
An isotropic radiator is an ideal antenna which radiates power with
unit gain uniformly in all directions, and is often used to reference
antenna gains in wireless systems. The effective isotropic radiated
power (EIRP) is defined as and represents the maximum radiated power
available from a transmitter in the direction of maximum antenna gain, as
compared to an isotropic radiator.
EIRP = Pt Gt(4)
In practice, effective radiated power (ERP) is used instead of EIRP to
denote the maximum radiated power as compared to a half-wave dipole
antenna (instead of an isotropic antenna). Since a dipole antenna has
a gain of 1.64 (2.15 dB above an isotrope), the ERP will be 2.15 dB
smaller than the EIRP for the same transmission system. In practice,
antenna gains are given in units of dBi (dB gain with respect to an
isotropic antenna) or dBd (dB gain with respect to a half-wave dipole)
[Stu81].
The path loss, which represents signal attenuation as a positive quantity
measured in dB, is defined as the difference (in dB) between the effective
transmitted power and the received power, and may or may not include
the effect of the antenna gains. The path loss for the free space model
when antenna gains are included is given by
Pt G G l2
PL (dB) = 10log = 10log t r2 2 (5)
Pr ( 4p ) d
When antenna gains are excluded, the antennas are assumed to have
unity gain, and path loss is given by
Pt l2
PL (dB) = 10log = 10log 2 2
(6)
Pr ( 4p ) d
The Friis free space model is only a valid predictor for Pr for values of
d which are in the far-field of the transmitting antenna. The far-field,
or Fraunhofer region, of a transmitting antenna is defined as the region
beyond the far-field distance df, which is related to the largest linear
dimension of the transmitter antenna aperture and the carrier wavelength.
The Fraunhofer distance is given by

2D 2
df =(7)
l
where D is the largest physical linear dimension of the antenna.
Additionally, to be in the far-field region, df must satisfy
df D (8)
and
df l (9)
Furthermore, it is clear that Equation (1) does not hold for d = 0. For
this reason, large-scale propagation models use a close-in distance,
d0, as a known received power reference point. The received power,
Pr(d), at any distance d > d0, may be related to Pr at d0. The value Pr(d0)
may be predicted from Equation (1), or may be measured in the radio
environment by taking the average received power at many points
located at a close-in radial distance d0 from the transmitter. The reference
distance must be chosen such that it lies in the far-field region, that is,
d0 df, and d0 is chosen to be smaller than any practical distance used
in the mobile communication system. Thus, using Equation (1), the
received power in free at a distance greater than d0 is given by
2
d
Pr ( d ) = Pr ( d0 ) 0 d d0 d f (10)
d
In mobile radio systems, it is not uncommon to find that Pr may change
by many orders of magnitude over a typical coverage area of several
square kilometers. Because of the large dynamic range of received
power levels, often dBm or dBW units are used to express received
power levels. Equation (10) may be expressed in units of dBm or dBW
by simply taking the logarithm of both sides and multiplying by 10. For
example, if Pr is in units of dBm, the received power is given by where
Pr(d0) is in units of watts.
P (d ) d
Pr ( d ) dBm = 10log r 0 + 20 log 0 d d0 d f (11)
0.001 W d
The reference distance d0 for practical system using low-gain
antennas in the 12 GHz region is typically chosen to be 1 m in indoor
environments and 100 m or 1 km in outdoor environments, so that the
numerator in Equations (10) and (11) is a multiple of 10. This makes
path loss computations easy in dB units.
Operating frequency, f = 900 MHz

l = c /f = 3 108 m /s / 900 106 Hz = 0.33 m

Fraunhofer distance, d f = 2D 2 /l = 2(1) 2 / 0.33 = 6 m
path loss PL (dB ) = 10 log [(l 2 )/(4p ) 2 d 2 ]
= 10 log [(0.33) 2 /(4 3.14) 2 36] = 47 dB
12. (a) (ii) Reflection from Perfect Conductors

Since electromagnetic energy cannot pass through a perfect conductor
a plane wave incident on a conductor has all of its energy reflected. As
the electric field at the surface of the conductor must be equal to zero at
all times in order to obey Maxwells equations, the reflected wave must
be equal in magnitude to the incident wave. For the case when E-field
polarization is in the plane of incidence, the boundary conditions require
that [Ram65]
qi = qr(1)
and
Ei = Er (E-field in plane of incidence) (2)
Similarly, for the case when the E-field is horizontally polarized, the
boundary conditions require that
qi = qr(4.31)
and
Ei = Er (E-field normal to plane of incidence) (4.32)
Referring to Equations (1) to (4), we see that for a perfect conductor,
G|| = 1, and G^ = 1 regardless of incident angle. Elliptical polarized
waves may be analyzed by using superposition.
Ground Reflection (Two-Ray) Model
In a mobile radio channel, a single direct path between the base station
and a mobile is seldom the only physical means for propagation, and
hence the free space propagation model of Equation (3) is in most cases
inaccurate when used alone. The two-ray ground reflection model shown
in Figure 3 is a useful propagation model that is based on geometric
optics, and considers both the direct path and a ground reflected
propagation path between transmitter and receiver. This model has been
found to be reasonably accurate for predicting the large-scale signal
strength over distances of several kilometers for mobile radio systems
that use tall towers (heights which exceed 50 m), as well as for line-of-
sight microcell channels in urban environments [Feu94].

In most mobile communication systems, the maximum TR

separation distance is at most only a few tens of kilometers, and the earth
may be assumed to be flat. The total received E-field, ETOT, is then a result
of the direct line-of-sight component, ELOS, and the ground reflected
component, Eg.
Figure 1 Two-ray ground reflection model.
Referring to Figure 1, ht is the height of the transmitter and hr is the

height of the receiver. If E0 is the free space E-field (in units of V/m) at a
reference distance d0 from the transmitter, then for d > d0, the free space
propagating E-field is given by
E0d 0 d
E (d , t ) = cos w c t (d > d 0 ) (5)
d c
where E(d, t) = E0d0/d represents the envelope of the E-field at d meters

from the transmitter.
Two propagating waves arrive at the receiver: the direct wave that travels
a distance d; and the reflected wave that travels a distance d. The E-field
due to the line-of-sight component at the receiver can be expressed as
E 0d 0 d
E LOS (d , t ) = cos w c t (6)
d c
and the E-field for the ground reflected wave, which has a propagation
distance of d, can be expressed as
E0 d0 d
E g ( d , t ) = cos w c t (7)
d c
According to laws of reflection in dielectrics given in Section 4.5.1

qi = q0(8)

and
Eg = G Ei(9)
Et = (1 + G) Ei (10)
where G is the reflection coefficient for ground. For small values of qi
(i.e., grazing incidence), the reflected wave is equal in magnitude and
180 out of phase with the incident wave, as shown in Example 4.4. The
resultant E-field, assuming perfect horizontal E-field polarization and
ground reflection (i.e., G^ = -1 and Et = 0), is the vector sum of ELOS and
Eg, and the resultant total E-field envelope is given by
|ETOT| = |ELOS + Eg|(11)
The electric field ETOT (d, t) can be expressed as the sum of Equations (6)
and (7)
E 0d 0 d Ed d
ETOT (d , t ) = cos w c t + ( 1) 0 0 cos w c t (12)
d c d c
Using the method of images, which is demonstrated by the geometry of
Figure 2, the path difference, , between the line-of-sight and the ground
reflected paths can be expressed as
= d d = ( ht + hr ) 2 + d 2 ( ht hr ) 2 + d 2 (13)
When the TR separation distance d is very large compared to ht + hr
Equation (13) can be simplified using a Taylor series approximation
2ht hr
= d d (14)
d
Figure 2 The method of images is used to find the path difference between
the line-of-sight and the ground reflected paths

Once the path difference is known, the phase difference qD between the
two E-field components and the time delay td between the arrival of the
two components can be easily computed using the following relations
2p w c
q = = (15)
l c
and
q
td = = (16)
c 2p f c
It should be noted that as d becomes large, the difference between the
distances d and d becomes very small, and the amplitudes of ELOS and
Eg are virtually identical and differ only in phase. That is
E 0d 0 Ed Ed
0 0 0 0 (17)
d d d
If the received E-field is evaluated at some time, say at t = d/c, Equation
(12) can be expressed as a phasor sum
d E d d d E0d 0
ETOT d , t = 0 0 cos w c cos 0 (18)
c d c d
Ed Ed
= 0 0 q 0 0
d d
E0d 0
[q 1]
d
where d is the distance over a flat earth between the bases of the transmitter
and receiver antennas. Referring to the phasor diagram of Figure 5 which
shows how the direct and ground reflected rays combine, the electric field
(at the receiver) at a distance d from the transmitter can be written as
2 2
E d E d
ETOT ( d ) = 0 0 (cos q 1) 2 + 0 0 sin 2 q (19)
d d
Figure 3 Phasor diagram showing the electric field components of the

line-of-sight, ground reflected, and total received E-fields, derived from
Equation(18)

E0 d0
ETOT ( d ) = 2 2 cos q (20)
d
Note that if the E-field is assumed to be in the plane of incidence (i.e.,
vertical polarization) then G|| = 1 and Equation (20) would have a +
instead of a .
Using trigonometric identities, Equation (20) can be expressed as
E0 d0 q
ETOT ( d ) = 2 sin (21)
d 2
Equation (21) is an important expression, as it provides the exact received
E-field for the two-ray ground reflection model. One notes that for
increasing distance from the transmitter, ETOT (d) decays in an oscillatory
fashion, with local maxima being 6 dB greater than the free space
value and the local minima plummeting to dB (the received E-field
cancels out to zero volts at certain values of, although in reality this
never happens). Once the distance d is sufficiently large, qD becomes p
and the received E-field, ETOT (d), then falls off asymtotically with
increasing distance. Note that Equation (21) may be simplified whenever
(qD/2) qD/2. This occurs when qD/2 is less than 0.3 radian. Using
Equations (14) and (15)
q 2pht hr
< 0.3 rad (22)
2 ld
which implies that Equation (21) may be simplified whenever
20p ht hr 20 ht hr
d> (23)
3l l
Thus, as long as d satisfies (23), the received E-field can be approxi
mated as
2E 0 d 0 2p ht hr k
ETOT (d ) 2 V/m (24)
d ld d
Where k is a constant related to E0, the antenna heights, and the
wavelength. This asymptotic behavior is identical for both the E-field
in the plane of incidence or normal to the plane of incidence. The free
space power received at d is related to the square of the electric field
through Equation (24). Combining Equations (24), the received power at
a distance d from the transmitter for the two-ray ground bounce model
can be expressed as
ht2 hr2
Pr = PG
t tGr (25)
d4

As seen from Equation (25) at large distances (d ht hr ), the received

power falls off with distance raised to the fourth power, or at a rate of
40 dB/decade. This is a much more rapid path loss than is experienced
in free space. Note also that at large values of d, the received power and
path loss become independent of frequency. The path loss for the two-ray
model (with antenna gains) can be expressed in dB as
PL (dB ) = 40 log d (10 log G t + 10 log G r + 20 log ht + 20 log hr ) (26)
At small TR separation distances, Equation (12) must be used to

compute the total E-field. When Equation (15) is evaluated for q = p ,
then d = ( 4 ht hr ) /l is where the ground appears in the first Fresnel zone
between the transmitter and receiver (Fresnel zones are treated in Section
4.7.1). The first Fresnel zone distance is a useful parameter in microcell
path loss models [Feu94].
12. (b) (i)This assumption the Doppler phase shift is3 fn (t)=

t 2p f Dn dt = 2p f Dn t , and the phase of the nth multipath component
becomes fn (t) = 2p fctn - 2p fDnt - f0.
We now make a key assumption: we assume that for the nth multipath
component the term fctn in fn (t) changes rapidly relative to all other
phase terms in this expression. This is a reasonable assumption since
fc is large and hence the term fctn can go through a 360 degree rotation
for a small change in multipath delay tn. Under this assumption fn (t) is
uniformly distributed on [-p, p]. Thus
E[r1 (t )] = E a n cos fn (t ) = E [a n ]E[cos fn (t )] = 0,

n n
Consider now the autocorrelation of the in-phase and quadrature

components. Using the independence of an and f the independence of fn
and fm, n m, and the uniform distribution of fn we get that
E[r1 (t )rQ (t )] = E a n cos fn (t ) a m sin fm (t ) (1)

n m
= E [a na m ]E [cos fn (t ) sin fm (t )]
n m
= E [a n2 ]E [cos fn (t ) sin fm (t )]
n
= 0.
Thus r1 (t) and rQ (t) are uncorrelated and, since they are jointly Gaussian
processes, this means they are independent.

Following a similar derivation as in (1) we obtain the autocorrelation of

r1 (t) as
A r1 (t ,t ) = E[r1 (t )r1 (t + t )] = E[a n2 ]E[cos fn (t ) cos fn (t + t )]. (2)

n
Now making the substitution fn (t ) = 2p f ct n 2p f D n t f0 . We get

E[cos fn (t ) cos fn (t + t )] = .5E[cos 2pf D nt ]
+.5E[cos( 4p f ct n + 4p f D nt n 4p f D n t (3)
-2p fD nt 2f0 )].
Since fctn changes rapidly relative to all other phase terms and is uniformly
distributed, the second expectation term in (3) goes to zero, and thus
A r1 (t ,t ) = .5 E [a n2 ]E [cos( 2pf D nt )] (4)

n
= .5 E[a n2 ]cos( 2put cos q n /l

l ),
n
since f Dn = u cos q n /l is assumed fixed. Note that Ar1 (t ,t ) depends only on

t , A r1 (t,t ) = A r1 (t ), and thus r1 (t ) is a wide-sense stationary (WSS)
random process.
Using a similar derivation we can show that the quadrature component
is also WSS with autocorrelation ArQ (t ) = Ar1 (t ) . In addition, the cross
correlation between the in-phase and quadrature components depends
only on the time difference t and is given by
A r1 , rQ (t ,t ) = A r1 , rQ (t ) = E[r1 (t )rQ (t + t )]
= 5 E [an2 ]sin( 2put cos q n /l )
= E [rQ (t )r1 (t + t )]. (5)

The power spectral densities (PSDS) of r1 (t) and rQ (t), denoted by Sr1 (f)
and SrQ (f), respectively, are obtained by taking the Fourier transform of
their respective autocorrelation functions relative to the delay parameter
t. Since these autocorrelation functions are equal, so are the PSDS. Thus
S r1 (f ) = S rQ (f ) = F[A r1 (t )]
(6)
pr 1
2pfD f fD
= 1 (f /fD 2 )

0 else

12. (b) (ii) Most radio propagation models are derived using a combination
of analytical and empirical methods. The empirical approach is based on
fitting curves or analytical expressions that recreate a set of measured data.
This has the advantage of implicitly taking into account all propagation
factors, both known and unknown, through actual field measurements.
However, the validity of an empirical model at transmission frequencies
or environments other than those used to derive the model can only be
established by additional measured data in the new environment at the
required transmission frequency. Over time, some classical propagation
models have emerged, which are now used to predict large-scale
coverage for mobile communication systems design. By using path loss
models to estimate the received signal level as a function of distance,
it becomes possible to predict the SNR for a mobile communication
system
13. (a) Refer 13. (b) (i) from Nov/Dec 2013
13. (b) (i)
LPF Decision
Circuit
Carrier Symbol
Received Recovered
BPF Recovery Timing Multiplex
Signal Signal
Circuit Recovery
90
Decision
LPF Circuit
Figure 1 Block diagram of a QPSK receiver
Figure 1 shows a block diagram of a coherent QPSK receiver. The

frontend bandpass filter removes the out-of-band noise and adjacent
channel interference. The filtered output is split into two parts, and
each part is coherently demodulated using the in-phase and quadrature
carriers. The coherent carriers used for demodulation are recovered from
the received signal using carrier recovery circuits of the type described
in Figure 1. The outputs of the demodulators are passed through decision
circuits which generate the in-phase and quadrature binary streams. The
two components are then multiplexed to reproduce the original binary
sequence.

Offset QPSK
The amplitude of a QPSK signal is ideally constant. However, when
QPSK signals are pulse shaped, they lose the constant envelope property.
The occasional phase shift of p radians can cause the signal envelope
to pass through zero for just an instant. Any kind of hardlimiting or
nonlinear amplification of the zero-crossings brings back the filtered
sidelobes since the fidelity of the signal at small voltage levels is lost
in transmission. To prevent the regeneration of sidelobes and spectral
widening, it is imperative that QPSK signals that use pulse shaping be
amplified only using linear amplifiers, which are less efficient. A modified
form of QPSK, called offset QPSK (OQPSK) or staggered QPSK is less
susceptible to these deleterious effects [Pas79] and supports more efficient
amplification. That is, OQPSK ensures there are fewer baseband signal
transitions applied to the RF amplifier, which helps eliminate spectrum
regrowth after amplification.
OQPSK signaling is similar to QPSK signaling, as represented by
Equation (6.77), except for the time alignment of the even and odd bit
streams. In QPSK signaling, the bit transitions of the even and odd bit
streams occur at the same time instants, but in OQPSK signaling, the
even and odd bit streams, and mI(t) and mQ(t), are offset in their relative
alignment by one bit period (half-symbol period). This is shown in the
waveforms of Figure 2.
Figure 2 The time offset waveforms that are applied to the in-phase and
quadrature arms of an OQPSK modulator. Notice that a half-symbol offset
is used.

Due to the time alignment of mI (t) and mQ (t) in standard QPSK, phase
transitions occur only once every Ts = 2Tbs, and will be a maximum of
180 if there is a change in the value of both mI (t) and mQ (t). However,
in OQPSK signaling, bit transitions (and, hence, phase transitions) occur
every Tb s. Since the transition instants of mI (t) and mQ (t) are offset,
at any given time only one of the two bit streams can change values.
This implies that the maximum phase shift of the transmitted signal
at any given time is limited to 90. Hence, by switching phases more
frequently (i.e., every Tb s instead of 2Tb s) OQPSK signaling eliminates
180phase transitions.
Since 180 phase transitions have been eliminated, bandlimiting of
(i.e., pulse shaping) OQPSK signals does not cause the signal envelope
to go to zero. Obviously, there will be some amount of ISI caused by
the bandlimiting process, especially at the 90 phase transition points.
But the envelope variations are considerably less, and hence hardlimiting
or nonlinear amplification of OQPSK signals does not regenerate
the high frequency sidelobes as much as in QPSK. Thus, spectral
occupancy is significantly reduced, while permitting more efficient RF
amplification.
The spectrum of an OQPSK signal is identical to that of a QPSK
signal, hence both signals occupy the same bandwidth. The staggered
alignment of the even and odd bit streams does not change the nature of
the spectrum. OQPSK retains its bandlimited nature even after nonlinear
amplification, and therefore is very attractive for mobile communication
systems where bandwidth efficiency and efficient nonlinear amplifiers
are critical for low power drain. Further, OQPSK signals also appear to
perform better than QPSK in the presence of phase jitter due to noisy
reference signals at the receiver [Chu87].
13. (b) (ii)
Figure 1 Block diagram of an MSK receiver

Figure 2 Power spectral density of a GMSK signal [from [Mur81]

IEEE]
Figure 3 Block diagram of a GMSK transmitter using direct FM

generation.
Figure 4 Block diagram of a GMSK receiver

14. (a) (i)
Figure 1 Graph of probability distributions of SNR =g threshold for M

branch selection diversity. The term represents the mean SNR on each
branch [from [Jak71] IEEE]
14. (a) (ii) Decision Feedback Equalization (DFE)
The basic idea behind decision feedback equalization is that once an
information symbol has been detected and decided upon, the ISI that it
induces on future symbols can be estimated and subtracted out before
detection of subsequent symbols [Pro89]. The DFE can be realized in
either the direct transversal form or as a lattice filter. The direct form
is shown in Figure 1. It consists of a feed forward filter (FFF) and a
feedback filter (FBF). The FBF is driven by decisions on the output of
the detector, and its coefficients can be adjusted to cancel the ISI on the
current symbol from past detected symbols. The equalizer has N1 + N2 +
1 taps in the feed forward filter and N3 taps in the feedback filter, and its
output can be expressed as:
N2 N3
d k =
n = N1
C n* y k n + Fi d k i (1)
i =1

Figure 1 Decision feedback equalizer (DFE)
where Cn* and yn are tap gains and the inputs, respectively, to the forward
filter, Fi* are tap gains for the feedback filter, and di(i < k) is the previous
decision made on the detected signal. That is, once d k is obtained using
Equation (1), dk is decided from it. Then, dk along with previous decisions
dk-1, dk-2, . are fed back into the equalizer, and d k +1 is obtained using
Equation (1).
The minimum mean squared error a DFE can achieve is [Pro89]
T p /T
N0
E[| e ( n )| 2 ]min = exp ln jwT dw (2)
2p | F (e )| + N 0
2
-p /T
It can be shown that the minimum MSE for a DFE in Equation (2) is
always smaller than that of an LTE in Equation (2) unless F (ejwT) is
a constant (i.e., when adaptive equalization is not needed) [Pro89]. If
there are nulls in |F (ejwT)|, a DFE has significantly smaller minimum
MSE than an LTE. Therefore, an LTE is well behaved when the channel
spectrum is comparatively flat, but if the channel is severely distorted or
exhibits nulls in the spectrum, the performance of an LTE deteriorates
and the mean squared error of a DFE is much better than a LTE. Also, an
LTE has difficulty equalizing a nonminimum phase channel, where the
strongest energy arrives after the first arriving signal component. Thus, a
DFE is more appropriate for severely distorted wireless channels.

The lattice implementation of the DFE is equivalent to a transversal

DFE having a feed forward filter of length N1 and a feedback filter of
length N2, where N1 > N2.
Another form of DFE proposed by Belfiore and Park [Bel79] is called
a predictive DFE, and is shown in Figure. It also consists of a feed
forward filter (FFF) as in the conventional DFE. However, the feedback
filter (FBF) is driven by an input sequence formed by the difference
of the output of the detector and the output of the feed forward filter.
Hence, the FBF here is called a noise predicator because it predicts
the noise and the residual ISI contained in the signal at the FFF output
and subtracts from it the detector output after some feedback delay. The
predictive DFE performs as well as the conventional DFE as the limit
in the number of taps in the FFF and the FBF approach infinity. The
FBF in the predictive DFE can also be realized as a lattice structure
[Zho90]. The RLS lattice algorithm can be used in this case to yield fast
convergence.
(b) (i) Block Codes and Finite Fields

Block codes are forward error correction (FEC) codes that enable
a limited number of errors to be detected and corrected without
retransmission. Block codes can be used to improve the performance of
a communications system when other means of improvement (such as
increasing transmitter power or using a more sophisticated demodulator)
are impractical.
In block codes, parity bits are added to blocks of message bits to make
codewords or code blocks. In a block encoder, k information bits are
encoded into n code bits. A total of nk redundant bits are added to the
k information bits for the purpose of detecting and correcting errors
[Lin83]. The block code is referred to as an (n, k) code, and the rate of
the code is defined as Rc = k/n and is equal to the rate of information
divided by the raw channel rate.
The ability of a block code to correct errors is a function of the code
distance. Many families of codes exist that provide varying degrees of
error protection [Cou93], [Hay94], [Lin83], [Skl93], and [Vit79].
Besides the code rate, other important parameters are the code distance
and the weight of particular codewords. These are defined below.
Distance of a Code The distance between two codewords is the
number of elements in which two codewords Ci and Cj differ
N
d (C i , C j ) = C i ,l C j ,l (mod ulo q ) (1)
l =1

where d is the distance between the codewords and q is the total number
of possible values of Ci and Cj. The length of each codeword is N
elements or characters. If the code used is binary, the distance is known
as the Hamming distance. The minimum distance dmin is the smallest
distance for the given codeword set and is given as
dmin = Min{d (Ci , C j )}(2)
Weight of a Code The weight of a codeword of length N is given by

the number of nonzero elements in the codeword. For a binary code, the
weight is basically the number of 1s in the codeword and is given as
N
w (C i ) = C i ,l (3)
l =1
Properties of Block Codes

Linearity Suppose Ci and Cj are two codewords in an (n, k) block
code. Let a1and a2 be any two elements selected from the alphabet. Then
the code is said to be linear if and only if a1C1 + a2C2 is also a code
word. A linear code must contain the all-zero code word. Consequently,
a constant-weight code is nonlinear.
Systematic A systematic code is one in which the parity bits are
appended to the end of the information bits. For an (n, k) code, the
first k bits are identical to the information bits, and the remaining n-k
bits of each code word are linear combinations of the k information
bits.
Cyclic Cyclic codes are a subset of the class of linear codes which
satisfy the following cyclic shift property: If C = [Cn-1, Cn-2, , C0] is
a codeword of a cyclic code, then[Cn-2, Cn-3, , C0, Cn-1], obtained by
a cyclic shift of the elements of C, is also a code word. That is, all cyclic
shifts of C are code words. As a consequence of the cyclic property, the
codes possess a considerable amount of structure which can be exploited
to greatly simplify the encoding and decoding operations.
14. (b) (ii) Refer Q. no. 14 (b) from Nov/Dec 2013
15. (a)The number of simultaneous users that can be accommodated in

GSM is given as
25 MHz
N = = 1000
( 200 KHz ) /8
Thus, GSM can accommodate 1000 simultaneous users.

1
(a) The time duration of a bit, Tb = = 3.692 s.
270.833kbps
(b) The time duration of a slot, Tslot = 156.25 Tb = 0.577 ms.
(c) The time duration of a frame, T = 8 Tslot = 4.615 ms.
f
(d)A user has to wait 4.615 ms, the arrival time of a new frame, for its
next transmission
A time slot has 6 + 8.25 + 26 + 2(58) = 156.25 bits
(a) Number of overhead bits, boh = 8(6) + 8(8.25) + 8(26) = 322 bits
(b) Number of bits/frame = 8 156.25 = 1250 bits/frame
(c) Frame rate: 270.833 kbps/1250 bits/frame = 216.66 frame/sec
(d) Time duration of a slot = 156.25 1/270.833 kbps = 576.92 ms
(e) Frame efficiency = hf = [1 (322/1250)] = 74.24%
Spread Spectrum Multiple Access

Spread Spectrum Multiple Access (SSMA) uses signals which have a
transmission bandwidth that is several orders of magnitude greater than
the minimum required RF bandwidth. A pseudo-noise (PN) sequence
(discussed in Chapter 6) converts a narrowband signal to a wideband
noise-like signal before transmission. SSMA also provides immunity to
multipath interference and robust multiple access capability. 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. It is exactly this situation that is of interest
to wireless system designers. There are two main types of spread spectrum
multiple access techniques; frequency hopped Multiple Access (FH) and
direct sequence multiple access (DS). Direct sequence multiple access is
also called code division multiple access (CDMA).
Frequency Hopped Multiple Access (FHMA)

Frequency Hopped Multiple Access (FHMA) is a digital multiple
access system in which the carrier frequencies of the individual users
are varied in a pseudorandom fashion within a wideband channel.
Figure 1 illustrates how FHMA allows multiple users to simultaneously
occupy the same spectrum at the same time, where each user dwells at
a specific narrowband channel at a particular instance of time, based on
the particular PN code of the user. The digital data of each user is broken
into uniform sized bursts which are transmitted on different channels
within the allocated spectrum band. The instantaneous bandwidth
of any one transmission burst is much smaller than the total spread
bandwidth. The pseudorandom change of the channel frequencies of the

Figure 1 Spread spectrum multiple access in which each channel is

assigned a unique PN code which is orthogonal or approximately orthogonal
to PN codes used by other users
user randomizes the occupancy of a specific channel at any given time,

thereby allowing for multiple access over a wide range of frequencies.
In the FH receiver, a locally generated PN code is used to synchronize
the receivers instantaneous frequency with that of the transmitter. At
any given point in time, a frequency hopped signal only occupies a
single, relatively narrow channel since narrowband FM or FSK is used.
The difference between FHMA and a traditional FDMA system is that
the frequency hopped signal changes channels at rapid intervals. If the
rate of change of the carrier frequency is greater than the symbol rate,
then the system is referred to as a fast frequency hopping system. If the
channel changes at a rate less than or equal to the symbol rate, it is called
slow frequency hopping. A fast frequency hopper may thus be thought
of as an FDMA system which employs frequency diversity. FHMA
systems often employ energy efficient constant envelope modulation.
Inexpensive receivers may be built to provide noncoherent detection of
FHMA. This implies that linearity is not an issue, and the power of
multiple users at the receiver does not degrade FHMA performance.
A frequency hopped system provides a level of security, especially
when a large number of channels are used, since an unintended (or an
intercepting) receiver that does not know the pseudorandom sequence of
frequency slots must retune rapidly to search for the signal it wishes to

intercept. In addition, the FH signal is somewhat immune to fading, since

error control coding and interleaving can be used to protect the frequency
hopped signal against deep fades which may occasionally occur during
the hopping sequence. Error control coding and interleaving can also be
combined to guard against erasures which can occur when two or more
users transmit on the same channel at the same time. Bluetooth and
HomeRF wireless technologies have adopted FHMA for power efficiency
and low cost implementation.


NOVEMBER/DECEMBER 2011
(Unsolved Question Paper)
Seventh Semester
1. Differentiate Cellular telephony and Cordless telephony.
2. When does a WLAN become a personal Area Network (PAN)?
3. Compute the Rayleigh distance of a square Antenna with 20 dB gain.
4. List out any two properties of wideband channel.
5. Draw the mathematical link model for analysis of modulation schemes.
6. What is OQPSK?
7. Mention any four common methods of micro diversity.
8. Define Hamming distance and Euclidean distance between two codes.
9. Discuss the principle of OFDM modulation scheme.
10. Give three important functional blocks of GSM system.
11. (a) (i) Compare and contrast wired and wireless communication. (8)
(ii)Discuss briefly about the requirements of services for a
wireless system. (8)
WC_Question_Nov-Dec 2011.indd 3 7/26/2012 4:38:11 PM

Or
(b) (i)Discuss in detail the constructive and destructive
interference. (8)
(ii)Explain how inter symbol interference is caused and how
it is eliminated. (8)
12. (a) (i)Describe any two methods of diffraction by multiple

screens. (8)
(ii) Discuss about ultra wide band channel. (8)
Or
(b) (i)Compare Coherence Bandwidth and Coherence time (8)

(ii)Discuss the mathematical formulation for narrowband
and wideband system, with relevant figures. (8)
13. (a)Compare the ratio of signal power to adjacent channel inter-

ference when using (i) raised cosine pulses (ii) root raised
cosine pulses with a = 0.5, when two considered signals have
center frequencies 0 and 1.25/T. (16)
Or
(b) (i)Discuss in detail any two demodulation techniques of

minimum shift keying method. (8)
(ii)Explain in detail about optimum receiver structure for
Non-coherent detection. (8)
14. (a) (i)Explain the Viterbi decoding scheme if the decoder input
sequence is 010 000 100 001 011 110 001. (16)
Or
(b) (i)With a neat block diagram discuss the structure of a deci-

sion feedback equalizer. (8)
(ii) Discuss linear predictive vocoder with block diagram. (8)

15. (a) (i)Explain the principle of direct sequence spread spectrum

technique. (8)
(ii)Discuss some methods to increase the capacity of wireless
communication system. (8)
Or
(b) (i) Explain in detail about the GSM logical channels. (8)
(ii) Explain the block diagram of IS-95 transmitter. (8)

APRIL/MAY 2011
Seventh Semester
1.What is the consequence of increased link performance in CDMA
systems?
2.Define signal to interface ratio in terms of distance between cells centres

and radius of cell in a cellular scenario.
3. How does microzone concept reduce hard off?
4.Differentiate fast fading and slow fading with respect to coherence time
and doppler spread.
5.Specify the type of model that can be used to predict path loss for GSM
frequency range.
6.What is the probability of error (BER) in QPSK modulation systems?

Comment on the performance.
7. Write the DPSK sequence for the data stream 110110.
8.Differentiate between PCM and DPCM with respect to error perfor-

mance, which source coding is suitable for speech signals.
9. What type of channel coding is used for QSM network?
10.Give the specification of ideal uplink and downlink characteristics of

W-CDMA.
WC_Question_Apr-May 2011.indd 6 7/26/2012 4:37:20 PM

Wireless Communication (Apr/May 2011) 3.7
11. (a)Explain the different technique to reduce CCI and ACI in a
seven cell reuse cellular system. Derive the signal to inter-
ference for the worst conditions. Also explain the effect of
pathloss on interference (16)
Or
(b) (i)Derive the probability of blocking in a trusted Lost Call
Cleared (LCC) and Last Call Delayed (LCD) systems. (6)
(ii)In a cellular system there are 12 clusters available with
120 cells with 10,000 subscribers. Each subcribers uses
the cell phone for one hour everyday. On an average 30
minutes of time is used during peak hours. Find the aver-
age and peak traffic for the entire cellular system and also
for one cell considering all callers are distributed evenly
over the system. (10)
12. (a) (i)What are the different indoor propagation models avail-
able for path loss calculation? Explain the log-distance
path loss model(s). (8)
(ii)Calculate the propagation loss for a ratio signal at 850
mHz. The transmitter antenna height is 30 m over a dis-
tance of 11 km. Use mobile propagation model and free
space model and compare them. (8)
Or
(b) (i)Explain the different statistical models available for mul-

tipath fading channels (8)
(ii) Explain the fading effects due to Doppler spread. (8)
13. (a)How does maximal ratio combining techniques combat deep

fades in wireless environment? Derive the expression for the
maximum gain of the transmitting antennas. (16)
Or
(b)Derive the BER for MSK with respect to symbol energy

and symbol time period. Why MSK is preferred in mobile

e nvironment. Draw the spectral characteristics of the MSK

modulation scheme. (16)
14. (a) (i)Derive the general expression that relates SNR due to
quantisation as a function of number of bits. (8)
(ii)Explain the different type of vocoders. (8)
Or
(b)Derive the radio capacity for FDMA in terms of path loss,

total allocated spectrum, channel Bandwidth and Carrier
to interface (CLP) ratio. Compare with capacity of cellular
CDMA. (16)
15. (a)Explain the functions of DECT. What are the various entities?
List the specifications for DECT Radio. (16)
Or
(b)What are the various GSM control channels? Explain the various
control channels by original a cellular call through the GSM net-
work. Draw the timing diagrams. (16)

NOVEMBER/DECEMBER 2010
Seventh Semester
1. Draw the frequency management allocation for GSM network.
2. What is meant by grade service?
3. What are the data rates for the cordless phone?
4.If S/T is a cellular network in 18 dB, what is the frequency reuse factor
and cluster size for a path loss component of 3?
5.What are the various methods for improving the capacity of cellular
networks?
6.Define large scale propagation?
7. Why is QPSK preferred for wireless communication?
8.What is the need for diversity in multipath propagation?
9.Find the number of channels available in a FDMA system, if the cel-

lular network is allocated 12.5 mHz for each simplex band, total band-
width is 12.5 mHz guard bandwidth is 10 kHz and channel bandwidth is
30 kHZ.
10. Draw the frame structure for the GSM network.

11. (a)Derive the tracking capacity in a cellular network which
employ cell splitting and cell sectoring techniques comes on
the improvement in CCI and ACI. (16)
Or
(b)Prove that the frequency reuse ratio is 3N . Where N is the
size of the cluster. Also derive the worst case S/I ratio as 17
dB for a reuse ratio of 4.6. (16)
12. (a)Derive the path loss for small scale propagation and large
scale propagation in a multipath wireless environment. What
is Doppler spread? (16)
Or
(b)Derive any one of the path loss models for outdoor environ-
ment and explain how it may be used to obtain the path loss
in urban and semi urban areas. (16)
13. (a)What are the different digital modulation techniques employed

in wireless communication? Derive the BER for BPSK digital
modulation scheme. (16)
Or
(b)What is the necessity of diversity techniques? Explain space

and time diversity and how it can reduce fading in a multipath
propagation. (16)
14. (a)Explain PCM and ADPCM and how it is utilized for encod-
ing and decoding in wireless networks. (16)
Or
(b)What are the multiple access techniques used in AMPS, GSM

and CDMA networks? Explain how capacity is improved
using special spectrum techniques. (10)

15. (a)Draw the architecture of the GSM and explain the different
air channels. (16)
Or
(b)Compare the operational features of WLL and Bluetooth. (16)

APRIL/MAY 2010
Seventh Semester
1. List any four advantages of third generation (3G) mobile networks.
2. Write down the formula for co-channel reuse ratio.
3. What is doppler shift?
4.What is meant by coherence bandwidth?
5.Name the four space diversity reception techniques?
6.Define absolute bandwidth.
7. What is meant by vocoders?
8.What is self-jamming in CDMA?
9. What is the purpose of SIM?
10.What are the five functional entities of a DECT system?
11. (a) (i) Explain in detail the various cellular components. (8)
(ii) Explain the fundamentals of Digital cellular systems. (8)
Or

Wireless Communication (Apr/May 2010) 3.13
(b) (i)Explain the process of operation of paging systems. (8)

(ii)Describe the steps involved in making a cellular telephone
call. (8)
12. (a)Derive the impulse response model of a multipath channel. (16)
Or
(b)Explain in detail about type of small scale fading. (16)
13. (a) (i) Write a note on RAKE Receiver, with block diagram. (10)
(ii) Explain frequency diversity and time diversity. (6)
Or
(b)Explain the concept of minimum shift keying and Gaussian

MSK. (16)
14. (a)Explain TDMA and discuss the time division multiple access
frame structure. (16)
Or
(b) (i)Draw the block diagram of RELP encoder. Explain the

function of coder. (6)
(ii)Enumerate the procedure for selecting the optimum exci-
tation signal. (10)
15. (a) (i)Discuss the features and services of GSM. (8)

(ii)Explain the GSM system architecture with neat sketch. (8)
Or
(b)Discuss the channels and cellular operation in advanced mobile

phone systems (AMPS). (16)


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Anna University

Solved Question Papers

B.E./B.Tech. 7th Semester

ECE_Semester-V_FM.indd iii 7/19/2012 10:32:36 AM

Published by Dorling Kindersley (India) Pvt. Ltd, licensees of Pearson Education in

M02_ECE_7th-SEMESTER_CH01(Nov-Dec-2013).indd 1 4/25/2014 8:17:09 PM

EEE_Sem-VI_Chennai_FM.indd iv 12/7/2012 6:40:43 PM

2. What is a multiple access technique?

3. State the difference between Narrowband and wideband systems.

4. Find the far-field distance for an antenna with maximum dimension of

6. What is fading and Doppler spread?

9. What is a PN sequence? Give its significance in spread spectrum

10. What is DECT?

M02_ECE_7th-SEMESTER_CH01(Nov-Dec-2013).indd 3 4/25/2014 8:17:09 PM

(b) Explain the principle of Cellular Networks and various types of

1 3. (a) (i) Explain the Nyquist criterion for ISI cancellation.

1 4. (a) Explain in detail about:

1 5. (a) Explain in detail about:

M02_ECE_7th-SEMESTER_CH01(Nov-Dec-2013).indd 4 4/25/2014 8:17:09 PM

3. Narrowband systems Here the available radio spectrum is divided

4. Operating frequency, f = 900 MHz

5. The error probability for GMSK is given by

M02_ECE_7th-SEMESTER_CH01(Nov-Dec-2013).indd 5 4/25/2014 8:17:10 PM

Doppler spread is a measure of the spectral broadening caused by the

7. Diversity is technique used to compensate for fading channel impairments

8. Equalization compensates for inter symbol interference (ISI) created by

9. The pseudo-noise (PN) sequence is the spreading codes which is a binary

10. The Digital European Cordless Telephone (DECT) is a universal cordless

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is quite sophisticated. Wide area paging systems consist of a network of

an airplane. This necessitates large transmitter powers (on the order of

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In first generation cordless telephone systems (manufactured in the

Figure 2 A cordless telephone system.

Cellular Telephone Systems

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a handoff enables a call to proceed uninterrupted when the user moves

Figure 3 A cellular system. The towers represent base stations which

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11. (b) Cellular Networks

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three of the six cell vertices (edge-excited cells). Normally, omnidirectional

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Soft Handoff: CDMA usessoft handoff. Soft handoff is beneficial

12. (a) (i) Factors Influencing Small-Scale Fading

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12. (a) (ii) Given:

12. (b) (i) The Three Basic Propagation Mechanisms

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Reflection occurs when a propagating electromagnetic wave impinges

sin(q B ) = 1/ 5 or, q B = sin 1 (1/ 5) = 26.56

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13. (a) (i) Nyquist Criterion for ISI Cancellation

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Figure 2Time waveforms of binary line codes: (a) unipolar NRZ;

where Ts is the symbol period, n is an integer, and K is a non-zero constant.

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and receiver to produce the desired Heff(f). Consider the impulse

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rectangular filter, satisfies the zero ISI condition. Mathematically, the

where Z(f) = Z(-f), and Z(f) = 0 for f f 0 1 / 2Ts . Expressed in

Figure 4 Transfer function of a Nyquist pulse-shaping filter at baseband.

13. (a) (ii) Raised Cosine Roll off Filter

1 3. (a) (i) Explain the Nyquist criterion for ISI cancellation.

13. (b) (ii)Performance of Digital Modulation in Slow Flat-Fading

r (t ) = a (t ) exp( jq (t ))s (t ) + n (t ) 0 t T (1)