# Lecture Notes on Mobile Communication

**A Curriculum Development Cell Project Under QIP, IIT Guwahati
**

Dr. Abhijit Mitra

Department of Electronics and Communication Engineering

Indian Institute of Technology Guwahati

Guwahati – 781039, India

November 2009

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Preface

It’s been many years that I’m teaching the subject “Mobile Communication”

(EC632) to the IIT Guwahati students and the current lecture notes intend to act

as a supplement to that course so that our students can have an access to this

course anytime. This course is mainly aimed toward senior year students of the

ECE discipline, and in particular, for the ﬁnal year BTech, ﬁrst year MTech and

PhD students. However, this does not necessarily imply that any other discipline

students can not study this course. Rather, they also should delve deeper into

this course since mobile communication is a familiar term to everyone nowadays.

Although the communication aspects of this subject depends on the fundamentals of

another interesting subject, communication engineering, I would strongly advocate

the engineering students to go through the same in order to grow up adequate

interest in this ﬁeld. In fact, the present lecture notes are designed in such a way

that even a non-ECE student also would get certain basic notions of this subject.

The entire lecture notes are broadly divided into 8 chapters, which, I consider to

be most rudimentary topics to know the basics of this subject. The advance level

topics are avoided intensionally so as to give space to the possibility of developing

another lecture note in that area. In fact, this area is so vast and changing so fast

over time, there is no limit of discussing the advanced level topics. The current focus

has been therefore to deal with those main topics which would give a senior student

suﬃcient exposure to this ﬁeld to carry out further study and/or research. Initially,

after dealing with the introductory concepts (i.e., what is mobile communication,

how a mobile call is made etc) and the evolution of mobile communication systems till

the present day status, the cellular engineering fundamentals are discussed at length

to make the students realize the importance of the practical engineering aspects of

this subject. Next, the diﬀerent kinds of mobile communication channels is taken

up and large scale path loss model as well as small scale fading eﬀects are dealt,

both with simulation and statistical approaches. To enhance the link performance

amidst the adverse channel conditions, the transmitter and receiver techniques are

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discussed next. It is further extended with three main signal processing techniques

at the receiver, namely, equalization, diversity and channel coding. Finally, diﬀerent

kinds of multiple access techniques are covered at length with the emphasis on how

several mobile communication techniques evolve via this. It should also be mentioned

that many ﬁgures in the lecture notes have been adopted from some standard text

books to keep the easy ﬂow of the understanding of the topics.

During the process of developing the lecture notes, I have received kind helps

from my friends, colleagues as well as my post graduate and doctoral students which

I must acknowledge at the onset. I’m fortunate to have a group of energetic students

who have helped me a lot. It is for them only I could ﬁnish this project, albeit a

bit late. My sincere acknowledgment should also go to my parents and my younger

brother who have nicely reciprocated my oblivion nature by their nourishing and

generous attitude toward me since my childhood.

Finally, about the satisfaction of the author. In general, an author becomes

happy if he/she sees that his/her creation could instill certain sparks in the reader’s

mind. The same is true for me too. Once Bertrand Russell said “Science may set

limits to knowledge, but should not set limits to imagination”. If the readers can

visualize the continuously changing technology in this ﬁeld after reading this lecture

notes and also can dream about a future career in the same, I’ll consider my en-

deavor to be successful. My best wishes to all the readers.

Abhijit Mitra

November 2009

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Contents

1 Introductory Concepts 1

1.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

1.2 Evolution of Mobile Radio Communications . . . . . . . . . . . . . . 1

1.3 Present Day Mobile Communication . . . . . . . . . . . . . . . . . . 3

1.4 Fundamental Techniques . . . . . . . . . . . . . . . . . . . . . . . . . 4

1.4.1 Radio Transmission Techniques . . . . . . . . . . . . . . . . . 5

1.5 How a Mobile Call is Actually Made? . . . . . . . . . . . . . . . . . 7

1.5.1 Cellular Concept . . . . . . . . . . . . . . . . . . . . . . . . . 7

1.5.2 Operational Channels . . . . . . . . . . . . . . . . . . . . . . 8

1.5.3 Making a Call . . . . . . . . . . . . . . . . . . . . . . . . . . 8

1.6 Future Trends . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10

1.7 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10

2 Modern Wireless Communication Systems 11

2.1 1G: First Generation Networks . . . . . . . . . . . . . . . . . . . . . 11

2.2 2G: Second Generation Networks . . . . . . . . . . . . . . . . . . . . 11

2.2.1 TDMA/FDD Standards . . . . . . . . . . . . . . . . . . . . . 12

2.2.2 CDMA/FDD Standard . . . . . . . . . . . . . . . . . . . . . 12

2.2.3 2.5G Mobile Networks . . . . . . . . . . . . . . . . . . . . . . 12

2.3 3G: Third Generation Networks . . . . . . . . . . . . . . . . . . . . . 13

2.3.1 3G Standards and Access Technologies . . . . . . . . . . . . . 14

2.3.2 3G W-CDMA (UMTS) . . . . . . . . . . . . . . . . . . . . . 14

2.3.3 3G CDMA2000 . . . . . . . . . . . . . . . . . . . . . . . . . . 16

2.3.4 3G TD-SCDMA . . . . . . . . . . . . . . . . . . . . . . . . . 18

2.4 Wireless Transmission Protocols . . . . . . . . . . . . . . . . . . . . 19

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2.4.1 Wireless Local Loop (WLL) and LMDS . . . . . . . . . . . . 19

2.4.2 Bluetooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

2.4.3 Wireless Local Area Networks (W-LAN) . . . . . . . . . . . . 20

2.4.4 WiMax . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

2.4.5 Zigbee . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

2.4.6 Wibree . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

2.5 Conclusion: Beyond 3G Networks . . . . . . . . . . . . . . . . . . . . 22

2.6 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

3 The Cellular Engineering Fundamentals 23

3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

3.2 What is a Cell? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

3.3 Frequency Reuse . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

3.4 Channel Assignment Strategies . . . . . . . . . . . . . . . . . . . . . 27

3.4.1 Fixed Channel Assignment (FCA) . . . . . . . . . . . . . . . 27

3.4.2 Dynamic Channel Assignment (DCA) . . . . . . . . . . . . . 27

3.5 Handoﬀ Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

3.5.1 Factors Inﬂuencing Handoﬀs . . . . . . . . . . . . . . . . . . 29

3.5.2 Handoﬀs In Diﬀerent Generations . . . . . . . . . . . . . . . 31

3.5.3 Handoﬀ Priority . . . . . . . . . . . . . . . . . . . . . . . . . 33

3.5.4 A Few Practical Problems in Handoﬀ Scenario . . . . . . . . 33

3.6 Interference & System Capacity . . . . . . . . . . . . . . . . . . . . . 34

3.6.1 Co-channel interference (CCI) . . . . . . . . . . . . . . . . . . 34

3.6.2 Adjacent Channel Interference (ACI) . . . . . . . . . . . . . . 37

3.7 Enhancing Capacity And Cell Coverage . . . . . . . . . . . . . . . . 38

3.7.1 The Key Trade-oﬀ . . . . . . . . . . . . . . . . . . . . . . . . 38

3.7.2 Cell-Splitting . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

3.7.3 Sectoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43

3.7.4 Microcell Zone Concept . . . . . . . . . . . . . . . . . . . . . 46

3.8 Trunked Radio System . . . . . . . . . . . . . . . . . . . . . . . . . . 47

3.9 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53

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4 Free Space Radio Wave Propagation 54

4.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54

4.2 Free Space Propagation Model . . . . . . . . . . . . . . . . . . . . . 55

4.3 Basic Methods of Propagation . . . . . . . . . . . . . . . . . . . . . . 57

4.3.1 Reﬂection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57

4.3.2 Diﬀraction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

4.3.3 Scattering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

4.4 Two Ray Reﬂection Model . . . . . . . . . . . . . . . . . . . . . . . . 59

4.5 Diﬀraction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63

4.5.1 Knife-Edge Diﬀraction Geometry . . . . . . . . . . . . . . . . 64

4.5.2 Fresnel Zones: the Concept of Diﬀraction Loss . . . . . . . . 66

4.5.3 Knife-edge diﬀraction model . . . . . . . . . . . . . . . . . . . 68

4.6 Link Budget Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . 69

4.6.1 Log-distance Path Loss Model . . . . . . . . . . . . . . . . . 69

4.6.2 Log Normal Shadowing . . . . . . . . . . . . . . . . . . . . . 70

4.7 Outdoor Propagation Models . . . . . . . . . . . . . . . . . . . . . . 70

4.7.1 Okumura Model . . . . . . . . . . . . . . . . . . . . . . . . . 70

4.7.2 Hata Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71

4.8 Indoor Propagation Models . . . . . . . . . . . . . . . . . . . . . . . 72

4.8.1 Partition Losses Inside a Floor (Intra-ﬂoor) . . . . . . . . . . 72

4.8.2 Partition Losses Between Floors (Inter-ﬂoor) . . . . . . . . . 73

4.8.3 Log-distance Path Loss Model . . . . . . . . . . . . . . . . . 73

4.9 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73

4.10 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73

5 Multipath Wave Propagation and Fading 75

5.1 Multipath Propagation . . . . . . . . . . . . . . . . . . . . . . . . . . 75

5.2 Multipath & Small-Scale Fading . . . . . . . . . . . . . . . . . . . . 75

5.2.1 Fading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76

5.2.2 Multipath Fading Eﬀects . . . . . . . . . . . . . . . . . . . . 76

5.2.3 Factors Inﬂuencing Fading . . . . . . . . . . . . . . . . . . . . 76

5.3 Types of Small-Scale Fading . . . . . . . . . . . . . . . . . . . . . . . 77

5.3.1 Fading Eﬀects due to Multipath Time Delay Spread . . . . . 77

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5.3.2 Fading Eﬀects due to Doppler Spread . . . . . . . . . . . . . 78

5.3.3 Doppler Shift . . . . . . . . . . . . . . . . . . . . . . . . . . . 79

5.3.4 Impulse Response Model of a Multipath Channel . . . . . . . 80

5.3.5 Relation Between Bandwidth and Received Power . . . . . . 82

5.3.6 Linear Time Varying Channels (LTV) . . . . . . . . . . . . . 84

5.3.7 Small-Scale Multipath Measurements . . . . . . . . . . . . . . 85

5.4 Multipath Channel Parameters . . . . . . . . . . . . . . . . . . . . . 87

5.4.1 Time Dispersion Parameters . . . . . . . . . . . . . . . . . . 87

5.4.2 Frequency Dispersion Parameters . . . . . . . . . . . . . . . . 89

5.5 Statistical models for multipath propagation . . . . . . . . . . . . . . 90

5.5.1 NLoS Propagation: Rayleigh Fading Model . . . . . . . . . . 91

5.5.2 LoS Propagation: Rician Fading Model . . . . . . . . . . . . 93

5.5.3 Generalized Model: Nakagami Distribution . . . . . . . . . . 94

5.5.4 Second Order Statistics . . . . . . . . . . . . . . . . . . . . . 95

5.6 Simulation of Rayleigh Fading Models . . . . . . . . . . . . . . . . . 96

5.6.1 Clarke’s Model: without Doppler Eﬀect . . . . . . . . . . . . 96

5.6.2 Clarke and Gans’ Model: with Doppler Eﬀect . . . . . . . . . 96

5.6.3 Rayleigh Simulator with Wide Range of Channel Conditions 97

5.6.4 Two-Ray Rayleigh Faded Model . . . . . . . . . . . . . . . . 97

5.6.5 Saleh and Valenzuela Indoor Statistical Model . . . . . . . . 98

5.6.6 SIRCIM/SMRCIM Indoor/Outdoor Statistical Models . . . . 98

5.7 Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

5.8 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

6 Transmitter and Receiver Techniques 101

6.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101

6.2 Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101

6.2.1 Choice of Modulation Scheme . . . . . . . . . . . . . . . . . . 102

6.2.2 Advantages of Modulation . . . . . . . . . . . . . . . . . . . . 102

6.2.3 Linear and Non-linear Modulation Techniques . . . . . . . . . 103

6.2.4 Amplitude and Angle Modulation . . . . . . . . . . . . . . . 104

6.2.5 Analog and Digital Modulation Techniques . . . . . . . . . . 104

6.3 Signal Space Representation of Digitally Modulated Signals . . . . . 104

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6.4 Complex Representation of Linear Modulated Signals and Band Pass

Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105

6.5 Linear Modulation Techniques . . . . . . . . . . . . . . . . . . . . . 106

6.5.1 Amplitude Modulation (DSBSC) . . . . . . . . . . . . . . . . 106

6.5.2 BPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107

6.5.3 QPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107

6.5.4 Oﬀset-QPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

6.5.5 π/4 DQPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . 110

6.6 Line Coding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110

6.7 Pulse Shaping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111

6.7.1 Nyquist pulse shaping . . . . . . . . . . . . . . . . . . . . . . 112

6.7.2 Raised Cosine Roll-Oﬀ Filtering . . . . . . . . . . . . . . . . 113

6.7.3 Realization of Pulse Shaping Filters . . . . . . . . . . . . . . 113

6.8 Nonlinear Modulation Techniques . . . . . . . . . . . . . . . . . . . . 114

6.8.1 Angle Modulation (FM and PM) . . . . . . . . . . . . . . . . 114

6.8.2 BFSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116

6.9 GMSK Scheme . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118

6.10 GMSK Generator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

6.11 Two Practical Issues of Concern . . . . . . . . . . . . . . . . . . . . 121

6.11.1 Inter Channel Interference . . . . . . . . . . . . . . . . . . . . 121

6.11.2 Power Ampliﬁer Nonlinearity . . . . . . . . . . . . . . . . . . 122

6.12 Receiver performance in multipath channels . . . . . . . . . . . . . . 122

6.12.1 Bit Error Rate and Symbol Error Rate . . . . . . . . . . . . . 123

6.13 Example of a Multicarrier Modulation: OFDM . . . . . . . . . . . . 123

6.13.1 Orthogonality of Signals . . . . . . . . . . . . . . . . . . . . . 125

6.13.2 Mathematical Description of OFDM . . . . . . . . . . . . . . 125

6.14 Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127

6.15 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128

7 Techniques to Mitigate Fading Eﬀects 129

7.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129

7.2 Equalization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130

7.2.1 A Mathematical Framework . . . . . . . . . . . . . . . . . . . 131

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7.2.2 Zero Forcing Equalization . . . . . . . . . . . . . . . . . . . . 132

7.2.3 A Generic Adaptive Equalizer . . . . . . . . . . . . . . . . . . 132

7.2.4 Choice of Algorithms for Adaptive Equalization . . . . . . . . 134

7.3 Diversity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136

7.3.1 Diﬀerent Types of Diversity . . . . . . . . . . . . . . . . . . . 137

7.4 Channel Coding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143

7.4.1 Shannon’s Channel Capacity Theorem . . . . . . . . . . . . . 143

7.4.2 Block Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144

7.4.3 Convolutional Codes . . . . . . . . . . . . . . . . . . . . . . . 152

7.4.4 Concatenated Codes . . . . . . . . . . . . . . . . . . . . . . . 155

7.5 Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156

7.6 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156

8 Multiple Access Techniques 157

8.1 Multiple Access Techniques for Wireless Communication . . . . . . . 157

8.1.1 Narrowband Systems . . . . . . . . . . . . . . . . . . . . . . . 158

8.1.2 Wideband Systems . . . . . . . . . . . . . . . . . . . . . . . . 158

8.2 Frequency Division Multiple Access . . . . . . . . . . . . . . . . . . . 159

8.2.1 FDMA/FDD in AMPS . . . . . . . . . . . . . . . . . . . . . 160

8.2.2 FDMA/TDD in CT2 . . . . . . . . . . . . . . . . . . . . . . . 160

8.2.3 FDMA and Near-Far Problem . . . . . . . . . . . . . . . . . 160

8.3 Time Division Multiple Access . . . . . . . . . . . . . . . . . . . . . 161

8.3.1 TDMA/FDD in GSM . . . . . . . . . . . . . . . . . . . . . . 161

8.3.2 TDMA/TDD in DECT . . . . . . . . . . . . . . . . . . . . . 162

8.4 Spread Spectrum Multiple Access . . . . . . . . . . . . . . . . . . . . 163

8.4.1 Frequency Hopped Multiple Access (FHMA) . . . . . . . . . 163

8.4.2 Code Division Multiple Access . . . . . . . . . . . . . . . . . 163

8.4.3 CDMA and Self-interference Problem . . . . . . . . . . . . . 164

8.4.4 CDMA and Near-Far Problem . . . . . . . . . . . . . . . . . 165

8.4.5 Hybrid Spread Spectrum Techniques . . . . . . . . . . . . . . 165

8.5 Space Division Multiple Access . . . . . . . . . . . . . . . . . . . . . 166

8.6 Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166

8.7 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167

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List of Figures

1.1 The worldwide mobile subscriber chart. . . . . . . . . . . . . . . . . 2

1.2 Basic mobile communication structure. . . . . . . . . . . . . . . . . . 3

1.3 The basic radio transmission techniques: (a) simplex, (b) half duplex

and (c) full duplex. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

1.4 (a) Frequency division duplexing and (b) time division duplexing. . . 6

1.5 Basic Cellular Structure. . . . . . . . . . . . . . . . . . . . . . . . . . 7

2.1 Data transmission with Bluetooth. . . . . . . . . . . . . . . . . . . . 20

3.1 Footprint of cells showing the overlaps and gaps. . . . . . . . . . . . 24

3.2 Frequency reuse technique of a cellular system. . . . . . . . . . . . . 25

3.3 Handoﬀ scenario at two adjacent cell boundary. . . . . . . . . . . . . 29

3.4 Handoﬀ process associated with power levels, when the user is going

from i-th cell to j-th cell. . . . . . . . . . . . . . . . . . . . . . . . . . 30

3.5 Handoﬀ process with a rectangular cell inclined at an angle θ. . . . . 31

3.6 First tier of co-channel interfering cells . . . . . . . . . . . . . . . . . 37

3.7 Splitting of congested seven-cell clusters. . . . . . . . . . . . . . . . . 41

3.8 A cell divided into three 120

o

sectors. . . . . . . . . . . . . . . . . . 43

3.9 A seven-cell cluster with 60

o

sectors. . . . . . . . . . . . . . . . . . . 44

3.10 The micro-cell zone concept. . . . . . . . . . . . . . . . . . . . . . . . 47

3.11 The buﬀerless J-channel trunked radio system. . . . . . . . . . . . . 49

3.12 Discrete-time Markov chain for the M/M/J/J trunked radio system. 49

4.1 Free space propagation model, showing the near and far ﬁelds. . . . 55

4.2 Two-ray reﬂection model. . . . . . . . . . . . . . . . . . . . . . . . . 59

4.3 Phasor diagram of electric ﬁelds. . . . . . . . . . . . . . . . . . . . . 61

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4.4 Equivalent phasor diagram of Figure 4.3. . . . . . . . . . . . . . . . . 61

4.5 Huygen’s secondary wavelets. . . . . . . . . . . . . . . . . . . . . . . 64

4.6 Diﬀraction through a sharp edge. . . . . . . . . . . . . . . . . . . . . 65

4.7 Fresnel zones. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66

4.8 Knife-edge Diﬀraction Model . . . . . . . . . . . . . . . . . . . . . . 68

5.1 Illustration of Doppler eﬀect. . . . . . . . . . . . . . . . . . . . . . . 79

5.2 A generic transmitted pulsed RF signal. . . . . . . . . . . . . . . . . 83

5.3 Relationship among diﬀerent channel functions. . . . . . . . . . . . . 85

5.4 Direct RF pulsed channel IR measurement. . . . . . . . . . . . . . . 86

5.5 Frequency domain channel IR measurement. . . . . . . . . . . . . . . 87

5.6 Two ray NLoS multipath, resulting in Rayleigh fading. . . . . . . . . 91

5.7 Rayleigh probability density function. . . . . . . . . . . . . . . . . . 93

5.8 Ricean probability density function. . . . . . . . . . . . . . . . . . . 93

5.9 Nakagami probability density function. . . . . . . . . . . . . . . . . . 94

5.10 Schematic representation of level crossing with a Rayleigh fading en-

velope at 10 Hz Doppler spread. . . . . . . . . . . . . . . . . . . . . 95

5.11 Clarke and Gan’s model for Rayleigh fading generation using quadra-

ture amplitude modulation with (a) RF Doppler ﬁlter, and, (b) base-

band Doppler ﬁlter. . . . . . . . . . . . . . . . . . . . . . . . . . . . 97

5.12 Rayleigh fading model to get both the ﬂat and frequency selective

channel conditions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98

5.13 Two-ray Rayleigh fading model. . . . . . . . . . . . . . . . . . . . . . 99

6.1 BPSK signal constellation. . . . . . . . . . . . . . . . . . . . . . . . . 107

6.2 QPSK signal constellation. . . . . . . . . . . . . . . . . . . . . . . . . 108

6.3 QPSK transmitter. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

6.4 DQPSK constellation diagram. . . . . . . . . . . . . . . . . . . . . . 109

6.5 Scematic of the line coding techniques. . . . . . . . . . . . . . . . . . 111

6.6 Rectangular Pulse . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112

6.7 Raised Cosine Pulse. . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

6.8 Phase tree of 1101000 CPFSK sequence. . . . . . . . . . . . . . . . . 118

6.9 Spectrum of MSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118

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6.10 GMSK generation scheme. . . . . . . . . . . . . . . . . . . . . . . . . 119

6.11 A simple GMSK receiver. . . . . . . . . . . . . . . . . . . . . . . . . 120

6.12 Spectrum of GMSK scheme. . . . . . . . . . . . . . . . . . . . . . . . 121

6.13 OFDM Transmitter and Receiver Block Diagram. . . . . . . . . . . . 127

7.1 A general framework of fading eﬀects and their mitigation techniques. 130

7.2 A generic adaptive equalizer. . . . . . . . . . . . . . . . . . . . . . . 133

7.3 Receiver selection diversity, with M receivers. . . . . . . . . . . . . . 137

7.4 Maximal ratio combining technique. . . . . . . . . . . . . . . . . . . 140

7.5 RAKE receiver. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142

7.6 A convolutional encoder with n=2 and k=1. . . . . . . . . . . . . . . 153

7.7 State diagram representation of a convolutional encoder. . . . . . . . 153

7.8 Tree diagram representation of a convolutional encoder. . . . . . . . 154

7.9 Trellis diagram of a convolutional encoder. . . . . . . . . . . . . . . . 154

7.10 Block diagram of a turbo encoder. . . . . . . . . . . . . . . . . . . . 155

8.1 The basic concept of FDMA. . . . . . . . . . . . . . . . . . . . . . . 159

8.2 The basic concept of TDMA. . . . . . . . . . . . . . . . . . . . . . . 162

8.3 The basic concept of CDMA. . . . . . . . . . . . . . . . . . . . . . . 164

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List of Tables

2.1 Main WCDMA parameters . . . . . . . . . . . . . . . . . . . . . . . 16

7.1 Finite ﬁeld elements for US-CDPD . . . . . . . . . . . . . . . . . . . 152

8.1 MA techniques in diﬀerent wireless communication systems . . . . . 158

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Chapter 1

Introductory Concepts

1.1 Introduction

Communication is one of the integral parts of science that has always been a focus

point for exchanging information among parties at locations physically apart. After

its discovery, telephones have replaced the telegrams and letters. Similarly, the term

‘mobile’ has completely revolutionized the communication by opening up innovative

applications that are limited to one’s imagination. Today, mobile communication

has become the backbone of the society. All the mobile system technologies have

improved the way of living. Its main plus point is that it has privileged a common

mass of society. In this chapter, the evolution as well as the fundamental techniques

of the mobile communication is discussed.

1.2 Evolution of Mobile Radio Communications

The ﬁrst wireline telephone system was introduced in the year 1877. Mobile com-

munication systems as early as 1934 were based on Amplitude Modulation (AM)

schemes and only certain public organizations maintained such systems. With the

demand for newer and better mobile radio communication systems during the World

War II and the development of Frequency Modulation (FM) technique by Edwin

Armstrong, the mobile radio communication systems began to witness many new

changes. Mobile telephone was introduced in the year 1946. However, during its

initial three and a half decades it found very less market penetration owing to high

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Figure 1.1: The worldwide mobile subscriber chart.

costs and numerous technological drawbacks. But with the development of the cel-

lular concept in the 1960s at the Bell Laboratories, mobile communications began to

be a promising ﬁeld of expanse which could serve wider populations. Initially, mobile

communication was restricted to certain oﬃcial users and the cellular concept was

never even dreamt of being made commercially available. Moreover, even the growth

in the cellular networks was very slow. However, with the development of newer and

better technologies starting from the 1970s and with the mobile users now connected

to the Public Switched Telephone Network (PSTN), there has been an astronomical

growth in the cellular radio and the personal communication systems. Advanced

Mobile Phone System (AMPS) was the ﬁrst U.S. cellular telephone system and it

was deployed in 1983. Wireless services have since then been experiencing a 50%

per year growth rate. The number of cellular telephone users grew from 25000 in

1984 to around 3 billion in the year 2007 and the demand rate is increasing day by

day. A schematic of the subscribers is shown in Fig. 1.1.

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Figure 1.2: Basic mobile communication structure.

1.3 Present Day Mobile Communication

Since the time of wireless telegraphy, radio communication has been used extensively.

Our society has been looking for acquiring mobility in communication since then.

Initially the mobile communication was limited between one pair of users on single

channel pair. The range of mobility was deﬁned by the transmitter power, type of

antenna used and the frequency of operation. With the increase in the number of

users, accommodating them within the limited available frequency spectrum became

a major problem. To resolve this problem, the concept of cellular communication

was evolved. The present day cellular communication uses a basic unit called cell.

Each cell consists of small hexagonal area with a base station located at the center

of the cell which communicates with the user. To accommodate multiple users

Time Division multiple Access (TDMA), Code Division Multiple Access (CDMA),

Frequency Division Multiple Access (FDMA) and their hybrids are used. Numerous

mobile radio standards have been deployed at various places such as AMPS, PACS,

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Figure 1.3: The basic radio transmission techniques: (a) simplex, (b) half duplex

and (c) full duplex.

GSM, NTT, PHS and IS-95, each utilizing diﬀerent set of frequencies and allocating

diﬀerent number of users and channels.

1.4 Fundamental Techniques

By deﬁnition, mobile radio terminal means any radio terminal that could be moved

during its operation. Depending on the radio channel, there can be three diﬀer-

ent types of mobile communication. In general, however, a Mobile Station (MS)

or subscriber unit communicates to a ﬁxed Base Station (BS) which in turn com-

municates to the desired user at the other end. The MS consists of transceiver,

control circuitry, duplexer and an antenna while the BS consists of transceiver and

channel multiplexer along with antennas mounted on the tower. The BS are also

linked to a power source for the transmission of the radio signals for communication

and are connected to a ﬁxed backbone network. Figure 1.2 shows a basic mobile

communication with low power transmitters/receivers at the BS, the MS and also

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the Mobile Switching Center (MSC). The MSC is sometimes also called Mobile Tele-

phone Switching Oﬃce (MTSO). The radio signals emitted by the BS decay as the

signals travel away from it. A minimum amount of signal strength is needed in

order to be detected by the mobile stations or mobile sets which are the hand-held

personal units (portables) or those installed in the vehicles (mobiles). The region

over which the signal strength lies above such a threshold value is known as the

coverage area of a BS. The ﬁxed backbone network is a wired network that links all

the base stations and also the landline and other telephone networks through wires.

1.4.1 Radio Transmission Techniques

Based on the type of channels being utilized, mobile radio transmission systems may

be classiﬁed as the following three categories which is also shown in Fig. 1.3:

• Simplex System: Simplex systems utilize simplex channels i.e., the commu-

nication is unidirectional. The ﬁrst user can communicate with the second

user. However, the second user cannot communicate with the ﬁrst user. One

example of such a system is a pager.

• Half Duplex System: Half duplex radio systems that use half duplex radio

channels allow for non-simultaneous bidirectional communication. The ﬁrst

user can communicate with the second user but the second user can commu-

nicate to the ﬁrst user only after the ﬁrst user has ﬁnished his conversation.

At a time, the user can only transmit or receive information. A walkie-talkie

is an example of a half duplex system which uses ‘push to talk’ and ‘release to

listen’ type of switches.

• Full Duplex System: Full duplex systems allow two way simultaneous com-

munications. Both the users can communicate to each other simultaneously.

This can be done by providing two simultaneous but separate channels to both

the users. This is possible by one of the two following methods.

– Frequency Division Duplexing (FDD): FDD supports two-way radio

communication by using two distinct radio channels. One frequency chan-

nel is transmitted downstream from the BS to the MS (forward channel).

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Figure 1.4: (a) Frequency division duplexing and (b) time division duplexing.

A second frequency is used in the upstream direction and supports trans-

mission from the MS to the BS (reverse channel). Because of the pairing of

frequencies, simultaneous transmission in both directions is possible. To

mitigate self-interference between upstream and downstream transmis-

sions, a minimum amount of frequency separation must be maintained

between the frequency pair, as shown in Fig. 1.4.

– Time Division Duplexing (TDD): TDD uses a single frequency band

to transmit signals in both the downstream and upstream directions.

TDD operates by toggling transmission directions over a time interval.

This toggling takes place very rapidly and is imperceptible to the user.

A full duplex mobile system can further be subdivided into two category: a

single MS for a dedicated BS, and many MS for a single BS. Cordless telephone

systems are full duplex communication systems that use radio to connect to a

portable handset to a single dedicated BS, which is then connected to a dedi-

cated telephone line with a speciﬁc telephone number on the Public Switched

Telephone Network (PSTN). A mobile system, in general, on the other hand,

is the example of the second category of a full duplex mobile system where

many users connect among themselves via a single BS.

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Figure 1.5: Basic Cellular Structure.

1.5 How a Mobile Call is Actually Made?

In order to know how a mobile call is made, we should ﬁrst look into the basics of

cellular concept and main operational channels involved in making a call. These are

given below.

1.5.1 Cellular Concept

Cellular telephone systems must accommodate a large number of users over a large

geographic area with limited frequency spectrum, i.e., with limited number of chan-

nels. If a single transmitter/ receiver is used with only a single base station, then

suﬃcient amount of power may not be present at a huge distance from the BS.

For a large geographic coverage area, a high powered transmitter therefore has to

be used. But a high power radio transmitter causes harm to environment. Mobile

communication thus calls for replacing the high power transmitters by low power

transmitters by dividing the coverage area into small segments, called cells. Each

cell uses a certain number of the available channels and a group of adjacent cells

together use all the available channels. Such a group is called a cluster. This cluster

can repeat itself and hence the same set of channels can be used again and again.

Each cell has a low power transmitter with a coverage area equal to the area of the

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cell. This technique of substituting a single high powered transmitter by several low

powered transmitters to support many users is the backbone of the cellular concept.

1.5.2 Operational Channels

In each cell, there are four types of channels that take active part during a mobile

call. These are:

• Forward Voice Channel (FVC): This channel is used for the voice trans-

mission from the BS to the MS.

• Reverse Voice Channel (RVC): This is used for the voice transmission

from the MS to the BS.

• Forward Control Channel (FCC): Control channels are generally used

for controlling the activity of the call, i.e., they are used for setting up calls

and to divert the call to unused voice channels. Hence these are also called

setup channels. These channels transmit and receive call initiation and service

request messages. The FCC is used for control signaling purpose from the BS

to MS.

• Reverse Control Channel (RCC): This is used for the call control purpose

from the MS to the BS. Control channels are usually monitored by mobiles.

1.5.3 Making a Call

When a mobile is idle, i.e., it is not experiencing the process of a call, then it searches

all the FCCs to determine the one with the highest signal strength. The mobile

then monitors this particular FCC. However, when the signal strength falls below

a particular threshold that is insuﬃcient for a call to take place, the mobile again

searches all the FCCs for the one with the highest signal strength. For a particular

country or continent, the control channels will be the same. So all mobiles in that

country or continent will search among the same set of control channels. However,

when a mobile moves to a diﬀerent country or continent, then the control channels

for that particular location will be diﬀerent and hence the mobile will not work.

Each mobile has a mobile identiﬁcation number (MIN). When a user wants to

make a call, he sends a call request to the MSC on the reverse control channel. He

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also sends the MIN of the person to whom the call has to be made. The MSC then

sends this MIN to all the base stations. The base station transmits this MIN and all

the mobiles within the coverage area of that base station receive the MIN and match

it with their own. If the MIN matches with a particular MS, that mobile sends an

acknowledgment to the BS. The BS then informs the MSC that the mobile is within

its coverage area. The MSC then instructs the base station to access speciﬁc unused

voice channel pair. The base station then sends a message to the mobile to move to

the particular channels and it also sends a signal to the mobile for ringing.

In order to maintain the quality of the call, the MSC adjusts the transmitted

power of the mobile which is usually expressed in dB or dBm. When a mobile moves

from the coverage area of one base station to the coverage area of another base sta-

tion i.e., from one cell to another cell, then the signal strength of the initial base

station may not be suﬃcient to continue the call in progress. So the call has to be

transferred to the other base station. This is called handoﬀ. In such cases, in order

to maintain the call, the MSC transfers the call to one of the unused voice channels

of the new base station or it transfers the control of the current voice channels to

the new base station.

Ex. 1: Suppose a mobile unit transmits 10 W power at a certain place. Express this

power in terms of dBm.

Solution: Usually, 1 mW power developed over a 100 Ω load is equivalently called

0 dBm power. 1 W is equivalent to 0 dB, i.e., 10 log

10

(1W) = 0dB. Thus,

1W = 10

3

mW = 30dBm = 0dB. This means, xdB = (x + 30)dBm. Hence,

10W = 10 log

10

(10W) = 10dB = 40dBm.

Ex. 2: Among a pager, a cordless phone and a mobile phone, which device would

have the (i) shortest, and, (ii) longest battery life? Justify.

Solution: The ‘pager’ would have the longest and the ‘mobile phone’ would have the

shortest battery life. (justiﬁcation is left on the readers)

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1.6 Future Trends

Tremendous changes are occurring in the area of mobile radio communications, so

much so that the mobile phone of yesterday is rapidly turning into a sophisticated

mobile device capable of more applications than PCs were capable of only a few

years ago. Rapid development of the Internet with its new services and applications

has created fresh challenges for the further development of mobile communication

systems. Further enhancements in modulation schemes will soon increase the In-

ternet access rates on the mobile from current 1.8 Mbps to greater than 10 Mbps.

Bluetooth is rapidly becoming a common feature in mobiles for local connections.

The mobile communication has provided global connectivity to the people at

a lower cost due to advances in the technology and also because of the growing

competition among the service providers. We would review certain major features

as well as standards of the mobile communication till the present day technology in

the next chapter.

1.7 References

1. T. S. Rappaport, Wireless Communications: Principles and Practice, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

2. K. Feher, Wireless Digital Communications: Modulation and Spread Spectrum

Applications. Upper Saddle River, NJ: Prentice Hall, 1995.

3. J. G. Proakis, Digital Communications, 4th ed. NY: McGraw Hill, 2000.

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Chapter 2

Modern Wireless

Communication Systems

At the initial phase, mobile communication was restricted to certain oﬃcial users and

the cellular concept was never even dreamt of being made commercially available.

Moreover, even the growth in the cellular networks was very slow. However, with

the development of newer and better technologies starting from the 1970s and with

the mobile users now connected to the PSTN, there has been a remarkable growth

in the cellular radio. However, the spread of mobile communication was very fast

in the 1990s when the government throughout the world provided radio spectrum

licenses for Personal Communication Service (PCS) in 1.8 - 2 GHz frequency band.

2.1 1G: First Generation Networks

The ﬁrst mobile phone system in the market was AMPS. It was the ﬁrst U.S. cellular

telephone system, deployed in Chicago in 1983. The main technology of this ﬁrst

generation mobile system was FDMA/FDD and analog FM.

2.2 2G: Second Generation Networks

Digital modulation formats were introduced in this generation with the main tech-

nology as TDMA/FDD and CDMA/FDD. The 2G systems introduced three popular

TDMA standards and one popular CDMA standard in the market. These are as

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follows:

2.2.1 TDMA/FDD Standards

(a) Global System for Mobile (GSM): The GSM standard, introduced by Groupe

Special Mobile, was aimed at designing a uniform pan-European mobile system. It

was the ﬁrst fully digital system utilizing the 900 MHz frequency band. The initial

GSM had 200 KHz radio channels, 8 full-rate or 16 half-rate TDMA channels per

carrier, encryption of speech, low speed data services and support for SMS for which

it gained quick popularity.

(b) Interim Standard 136 (IS-136): It was popularly known as North American

Digital Cellular (NADC) system. In this system, there were 3 full-rate TDMA users

over each 30 KHz channel. The need of this system was mainly to increase the

capacity over the earlier analog (AMPS) system.

(c) Paciﬁc Digital Cellular (PDC): This standard was developed as the counter-

part of NADC in Japan. The main advantage of this standard was its low transmis-

sion bit rate which led to its better spectrum utilization.

2.2.2 CDMA/FDD Standard

Interim Standard 95 (IS-95): The IS-95 standard, also popularly known as CDMA-

One, uses 64 orthogonally coded users and codewords are transmitted simultaneously

on each of 1.25 MHz channels. Certain services that have been standardized as a

part of IS-95 standard are: short messaging service, slotted paging, over-the-air

activation (meaning the mobile can be activated by the service provider without

any third party intervention), enhanced mobile station identities etc.

2.2.3 2.5G Mobile Networks

In an eﬀort to retroﬁt the 2G standards for compatibility with increased throughput

rates to support modern Internet application, the new data centric standards were

developed to be overlaid on 2G standards and this is known as 2.5G standard.

Here, the main upgradation techniques are:

• supporting higher data rate transmission for web browsing

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• supporting e-mail traﬃc

• enabling location-based mobile service

2.5G networks also brought into the market some popular application, a few of

which are: Wireless Application Protocol (WAP), General Packet Radio Service

(GPRS), High Speed Circuit Switched Dada (HSCSD), Enhanced Data rates for

GSM Evolution (EDGE) etc.

2.3 3G: Third Generation Networks

3G is the third generation of mobile phone standards and technology, supersed-

ing 2.5G. It is based on the International Telecommunication Union (ITU) family

of standards under the International Mobile Telecommunications-2000 (IMT-2000).

ITU launched IMT-2000 program, which, together with the main industry and stan-

dardization bodies worldwide, targets to implement a global frequency band that

would support a single, ubiquitous wireless communication standard for all coun-

tries,to provide the framework for the deﬁnition of the 3G mobile systems.Several

radio access technologies have been accepted by ITU as part of the IMT-2000 frame-

work.

3G networks enable network operators to oﬀer users a wider range of more ad-

vanced services while achieving greater network capacity through improved spectral

eﬃciency. Services include wide-area wireless voice telephony, video calls, and broad-

band wireless data, all in a mobile environment. Additional features also include

HSPA data transmission capabilities able to deliver speeds up to 14.4Mbit/s on the

down link and 5.8Mbit/s on the uplink.

3G networks are wide area cellular telephone networks which evolved to incor-

porate high-speed internet access and video telephony. IMT-2000 deﬁnes a set of

technical requirements for the realization of such targets, which can be summarized

as follows:

• high data rates: 144 kbps in all environments and 2 Mbps in low-mobility and

indoor environments

• symmetrical and asymmetrical data transmission

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• circuit-switched and packet-switched-based services

• speech quality comparable to wire-line quality

• improved spectral eﬃciency

• several simultaneous services to end users for multimedia services

• seamless incorporation of second-generation cellular systems

• global roaming

• open architecture for the rapid introduction of new services and technology.

2.3.1 3G Standards and Access Technologies

As mentioned before, there are several diﬀerent radio access technologies deﬁned

within ITU, based on either CDMA or TDMA technology. An organization called

3rd Generation Partnership Project (3GPP) has continued that work by deﬁning a

mobile system that fulﬁlls the IMT-2000 standard. This system is called Universal

Mobile Telecommunications System (UMTS). After trying to establish a single 3G

standard, ITU ﬁnally approved a family of ﬁve 3G standards, which are part of the

3G framework known as IMT-2000:

• W-CDMA

• CDMA2000

• TD-SCDMA

Europe, Japan, and Asia have agreed upon a 3G standard called the Universal

Mobile Telecommunications System (UMTS), which is WCDMA operating at 2.1

GHz. UMTS and WCDMA are often used as synonyms. In the USA and other

parts of America, WCDMA will have to use another part of the radio spectrum.

2.3.2 3G W-CDMA (UMTS)

WCDMA is based on DS-CDMA (direct sequencecode division multiple access) tech-

nology in which user-information bits are spread over a wide bandwidth (much

larger than the information signal bandwidth) by multiplying the user data with

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the spreading code. The chip (symbol rate) rate of the spreading sequence is 3.84

Mcps, which, in the WCDMA system deployment is used together with the 5-MHz

carrier spacing. The processing gain term refers to the relationship between the

signal bandwidth and the information bandwidth. Thus, the name wideband is

derived to diﬀerentiate it from the 2G CDMA (IS-95), which has a chip rate of

1.2288 Mcps. In a CDMA system, all users are active at the same time on the same

frequency and are separated from each other with the use of user speciﬁc spreading

codes.

The wide carrier bandwidth of WCDMA allows supporting high user-data rates

and also has certain performance beneﬁts, such as increased multipath diversity.

The actual carrier spacing to be used by the operator may vary on a 200-kHz grid

between approximately 4.4 and 5 MHz, depending on spectrum arrangement and

the interference situation.

In WCDMA each user is allocated frames of 10 ms duration, during which the

user-data rate is kept constant. However, the data rate among the users can change

from frame to frame. This fast radio capacity allocation (or the limits for variation in

the uplink) is controlled and coordinated by the radio resource management (RRM)

functions in the network to achieve optimum throughput for packet data services

and to ensure suﬃcient quality of service (QoS) for circuit-switched users. WCDMA

supports two basic modes of operation: FDD and TDD. In the FDD mode, separate

5-MHz carrier frequencies with duplex spacing are used for the uplink and downlink,

respectively, whereas in TDD only one 5-MHz carrier is time shared between the up-

link and the downlink. WCDMA uses coherent detection based on the pilot symbols

and/or common pilot. WCDMA allows many performance- enhancement methods

to be used, such as transmit diversity or advanced CDMA receiver concepts.Table

summaries the main WCDMA parameters.

The support for handovers (HO) between GSM and WCDMA is part of the ﬁrst

standard version. This means that all multi-mode WCDMA/GSM terminals will

support measurements from the one system while camped on the other one. This

allows networks using both WCDMA and GSM to balance the load between the

networks and base the HO on actual measurements from the terminals for diﬀerent

radio conditions in addition to other criteria available.

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Table 2.1: Main WCDMA parameters

Multiple access method DS-CDMA

Duplexing method Frequency division duplex/time division

duplex

Base station synchronisation Asynchronous operation

Chip rate 3.84 Mcps

Frame length 10 ms

Service multiplexing Multiple services with diﬀerent quality of

service requirements multiplexed on one

connection

Multi-rate concept Variable spreading factor and multicode

Detection Coherent using pilot symbols or common

pilot

Multi-user detection, smart antennas Supported by the standard, optional in the

implementation

The world’s ﬁrst commercial W-CDMA service, FoMA, was launched by NTT

DoCoMo in Japan in 2001. FoMA is the short name for Freedom of Mobile Mul-

timedia Access, is the brand name for the 3G services being oﬀered by Japanese

mobile phone operator NTT DoCoMo. Elsewhere, W-CDMA deployments have

been exclusively UMTS based.

UMTS or W-CDMA, assures backward compatibility with the second generation

GSM, IS-136 and PDC TDMA technologies, as well as all 2.5G TDMA technologies.

The network structure and bit level packaging of GSM data is retained by W-CDMA,

with additional capacity and bandwidth provided by a new CDMA air interface.

2.3.3 3G CDMA2000

Code division multiple access 2000 is the natural evolution of IS-95 (cdmaOne). It

includes additional functionality that increases its spectral eﬃciency and data rate

capability.(code division multiple access) is a mobile digital radio technology where

channels are deﬁned with codes (PN sequences). CDMA permits many simultaneous

transmitters on the same frequency channel. Since more phones can be served by

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fewer cell sites, CDMA-based standards have a signiﬁcant economic advantage over

TDMA- or FDMA-based standards. This standard is being developed by Telecom-

munications Industry Association (TIA) of US and is is standardized by 3GPP2.

The main CDMA2000 standards are: CDMA2000 1xRTT,CDMA2000 1xEV and

CDMA2000 EV-DV. These are the approved radio interfaces for the ITU’s IMT-2000

standard. In the following, a brief discussion about all these standards is given.

CDMA2000 1xRTT: RTT stands for Radio Transmission Technology and the

designation ”1x”, meaning ”1 times Radio Transmission Technology”, indicates the

same RF bandwidth as IS-95.The main features of CDMA2000 1X are as follows:

• Supports an instantaneous data rate upto 307kpbs for a user in packet mode

and a typical throughput rates of 144kbps per user,depending on the number

of user, the velociy of user and the propagating conditions.

• Supports up to twice as many voice users a the 2G CDMA standard

• Provides the subscriber unit with upto two times the standby time for longer

lasting battery life.

CDMA2000 EV: This is an evolutionary advancement of CDMA with the

following characteristics:

• Provides CDMA carriers with the option of installing radio channels with data

only (CDMA2000 EV-DO) and with data and voice (CDMA2000 EV-DV) .

• The cdma2000 1xEV-DO supports greater than 2.4Mbps of instantaneous

high-speed packet throughput per user on a CDMA channel, although the

user data rates are much lower and highly dependent on other factors.

• CDMA2000 EV-DV can oﬀer data rates upto 144kbps with about twice as

many voice channels as IS-95B.

CDMA2000 3x is (also known as EV-DO Rev B) is a multi-carrier evolution.

• It has higher rates per carrier (up to 4.9 Mbit/s on the downlink per carrier).

Typical deployments are expected to include 3 carriers for a peak rate of 14.7

Mbit/s.Higher rates are possible by bundling multiple channels together. It

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enhances the user experience and enables new services such as high deﬁnition

video streaming.

• Uses statistical multiplexing across channels to further reduce latency, en-

hancing the experience for latency-sensitive services such as gaming, video

telephony, remote console sessions and web browsing.

• It provides increased talk-time and standby time.

• The interference from the adjacent sectors is reduced by hybrid frequency re-

use and improves the rates that can be oﬀered, especially to users at the edge

of the cell.

• It has eﬃcient support for services that have asymmetric download and upload

requirements (i.e. diﬀerent data rates required in each direction) such as ﬁle

transfers, web browsing, and broadband multimedia content delivery.

2.3.4 3G TD-SCDMA

Time Division-Synchronous Code Division Multiple Access, or TD-SCDMA, is a

3G mobile telecommunications standard, being pursued in the People’s Republic of

China by the Chinese Academy of Telecommunications Technology (CATT). This

proposal was adopted by ITU as one of the 3G options in late 1999. TD-SCDMA is

based on spread spectrum technology.

TD-SCDMA uses TDD, in contrast to the FDD scheme used by W-CDMA.

By dynamically adjusting the number of timeslots used for downlink and uplink,

the system can more easily accommodate asymmetric traﬃc with diﬀerent data

rate requirements on downlink and uplink than FDD schemes. Since it does not

require paired spectrum for downlink and uplink, spectrum allocation ﬂexibility is

also increased. Also, using the same carrier frequency for uplink and downlink means

that the channel condition is the same on both directions, and the base station can

deduce the downlink channel information from uplink channel estimates, which is

helpful to the application of beamforming techniques.

TD-SCDMA also uses TDMA in addition to the CDMA used in WCDMA. This

reduces the number of users in each timeslot, which reduces the implementation

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complexity of multiuser detection and beamforming schemes, but the non-continuous

transmission also reduces coverage (because of the higher peak power needed), mo-

bility (because of lower power control frequency) and complicates radio resource

management algorithms.

The ”S” in TD-SCDMA stands for ”synchronous”, which means that uplink sig-

nals are synchronized at the base station receiver, achieved by continuous timing

adjustments. This reduces the interference between users of the same timeslot using

diﬀerent codes by improving the orthogonality between the codes, therefore increas-

ing system capacity, at the cost of some hardware complexity in achieving uplink

synchronization.

2.4 Wireless Transmission Protocols

There are several transmission protocols in wireless manner to achieve diﬀerent

application oriented tasks. Below, some of these applications are given.

2.4.1 Wireless Local Loop (WLL) and LMDS

Microwave wireless links can be used to create a wireless local loop. The local loop

can be thought of as the ”last mile” of the telecommunication network that resides

between the central oﬃce (CO) and the individual homes and business in close

proximity to the CO. An advantage of WLL technology is that once the wireless

equipment is paid for, there are no additional costs for transport between the CO

and the customer premises equipment. Many new services have been proposed and

this includes the concept of Local Multipoint Distribution Service (LMDS), which

provides broadband telecommunication access in the local exchange.

2.4.2 Bluetooth

• Facilitates ad-hoc data transmission over short distances from ﬁxed and mobile

devices as shown in Figure 2.1

• Uses a radio technology called frequency hopping spread spectrum. It chops up

the data being sent and transmits chunks of it on up to 79 diﬀerent frequencies.

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Figure 2.1: Data transmission with Bluetooth.

In its basic mode, the modulation is Gaussian frequency shift keying (GFSK).

It can achieve a gross data rate of 1 Mb/s

• Primarily designed for low power consumption, with a short range (power-

class-dependent: 1 meter, 10 meters, 100 meters) based on low-cost transceiver

microchips in each device

2.4.3 Wireless Local Area Networks (W-LAN)

• IEEE 802.11 WLAN uses ISM band (5.275-5.825GHz)

• Uses 11Mcps DS-SS spreading and 2Mbps user data rates (will fallback to

1Mbps in noisy conditions)

• IEEE 802.11a stndard provides upto 54Mbps throughput in the 5GHz band.

The DS-SS IEEE 802.11b has been called Wi-Fi. Wi-Fi networks have limited

range. A typical Wi-Fi home router using 802.11b or 802.11g with a stock

antenna might have a range of 32 m (120 ft) indoors and 95 m (300 ft) outdoors.

Range also varies with frequency band.

• IEEE 802.11g uses Complementary Code Keying Orthogonal Frequency Divi-

sion Multiplexing (CCK-OFDM) standards in both 2.4GHz and 5GHz bands.

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2.4.4 WiMax

• Provides upto 70 Mb/sec symmetric broadband speed without the need for

cables. The technology is based on the IEEE 802.16 standard (also called

WirelessMAN)

• WiMAX can provide broadband wireless access (BWA) up to 30 miles (50 km)

for ﬁxed stations, and 3 - 10 miles (5 - 15 km) for mobile stations. In contrast,

the WiFi/802.11 wireless local area network standard is limited in most cases

to only 100 - 300 feet (30 - 100m)

• The 802.16 speciﬁcation applies across a wide range of the RF spectrum, and

WiMAX could function on any frequency below 66 GHz (higher frequencies

would decrease the range of a Base Station to a few hundred meters in an

urban environment).

2.4.5 Zigbee

• ZigBee is the speciﬁcation for a suite of high level communication protocols us-

ing small, low-power digital radios based on the IEEE 802.15.4-2006 standard

for wireless personal area networks (WPANs), such as wireless headphones

connecting with cell phones via short-range radio.

• This technology is intended to be simpler and cheaper. ZigBee is targeted at

radio-frequency (RF) applications that require a low data rate, long battery

life, and secure networking.

• ZigBee operates in the industrial, scientiﬁc and medical (ISM) radio bands;

868 MHz in Europe, 915 MHz in countries such as USA and Australia, and

2.4 GHz in most worldwide.

2.4.6 Wibree

• Wibree is a digital radio technology (intended to become an open standard of

wireless communications) designed for ultra low power consumption (button

cell batteries) within a short range (10 meters / 30 ft) based around low-cost

transceiver microchips in each device.

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• Wibree is known as Bluetooth with low energy technology.

• It operates in 2.4 GHz ISM band with physical layer bit rate of 1 Mbps.

2.5 Conclusion: Beyond 3G Networks

Beyond 3G networks, or 4G (Fourth Generation), represent the next complete evo-

lution in wireless communications. A 4G system will be able to provide a compre-

hensive IP solution where voice, data and streamed multimedia can be given to users

at higher data rates than previous generations.There is no formal deﬁnition for 4G ;

however, there are certain objectives that are projected for 4G. It will be capable of

providing between 100 Mbit/s and 1 Gbit/s speeds both indoors and outdoors, with

premium quality and high security. It would also support systems like multicarrier

communication, MIMO and UWB.

2.6 References

1. T. S. Rappaport, Wireless Communications: Principles and Practice, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

2. W. C. Lee, Mobile Communications Engineering, 2nd ed. New Delhi: Tata

McGraw-Hill, 2008.

3. R. Pandya, Mobile and Personal Communication Systems and Services, 4th

ed. New Delhi: PHI, 2004.

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Chapter 3

The Cellular Engineering

Fundamentals

3.1 Introduction

In Chapter 1, we have seen that the technique of substituting a single high power

transmitter by several low power transmitters to support many users is the backbone

of the cellular concept. In practice, the following four parameters are most important

while considering the cellular issues: system capacity, quality of service, spectrum

eﬃciency and power management. Starting from the basic notion of a cell, we would

deal with these parameters in the context of cellular engineering in this chapter.

3.2 What is a Cell?

The power of the radio signals transmitted by the BS decay as the signals travel

away from it. A minimum amount of signal strength (let us say, x dB) is needed in

order to be detected by the MS or mobile sets which may the hand-held personal

units or those installed in the vehicles. The region over which the signal strength

lies above this threshold value x dB is known as the coverage area of a BS and

it must be a circular region, considering the BS to be isotropic radiator. Such a

circle, which gives this actual radio coverage, is called the foot print of a cell (in

reality, it is amorphous). It might so happen that either there may be an overlap

between any two such side by side circles or there might be a gap between the

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Figure 3.1: Footprint of cells showing the overlaps and gaps.

coverage areas of two adjacent circles. This is shown in Figure 3.1. Such a circular

geometry, therefore, cannot serve as a regular shape to describe cells. We need a

regular shape for cellular design over a territory which can be served by 3 regular

polygons, namely, equilateral triangle, square and regular hexagon, which can cover

the entire area without any overlap and gaps. Along with its regularity, a cell must

be designed such that it is most reliable too, i.e., it supports even the weakest mobile

with occurs at the edges of the cell. For any distance between the center and the

farthest point in the cell from it, a regular hexagon covers the maximum area. Hence

regular hexagonal geometry is used as the cells in mobile communication.

3.3 Frequency Reuse

Frequency reuse, or, frequency planning, is a technique of reusing frequencies and

channels within a communication system to improve capacity and spectral eﬃciency.

Frequency reuse is one of the fundamental concepts on which commercial wireless

systems are based that involve the partitioning of an RF radiating area into cells.

The increased capacity in a commercial wireless network, compared with a network

with a single transmitter, comes from the fact that the same radio frequency can be

reused in a diﬀerent area for a completely diﬀerent transmission.

Frequency reuse in mobile cellular systems means that frequencies allocated to

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Figure 3.2: Frequency reuse technique of a cellular system.

the service are reused in a regular pattern of cells, each covered by one base station.

The repeating regular pattern of cells is called cluster. Since each cell is designed

to use radio frequencies only within its boundaries, the same frequencies can be

reused in other cells not far away without interference, in another cluster. Such cells

are called ‘co-channel’ cells. The reuse of frequencies enables a cellular system to

handle a huge number of calls with a limited number of channels. Figure 3.2 shows

a frequency planning with cluster size of 7, showing the co-channels cells in diﬀerent

clusters by the same letter. The closest distance between the co-channel cells (in

diﬀerent clusters) is determined by the choice of the cluster size and the layout of

the cell cluster. Consider a cellular system with S duplex channels available for

use and let N be the number of cells in a cluster. If each cell is allotted K duplex

channels with all being allotted unique and disjoint channel groups we have S = KN

under normal circumstances. Now, if the cluster are repeated M times within the

total area, the total number of duplex channels, or, the total number of users in the

system would be T = MS = KMN. Clearly, if K and N remain constant, then

T ∝ M (3.1)

and, if T and K remain constant, then

N ∝

1

M

. (3.2)

Hence the capacity gain achieved is directly proportional to the number of times

a cluster is repeated, as shown in (3.1), as well as, for a ﬁxed cell size, small N

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decreases the size of the cluster with in turn results in the increase of the number

of clusters (3.2) and hence the capacity. However for small N, co-channel cells are

located much closer and hence more interference. The value of N is determined by

calculating the amount of interference that can be tolerated for a suﬃcient quality

communication. Hence the smallest N having interference below the tolerated limit

is used. However, the cluster size N cannot take on any value and is given only by

the following equation

N = i

2

+ij +j

2

, i ≥ 0, j ≥ 0, (3.3)

where i and j are integer numbers.

Ex. 1: Find the relationship between any two nearest co-channel cell distance D

and the cluster size N.

Solution: For hexagonal cells, it can be shown that the distance between two adjacent

cell centers =

√

3R, where R is the radius of any cell. The normalized co-channel

cell distance D

n

can be calculated by traveling ’i’ cells in one direction and then

traveling ’j’ cells in anticlockwise 120

o

of the primary direction. Using law of vector

addition,

D

2

n

= j

2

cos

2

(30

o

) + (i +j sin(30

o

))

2

(3.4)

which turns out to be

D

n

=

_

i

2

+ij +j

2

=

√

N. (3.5)

Multiplying the actual distance

√

3R between two adjacent cells with it, we get

D = D

n

√

3R =

√

3NR. (3.6)

Ex. 2: Find out the surface area of a regular hexagon with radius R, the surface

area of a large hexagon with radius D, and hence compute the total number of cells

in this large hexagon.

Hint: In general, this large hexagon with radius D encompasses the center cluster of

N cells and one-third of the cells associated with six other peripheral large hexagons.

Thus, the answer must be N + 6(

N

3

) = 3N.

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3.4 Channel Assignment Strategies

With the rapid increase in number of mobile users, the mobile service providers

had to follow strategies which ensure the eﬀective utilization of the limited radio

spectrum. With increased capacity and low interference being the prime objectives,

a frequency reuse scheme was helpful in achieving this objectives. A variety of

channel assignment strategies have been followed to aid these objectives. Channel

assignment strategies are classiﬁed into two types: ﬁxed and dynamic, as discussed

below.

3.4.1 Fixed Channel Assignment (FCA)

In ﬁxed channel assignment strategy each cell is allocated a ﬁxed number of voice

channels. Any communication within the cell can only be made with the designated

unused channels of that particular cell. Suppose if all the channels are occupied,

then the call is blocked and subscriber has to wait. This is simplest of the channel

assignment strategies as it requires very simple circuitry but provides worst channel

utilization. Later there was another approach in which the channels were borrowed

from adjacent cell if all of its own designated channels were occupied. This was

named as borrowing strategy. In such cases the MSC supervises the borrowing pro-

cess and ensures that none of the calls in progress are interrupted.

3.4.2 Dynamic Channel Assignment (DCA)

In dynamic channel assignment strategy channels are temporarily assigned for use

in cells for the duration of the call. Each time a call attempt is made from a cell the

corresponding BS requests a channel from MSC. The MSC then allocates a channel

to the requesting the BS. After the call is over the channel is returned and kept in

a central pool. To avoid co-channel interference any channel that in use in one cell

can only be reassigned simultaneously to another cell in the system if the distance

between the two cells is larger than minimum reuse distance. When compared to the

FCA, DCA has reduced the likelihood of blocking and even increased the trunking

capacity of the network as all of the channels are available to all cells, i.e., good

quality of service. But this type of assignment strategy results in heavy load on

switching center at heavy traﬃc condition.

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Ex. 3: A total of 33 MHz bandwidth is allocated to a FDD cellular system with

two 25 KHz simplex channels to provide full duplex voice and control channels.

Compute the number of channels available per cell if the system uses (i) 4 cell, (ii)

7 cell, and (iii) 8 cell reuse technique. Assume 1 MHz of spectrum is allocated to

control channels. Give a distribution of voice and control channels.

Solution: One duplex channel = 2 x 25 = 50 kHz of spectrum. Hence the total

available duplex channels are = 33 MHz / 50 kHz = 660 in number. Among these

channels, 1 MHz / 50 kHz = 20 channels are kept as control channels.

(a) For N = 4, total channels per cell = 660/4 = 165.

Among these, voice channels are 160 and control channels are 5 in number.

(b) For N = 7, total channels per cell are 660/7 ≈ 94. Therefore, we have to go for

a more exact solution. We know that for this system, a total of 20 control channels

and a total of 640 voice channels are kept. Here, 6 cells can use 3 control channels

and the rest two can use 2 control channels each. On the other hand, 5 cells can use

92 voice channels and the rest two can use 90 voice channels each. Thus the total

solution for this case is:

6 x 3 + 1 x 2 = 20 control channels, and,

5 x 92 + 2 x 90 = 640 voice channels.

This is one solution, there might exist other solutions too.

(c) The option N = 8 is not a valid option since it cannot satisfy equation (3.3) by

two integers i and j.

3.5 Handoﬀ Process

When a user moves from one cell to the other, to keep the communication between

the user pair, the user channel has to be shifted from one BS to the other without

interrupting the call, i.e., when a MS moves into another cell, while the conversation

is still in progress, the MSC automatically transfers the call to a new FDD channel

without disturbing the conversation. This process is called as handoﬀ. A schematic

diagram of handoﬀ is given in Figure 3.3.

Processing of handoﬀ is an important task in any cellular system. Handoﬀs

must be performed successfully and be imperceptible to the users. Once a signal

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Figure 3.3: Handoﬀ scenario at two adjacent cell boundary.

level is set as the minimum acceptable for good voice quality (P

r

min

), then a slightly

stronger level is chosen as the threshold (P

r

H

)at which handoﬀ has to be made, as

shown in Figure 3.4. A parameter, called power margin, deﬁned as

∆ = P

r

H

−P

r

min

(3.7)

is quite an important parameter during the handoﬀ process since this margin ∆ can

neither be too large nor too small. If ∆ is too small, then there may not be enough

time to complete the handoﬀ and the call might be lost even if the user crosses the

cell boundary.

If ∆ is too high o the other hand, then MSC has to be burdened with unnecessary

handoﬀs. This is because MS may not intend to enter the other cell. Therefore ∆

should be judiciously chosen to ensure imperceptible handoﬀs and to meet other

objectives.

3.5.1 Factors Inﬂuencing Handoﬀs

The following factors inﬂuence the entire handoﬀ process:

(a) Transmitted power: as we know that the transmission power is diﬀerent for dif-

ferent cells, the handoﬀ threshold or the power margin varies from cell to cell.

(b) Received power: the received power mostly depends on the Line of Sight (LoS)

path between the user and the BS. Especially when the user is on the boundary of

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Figure 3.4: Handoﬀ process associated with power levels, when the user is going

from i-th cell to j-th cell.

the two cells, the LoS path plays a critical role in handoﬀs and therefore the power

margin ∆ depends on the minimum received power value from cell to cell.

(c) Area and shape of the cell: Apart from the power levels, the cell structure also

a plays an important role in the handoﬀ process.

(d) Mobility of users: The number of mobile users entering or going out of a partic-

ular cell, also ﬁxes the handoﬀ strategy of a cell.

To illustrate the reasons (c) and (d), let us consider a rectangular cell with sides R

1

and R

2

inclined at an angle θ with horizon, as shown in the Figure 3.5. Assume N

1

users are having handoﬀ in horizontal direction and N

2

in vertical direction per unit

length.

The number of crossings along R

1

side is : (N

1

cosθ +N

2

sinθ)R

1

and the number of

crossings along R

2

side is : (N

1

sinθ +N

2

cosθ)R

2

.

Then the handoﬀ rate λ

H

can be written as

λ

H

= (N

1

cosθ +N

2

sinθ)R

1

+ (N

1

sinθ +N

2

cosθ)R

2

. (3.8)

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Figure 3.5: Handoﬀ process with a rectangular cell inclined at an angle θ.

Now, given the ﬁxed area A = R

1

R

2

, we need to ﬁnd λ

min

H

for a given θ. Replacing

R

1

by

A

R

2

and equating

dλ

H

dR

1

to zero, we get

R

2

1

= A(

N

1

sinθ +N

2

cosθ

N

1

cosθ +N

2

sinθ

). (3.9)

Similarly, for R

2

, we get

R

2

2

= A(

N

1

cosθ +N

2

sinθ

N

1

sinθ +N

2

cosθ

). (3.10)

From the above equations, we have λ

H

= 2

_

A(N

1

N

2

+ (N

2

1

+N

2

2

)cosθsinθ) which

means it it minimized at θ = 0

o

. Hence λ

min

H

= 2

√

AN

1

N

2

. Putting the value of θ

in (3.9) or (3.10), we have

R

1

R

2

=

N

1

N

2

. This has two implications: (i) that handoﬀ is

minimized if rectangular cell is aligned with X-Y axis, i.e., θ = 0

o

, and, (ii) that the

number of users crossing the cell boundary is inversely proportional to the dimension

of the other side of the cell. The above analysis has been carried out for a simple

square cell and it changes in more complicated way when we consider a hexagonal

cell.

3.5.2 Handoﬀs In Diﬀerent Generations

In 1G analog cellular systems, the signal strength measurements were made by

the BS and in turn supervised by the MSC. The handoﬀs in this generation can

be termed as Network Controlled Hand-Oﬀ (NCHO). The BS monitors the signal

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strengths of voice channels to determine the relative positions of the subscriber.

The special receivers located on the BS are controlled by the MSC to monitor the

signal strengths of the users in the neighboring cells which appear to be in need

of handoﬀ. Based on the information received from the special receivers the MSC

decides whether a handoﬀ is required or not. The approximate time needed to make

a handoﬀ successful was about 5-10 s. This requires the value of ∆ to be in the

order of 6dB to 12dB.

In the 2G systems, the MSC was relieved from the entire operation. In this

generation, which started using the digital technology, handoﬀ decisions were mobile

assisted and therefore it is called Mobile Assisted Hand-Oﬀ (MAHO). In MAHO,

the mobile center measures the power changes received from nearby base stations

and notiﬁes the two BS. Accordingly the two BS communicate and channel transfer

occurs. As compared to 1G, the circuit complexity was increased here whereas the

delay in handoﬀ was reduced to 1-5 s. The value of ∆ was in the order of 0-5 dB.

However, even this amount of delay could create a communication pause.

In the current 3G systems, the MS measures the power from adjacent BS and

automatically upgrades the channels to its nearer BS. Hence this can be termed as

Mobile Controlled Hand-Oﬀ (MCHO). When compared to the other generations,

delay during handoﬀ is only 100 ms and the value of ∆ is around 20 dBm. The

Quality Of Service (QoS) has improved a lot although the complexity of the circuitry

has further increased which is inevitable.

All these types of handoﬀs are usually termed as hard handoﬀ as there is a shift

in the channels involved. There is also another kind of handoﬀ, called soft handoﬀ,

as discussed below.

Handoﬀ in CDMA: In spread spectrum cellular systems, the mobiles share the same

channels in every cell. The MSC evaluates the signal strengths received from diﬀerent

BS for a single user and then shifts the user from one BS to the other without actually

changing the channel. These types of handoﬀs are called as soft handoﬀ as there is

no change in the channel.

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3.5.3 Handoﬀ Priority

While assigning channels using either FCA or DCA strategy, a guard channel concept

must be followed to facilitate the handoﬀs. This means, a fraction of total available

channels must be kept for handoﬀ requests. But this would reduce the carried

traﬃc and only fewer channels can be assigned for the residual users of a cell. A

good solution to avoid such a dead-lock is to use DCA with handoﬀ priority (demand

based allocation).

3.5.4 A Few Practical Problems in Handoﬀ Scenario

(a) Diﬀerent speed of mobile users: with the increase of mobile users in urban areas,

microcells are introduced in the cells to increase the capacity (this will be discussed

later in this chapter). The users with high speed frequently crossing the micro-cells

become burdened to MSC as it has to take care of handoﬀs. Several schemes thus

have been designed to handle the simultaneous traﬃc of high speed and low speed

users while minimizing the handoﬀ intervention from the MSC, one of them being

the ‘Umbrella Cell’ approach. This technique provides large area coverage to high

speed users while providing small area coverage to users traveling at low speed. By

using diﬀerent antenna heights and diﬀerent power levels, it is possible to provide

larger and smaller cells at a same location. As illustrated in the Figure 3.6, umbrella

cell is co-located with few other microcells. The BS can measure the speed of the

user by its short term average signal strength over the RVC and decides which cell

to handle that call. If the speed is less, then the corresponding microcell handles

the call so that there is good corner coverage. This approach assures that handoﬀs

are minimized for high speed users and provides additional microcell channels for

pedestrian users.

(b) Cell dragging problem: this is another practical problem in the urban area with

additional microcells. For example, consider there is a LOS path between the MS

and BS1 while the user is in the cell covered by BS2. Since there is a LOS with the

BS1, the signal strength received from BS1 would be greater than that received from

BS2. However, since the user is in cell covered by BS2, handoﬀ cannot take place

and as a result, it experiences a lot of interferences. This problem can be solved by

judiciously choosing the handoﬀ threshold along with adjusting the coverage area.

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(c) Inter-system handoﬀ: if one user is leaving the coverage area of one MSC and is

entering the area of another MSC, then the call might be lost if there is no handoﬀ in

this case too. Such a handoﬀ is called inter-system handoﬀ and in order to facilitate

this, mobiles usually have roaming facility.

3.6 Interference & System Capacity

Susceptibility and interference problems associated with mobile communications

equipment are because of the problem of time congestion within the electromag-

netic spectrum. It is the limiting factor in the performance of cellular systems. This

interference can occur from clash with another mobile in the same cell or because

of a call in the adjacent cell. There can be interference between the base stations

operating at same frequency band or any other non-cellular system’s energy leaking

inadvertently into the frequency band of the cellular system. If there is an interfer-

ence in the voice channels, cross talk is heard will appear as noise between the users.

The interference in the control channels leads to missed and error calls because of

digital signaling. Interference is more severe in urban areas because of the greater

RF noise and greater density of mobiles and base stations. The interference can be

divided into 2 parts: co-channel interference and adjacent channel interference.

3.6.1 Co-channel interference (CCI)

For the eﬃcient use of available spectrum, it is necessary to reuse frequency band-

width over relatively small geographical areas. However, increasing frequency reuse

also increases interference, which decreases system capacity and service quality. The

cells where the same set of frequencies is used are call co-channel cells. Co-channel

interference is the cross talk between two diﬀerent radio transmitters using the same

radio frequency as is the case with the co-channel cells. The reasons of CCI can be

because of either adverse weather conditions or poor frequency planning or overly-

crowded radio spectrum.

If the cell size and the power transmitted at the base stations are same then CCI

will become independent of the transmitted power and will depend on radius of the

cell (R) and the distance between the interfering co-channel cells (D). If D/R ratio

is increased, then the eﬀective distance between the co-channel cells will increase

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and interference will decrease. The parameter Q is called the frequency reuse ratio

and is related to the cluster size. For hexagonal geometry

Q = D/R =

√

3N. (3.11)

From the above equation, small of ‘Q’ means small value of cluster size ‘N’ and

increase in cellular capacity. But large ‘Q’ leads to decrease in system capacity

but increase in transmission quality. Choosing the options is very careful for the

selection of ‘N’, the proof of which is given in the ﬁrst section.

The Signal to Interference Ratio (SIR) for a mobile receiver which monitors the

forward channel can be calculated as

S

I

=

S

i

0

i=1

I

i

(3.12)

where i

0

is the number of co-channel interfering cells, S is the desired signal power

from the baseband station and I

i

is the interference power caused by the i-th interfer-

ing co-channel base station. In order to solve this equation from power calculations,

we need to look into the signal power characteristics. The average power in the

mobile radio channel decays as a power law of the distance of separation between

transmitter and receiver. The expression for the received power P

r

at a distance d

can be approximately calculated as

P

r

= P

0

(

d

d

0

)

−n

(3.13)

and in the dB expression as

P

r

(dB) = P

0

(dB) −10nlog(

d

d

0

) (3.14)

where P

0

is the power received at a close-in reference point in the far ﬁeld region at

a small distance do from the transmitting antenna, and ‘n’ is the path loss exponent.

Let us calculate the SIR for this system. If D

i

is the distance of the i-th interferer

from the mobile, the received power at a given mobile due to i-th interfering cell

is proportional to (D

i

)

−n

(the value of ’n’ varies between 2 and 4 in urban cellular

systems).

Let us take that the path loss exponent is same throughout the coverage area

and the transmitted power be same, then SIR can be approximated as

S

I

=

R

−n

i

0

i=1

D

−n

i

(3.15)

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where the mobile is assumed to be located at R distance from the cell center. If

we consider only the ﬁrst layer of interfering cells and we assume that the interfer-

ing base stations are equidistant from the reference base station and the distance

between the cell centers is ’D’ then the above equation can be converted as

S

I

=

(D/R)

n

i

0

=

(

√

3N)

n

i

0

(3.16)

which is an approximate measure of the SIR. Subjective tests performed on AMPS

cellular system which uses FM and 30 kHz channels show that suﬃcient voice quality

can be obtained by SIR being greater than or equal to 18 dB. If we take n=4

, the value of ’N’ can be calculated as 6.49. Therefore minimum N is 7. The

above equations are based on hexagonal geometry and the distances from the closest

interfering cells can vary if diﬀerent frequency reuse plans are used.

We can go for a more approximate calculation for co-channel SIR. This is the

example of a 7 cell reuse case. The mobile is at a distance of D-R from 2 closest

interfering cells and approximately D+R/2, D, D-R/2 and D+R distance from other

interfering cells in the ﬁrst tier. Taking n = 4 in the above equation, SIR can be

approximately calculated as

S

I

=

R

−4

2(D −R)

−4

+ (D +R)

−4

+ (D)

−4

+ (D +R/2)

−4

+ (D −R/2)

−4

(3.17)

which can be rewritten in terms frequency reuse ratio Q as

S

I

=

1

2(Q−1)

−4

+ (Q+ 1)

−4

+ (Q)

−4

+ (Q+ 1/2)

−4

+ (Q−1/2)

−4

. (3.18)

Using the value of N equal to 7 (this means Q = 4.6), the above expression yields

that worst case SIR is 53.70 (17.3 dB). This shows that for a 7 cell reuse case the

worst case SIR is slightly less than 18 dB. The worst case is when the mobile is at

the corner of the cell i.e., on a vertex as shown in the Figure 3.6. Therefore N = 12

cluster size should be used. But this reduces the capacity by 7/12 times. Therefore,

co-channel interference controls link performance, which in a way controls frequency

reuse plan and the overall capacity of the cellular system. The eﬀect of co-channel

interference can be minimized by optimizing the frequency assignments of the base

stations and their transmit powers. Tilting the base-station antenna to limit the

spread of the signals in the system can also be done.

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Figure 3.6: First tier of co-channel interfering cells

3.6.2 Adjacent Channel Interference (ACI)

This is a diﬀerent type of interference which is caused by adjacent channels i.e.

channels in adjacent cells. It is the signal impairment which occurs to one frequency

due to presence of another signal on a nearby frequency. This occurs when imperfect

receiver ﬁlters allow nearby frequencies to leak into the passband. This problem is

enhanced if the adjacent channel user is transmitting in a close range compared to

the subscriber’s receiver while the receiver attempts to receive a base station on the

channel. This is called near-far eﬀect. The more adjacent channels are packed into

the channel block, the higher the spectral eﬃciency, provided that the performance

degradation can be tolerated in the system link budget. This eﬀect can also occur

if a mobile close to a base station transmits on a channel close to one being used

by a weak mobile. This problem might occur if the base station has problem in

discriminating the mobile user from the ”bleed over” caused by the close adjacent

channel mobile.

Adjacent channel interference occurs more frequently in small cell clusters and heav-

ily used cells. If the frequency separation between the channels is kept large this

interference can be reduced to some extent. Thus assignment of channels is given

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such that they do not form a contiguous band of frequencies within a particular

cell and frequency separation is maximized. Eﬃcient assignment strategies are very

much important in making the interference as less as possible. If the frequency fac-

tor is small then distance between the adjacent channels cannot put the interference

level within tolerance limits. If a mobile is 10 times close to the base station than

other mobile and has energy spill out of its passband, then SIR for weak mobile is

approximately

S

I

= 10

−n

(3.19)

which can be easily found from the earlier SIR expressions. If n = 4, then SIR is

−52 dB. Perfect base station ﬁlters are needed when close-in and distant users share

the same cell. Practically, each base station receiver is preceded by a high Q cavity

ﬁlter in order to remove adjacent channel interference. Power control is also very

much important for the prolonging of the battery life for the subscriber unit but also

reduces reverse channel SIR in the system. Power control is done such that each

mobile transmits the lowest power required to maintain a good quality link on the

reverse channel.

3.7 Enhancing Capacity And Cell Coverage

3.7.1 The Key Trade-oﬀ

Previously, we have seen that the frequency reuse technique in cellular systems

allows for almost boundless expansion of geographical area and the number of mobile

system users who could be accommodated. In designing a cellular layout, the two

parameters which are of great signiﬁcance are the cell radius R and the cluster size

N, and we have also seen that co-channel cell distance D =

√

3NR. In the following,

a brief description of the design trade-oﬀ is given, in which the above two parameters

play a crucial role.

The cell radius governs both the geographical area covered by a cell and also

the number of subscribers who can be serviced, given the subscriber density. It is

easy to see that the cell radius must be as large as possible. This is because, every

cell requires an investment in a tower, land on which the tower is placed, and radio

transmission equipment and so a large cell size minimizes the cost per subscriber.

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Eventually, the cell radius is determined by the requirement that adequate signal

to noise ratio be maintained over the coverage area. The SNR is determined by

several factors such as the antenna height, transmitter power, receiver noise ﬁgure

etc. Given a cell radius R and a cluster size N, the geographic area covered by a

cluster is

A

cluster

= NA

cell

= N3

√

3R

2

/2. (3.20)

If the total serviced area is A

total

, then the number of clusters M that could be

accommodated is given by

M = A

total

/A

cluster

= A

total

/(N3

√

3R

2

/2). (3.21)

Note that all of the available channels N, are reused in every cluster. Hence, to make

the maximum number of channels available to subscribers, the number of clusters

M should be large, which, by Equation (3.21), shows that the cell radius should

be small. However, cell radius is determined by a trade-oﬀ: R should be as large

as possible to minimize the cost of the installation per subscriber, but R should

be as small as possible to maximize the number of customers that the system can

accommodate. Now, if the cell radius R is ﬁxed, then the number of clusters could be

maximized by minimizing the size of a cluster N. We have seen earlier that the size

of a cluster depends on the frequency reuse ratio Q. Hence, in determining the value

of N, another trade-oﬀ is encountered in that N must be small to accommodate

large number of subscribers, but should be suﬃciently large so as to minimize the

interference eﬀects.

Now, we focus on the issues regarding system expansion. The history of cellular

phones has been characterized by a rapid growth and expansion in cell subscribers.

Though a cellular system can be expanded by simply adding cells to the geographical

area, the way in which user density can be increased is also important to look at.

This is because it is not always possible to counter the increasing demand for cellular

systems just by increasing the geographical coverage area due to the limitations in

obtaining new land with suitable requirements. We discuss here two methods for

dealing with an increasing subscriber density: Cell Splitting and Sectoring. The

other method, microcell zone concept can treated as enhancing the QoS in a cellular

system.

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The basic idea of adopting the cellular approach is to allow space for the growth

of mobile users. When a new system is deployed, the demand for it is fairly low and

users are assumed to be uniformly distributed over the service area. However, as new

users subscribe to the cellular service, the demand for channels may begin to exceed

the capacity of some base stations. As discussed previously,the number of channels

available to customers (equivalently, the channel density per square kilometer) could

be increased by decreasing the cluster size. However, once a system has been initially

deployed, a system-wide reduction in cluster size may not be necessary since user

density does not grow uniformly in all parts of the geographical area. It might be

that an increase in channel density is required only in speciﬁc parts of the system

to support an increased demand in those areas. Cell-splitting is a technique which

has the capability to add new smaller cells in speciﬁc areas of the system.

3.7.2 Cell-Splitting

Cell Splitting is based on the cell radius reduction and minimizes the need to modify

the existing cell parameters. Cell splitting involves the process of sub-dividing a

congested cell into smaller cells, each with its own base station and a corresponding

reduction in antenna size and transmitting power. This increases the capacity of

a cellular system since it increases the number of times that channels are reused.

Since the new cells have smaller radii than the existing cells, inserting these smaller

cells, known as microcells, between the already existing cells results in an increase

of capacity due to the additional number of channels per unit area. There are few

challenges in increasing the capacity by reducing the cell radius. Clearly, if cells

are small, there would have to be more of them and so additional base stations

will be needed in the system. The challenge in this case is to introduce the new

base stations without the need to move the already existing base station towers.

The other challenge is to meet the generally increasing demand that may vary quite

rapidly between geographical areas of the system. For instance, a city may have

highly populated areas and so the demand must be supported by cells with the

smallest radius. The radius of cells will generally increase as we move from urban to

sub urban areas, because the user density decreases on moving towards sub-urban

areas. The key factor is to add as minimum number of smaller cells as possible

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Figure 3.7: Splitting of congested seven-cell clusters.

wherever an increase in demand occurs. The gradual addition of the smaller cells

implies that, at least for a time, the cellular system operates with cells of more than

one size.

Figure 3.7 shows a cellular layout with seven-cell clusters. Consider that the cells

in the center of the diagram are becoming congested, and cell A in the center has

reached its maximum capacity. Figure also shows how the smaller cells are being

superimposed on the original layout. The new smaller cells have half the cell radius

of the original cells. At half the radius, the new cells will have one-fourth of the area

and will consequently need to support one-fourth the number of subscribers. Notice

that one of the new smaller cells lies in the center of each of the larger cells. If

we assume that base stations are located in the cell centers, this allows the original

base stations to be maintained even in the new system layout. However, new base

stations will have to be added for new cells that do not lie in the center of the larger

cells. The organization of cells into clusters is independent of the cell radius, so that

the cluster size can be the same in the small-cell layout as it was in the large-cell

layout. Also the signal-to-interference ratio is determined by cluster size and not by

cell radius. Consequently, if the cluster size is maintained, the signal-to-interference

ratio will be the same after cell splitting as it was before. If the entire system is

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replaced with new half-radius cells, and the cluster size is maintained, the number

of channels per cell will be exactly as it was before, and the number of subscribers

per cell will have been reduced.

When the cell radius is reduced by a factor, it is also desirable to reduce the

transmitted power. The transmit power of the new cells with radius half that of the

old cells can be found by examining the received power PR at the new and old cell

boundaries and setting them equal. This is necessary to maintain the same frequency

re-use plan in the new cell layout as well. Assume that PT1 and PT2 are the transmit

powers of the larger and smaller base stations respectively. Then, assuming a path

loss index n=4, we have power received at old cell boundary = P

T1

/R

4

and the

power received at new cell boundary = P

T2

/(R/2)

4

. On equating the two received

powers, we get P

T2

= P

T1

/ 16. In other words, the transmit power must be reduced

by 12 dB in order to maintain the same S/I with the new system lay-out.

At the beginning of this channel splitting process, there would be fewer channels

in the smaller power groups. As the demand increases, more and more channels need

to be accommodated and hence the splitting process continues until all the larger

cells have been replaced by the smaller cells, at which point splitting is complete

within the region and the entire system is rescaled to have a smaller radius per cell.

If a cellular layout is replaced entirety by a new layout with a smaller cell radius,

the signal-to-interference ratio will not change, provided the cluster size does not

change. Some special care must be taken, however, to avoid co-channel interference

when both large and small cell radii coexist. It turns out that the only way to

avoid interference between the large-cell and small-cell systems is to assign entirely

diﬀerent sets of channels to the two systems. So, when two sizes of cells co-exist in

a system, channels in the old cell must be broken down into two groups, one that

corresponds to larger cell reuse requirements and the other which corresponds to the

smaller cell reuse requirements. The larger cell is usually dedicated to high speed

users as in the umbrella cell approach so as to minimize the number of hand-oﬀs.

Ex. 4: When the AMPS cellular system was ﬁrst deployed, the aim of the

system designers was to guarantee coverage. Initially the number of users was not

signiﬁcant. Consequently cells were conﬁgured with an eight-mile radius, and a

12-cell cluster size was chosen. The cell radius was chosen to guarantee a 17 dB

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Figure 3.8: A cell divided into three 120

o

sectors.

signal-to-noise ratio over 90% of the coverage area. Although a 12-cell cluster size

provided more than adequate co-channel separation to meet a requirement for a

17 dB signal-to-interference ratio in an interference-limited environment, it did not

provide adequate frequency reuse to service an explosively growing customer base.

The system planners reasoned that a subsequent shift to a 7-cell cluster size would

provide an adequate number of channels. It was estimated that a 7-cell cluster

size should provide an adequate 18.7 dB signal-to-interference ratio. The margin,

however, is slim, and the 17 dB signal-to-interference ratio requirement could not

be met over 90 % of the coverage area.

3.7.3 Sectoring

Sectoring is basically a technique which can increase the SIR without necessitating

an increase in the cluster size. Till now, it has been assumed that the base station is

located in the center of a cell and radiates uniformly in all the directions behaving as

an omni-directional antenna. However it has been found that the co-channel inter-

ference in a cellular system may be decreased by replacing a single omni-directional

antenna at the base station by several directional antennas, each radiating within a

speciﬁed sector. In the Figure 3.8, a cell is shown which has been split into three

120

o

sectors. The base station feeds three 120

o

directional antennas, each of which

radiates into one of the three sectors. The channel set serving this cell has also been

divided, so that each sector is assigned one-third of the available number cell of

channels. This technique for reducing co-channel interference wherein by using suit-

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Figure 3.9: A seven-cell cluster with 60

o

sectors.

able directional antennas, a given cell would receive interference and transmit with

a fraction of available co-channel cells is called ’sectoring’. In a seven-cell-cluster

layout with 120

o

sectored cells, it can be easily understood that the mobile units in

a particular sector of the center cell will receive co-channel interference from only

two of the ﬁrst-tier co-channel base stations, rather than from all six. Likewise, the

base station in the center cell will receive co-channel interference from mobile units

in only two of the co-channel cells. Hence the signal to interference ratio is now

modiﬁed to

S

I

=

(

√

3N)

n

2

(3.22)

where the denominator has been reduced from 6 to 2 to account for the reduced

number of interfering sources. Now, the signal to interference ratio for a seven-cell

cluster layout using 120

o

sectored antennas can be found from equation (3.24) to be

23.4 dB which is a signiﬁcant improvement over the Omni-directional case where the

worst-case S/I is found to be 17 dB (assuming a path-loss exponent, n=4). Some

cellular systems divide the cells into 60

o

sectors. Similar analysis can be performed

on them as well.

Ex. 5: A cellular system having a seven-cell cluster layout with omni-directional

antennas has been performing satisfactorily for a required signal to interference ratio

of 15 dB. However due to the need for increasing the number of available channels, a

60

o

sectoring of the cells has been introduced. By what percentage can the number

of channels N

total

be increased assuming a path-loss component n=4?

Solution: The seven-cell cluster layout with 60

o

sectoring is shown in the Figure 3.9.

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It is easy to see that the shaded region in the center receives interference from just

one ﬁrst-tier cell and hence the signal to interference ratio can be obtained suitably

as

S

I

=

(

√

3N)

n

1

=

(

_

(3)(7))

4

1

= 26.4dB. (3.23)

Since the SIR exceeds 15 dB, one can try reducing the cluster size from seven to

four. Now, the SIR for this reduced cluster size layout can be found to be

S

I

=

(

√

3N)

n

1

=

(

_

(3)(4))

4

1

= 21.6dB. (3.24)

The S/I ratio is still above the requirement and so a further reduction in the cell

cluster size is possible. For a 3-cell cluster layout, there are two interfering sources

and hence the S/I ratio is found to be

S

I

=

(

√

3N)

n

1

=

(

√

33)

4

2

= 16.07dB. (3.25)

This is just above the adequate S/I ratio and further reduction in cluster size is

not possible. So, a 3-cluster cell layout could be used for meeting the growth re-

quirements. Thus, when the cluster size is reduced from 7 to 3, the total number of

channels increased by a factor of 7/3.

The calculations in the above example are actually an idealization for several

reasons. Firstly, practical antennas have side lobes and cannot be used to focus a

transmitted beam into a perfect 120

o

sector or 60

o

sector. Due to this, additional

interference will be introduced. Next, it is also a cause of concern that a given

number of channels are not able to support as many subscribers when the pool

of channels is divided into small groups. This is due to a reduction in Trunking

Eﬃciency, a term which will be explained later on. Because sectoring involves using

more than one antenna per base station, the available channels in the cell are divided

and dedicated to a speciﬁc antenna. This breaks the available set of channels into

smaller sets, thus reducing the trunking eﬃciency. Moreover, dividing a cell into

sectors requires that a call in progress will have to be handed oﬀ (that is, assigned

a new channel) when a mobile unit travels into a new sector. This increases the

complexity of the system and also the load on the mobile switching center/base

station.

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3.7.4 Microcell Zone Concept

The increased number of handoﬀs required when sectoring is employed results in an

increased load on the switching and control link elements of the mobile system. To

overcome this problem, a new microcell zone concept has been proposed. As shown

in Figure 3.10, this scheme has a cell divided into three microcell zones, with each

of the three zone sites connected to the base station and sharing the same radio

equipment. It is necessary to note that all the microcell zones, within a cell, use the

same frequency used by that cell; that is no handovers occur between microcells.

Thus when a mobile user moves between two microcell zones of the cell, the BS

simply switches the channel to a diﬀerent zone site and no physical re-allotment of

channel takes place.

Locating the mobile unit within the cell: An active mobile unit sends a signal to all

zone sites, which in turn send a signal to the BS. A zone selector at the BS uses that

signal to select a suitable zone to serve the mobile unit - choosing the zone with the

strongest signal.

Base Station Signals: When a call is made to a cellular phone, the system already

knows the cell location of that phone. The base station of that cell knows in which

zone, within that cell, the cellular phone is located. Therefore when it receives the

signal, the base station transmits it to the suitable zone site. The zone site receives

the cellular signal from the base station and transmits that signal to the mobile

phone after ampliﬁcation. By conﬁning the power transmitted to the mobile phone,

co-channel interference is reduced between the zones and the capacity of system is

increased.

Beneﬁts of the micro-cell zone concept: 1) Interference is reduced in this case as

compared to the scheme in which the cell size is reduced.

2) Handoﬀs are reduced (also compared to decreasing the cell size) since the micro-

cells within the cell operate at the same frequency; no handover occurs when the

mobile unit moves between the microcells.

3) Size of the zone apparatus is small. The zone site equipment being small can be

mounted on the side of a building or on poles.

4) System capacity is increased. The new microcell knows where to locate the mo-

bile unit in a particular zone of the cell and deliver the power to that zone. Since

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Figure 3.10: The micro-cell zone concept.

the signal power is reduced, the microcells can be closer and result in an increased

system capacity. However, in a microcellular system, the transmitted power to a

mobile phone within a microcell has to be precise; too much power results in inter-

ference between microcells, while with too little power the signal might not reach

the mobile phone.This is a drawback of microcellular systems, since a change in the

surrounding (a new building, say, within a microcell) will require a change of the

transmission power.

3.8 Trunked Radio System

In the previous sections, we have discussed the frequency reuse plan, the design

trade-oﬀs and also explored certain capacity expansion techniques like cell-splitting

and sectoring. Now, we look at the relation between the number of radio channels

a cell contains and the number of users a cell can support. Cellular systems use

the concept of trunking to accommodate a large number of users in a limited radio

spectrum. It was found that a central oﬃce associated with say, 10,000 telephones

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requires about 50 million connections to connect every possible pair of users. How-

ever, a worst case maximum of 5000 connections need to be made among these

telephones at any given instant of time, as against the possible 50 million connec-

tions. In fact, only a few hundreds of lines are needed owing to the relatively short

duration of a call. This indicates that the resources are shared so that the number of

lines is much smaller than the number of possible connections. A line that connects

switching oﬃces and that is shared among users on an as-needed basis is called a

trunk.

The fact that the number of trunks needed to make connections between oﬃces

is much smaller than the maximum number that could be used suggests that at

times there might not be suﬃcient facilities to allow a call to be completed. A call

that cannot be completed owing to a lack of resources is said to be blocked. So one

important to be answered in mobile cellular systems is: How many channels per cell

are needed in a cellular telephone system to ensure a reasonably low probability that

a call will be blocked?

In a trunked radio system, a channel is allotted on per call basis. The perfor-

mance of a radio system can be estimated in a way by looking at how eﬃciently the

calls are getting connected and also how they are being maintained at handoﬀs.

Some of the important factors to take into consideration are (i) Arrival statistics,

(ii)Service statistics, (iii)Number of servers/channels.

Let us now consider the following assumptions for a buﬀerless system handling ’L’

users as shown in Figure 3.11:

(i) The number of users L is large when compared to 1.

(ii) Arrival statistics is Poisson distributed with a mean parameter λ.

(iii) Duration of a call is exponentially distributed with a mean rate µ

1

.

(iv) Residence time of each user is exponentially distributed with a rate parameter

µ

2

.

(v) The channel holding rate therefore is exponentially distributed with a parameter

µ = µ

1

+µ

2

.

(vi) There is a total of ’J’ number of channels (J ≤ L).

To analyze such a system, let us recapitulate a queuing system in brief. Consider an

M/M/m/m system which is an m-server loss system. The name M/M/m/m reﬂects

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Figure 3.11: The buﬀerless J-channel trunked radio system.

Figure 3.12: Discrete-time Markov chain for the M/M/J/J trunked radio system.

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standard queuing theory nomenclature whereby:

(i) the ﬁrst letter indicates the nature of arrival process(e.g. M stands for memory-

less which here means a Poisson process).

(ii) the second letter indicates the nature of probability distribution of service

times.(e.g M stands for exponential distribution). In all cases,successive inter ar-

rival times and service times are assumed to be statistically independent of each

other.

(iii) the third letter indicates the number of servers.

(iv) the last letter indicates that if an arrival ﬁnds all ’m’ users to be busy, then it

will not enter the system and is lost.

In view of the above, the buﬀerless system as shown in Figure 3.11 can be modeled

as M/M/J/J system and the discrete-time Markov chain of this system is shown in

Figure 3.12.

Trunking mainly exploits the statistical behavior of users so that a ﬁxed number

of channels can be used to accommodate a large, random user community. As the

number of telephone lines decrease, it becomes more likely that all channels are

busy for a particular user. As a result, the call gets rejected and in some systems,

a queue may be used to hold the caller’s request until a channel becomes available.

In the telephone system context the term Grade of Service (GoS) is used to mean

the probability that a user’s request for service will be blocked because a required

facility, such as a trunk or a cellular channel, is not available. For example, a GoS of

2 % implies that on the average a user might not be successful in placing a call on 2

out of every 100 attempts. In practice the blocking frequency varies with time. One

would expect far more call attempts during business hours than during the middle of

the night. Telephone operating companies maintain usage records and can identify a

”busy hour”, that is, the hour of the day during which there is the greatest demand

for service. Typically, telephone systems are engineered to provide a speciﬁed grade

of service during a speciﬁed busy hour.

User calling can be modeled statistically by two parameters: the average number

of call requests per unit time λ

user

and the average holding time H. The parameter

λ

user

is also called the average arrival rate, referring to the rate at which calls from

a single user arrive. The average holding time is the average duration of a call. The

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product:

A

user

= λ

user

H (3.26)

that is, the product of the average arrival rate and the average holding time–is called

the oﬀered traﬃc intensity or oﬀered load. This quantity represents the average

traﬃc that a user provides to the system. Oﬀered traﬃc intensity is a quantity that

is traditionally measured in Erlangs. One Erlang represents the amount of traﬃc

intensity carried by a channel that is completely occupied. For example, a channel

that is occupied for thirty minutes during an hour carries 0.5 Erlang of traﬃc.

Call arrivals or requests for service are modeled as a Poisson random process. It

is based on the assumption that there is a large pool of users who do not cooperate

in deciding when to place calls. Holding times are very well predicted using an

exponential probability distribution. This implies that calls of long duration are

much less frequent than short calls. If the traﬃc intensity oﬀered by a single user is

A

user

, then the traﬃc intensity oﬀered by N users is A = NA

user

. The purpose of

the statistical model is to relate the oﬀered traﬃc intensity A, the grade of service

P

b

, and the number of channels or trunks C needed to maintain the desired grade

of service.

Two models are widely used in traﬃc engineering to represent what happens

when a call is blocked. The blocked calls cleared model assumes that when a channel

or trunk is not available to service an arriving call, the call is cleared from the

system. The second model is known as blocked calls delayed. In this model a call

that cannot be serviced is placed on a queue and will be serviced when a channel or

trunk becomes available.

Use of the blocked-calls-cleared statistical model leads to the Erlang B formula

that relates oﬀered traﬃc intensity A, grade of service P

b

, and number of channels

K. The Erlang B formula is:

P

b

=

A

K

/K!

K

n=0

A

n

/n!

(3.27)

When the blocked-calls-delayed model is used, the ”grade of service” refers to the

probability that a call will be delayed. In this case the statistical model leads to the

Erlang C formula,

P[delay] =

A

K

/[(K −A)(K −1)]!

A

K

/[(K −A)(K −1)]! +

K

n=0

A

n

/n!

. (3.28)

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Ex. 6: In a certain cellular system, an average subscriber places two calls per

hour during a busy hour and the average holding time is 3 min. Each cell has 100

channels. If the blocked calls are cleared, how many subscribers can be serviced by

each cell at 2 % GoS?

Solution: Using Erlang B table, it can be seen that for C = 100 and GoS = P

b

= 2%,

the total oﬀered load A=87.972 Erlangs. Since an individual subscriber oﬀers a load

of A

user

= (2 calls / 60 min)3 min = 0.1 Erlang, the maximum number of subscribers

served is

N = A/A

user

= 87.972/0.1 ≈ 880. (3.29)

Ex. 4: In the previous example, suppose that the channels have been divided into

two groups of 50 channels each. Each subscriber is assigned to a group and can be

served only by that group. How many subscribers can be served by the two group

cell?

Solution: Using the Erlang B table with C = 50 and GOS = P

b

= 2%, the total

oﬀered load per group is

A = 40.255Erlangs (3.30)

Thus the maximum number of users per group is

N

group

= A/A

user

≈ 403. (3.31)

Thus, counting both the groups, maximum number of users in the two group cell is

806.

The above example indicates that the number of subscribers that can be sup-

ported by a given number of channels decreases as the pool of channels is sub-divided.

We can express this in terms of the trunking eﬃciency, deﬁned as the carrier load

per channel, that is,

ξ = (1 −P

b

)A/C. (3.32)

This explains why the sectoring of a cell into either 120

o

or 60

o

sectors reduces

the trunking eﬃciency of the system. Thus the system growth due to sectoring is

impacted by trunking eﬃciency considerations.

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3.9 References

1. T. S. Rappaport, Wireless Communications: Principles and Practice, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

2. K. Feher, Wireless Digital Communications: Modulation and Spread Spectrum

Applications. Upper Saddle River, NJ: Prentice Hall, 1995.

3. S. Haykin and M. Moher, Modern Wireless Communications. Singapore: Pear-

son Education, Inc., 2002.

4. J. W. Mark and W. Zhuang, Wireless Communications and Networking. New

Delhi: PHI, 2005.

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Chapter 4

Free Space Radio Wave

Propagation

4.1 Introduction

There are two basic ways of transmitting an electro-magnetic (EM) signal, through a

guided medium or through an unguided medium. Guided mediums such as coaxial

cables and ﬁber optic cables, are far less hostile toward the information carrying

EM signal than the wireless or the unguided medium. It presents challenges and

conditions which are unique for this kind of transmissions. A signal, as it travels

through the wireless channel, undergoes many kinds of propagation eﬀects such as

reﬂection, diﬀraction and scattering, due to the presence of buildings, mountains and

other such obstructions. Reﬂection occurs when the EM waves impinge on objects

which are much greater than the wavelength of the traveling wave. Diﬀraction

is a phenomena occurring when the wave interacts with a surface having sharp

irregularities. Scattering occurs when the medium through the wave is traveling

contains objects which are much smaller than the wavelength of the EM wave.

These varied phenomena’s lead to large scale and small scale propagation losses. Due

to the inherent randomness associated with such channels they are best described

with the help of statistical models. Models which predict the mean signal strength

for arbitrary transmitter receiver distances are termed as large scale propagation

models. These are termed so because they predict the average signal strength for

large Tx-Rx separations, typically for hundreds of kilometers.

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Figure 4.1: Free space propagation model, showing the near and far ﬁelds.

4.2 Free Space Propagation Model

Although EM signals when traveling through wireless channels experience fading

eﬀects due to various eﬀects, but in some cases the transmission is with a direct

line of sight such as in satellite communication. Free space model predicts that

the received power decays as negative square root of the distance. Friis free space

equation is given by

P

r

(d) =

P

t

G

t

G

r

λ

2

(4π)

2

d

2

L

(4.1)

where P

t

is the transmitted power, P

r

(d) is the received power, G

t

is the transmitter

antenna gain, G

r

is the receiver antenna gain, d is the Tx-Rx separation and L is the

system loss factor depended upon line attenuation, ﬁlter losses and antenna losses

and not related to propagation. The gain of the antenna is related to the eﬀective

aperture of the antenna which in turn is dependent upon the physical size of the

antenna as given below

G = 4πA

e

/λ

2

. (4.2)

The path loss, representing the attenuation suﬀered by the signal as it travels

through the wireless channel is given by the diﬀerence of the transmitted and re-

ceived power in dB and is expressed as:

PL(dB) = 10 log P

t

/P

r

. (4.3)

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The ﬁelds of an antenna can broadly be classiﬁed in two regions, the far ﬁeld and

the near ﬁeld. It is in the far ﬁeld that the propagating waves act as plane waves

and the power decays inversely with distance. The far ﬁeld region is also termed

as Fraunhofer region and the Friis equation holds in this region. Hence, the Friis

equation is used only beyond the far ﬁeld distance, d

f

, which is dependent upon the

largest dimension of the antenna as

d

f

= 2D

2

/λ. (4.4)

Also we can see that the Friis equation is not deﬁned for d=0. For this reason, we

use a close in distance, d

o

, as a reference point. The power received, P

r

(d), is then

given by:

P

r

(d) = P

r

(d

o

)(d

o

/d)

2

. (4.5)

Ex. 1: Find the far ﬁeld distance for a circular antenna with maximum dimension

of 1 m and operating frequency of 900 MHz.

Solution: Since the operating frequency f = 900 Mhz, the wavelength

λ =

3 10

8

m/s

900 10

6

Hz

m

. Thus, with the largest dimension of the antenna, D=1m, the far ﬁeld distance is

d

f

=

2D

2

λ

=

2(1)

2

0.33

= 6m

.

Ex. 2: A unit gain antenna with a maximum dimension of 1 m produces 50 W

power at 900 MHz. Find (i) the transmit power in dBm and dB, (ii) the received

power at a free space distance of 5 m and 100 m.

Solution:

(i) Tx power = 10log(50) = 17 dB = (17+30) dBm = 47 dBm

(ii) d

f

=

2×D

2

λ

=

2×1

2

1/3

= 6m

Thus the received power at 5 m can not be calculated using free space distance

formula.

At 100 m ,

P

R

=

P

T

G

T

G

R

λ

2

4πd

2

=

50 1 (1/3)

2

4π100

2

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= 3.5 10

−3

mW

P

R

(dBm) = 10logP

r

(mW) = −24.5dBm

4.3 Basic Methods of Propagation

Reﬂection, diﬀraction and scattering are the three fundamental phenomena that

cause signal propagation in a mobile communication system, apart from LoS com-

munication. The most important parameter, predicted by propagation models based

on above three phenomena, is the received power. The physics of the above phe-

nomena may also be used to describe small scale fading and multipath propagation.

The following subsections give an outline of these phenomena.

4.3.1 Reﬂection

Reﬂection occurs when an electromagnetic wave falls on an object, which has very

large dimensions as compared to the wavelength of the propagating wave. For ex-

ample, such objects can be the earth, buildings and walls. When a radio wave falls

on another medium having diﬀerent electrical properties, a part of it is transmitted

into it, while some energy is reﬂected back. Let us see some special cases. If the

medium on which the e.m. wave is incident is a dielectric, some energy is reﬂected

back and some energy is transmitted. If the medium is a perfect conductor, all

energy is reﬂected back to the ﬁrst medium. The amount of energy that is reﬂected

back depends on the polarization of the e.m. wave.

Another particular case of interest arises in parallel polarization, when no re-

ﬂection occurs in the medium of origin. This would occur, when the incident angle

would be such that the reﬂection coeﬃcient is equal to zero. This angle is the

Brewster’s angle. By applying laws of electro-magnetics, it is found to be

sin(θ

B

) =

_

1

1

+

2

. (4.6)

Further, considering perfect conductors, the electric ﬁeld inside the conductor is

always zero. Hence all energy is reﬂected back. Boundary conditions require that

θ

i

= θ

r

(4.7)

and

E

i

= E

r

(4.8)

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for vertical polarization, and

E

i

= −E

r

(4.9)

for horizontal polarization.

4.3.2 Diﬀraction

Diﬀraction is the phenomenon due to which an EM wave can propagate beyond the

horizon, around the curved earth’s surface and obstructions like tall buildings. As

the user moves deeper into the shadowed region, the received ﬁeld strength decreases.

But the diﬀraction ﬁeld still exists an it has enough strength to yield a good signal.

This phenomenon can be explained by the Huygen’s principle, according to

which, every point on a wavefront acts as point sources for the production of sec-

ondary wavelets, and they combine to produce a new wavefront in the direction of

propagation. The propagation of secondary wavelets in the shadowed region results

in diﬀraction. The ﬁeld in the shadowed region is the vector sum of the electric ﬁeld

components of all the secondary wavelets that are received by the receiver.

4.3.3 Scattering

The actual received power at the receiver is somewhat stronger than claimed by the

models of reﬂection and diﬀraction. The cause is that the trees, buildings and lamp-

posts scatter energy in all directions. This provides extra energy at the receiver.

Roughness is tested by a Rayleigh criterion, which deﬁnes a critical height h

c

of

surface protuberances for a given angle of incidence θ

i

, given by,

h

c

=

λ

8sinθ

i

. (4.10)

A surface is smooth if its minimum to maximum protuberance h is less than h

c

,

and rough if protuberance is greater than h

c

. In case of rough surfaces, the surface

reﬂection coeﬃcient needs to be multiplied by a scattering loss factor ρ

S

, given by

ρ

S

= exp(−8(

πσ

h

sinθ

i

λ

)

2

) (4.11)

where σ

h

is the standard deviation of the Gaussian random variable h. The following

result is a better approximation to the observed value

ρ

S

= exp(−8(

πσ

h

sinθ

i

λ

)

2

)I

0

[−8(

πσ

h

sinθ

i

λ

)

2

] (4.12)

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Figure 4.2: Two-ray reﬂection model.

which agrees very well for large walls made of limestone. The equivalent reﬂection

coeﬃcient is given by,

Γ

rough

= ρ

S

Γ. (4.13)

4.4 Two Ray Reﬂection Model

Interaction of EM waves with materials having diﬀerent electrical properties than

the material through which the wave is traveling leads to transmitting of energy

through the medium and reﬂection of energy back in the medium of propagation.

The amount of energy reﬂected to the amount of energy incidented is represented

by Fresnel reﬂection coeﬃcient Γ, which depends upon the wave polarization, angle

of incidence and frequency of the wave. For example, as the EM waves can not pass

through conductors, all the energy is reﬂected back with angle of incidence equal to

the angle of reﬂection and reﬂection coeﬃcient Γ = −1. In general, for parallel and

perpendicular polarizations, Γ is given by:

Γ

||

= E

r

/E

i

= η

2

sin θ

t

−η

1

sin θ

i

/η

2

sin θ

t

+η

1

sin θ

i

(4.14)

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Γ

⊥

= E

r

/E

i

= η

2

sin θ

i

−η

1

sin θ

t

/η

2

sin θ

i

+η

1

sin θ

t

. (4.15)

Seldom in communication systems we encounter channels with only LOS paths and

hence the Friis formula is not a very accurate description of the communication link.

A two-ray model, which consists of two overlapping waves at the receiver, one direct

path and one reﬂected wave from the ground gives a more accurate description as

shown in Figure 4.2. A simple addition of a single reﬂected wave shows that power

varies inversely with the forth power of the distance between the Tx and the Rx.

This is deduced via the following treatment. From Figure 4.2, the total transmitted

and received electric ﬁelds are

E

TOT

T

= E

i

+E

LOS

, (4.16)

E

TOT

R

= E

g

+E

LOS

. (4.17)

Let E

0

is the free space electric ﬁeld (in V/m) at a reference distance d

0

. Then

E(d, t) =

E

0

d

0

d

cos(ω

c

t −φ) (4.18)

where

φ = ω

c

d

c

(4.19)

and d > d

0

. The envelop of the electric ﬁeld at d meters from the transmitter at

any time t is therefore

[E(d, t)[ =

E

0

d

0

d

. (4.20)

This means the envelop is constant with respect to time.

Two propagating waves arrive at the receiver, one LOS wave which travels a

distance of d

and another ground reﬂected wave, that travels d

. Mathematically,

it can be expressed as:

E(d

, t) =

E

0

d

0

d

cos(ω

c

t −φ

) (4.21)

where

φ

= ω

c

d

c

(4.22)

and

E(d

, t) =

E

0

d

0

d

cos(ω

c

t −φ

) (4.23)

where

φ

= ω

c

d

c

. (4.24)

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Figure 4.3: Phasor diagram of electric

ﬁelds.

Figure 4.4: Equivalent phasor diagram of

Figure 4.3.

According to the law of reﬂection in a dielectric, θ

i

= θ

0

and E

g

= ΓE

i

which means

the total electric ﬁeld,

E

t

= E

i

+E

g

= E

i

(1 + Γ). (4.25)

For small values of θ

i

, reﬂected wave is equal in magnitude and 180

o

out of phase

with respect to incident wave. Assuming perfect horizontal electric ﬁeld polarization,

i.e.,

Γ

⊥

= −1 =⇒E

t

= (1 −1)E

i

= 0, (4.26)

the resultant electric ﬁeld is the vector sum of E

LOS

and E

g

. This implies that,

E

TOT

R

= [E

LOS

+E

g

[. (4.27)

It can be therefore written that

E

TOT

R

(d, t) =

E

0

d

0

d

cos(ω

c

t −φ

) + (−1)

E

0

d

0

d

cos(ω

c

t −φ

) (4.28)

In such cases, the path diﬀerence is

∆ = d

−d

=

_

(h

t

+h

r

)

2

+d

2

−

_

(h

t

−h

r

)

2

+d

2

. (4.29)

However, when T-R separation distance is very large compared to (h

t

+h

r

), then

∆ ≈

2h

t

h

r

d

(4.30)

Ex 3: Prove the above two equations, i.e., equation (4.29) and (4.30).

Once the path diﬀerence is known, the phase diﬀerence is

θ

∆

=

2π∆

λ

=

∆ω

c

λ

(4.31)

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and the time diﬀerence,

τ

d

=

∆

c

=

θ

∆

2πf

c

. (4.32)

When d is very large, then ∆ becomes very small and therefore E

LOS

and E

g

are

virtually identical with only phase diﬀerence,i.e.,

[

E

0

d

0

d

[ ≈ [

E

0

d

0

d

[ ≈ [

E

0

d

0

d

[. (4.33)

Say, we want to evaluate the received E-ﬁeld at any t =

d

c

. Then,

E

TOT

R

(d, t =

d

c

) =

E

0

d

0

d

cos(ω

c

d

c

−ω

c

d

c

) −

E

0

d

0

d

cos(ω

c

d

c

−ω

c

d

c

) (4.34)

=

E

0

d

0

d

cos(

∆ω

c

c

) −

E

0

d

0

d

cos(0

o

) (4.35)

=

E

0

d

0

d

θ

∆

−

E

0

d

0

d

(4.36)

≈

E

0

d

0

d

(

θ

∆

−1). (4.37)

Using phasor diagram concept for vector addition as shown in Figures 4.3 and 4.4,

we get

[E

TOT

R

(d)[ =

¸

(

E

0

d

0

d

+

E

0

d

0

d

cos(θ

∆

))

2

+ (

E

0

d

0

d

sin(θ

∆

))

2

(4.38)

=

E

0

d

0

d

_

(cos(θ

∆

) −1)

2

+sin

2

(θ

∆

) (4.39)

=

E

0

d

0

d

_

2 −2cosθ

∆

(4.40)

= 2

E

0

d

0

d

sin(

θ

∆

2

). (4.41)

For

θ

∆

2

< 0.5rad, sin(

θ

∆

2

) ≈

θ

∆

2

. Using equation (4.31) and further equation (4.30),

we can then approximate that

sin(

θ

∆

2

) ≈

π

λ

∆ =

2πh

t

h

r

λd

< 0.5rad. (4.42)

This raises the wonderful concept of ‘cross-over distance’ d

c

, deﬁned as

d > d

c

=

20πh

t

h

r

5λ

=

4πh

t

h

r

λ

. (4.43)

The corresponding approximate received electric ﬁeld is

E

TOT

R

(d) ≈ 2

E

0

d

0

d

2πh

t

h

r

λd

= k

h

t

h

r

d

2

. (4.44)

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Therefore, using equation (4.43) in (4.1), we get the received power as

P

r

=

P

t

G

t

G

r

h

2

t

h

2

r

Ld

4

. (4.45)

The cross-over distance shows an approximation of the distance after which the

received power decays with its fourth order. The basic diﬀerence between equation

(4.1) and (4.45) is that when d < d

c

, equation (4.1) is suﬃcient to calculate the

path loss since the two-ray model does not give a good result for a short distance

due to the oscillation caused by the constructive and destructive combination of the

two rays, but whenever we distance crosses the ‘cross-over distance’, the power falls

oﬀ rapidly as well as two-ray model approximation gives better result than Friis

equation.

Observations on Equation (4.45): The important observations from this

equation are:

1. This equation gives fair results when the T-R separation distance crosses the

cross-over distance.

1. In that case, the power decays as the fourth power of distance

P

r

(d) =

K

d

4

, (4.46)

with K being a constant.

2. Path loss is independent of frequency (wavelength).

3. Received power is also proportional to h

2

t

and h

2

r

, meaning, if height of any of the

antennas is increased, received power increases.

4.5 Diﬀraction

Diﬀraction is the phenomena that explains the digression of a wave from a straight

line path, under the inﬂuence of an obstacle, so as to propagate behind the obstacle.

It is an inherent feature of a wave be it longitudinal or transverse. For e.g the

sound can be heard in a room, where the source of the sound is another room

without having any line of sight. The similar phenomena occurs for light also but

the diﬀracted light intensity is not noticeable. This is because the obstacle or slit

need to be of the order of the wavelength of the wave to have a signiﬁcant eﬀect.

Thus radiation from a point source radiating in all directions can be received at any

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Figure 4.5: Huygen’s secondary wavelets.

point, even behind an obstacle (unless it is not completely enveloped by it), as shown

in Figure 4.5. Though the intensity received gets smaller as receiver is moved into the

shadowed region. Diﬀraction is explained by Huygens-Fresnel principle which states

that all points on a wavefront can be considered as the point source for secondary

wavelets which form the secondary wavefront in the direction of the prorogation.

Normally, in absence of an obstacle, the sum of all wave sources is zero at a point

not in the direct path of the wave and thus the wave travels in the straight line. But

in the case of an obstacle, the eﬀect of wave source behind the obstacle cannot be

felt and the sources around the obstacle contribute to the secondary wavelets in the

shadowed region, leading to bending of wave. In mobile communication, this has a

great advantage since, by diﬀraction (and scattering, reﬂection), the receiver is able

to receive the signal even when not in line of sight of the transmitter. This we show

in the subsection given below.

4.5.1 Knife-Edge Diﬀraction Geometry

As shown in Figure 4.6, consider that there’s an impenetrable obstruction of hight

h at a distance of d

1

from the transmitter and d

2

from the receiver. The path

diﬀerence between direct path and the diﬀracted path is

δ =

_

d

2

1

+h

2

+

_

d

2

2

+h

2

−(d

1

+d

2

) (4.47)

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Figure 4.6: Diﬀraction through a sharp edge.

which can be further simpliﬁed as

δ = d

1

(1 +h

2

/2d

2

1

) +d

2

(1 +h

2

/2d

2

2

) −(d

1

+d

2

)

= h

2

/(2d

1

) +h

2

/(2d

2

) = h

2

(d

1

+d

2

)/(2d

1

d

2

). (4.48)

Thus the phase diﬀerence equals

φ = 2πδ/λ = 2πh

2

(d

1

+d

2

)/λ2(d

1

d

2

). (4.49)

With the following considerations that

α = β +γ (4.50)

and

α ≈ tanα (4.51)

we can write,

αtanα = tanβ + tanγ = h/d

1

+h/d

2

= h(d

1

+d

2

)/d

1

d

2

. (4.52)

In order to normalize this, we usually use a Fresnel-Kirchoﬀ diﬀraction parameter

v, expressed as

v = h

_

2(d

1

+d

2

)/(λd

1

d

2

) = α

_

(2d

1

d

2

)/(λ(d

1

+d

2

)) (4.53)

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Figure 4.7: Fresnel zones.

and therefore the phase diﬀerence becomes

φ = πv

2

/2. (4.54)

From this, we can observe that: (i) phase diﬀerence is a function of the height of

the obstruction, and also, (ii) phase diﬀerence is a function of the position of the

obstruction from transmitter and receiver.

4.5.2 Fresnel Zones: the Concept of Diﬀraction Loss

As mentioned before, the more is the object in the shadowed region greater is the

diﬀraction loss of the signal. The eﬀect of diﬀraction loss is explained by Fresnel

zones as a function of the path diﬀerence. The successive Fresnel zones are limited

by the circular periphery through which the path diﬀerence of the secondary waves

is nλ/2 greater than total length of the LOS path, as shown in Figure 4.7. Thus

successive Fresnel zones have phase diﬀerence of π which means they alternatively

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provide constructive and destructive interference to the received the signal. The

radius of the each Fresnel zone is maximum at middle of transmitter and receiver

(i.e. when d

1

= d

2

) and decreases as moved to either side. It is seen that the loci

of a Fresnel zone varied over d1 and d2 forms an ellipsoid with the transmitter and

receiver at its focii. Now, if there’s no obstruction, then all Fresnel zones result in

only the direct LOS prorogation and no diﬀraction eﬀects are observed. But if an

obstruction is present, depending on its geometry, it obstructs contribution from

some of the secondary wavelets, resulting in diﬀraction and also the loss of energy,

which is the vector sum of energy from unobstructed sources. please note that height

of the obstruction can be positive zero and negative also. The diﬀraction losses are

minimum as long as obstruction doesn’t block volume of the 1st Fresnel zone. As a

rule of thumb, diﬀraction eﬀects are negligible beyond 55% of 1st Fresnel zone.

Ex 4: Calculate the ﬁrst Fresnel zone obstruction height maximum for f = 800

MHz.

Solution:

λ =

c

f

=

3 10

8

8 10

2

10

6

=

3

8

m

H =

_

λ(d

1

+d

2

)

d

1

+d

2

H

1

=

_

3

8

250×250

500

= 6.89m

Thus H

1

= 10 + 6.89 = 16.89m

(b)

H

2

=

¸

3

8

100 400

500

= 10

_

(0.3) = 5.48m

Thus

H

2

= 10 + 5.6 = 15.48m

. To have good power strength, obstacle should be within the 60% of the ﬁrst fresnel

zone.

Ex 5: Given f=900 MHz, d

1

= d

2

= 1 km, h = 25m, where symbols have usual

meaning. Compute the diﬀraction loss. Also ﬁnd out in which Fresnel zone the tip

of the obstruction lies.

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Figure 4.8: Knife-edge Diﬀraction Model

Given,

G

d

(dB) = 20 log(0.5 −0.62v) −1 < v <= 0

G

d

(dB) = 20 log(0.225/v) v > 2.24

Solution:

v = h

¸

2(d

1

+d

2

)

λd

1

d

2

= 25

¸

2 2000

1

3

10

= 2.74

G

d

(dB) = 20 log(

225

v

) = −21.7dB

Since loss = -G

d

(dB) = 21.7 dB

n =

(2.74)

2

2

= 3.5

Thus n=4.

4.5.3 Knife-edge diﬀraction model

Knife-edge diﬀraction model is one of the simplest diﬀraction model to estimate the

diﬀraction loss. It considers the object like hill or mountain as a knife edge sharp

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object. The electric ﬁeld strength, E

d

of a knife-edge diﬀracted wave is given by

E

d

/E

o

= F(v) = (1 +j)/2

_

∞

v

(exp((−jπt

2

)/2)dt. (4.55)

The diﬀraction gain due to presence of knife edge can be given as

G

d

(db) = 20log[F(v)[ (4.56)

G

d

(db) = 0v <= −1 (4.57)

G

d

(db) = 20log(0.5 −0.62) −1 <= v <= 0 (4.58)

G

d

(db) = 20log(0.5exp(−0.95v)) 0 <= v <= 1 (4.59)

G

d

(db) = 20log(0.4 −sqrt(0.1184 −(0.38 −0.1v

2

))) 1 <= v <= 2.4 (4.60)

G

d

(db) = 20log(0.225/v) v > 2.4 (4.61)

When there are more than one obstruction, then the equivalent model can be found

by one knife-edge diﬀraction model as shown in Figure 4.8.

4.6 Link Budget Analysis

4.6.1 Log-distance Path Loss Model

According to this model the received power at distance d is given by,

PL(d)(

d

d

0

)

n

=⇒PL(dB) = PL(d

0

) + 10nlog(

d

d

0

) (4.62)

The value of n varies with propagation environments. The value of n is 2 for free

space. The value of n varies from 4 to 6 for obstruction of building, and 3 to 5 for

urban scenarios. The important factor is to select the correct reference distance d

0

.

For large cell area it is 1 Km, while for micro-cell system it varies from 10m-1m.

Limitations:

Surrounding environmental clutter may be diﬀerent for two locations having

the same transmitter to receiver separation. Moreover it does not account for the

shadowing eﬀects.

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4.6.2 Log Normal Shadowing

The equation for the log normal shadowing is given by,

PL(dB) = PL(dB) +X

σ

= PL(d

0

) + 10nlog(

d

d

0

) +X

σ

(4.63)

where X

σ

is a zero mean Gaussian distributed random variable in dB with standard

deviation σ also in dB. In practice n and σ values are computed from measured

data.

Average received power

The ‘Q’ function is given by,

Q(z) = 0.5(1 −erf(

z

√

2

)) (4.64)

and

Q(z) = 1 −Q(−z) (4.65)

So the probability that the received signal level (in dB) will exceed a certain value

γ is

P(P

d

> γ) = Q(

γ −P

r

σ

). (4.66)

4.7 Outdoor Propagation Models

There are many empirical outdoor propagation models such as Longley-Rice model,

Durkin’s model, Okumura model, Hata model etc. Longley-Rice model is the most

commonly used model within a frequency band of 40 MHz to 100 GHz over diﬀerent

terrains. Certain modiﬁcations over the rudimentary model like an extra urban

factor (UF) due to urban clutter near the reciever is also included in this model.

Below, we discuss some of the outdoor models, followed by a few indoor models too.

4.7.1 Okumura Model

The Okumura model is used for Urban Areas is a Radio propagation model that is

used for signal prediction.The frequency coverage of this model is in the range of

200 MHz to 1900 MHz and distances of 1 Km to 100 Km.It can be applicable for

base station eﬀective antenna heights (h

t

) ranging from 30 m to 1000 m.

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Okumura used extensive measurements of base station-to-mobile signal attenua-

tion throughout Tokyo to develop a set of curves giving median attenuation relative

to free space (A

mu

) of signal propogation in irregular terrain. The empirical path-

loss formula of Okumura at distance d parameterized by the carrier frequency f

c

is

given by

P

L

(d)dB = L(f

c

, d) +A

mu

(f

c

, d) −G(h

t

) −G(h

r

) −G

AREA

(4.67)

where L(f

c

, d) is free space path loss at distance d and carrier frequency f

c

, A

mu

(f

c

, d)

is the median attenuation in addition to free-space path loss across all environments,G(h

t

)

is the base station antenna height gain factor,G(h

r

) is the mobile antenna height gain

factor,G

AREA

is the gain due to type of environment. The values of A

mu

(f

c

, d) and

G

AREA

are obtained from Okumura’s empirical plots. Okumura derived empirical

formulas for G(h

t

) and G(h

r

) as follows:

G(h

t

) = 20 log

10

(h

t

/200), 30m < h

t

< 1000m (4.68)

G(h

r

) = 10 log

10

(h

r

/3), h

r

≤ 3m (4.69)

G(h

r

) = 20 log

10

(h

r

/3), 3m < h

r

< 10m (4.70)

Correlation factors related to terrain are also developed in order to improve the

models accuracy. Okumura’s model has a 10-14 dB empirical standard deviation

between the path loss predicted by the model and the path loss associated with one

of the measurements used to develop the model.

4.7.2 Hata Model

The Hata model is an empirical formulation of the graphical path-loss data provided

by the Okumura and is valid over roughly the same range of frequencies, 150-1500

MHz. This empirical formula simpliﬁes the calculation of path loss because it is

closed form formula and it is not based on empirical curves for the diﬀerent param-

eters. The standard formula for empirical path loss in urban areas under the Hata

model is

P

L,urban

(d)dB = 69.55+26.16 log

10

(f

c

)−13.82 log

10

(h

t

)−a(h

r

)+(44.9−6.55 log

10

(h

t

)) log

10

(d)

(4.71)

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The parameters in this model are same as in the Okumura model,and a(h

r

) is a

correction factor for the mobile antenna height based on the size of coverage area.For

small to medium sized cities this factor is given by

a(h

r

) = (1.11 log

10

(f

c

) −0.7)h

r

−(1.56 log

10

(f

c

) −0.8)dB

and for larger cities at a frequencies f

c

> 300 MHz by

a(h

r

) = 3.2(log

10

(11.75h

r

))

2

−4.97dB

else it is

a(h

r

) = 8.29(log

10

(1.54h

r

))

2

−1.1dB

Corrections to the urban model are made for the suburban, and is given by

P

L,suburban

(d)dB = P

L,urban

(d)dB −2(log

10

(f

c

/28))

2

−5.4 (4.72)

Unlike the Okumura model,the Hata model does not provide for any speciﬁc path-

correlation factors. The Hata model well approximates the Okumura model for

distances d > 1 Km. Hence it is a good model for ﬁrst generation cellular systems,

but it does not model propogation well in current cellular systems with smaller cell

sizes and higher frequencies. Indoor environments are also not captured by the Hata

model.

4.8 Indoor Propagation Models

The indoor radio channel diﬀers from the traditional mobile radio channel in ways

- the distances covered are much smaller ,and the variability of the environment

is much greater for smaller range of Tx-Rx separation distances.Features such as

lay-out of the building,the construction materials,and the building type strongly in-

ﬂuence the propagation within the building.Indoor radio propagation is dominated

by the same mechanisms as outdoor: reﬂection, diﬀraction and scattering with vari-

able conditions. In general,indoor channels may be classiﬁed as either line-of-sight

or obstructed.

4.8.1 Partition Losses Inside a Floor (Intra-ﬂoor)

The internal and external structure of a building formed by partitions and obstacles

vary widely.Partitions that are formed as a part of building structure are called

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hard partitions , and partitions that may be moved and which do not span to

the ceiling are called soft partitions. Partitions vary widely in their physical and

electrical characteristics,making it diﬃcult to apply general models to speciﬁc indoor

installations.

4.8.2 Partition Losses Between Floors (Inter-ﬂoor)

The losses between ﬂoors of a building are determined by the external dimensions

and materials of the building,as well as the type of construction used to create the

ﬂoors and the external surroundings. Even the number of windows in a building

and the presence of tinting can impact the loss between ﬂoors.

4.8.3 Log-distance Path Loss Model

It has been observed that indoor path loss obeys the distance power law given by

PL(dB) = PL(d

0

) + 10nlog

10

(d/d

0

) +X

σ

(4.73)

where n depends on the building and surrounding type, and X

σ

represents a normal

random variable in dB having standard deviation of σ dB.

4.9 Summary

In this chapter, three principal propagation models have been identiﬁed: free-space

propagation, reﬂection and diﬀraction, which are common terrestrial models and

these mainly explains the large scale path loss. Regarding path-loss, one important

factor introduced in this chapter is log-distance path loss model. These, however,

may be insigniﬁcant when we consider the small-scale rapid path losses. This is

discussed in the next chapter.

4.10 References

1. T. S. Rappaport, Wireless Communications: Principles and Practice, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

2. S. Haykin and M. Moher, Modern Wireless Communications. Singapore: Pear-

son Education, Inc., 2002.

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3. J. W. Mark and W. Zhuang, Wireless Communications and Networking. New

Delhi: PHI, 2005.

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Chapter 5

Multipath Wave Propagation

and Fading

5.1 Multipath Propagation

In wireless telecommunications, multipath is the propagation phenomenon that re-

sults in radio signals reaching the receiving antenna by two or more paths. Causes

of multipath include atmospheric ducting, ionospheric reﬂection and refraction, and

reﬂection from water bodies and terrestrial objects such as mountains and buildings.

The eﬀects of multipath include constructive and destructive interference, and phase

shifting of the signal. In digital radio communications (such as GSM) multipath can

cause errors and aﬀect the quality of communications. We discuss all the related

issues in this chapter.

5.2 Multipath & Small-Scale Fading

Multipath signals are received in a terrestrial environment, i.e., where diﬀerent forms

of propagation are present and the signals arrive at the receiver from transmitter via

a variety of paths. Therefore there would be multipath interference, causing multi-

path fading. Adding the eﬀect of movement of either Tx or Rx or the surrounding

clutter to it, the received overall signal amplitude or phase changes over a small

amount of time. Mainly this causes the fading.

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5.2.1 Fading

The term fading, or, small-scale fading, means rapid ﬂuctuations of the amplitudes,

phases, or multipath delays of a radio signal over a short period or short travel

distance. This might be so severe that large scale radio propagation loss eﬀects

might be ignored.

5.2.2 Multipath Fading Eﬀects

In principle, the following are the main multipath eﬀects:

1. Rapid changes in signal strength over a small travel distance or time interval.

2. Random frequency modulation due to varying Doppler shifts on diﬀerent mul-

tipath signals.

3. Time dispersion or echoes caused by multipath propagation delays.

5.2.3 Factors Inﬂuencing Fading

The following physical factors inﬂuence small-scale fading in the radio propagation

channel:

(1) Multipath propagation – Multipath is the propagation phenomenon that re-

sults in radio signals reaching the receiving antenna by two or more paths.

The eﬀects of multipath include constructive and destructive interference, and

phase shifting of the signal.

(2) 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.

(3) Speed of surrounding objects – If objects in the radio channel are in mo-

tion, 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 fading.

(4) Transmission Bandwidth of the signal – If the transmitted radio signal

bandwidth is greater than the “bandwidth” of the multipath channel (quanti-

ﬁed by coherence bandwidth), the received signal will be distorted.

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5.3 Types of Small-Scale Fading

The type of fading experienced by the signal through a mobile channel depends

on the relation between the signal parameters (bandwidth, symbol period) and the

channel parameters (rms delay spread and Doppler spread). Hence we have four

diﬀerent types of fading. There are two types of fading due to the time dispersive

nature of the channel.

5.3.1 Fading Eﬀects due to Multipath Time Delay Spread

Flat Fading

Such types of fading occurs when the bandwidth of the transmitted signal is less than

the coherence bandwidth of the channel. Equivalently if the symbol period of the

signal is more than the rms delay spread of the channel, then the fading is ﬂat fading.

So we can say that ﬂat fading occurs when

B

S

¸B

C

(5.1)

where B

S

is the signal bandwidth and B

C

is the coherence bandwidth. Also

T

S

¸σ

τ

(5.2)

where T

S

is the symbol period and σ

τ

is the rms delay spread. And in such a case,

mobile channel has a constant gain and linear phase response over its bandwidth.

Frequency Selective Fading

Frequency selective fading occurs when the signal bandwidth is more than the co-

herence bandwidth of the mobile radio channel or equivalently the symbols duration

of the signal is less than the rms delay spread.

B

S

¸B

C

(5.3)

and

T

S

¸σ

τ

(5.4)

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At the receiver, we obtain multiple copies of the transmitted signal, all attenuated

and delayed in time. The channel introduces inter symbol interference. A rule of

thumb for a channel to have ﬂat fading is if

σ

τ

T

S

≤ 0.1 (5.5)

5.3.2 Fading Eﬀects due to Doppler Spread

Fast Fading

In a fast fading channel, the channel impulse response changes rapidly within the

symbol duration of the signal. Due to Doppler spreading, signal undergoes frequency

dispersion leading to distortion. Therefore a signal undergoes fast fading if

T

S

¸T

C

(5.6)

where T

C

is the coherence time and

B

S

¸B

D

(5.7)

where B

D

is the Doppler spread. Transmission involving very low data rates suﬀer

from fast fading.

Slow Fading

In such a channel, the rate of the change of the channel impulse response is much

less than the transmitted signal. We can consider a slow faded channel a channel in

which channel is almost constant over atleast one symbol duration. Hence

T

S

¸T

C

(5.8)

and

B

S

¸B

D

(5.9)

We observe that the velocity of the user plays an important role in deciding whether

the signal experiences fast or slow fading.

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Figure 5.1: Illustration of Doppler eﬀect.

5.3.3 Doppler Shift

The Doppler eﬀect (or Doppler shift) is the change in frequency of a wave for an

observer moving relative to the source of the wave. In classical physics (waves in

a medium), the relationship between the observed frequency f and the emitted

frequency f

o

is given by:

f =

_

v ±v

r

v ±v

s

_

f

0

(5.10)

where v is the velocity of waves in the medium, v

s

is the velocity of the source

relative to the medium and v

r

is the velocity of the receiver relative to the medium.

In mobile communication, the above equation can be slightly changed according

to our convenience since the source (BS) is ﬁxed and located at a remote elevated

level from ground. The expected Doppler shift of the EM wave then comes out to

be ±

v

r

c

f

o

or, ±

v

r

λ

. As the BS is located at an elevated place, a cos φ factor would

also be multiplied with this. The exact scenario, as given in Figure 5.1, is illustrated

below.

Consider a mobile moving at a constant velocity v, along a path segment length

d between points A and B, while it receives signals from a remote BS source S. The

diﬀerence in path lengths traveled by the wave from source S to the mobile at points

A and B is ∆l = d cos θ = v∆t cos θ, where ∆t is the time required for the mobile

to travel from A to B, and θ is assumed to be the same at points A and B since the

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source is assumed to be very far away. The phase change in the received signal due

to the diﬀerence in path lengths is therefore

∆ϕ =

2π∆l

λ

=

2πv∆t

λ

cos θ (5.11)

and hence the apparent change in frequency, or Doppler shift (f

d

) is

f

d

=

1

2π

.

∆ϕ

∆t

=

v

λ

. cos θ. (5.12)

Example 1

An aircraft is heading towards a control tower with 500 kmph, at an elevation of

20

◦

. Communication between aircraft and control tower occurs at 900 MHz. Find

out the expected Doppler shift.

Solution As given here,

v = 500 kmph

the horizontal component of the velocity is

v

**= v cos θ = 500 cos 20
**

◦

= 130 m/s

Hence, it can be written that

λ =

900 10

6

3 10

8

=

1

3

m

f

d

=

130

1

/

3

= 390 Hz

If the plane banks suddenly and heads for other direction, the Doppler shift change

will be 390 Hz to −390 Hz.

5.3.4 Impulse Response Model of a Multipath Channel

Mobile radio channel may be modeled as a linear ﬁlter with time varying impulse

response in continuous time. To show this, consider time variation due to receiver

motion and time varying impulse response h(d, t) and x(t), the transmitted signal.

The received signal y(d, t) at any position d would be

y(d, t) = x(t) ∗ h(d, t) =

_

∞

−∞

x(τ) h(d, t −τ) dτ (5.13)

For a causal system: h(d, t) = 0, for t < 0 and for a stable system

_

∞

−∞

[h(d, t)[ dt <

∞

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Applying causality condition in the above equation, h(d, t − τ) = 0 for t − τ < 0

⇒τ > t, i.e., the integral limits are changed to

y(d, t) =

_

t

−∞

x(τ) h(d, t −τ) dτ.

Since the receiver moves along the ground at a constant velocity v, the position of

the receiver is d = vt, i.e.,

y(vt, t) =

_

t

−∞

x(τ) h(vt, t −τ) dτ.

Since v is a constant, y(vt, t) is just a function of t. Therefore the above equation

can be expressed as

y(t) =

_

t

−∞

x(τ) h(vt, t −τ) dτ = x(t) ∗ h(vt, t) = x(t) ∗ h(d, t) (5.14)

It is useful to discretize the multipath delay axis τ of the impulse response into equal

time delay segments called excess delay bins, each bin having a time delay width

equal to ( τ

i+1

−τ

i

) = ∆τ and τ

i

= i∆τ for i ∈ ¦0, 1, 2, ..N −1¦, where N represents

the total number of possible equally-spaced multipath components, including the

ﬁrst arriving component. The useful frequency span of the model is

2

/

∆τ

. The

model may be used to analyze transmitted RF signals having bandwidth less than

2

/

∆τ

.

If there are N multipaths, maximum excess delay is given by N∆τ.

¦y(t) = x(t) ∗ h(t, τ

i

)[i = 0, 1, ...N −1¦ (5.15)

Bandpass channel impulse response model is

x(t) →h(t, τ) = Re¦h

b

(t, τ)e

jω

c

t

→y(t) = Re¦r(t)e

jω

c

t

¦ (5.16)

Baseband equivalent channel impulse response model is given by

c(t) →

1

2

h

b

(t, τ) →r(t) = c(t) ∗

1

2

h

b

(t, τ) (5.17)

Average power is

x

2

(t) =

1

2

[c(t)[

2

(5.18)

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The baseband impulse response of a multipath channel can be expressed as

h

b

(t, τ) =

N−1

i=0

a

i

(t, τ) exp[j(2πf

c

τ

i

(t) +ϕ

i

(t, τ))]δ(τ −τ

i

(t)) (5.19)

where a

i

(t, τ) and τ

i

(t) are the real amplitudes and excess delays, respectively, of

the ith multipath component at time t. The phase term 2πf

c

τ

i

(t) +ϕ

i

(t, τ) in the

above equation represents the phase shift due to free space propagation of the ith

multipath component, plus any additional phase shifts which are encountered in the

channel.

If the channel impulse response is wide sense stationary over a small-scale time or

distance interval, then

h

b

(τ) =

N−1

i=0

a

i

exp[jθ

i

]δ(τ −τ

i

) (5.20)

For measuring h

b

(τ), we use a probing pulse to approximate δ(t) i.e.,

p(t) ≈ δ(t −τ) (5.21)

Power delay proﬁle is taken by spatial average of [h

b

(t, τ)[

2

over a local area. The

received power delay proﬁle in a local area is given by

p(τ) ≈ k[h

b

(t; τ)[

2

. (5.22)

5.3.5 Relation Between Bandwidth and Received Power

In actual wireless communications, impulse response of a multipath channel is mea-

sured using channel sounding techniques. Let us consider two extreme channel

sounding cases.

Consider a pulsed, transmitted RF signal

x(t) = Re¦p(t)e

j2πf

c

t

¦ (5.23)

where p(t) =

_

4τ

max

T

bb

for 0 ≤ t ≤ T

bb

and 0 elsewhere. The low pass channel output

is

r(t) =

1

2

N−1

i=0

a

i

exp[jθ

i

]p(t −τ

i

)

=

N−1

i=0

a

i

exp[jθ

i

].

_

τ

max

T

bb

rect(t −

T

b

2

−τ

i

).

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Figure 5.2: A generic transmitted pulsed RF signal.

The received power at any time t

0

is

[r(t

0

)[

2

=

1

τ

max

τ

max

_

0

r(t)r

∗

(t)dt

=

1

τ

max

τ

max

_

0

1

4

_

N−1

k=0

a

2

k

(t

0

)p

2

(t −τ

k

)

_

dt

=

1

τ

max

N−1

k=0

a

2

k

(t

0

)

τ

max

_

0

__

τ

max

T

bb

rect(t −

T

b

2

−τ

i

)

_

2

dt

=

N−1

k=0

a

2

k

(t

0

).

Interpretation: If the transmitted signal is able to resolve the multipaths, then

average small-scale receiver power is simply sum of average powers received from

each multipath components.

E

a,θ

[P

WB

] = E

a,θ

[

N−1

i=0

[a

i

exp(jθ

i

)[

2

] ≈

N−1

i=0

a

2

i

(5.24)

Now instead of a pulse, consider a CW signal, transmitted into the same channel

and for simplicity, let the envelope be c(t) = 2. Then

r(t) =

N−1

i=0

a

i

exp[jθ

i

(t, τ)] (5.25)

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and the instantaneous power is

[r(t)[

2

= [

N−1

i=0

a

i

exp[jθ

i

(t, τ)][

2

(5.26)

Over local areas, a

i

varies little but θ

i

varies greatly resulting in large ﬂuctuations.

E

a,θ

[P

CW

] = E

a,θ

[

N−1

i=0

[a

i

exp(jθ

i

)[

2

]

≈

N−1

i=0

a

2

i

+ 2

N−1

i=0

N

i,j=i

r

ij

cos(θ

i

−θ

j

)

where r

ij

= E

a

[a

i

a

j

].

If, r

ij

= cos(θ

i

−θ

j

) = 0, then E

a,θ

[P

CW

] = E

a,θ

[P

WB

]. This occurs if multipath

components are uncorrelated or if multipath phases are i.i.d over [0, 2π].

Bottomline:

1. If the signal bandwidth is greater than multipath channel bandwidth then

fading eﬀects are negligible

2. If the signal bandwidth is less than the multipath channel bandwidth, large

fading occurs due to phase shift of unresolved paths.

5.3.6 Linear Time Varying Channels (LTV)

The time variant transfer function(TF) of an LTV channel is FT of h(t, τ) w.r.t. τ.

H(f, t) = FT[h(τ, t)] =

_

∞

−∞

h(τ, t)e

−j2πfτ

dτ (5.27)

h(τ, t) = FT

−1

[H(f, t)] =

_

∞

−∞

H(f, t)e

j2πfτ

df (5.28)

The received signal

r(t) =

_

∞

−∞

R(f, t)e

j2πft

df (5.29)

where R(f, t) = H(f, t)X(f).

For ﬂat fading channel, h(τ, t) = Z(t)δ(τ −τ

i

) where Z(t) =

α

n

(t)e

−j2πf

c

τ

n

(t)

. In

this case, the received signal is

r(t) =

_

∞

−∞

h(τ, t)x(t −τ) dτ = Z(t)x(t −τ

i

) (5.30)

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Figure 5.3: Relationship among diﬀerent channel functions.

where the channel becomes multiplicative.

Doppler spread functions:

H(f, ν) = FT[H(f, t)] =

_

∞

−∞

H(f, t)e

−j2πνt

dt (5.31)

and

H(f, t) = FT

−1

[H(f, ν)] =

_

∞

−∞

H(f, ν)e

j2πνt

dν (5.32)

Delay Doppler spread:

H(τ, ν) = FT[h(τ, t)] =

_

∞

−∞

h(τ, t)e

−j2πνt

dt (5.33)

5.3.7 Small-Scale Multipath Measurements

Direct RF Pulse System

A wideband pulsed bistatic radar usually transmits a repetitive pulse of width T

bb

s,

and uses a receiver with a wide bandpass ﬁlter (BW =

2

T

bb

Hz). The signal is then

ampliﬁed, envelope detected, and displayed and stored on a high speed oscilloscope.

Immediate measurements of the square of the channel impulse response convolved

with the probing pulse can be taken. If the oscilloscope is set on averaging mode,

then this system provides a local average power delay proﬁle.

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Figure 5.4: Direct RF pulsed channel IR measurement.

This system is subject to interference noise. If the ﬁrst arriving signal is blocked

or fades, severe fading occurs, and it is possible the system may not trigger properly.

Frequency Domain Channel Sounding

In this case we measure the channel in the frequency domain and then convert it

into time domain impulse response by taking its inverse discrete Fourier transform

(IDFT). A vector network analyzer controls a swept frequency synthesizer. An S-

parameter test set is used to monitor the frequency response of the channel. The

sweeper scans a particular frequency band, centered on the carrier, by stepping

through discrete frequencies. The number and spacing of the frequency step impacts

the time resolution of the impulse response measurement. For each frequency step,

the S-parameter test set transmits a known signal level at port 1 and monitors

the received signal at port 2. These signals allow the analyzer to measure the

complex response, S

21

(ω), of the channel over the measured frequency range. The

S

21

(ω) measure is the measure of the signal ﬂow from transmitter antenna to receiver

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Figure 5.5: Frequency domain channel IR measurement.

antenna (i.e., the channel).

This system is suitable only for indoor channel measurements. This system is

also non real-time. Hence, it is not suitable for time-varying channels unless the

sweep times are fast enough.

5.4 Multipath Channel Parameters

To compare the diﬀerent multipath channels and to quantify them, we deﬁne some

parameters. They all can be determined from the power delay proﬁle. These pa-

rameters can be broadly divided in to two types.

5.4.1 Time Dispersion Parameters

These parameters include the mean excess delay,rms delay spread and excess delay

spread. The mean excess delay is the ﬁrst moment of the power delay proﬁle and is

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deﬁned as

¯ τ =

a

2

k

τ

k

a

2

k

=

P(τ

k

)τ

k

P(τ

k

)

(5.34)

where a

k

is the amplitude, τ

k

is the excess delay and P(τ

k

) is the power of the

individual multipath signals.

The mean square excess delay spread is deﬁned as

¯

τ

2

=

P(τ

k

)τ

2

k

P(τ

k

)

(5.35)

Since the rms delay spread is the square root of the second central moment of the

power delay proﬁle, it can be written as

σ

τ

=

_

¯

τ

2

−(¯ τ)

2

(5.36)

As a rule of thumb, for a channel to be ﬂat fading the following condition must be

satisﬁed

σ

τ

T

S

≤ 0.1 (5.37)

where T

S

is the symbol duration. For this case, no equalizer is required at the

receiver.

Example 2

1. Sketch the power delay proﬁle and compute RMS delay spread for the follow-

ing:

P(τ) =

1

n=0

δ(τ −n 10

−6

) (in watts)

2. If BPSK modulation is used, what is the maximum bit rate that can be sent

through the channel without needing an equalizer?

Solution

1. P(0) = 1 watt, P(1) = 1 watt

τ =

(1)(0) + (1)(1)

1 + 1

= 0.5µs

τ

2

= 0.5µs

2

σ

τ

= 0.5µs

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2. For ﬂat fading channel, we need

σ

τ

T

s

0.1 ⇒R

s

=

1

T

s

= 0.2 10

4

= 200 kbps

For BPSK we need R

b

= R

s

= 200 kbps

Example 3 A simple delay spread bound: Feher’s upper bound

Consider a simple worst-case delay spread scenario as shown in ﬁgure below.

Here d

min

= d

0

and d

max

= d

i

+d

r

Transmitted power = P

T

, Minimum received power = P

R

min

= P

Threshold

P

R

min

P

T

= G

T

G

R

(

λ

4πd

max

)

2

Put G

T

= G

R

= 1 i.e., considering omni-directional unity gain antennas

d

max

= (

λ

4π

)(

P

T

P

R

min

)

1

2

τ

max

=

d

max

c

= (

λ

4πc

)(

P

T

P

R

min

)

1

2

τ

max

= (

1

4πf

)(

P

T

P

R

min

)

1

2

5.4.2 Frequency Dispersion Parameters

To characterize the channel in the frequency domain, we have the following param-

eters.

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(1) Coherence bandwidth: it is a statistical measure of the range of frequencies

over which the channel can be considered to pass all the frequency components with

almost equal gain and linear phase. When this condition is satisﬁed then we say the

channel to be ﬂat.

Practically, coherence bandwidth is the minimum separation over which the two

frequency components are aﬀected diﬀerently. If the coherence bandwidth is con-

sidered to be the bandwidth over which the frequency correlation function is above

0.9, then it is approximated as

B

C

≈

1

50σ

τ

. (5.38)

However, if the coherence bandwidth is considered to be the bandwidth over which

the frequency correlation function is above 0.5, then it is deﬁned as

B

C

≈

1

5σ

τ

. (5.39)

The coherence bandwidth describes the time dispersive nature of the channel in the

local area. A more convenient parameter to study the time variation of the channel

is the coherence time. This variation may be due to the relative motion between the

mobile and the base station or the motion of the objects in the channel.

(2) Coherence time: this is a statistical measure of the time duration over which

the channel impulse response is almost invariant. When channel behaves like this,

it is said to be slow faded. Essentially it is the minimum time duration over which

two received signals are aﬀected diﬀerently. For an example, if the coherence time

is considered to be the bandwidth over which the time correlation is above 0.5, then

it can be approximated as

T

C

≈

9

16πf

m

(5.40)

where f

m

is the maximum doppler spread given be f

m

=

ν

λ

.

Another parameter is the Doppler spread (B

D

) which is the range of frequencies

over which the received Doppler spectrum is non zero.

5.5 Statistical models for multipath propagation

Many multipath models have been proposed to explain the observed statistical na-

ture of a practical mobile channel. Both the ﬁrst order and second order statistics

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Figure 5.6: Two ray NLoS multipath, resulting in Rayleigh fading.

have been examined in order to ﬁnd out the eﬀective way to model and combat the

channel eﬀects. The most popular of these models are Rayleigh model, which de-

scribes the NLoS propagation. The Rayleigh model is used to model the statistical

time varying nature of the received envelope of a ﬂat fading envelope. Below, we

discuss about the main ﬁrst order and second order statistical models.

5.5.1 NLoS Propagation: Rayleigh Fading Model

Let there be two multipath signals S

1

and S

2

received at two diﬀerent time instants

due to the presence of obstacles as shown in Figure 5.6. Now there can either be

constructive or destructive interference between the two signals.

Let E

n

be the electric ﬁeld and Θ

n

be the relative phase of the various multipath

signals.So we have

˜

E =

N

n=1

E

n

e

jθ

n

(5.41)

Now if N→ ∞(i.e. are suﬃciently large number of multipaths) and all the E

n

are

IID distributed, then by Central Limit Theorem we have,

lim

N→∞

˜

E = lim

N→∞

N

n=1

E

n

e

jθ

n

(5.42)

= Z

r

+jZ

i

= Re

jφ

(5.43)

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where Z

r

and Z

i

are Gaussian Random variables. For the above case

R =

_

Z

2

r

+Z

2

i

(5.44)

and

φ = tan

−1

Z

i

Z

r

(5.45)

For all practical purposes we assume that the relative phase Θ

n

is uniformaly dis-

tributed.

E[e

jθ

n

] =

1

2π

2π

_

0

e

jθ

dθ = 0 (5.46)

It can be seen that E

n

and Θ

n

are independent. So,

E[

˜

E] = E[

E

n

e

jθ

n

] = 0 (5.47)

E[

¸

¸

¸

˜

E

¸

¸

¸

2

] = E[

E

n

e

jθ

n

E

∗

n

e

−jθ

n

] = E[

m

n

E

n

E

m

e

j(θ

n

−θ

m

)

] =

N

n=1

E

2

n

= P

0

(5.48)

where P

0

is the total power obtained. To ﬁnd the Cumulative Distribution Func-

tion(CDF) of R, we proceed as follows.

F

R

(r) = P

r

(R ≤ r) =

_

A

_

f

Z

i

,Z

r

(z

i

, z

r

)dz

i

dz

r

(5.49)

where A is determined by the values taken by the dummy variable r. Let Z

i

and Z

r

be zero mean Gaussian RVs. Hence the CDF can be written as

F

R

(r) =

_

A

_

1

√

2πσ

2

e

−(Z

2

r

+Z

2

i

)

2σ

2

dZ

i

dZ

r

(5.50)

Let Z

r

= p cos(Θ) and Z

i

= p sin(Θ) So we have

F

R

(r) =

2π

_

0

2π

_

0

1

√

2πσ

2

e

−p

2

2σ

2

pdpdθ (5.51)

= 1 −e

−r

2

2σ

2

(5.52)

Above equation is valid for all r ≥ 0. The pdf can be written as

f

R

(r) =

r

σ

2

e

−

r

2

2σ

2

(5.53)

and is shown in Figure 5.7 with diﬀerent σ values. This equation too is valid for all

r ≥ 0. Above distribution is known as Rayleigh distribution and it has been derived

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Figure 5.7: Rayleigh probability density function.

for slow fading. However, if f

D

¸ 1 Hz, we call it as Quasi-stationary Rayleigh

fading. We observe the following:

E[R] =

_

π

2

σ (5.54)

E[R

2

] = 2σ

2

(5.55)

var[R] = (2 −

π

2

)σ

2

(5.56)

median[R] = 1.77σ. (5.57)

5.5.2 LoS Propagation: Rician Fading Model

Rician Fading is the addition to all the normal multipaths a direct LOS path.

Figure 5.8: Ricean probability density function.

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f

R

(r) =

r

σ

2

e

−(r

2

+A

2

)

2σ

2

I

0

(

A

r

σ

2

) (5.58)

for all A ≥ 0 and r ≥ 0. Here A is the peak amplitude of the dominant signal and

I

0

(.) is the modiﬁed Bessel function of the ﬁrst kind and zeroth order.

A factor K is deﬁned as

K

dB

= 10 log

A

2

2σ

2

(5.59)

As A →0 then K

dB

→∞.

5.5.3 Generalized Model: Nakagami Distribution

A generalization of the Rayleigh and Rician fading is the Nakagami distribution.

Figure 5.9: Nakagami probability density function.

Its pdf is given as,

f

R

(r) =

2r

m−1

Γ(m)

(

m

m

Ω

m

)e

−mr

2

Ω

(5.60)

where,

Γ(m) is the gamma function

Ω is the average signal power and

m is the fading factor.It is always greater than or equal to 0.5.

When m=1, Nakagami model is the Rayleigh model.

When

m =

(M + 1)

2

2M + 1

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Figure 5.10: Schematic representation of level crossing with a Rayleigh fading enve-

lope at 10 Hz Doppler spread.

where

M =

A

2σ

Nakagami fading is the Rician fading.

As m →∞ Nakagami fading is the impulse channel and no fading occurs.

5.5.4 Second Order Statistics

To design better error control codes, we have two important second order param-

eters of fading model, namely the level crossing rate (LCR) and average fade

duration (AFD). These parameters can be utilized to assess the speed of the user

by measuring them through the reverse channel. The LCR is the expected rate at

which the Rayleigh fading envelope normalized to the local rms amplitude crosses a

speciﬁc level ’R’ in a positive going direction.

N

R

=

∞

_

0

˙ rp(R, ˙ r)d ˙ r =

√

2πf

D

ρe

−ρ

2

(5.61)

where ˙ r is the time derivative of r(t), f

D

is the maximum Doppler shift and ρ is the

value of the speciﬁed level R, normalized to the local rms amplitude of the fading

envelope.

The other important parameter, AFD, is the average period time for which the

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receiver power is below a speciﬁed level R.

¯ τ =

1

N

r

P

r

(r ≤ R) (5.62)

As

P

r

(r ≤ R) =

R

_

0

p(r)dr = 1 −e

−ρ

2

, (5.63)

therefore,

¯ τ =

1 −e

−ρ

2

√

2πf

D

ρe

−ρ

2

(5.64)

=

e

−ρ

2

−1

√

2πf

D

ρ

. (5.65)

Apart from LCR, another parameter is fading rate, which is deﬁned as the number of

times the signal envelope crosses the middle value (r

m

) in a positive going direction

per unit time. The average rate is expressed as

N(r

m

) =

2v

λ

. (5.66)

Another statistical parameter, sometimes used in the mobile communication, is

called as depth of fading. It is deﬁned as the ratio between the minimum value

and the mean square value of the faded signal. Usually, an average value of 10% as

depth of fading gives a marginal fading scenario.

5.6 Simulation of Rayleigh Fading Models

5.6.1 Clarke’s Model: without Doppler Eﬀect

In it, two independent Gaussian low pass noise sources are used to produce in-phase

and quadrature fading branches. This is the basic model and is useful for slow fading

channel. Also the Doppler eﬀect is not accounted for.

5.6.2 Clarke and Gans’ Model: with Doppler Eﬀect

In this model, the output of the Clarke’s model is passed through Doppler ﬁlter in

the RF or through two initial baseband Doppler ﬁlters for baseband processing as

shown in Figure 5.11. Here, the obtained Rayleigh output is ﬂat faded signal but

not frequency selective.

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Figure 5.11: Clarke and Gan’s model for Rayleigh fading generation using quadra-

ture amplitude modulation with (a) RF Doppler ﬁlter, and, (b) baseband Doppler

ﬁlter.

5.6.3 Rayleigh Simulator with Wide Range of Channel Conditions

To get a frequency selective output we have the following simulator through which

both the frequency selective and ﬂat faded Rayleigh signal may be obtained. This

is achieved through varying the parameters a

i

and τ

i

, as given in Figure 5.12.

5.6.4 Two-Ray Rayleigh Faded Model

The above model is, however, very complex and diﬃcult to implement. So, we have

the two ray Rayleigh fading model which can be easily implemented in software as

shown in Figure 5.13.

h

b

(t) = α

1

e

jφ

1

δ(t) +α

2

e

jφ

2

δ(t −τ) (5.67)

where α

1

and α

2

are independent Rayleigh distributed and φ

1

and φ

2

are indepen-

dent and uniformaly distributed over 0 to 2π. By varying τ it is possible to create

a wide range of frequency selective fading eﬀects.

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Figure 5.12: Rayleigh fading model to get both the ﬂat and frequency selective

channel conditions.

5.6.5 Saleh and Valenzuela Indoor Statistical Model

This method involved averaging the square law detected pulse response while sweep-

ing the frequency of the transmitted pulse. The model assumes that the multipath

components arrive in clusters. The amplitudes of the received components are in-

dependent Rayleigh random variables with variances that decay exponentially with

cluster delay as well as excess delay within a cluster. The clusters and multipath

components within a cluster form Poisson arrival processes with diﬀerent rates.

5.6.6 SIRCIM/SMRCIM Indoor/Outdoor Statistical Models

SIRCIM (Simulation of Indoor Radio Channel Impulse-response Model) generates

realistic samples of small-scale indoor channel impulse response measurements. Sub-

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Figure 5.13: Two-ray Rayleigh fading model.

sequent work by Huang produced SMRCIM (Simulation of Mobile Radio Channel

Impulse-response Model), a similar program that generates small-scale urban cellu-

lar and micro-cellular channel impulse responses.

5.7 Conclusion

In this chapter, the main channel impairment, i.e., fading, has been introduced which

becomes so severe sometimes that even the large scale path loss becomes insigniﬁcant

in comparison to it. Some statistical propagation models have been presented based

on the fading characteristics. Mainly the frequency selective fading, fast fading and

deep fading can be considered the major obstruction from the channel severity view

point. Certain eﬃcient signal processing techniques to mitigate these eﬀects will be

discussed in Chapter 7.

5.8 References

1. T. S. Rappaport, Wireless Communications: Principles and Practice, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

2. S. Haykin and M. Moher, Modern Wireless Communications. Singapore: Pear-

son Education, Inc., 2002.

3. J. W. Mark and W. Zhuang, Wireless Communications and Networking. New

Delhi: PHI, 2005.

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4. K. Feher, Wireless Digital Communications: Modulation and Spread Spectrum

Applications. Upper Saddle River, NJ: Prentice Hall, 1995.

5. R. Blake, Wireless Communications Technology. Delmar, Singapore: Thom-

son Asia Pvt Ltd, 2004.

6. D. P. Agarwal and Q-A. Zeng, Introduction to Wireless and Mobile Systems.

Nelson, India: Thomson Learning, 2007.

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Chapter 6

Transmitter and Receiver

Techniques

6.1 Introduction

Electrical communication transmitter and receiver techniques strive toward obtain-

ing reliable communication at a low cost, with maximum utilization of the channel

resources. The information transmitted by the source is received by the destina-

tion via a physical medium called a channel. This physical medium, which may be

wired or wireless, introduces distortion, noise and interference in the transmitted

information bearing signal. To counteract these eﬀects is one of the requirements

while designing a transmitter and receiver end technique. The other requirements

are power and bandwidth eﬃciency at a low implementation complexity.

6.2 Modulation

Modulation is a process of encoding information from a message source in a man-

ner suitable for transmission. It involves translating a baseband message signal to

a passband signal. The baseband signal is called the modulating signal and the

passband signal is called the modulated signal. Modulation can be done by varying

certain characteristics of carrier waves according to the message signal. Demodu-

lation is the reciprocal process of modulation which involves extraction of original

baseband signal from the modulated passband signal.

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6.2.1 Choice of Modulation Scheme

Several factors inﬂuence the choice of a digital modulation scheme. A desirable

modulation scheme provides low bit error rates at low received signal to noise ratios,

performs well in multipath and fading conditions, occupies a minimum of bandwidth,

and is easy and cost-eﬀective to implement. The performance of a modulation

scheme is often measured in terms of its power eﬃciency and bandwidth eﬃciency.

Power eﬃciency describes the ability of a modulation technique to preserve the

ﬁdelity of the digital message at low power levels. In a digital communication system,

in order to increase noise immunity, it is necessary to increase the signal power.

Bandwidth eﬃciency describes the ability of a modulation scheme to accommodate

data within a limited bandwidth.

The system capacity of a digital mobile communication system is directly related

to the bandwidth eﬃciency of the modulation scheme, since a modulation with a

greater value of η

b

(=

R

B

) will transmit more data in a given spectrum allocation.

There is a fundamental upper bound on achievable bandwidth eﬃciency. Shan-

non’s channel coding theorem states that for an arbitrarily small probability of error,

the maximum possible bandwidth eﬃciency is limited by the noise in the channel,

and is given by the channel capacity formula

η

Bmax

=

C

B

= log

2

(1 +

S

N

) (6.1)

6.2.2 Advantages of Modulation

1. Facilitates multiple access: By translating the baseband spectrum of signals

from various users to diﬀerent frequency bands, multiple users can be accom-

modated within a band of the electromagnetic spectrum.

2. Increases the range of communication: Low frequency baseband signals suﬀer

from attenuation and hence cannot be transmitted over long distances. So

translation to a higher frequency band results in long distance transmission.

3. Reduction in antenna size: The antenna height and aperture is inversely pro-

portional to the radiated signal frequency and hence high frequency signal

radiation result in smaller antenna size.

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6.2.3 Linear and Non-linear Modulation Techniques

The mathematical relation between the message signal (applied at the modulator

input) and the modulated signal (obtained at the modulator output) decides whether

a modulation technique can be classiﬁed as linear or non-linear. If this input-output

relation satisﬁes the principle of homogeneity and superposition then the modulation

technique is said to be linear. The principle of homogeneity states that if the input

signal to a system (in our case the system is a modulator) is scaled by a factor then

the output must be scaled by the same factor. The principle of superposition states

that the output of a linear system due to many simultaneously applied input signals

is equal to the summation of outputs obtained when each input is applied one at a

time.

For example an amplitude modulated wave consists of the addition two terms: the

message signal multiplied with the carrier and the carrier itself. If m(t) is the

message signal and s

AM

(t) is the modulated signal given by:

s

AM

(t) = A

c

[1 +km(t)] cos(2πf

c

t) (6.2)

Then,

1. From the principle of homogeneity: Let us scale the input by a factor a. So

m(t) = am

1

(t) and the corresponding output becomes :

s

AM1

(t) = A

c

[1 +am

1

(t)] cos(2πf

c

t) (6.3)

,= as

AM1

(t)

2. From the principle of superposition: Let m(t) = m

1

(t) + m

2

(t) be applied

simultaneously at the input of the modulator. The resulting output is:

s

AM

(t) = A

c

[1 +m

1

(t) +m

2

(t)] cos(2πf

c

t) (6.4)

,= s

AM1

(t) +s

AM2

(t)

= A

c

[2 +m

1

(t) +m

2

(t)] cos(2πf

c

t)

Here, s

AM1

(t) and s

AM2

(t) are the outputs obtained when m

1

(t) and m

2

(t)

are applied one at a time.

Hence AM is a nonlinear technique but DSBSC modulation is a linear technique

since it satisﬁes both the above mentioned principles.

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6.2.4 Amplitude and Angle Modulation

Depending on the parameter of the carrier (amplitude or angle) that is changed

in accordance with the message signal, a modulation scheme can be classiﬁed as

an amplitude or angle modulation. Amplitude modulation involves variation of

amplitude of the carrier wave with changes in the message signal. Angle modulation

varies a sinusoidal carrier signal in such a way that the angle of the carrier is varied

according to the amplitude of the modulating baseband signal.

6.2.5 Analog and Digital Modulation Techniques

The nature of the information generating source classiﬁes a modulation technique as

an analog or digital modulation technique. When analog messages generated from

a source passe through a modulator, the resulting amplitude or angle modulation

technique is called analog modulation. When digital messages undergo modulation

the resulting modulation technique is called digital modulation.

6.3 Signal Space Representation of Digitally Modulated

Signals

Any arbitrary signal can be expressed as the linear combination of a set of orthog-

onal signals or equivalently as a point in an M dimensional signal space, where M

denotes the cardinality of the set of orthogonal signals. These orthogonal signals are

normalized with respect to their energy content to yield an orthonormal signal set

having unit energy. These orthonormal signals are independent of each other and

form a basis set of the signal space.

Generally a digitally modulated signal s(t), having a symbol duration T, is ex-

pressed as a linear combination of two orthonormal signals φ

1

(t) and φ

2

(t), consti-

tuting the two orthogonal axis in this two dimensional signal space and is expressed

mathematically as,

s(t) = s

1

φ

1

(t) +s

2

φ

2

(t) (6.5)

where φ

1

(t) and φ

2

(t) are given by,

φ

1

(t) =

_

2

T

cos(2πf

c

t) (6.6)

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φ

2

(t) =

_

2

T

cos(2πf

c

t) (6.7)

The coeﬃcients s

1

and s

2

form the coordinates of the signal s(t) in the two dimen-

sional signal space.

6.4 Complex Representation of Linear Modulated Sig-

nals and Band Pass Systems

A band-pass signal s(t) can be resolved in terms of two sinusoids in phase quadrature

as follows:

s(t) = s

I

(t)cos(2πf

c

t) −s

Q

(t)sin(2πf

c

t) (6.8)

Hence s

I

(t) and s

Q

(t) are known as the in-phase and quadrature-phase components

respectively. When s

I

(t) and s

Q

(t) are incorporated in the formation of the following

complex signal,

˜ s(t) = s

I

(t) +s

Q

(t) (6.9)

then s(t) can be expressed in a more compact form as:

s(t) = Re¦˜ s(t)e

(j2πf

c

t)

¦ (6.10)

where ˜ s(t) is called the complex envelope of s(t).

Analogously, band-pass systems characterized by an impulse response h(t) can

be expressed in terms of its in-phase and quadrature-phase components as:

h(t) = h

I

(t)cos(2πf

c

t) −h

Q

(t)sin(2πf

c

t) (6.11)

The complex baseband model for the impulse response therefore becomes,

˜

h(t) = h

I

(t) +h

Q

(t) (6.12)

h(t) can therefore be expressed in terms of its complex envelope as

h(t) = Re¦

˜

h(t)e

j2πf

c

t

¦. (6.13)

When s(t) passes through h(t), then in the complex baseband domain, the output

˜ r(t) of the bandpass system is given by the following convolution

˜ r(t) =

1

2

˜ s(t) ⊗

˜

h(t) (6.14)

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6.5 Linear Modulation Techniques

6.5.1 Amplitude Modulation (DSBSC)

Generally, in amplitude modulation, the amplitude of a high frequency carrier signal,

cos(2πf

c

t), is varied in accordance to the instantaneous amplitude of the modulat-

ing message signal m(t). The resulting modulated carrier or AM signal can be

represented as:

s

AM

(t) = A

c

[1 +km(t)] cos(2πf

c

t). (6.15)

The modulation index k of an AM signal is deﬁned as the ratio of the peak message

signal amplitude to the peak carrier amplitude. For a sinusoidal modulating signal

m(t) =

A

m

A

c

cos(2πf

m

t), the modulation index is given by

k =

A

m

A

c

. (6.16)

This is a nonlinear technique and can be made linear by multiplying the carrier with

the message signal.The resulting modulation scheme is known as DSBSC modula-

tion. In DSBSC the amplitude of the transmitted signal, s(t), varies linearly with

the modulating digital signal, m(t). Linear modulation techniques are bandwidth

eﬃcient and hence are very attractive for use in wireless communication systems

where there is an increasing demand to accommodate more and more users within

a limited spectrum. The transmitted signal DSBSC signal s(t) can be expressed as:

s(t) = Am(t)exp(j2πf

c

t). (6.17)

If m(t) is scaled by a factor of a, then s(t), the output of the modulator, is also

scaled by the same factor as seen from the above equation. Hence the principle of

homogeneity is satisﬁed. Moreover,

s

12

(t) = A[m

1

(t) +m

2

(t)]cos(2πf

c

t) (6.18)

= Am

1

(t)cos(2πf

c

t) +Am

2

(t)cos(2πf

c

t)

= s

1

(t) +s

2

(t)

where A is the carrier amplitude and f

c

is the carrier frequency. Hence the principle

of superposition is also satisﬁed. Thus DSBSC is a linear modulation technique.

AM demodulation techniques may be broadly divided into two categories: co-

herent and non-coherent demodulation. Coherent demodulation requires knowledge

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Figure 6.1: BPSK signal constellation.

of the transmitted carrier frequency and phase at the receiver, whereas non-coherent

detection requires no phase information.

6.5.2 BPSK

In binary phase shift keying (BPSK), the phase of a constant amplitude carrier

signal is switched between two values according to the two possible signals m1 and

m2 corresponding to binary 1 and 0, respectively. Normally, the two phases are

separated by 180

o

. If the sinusoidal carrier has an amplitude A, and energy per bit

E

o

=

1

2

A

2

c

T

b

then the transmitted BPSK signal is

s

BPSK

(t) = m(t)

¸

2E

b

T

b

cos(2πf

c

t +θ

c

). (6.19)

A typical BPSK signal constellation diagram is shown in Figure 6.1.

The probability of bit error for many modulation schemes in an AWGN channel

is found using the Q-function of the distance between the signal points. In case of

BPSK,

P

eBPSK

= Q(

¸

2E

b

N

0

). (6.20)

6.5.3 QPSK

The Quadrature Phase Shift Keying (QPSK) is a 4-ary PSK signal. The phase of

the carrier in the QPSK takes 1 of 4 equally spaced shifts. Although QPSK can

be viewed as a quaternary modulation, it is easier to see it as two independently

modulated quadrature carriers. With this interpretation, the even (or odd) bits are

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Figure 6.2: QPSK signal constellation.

Figure 6.3: QPSK transmitter.

used to modulate the in-phase component of the carrier, while the odd (or even)

bits are used to modulate the quadrature-phase component of the carrier.

The QPSK transmitted signal is deﬁned by:

s

i

(t) = Acos(ωt + (i −1)π/2), i = (1, 2, 3, 4) (6.21)

and the constellation disgram is shown in Figure 6.2.

6.5.4 Oﬀset-QPSK

As in QPSK, as shown in Figure 6.3, the NRZ data is split into two streams of odd

and even bits. Each bit in these streams has a duration of twice the bit duration,

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Figure 6.4: DQPSK constellation diagram.

T

b

, of the original data stream. These odd (d

1

(t)) and even bit streams (d

2

(t)) are

then used to modulate two sinusoidals in phase quadrature,and hence these data

streams are also called the in-phase and and quadrature phase components. After

modulation they are added up and transmitted. The constellation diagram of Oﬀset-

QPSK is the same as QPSK. Oﬀset-QPSK diﬀers from QPSK in that the d

1

(t) and

d

2

(t) are aligned such that the timing of the pulse streams are oﬀset with respect

to each other by T

b

seconds. From the constellation diagram it is observed that a

signal point in any quadrant can take a value in the diagonally opposite quadrant

only when two pulses change their polarities together leading to an abrupt 180 degree

phase shift between adjacent symbol slots. This is prevented in O-QPSK and the

allowed phase transitions are ± 90 degree.

Abrupt phase changes leading to sudden changes in the signal amplitude in the

time domain corresponds to signiﬁcant out of band high frequency components in

the frequency domain. Thus to reduce these sidelobes spectral shaping is done at

baseband. When high eﬃciency power ampliﬁers, whose non-linearity increases as

the eﬃciency goes high, are used then due to distortion, harmonics are generated

and this leads to what is known as spectral regrowth. Since sudden 180 degree phase

changes cannot occur in OQPSK, this problem is reduced to a certain extent.

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6.5.5 π/4 DQPSK

The data for π/4 DQPSK like QPSK can be thought to be carried in the phase of a

single modulated carrier or on the amplitudes of a pair of quadrature carriers. The

modulated signal during the time slot of kT < t < (k + 1)T given by:

s(t) = cos(2πf

c

t +ψ

k+1

) (6.22)

Here, ψ

k+1

= ψ

k

+ ∆ψ

k

and ∆ψ

k

can take values π/4 for 00, 3π/4 for 01, −3π/4

for 11 and −π/4 for 10. This corresponds to eight points in the signal constellation

but at any instant of time only one of the four points are possible: the four points

on axis or the four points oﬀ axis. The constellation diagram along with possible

transitions are shown in Figure 6.4.

6.6 Line Coding

Speciﬁc waveforms are required to represent a zero and a one uniquely so that a

sequence of bits is coded into electrical pulses. This is known as line coding. There

are various ways to accomplish this and the diﬀerent forms are summarized below.

1. Non-return to zero level (NRZ-L): 1 forces a a high while 0 forces a low.

2. Non-return to zero mark (NRZ-M): 1 forces negative and positive transitions

while 0 causes no transitions.

3. Non-return to zero space (NRZ-S): 0 forces negative and positive transitions

while 1 causes no transitions.

4. Return to zero (RZ): 1 goes high for half a period while 0 remains at zero

state.

5. Biphase-L: Manchester 1 forces positive transition while 0 forces negative tran-

sition. In case of consecutive bits of same type a transition occurs in the

beginning of the bit period.

6. Biphase-M: There is always a transition in the beginning of a bit interval. 1

forces a transition in the middle of the bit while 0 does nothing.

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Figure 6.5: Scematic of the line coding techniques.

7. Biphase-S: There is always a transition in the beginning of a bit interval. 0

forces a transition in the middle of the bit while 1 does nothing.

8. Diﬀerential Manchester: There is always a transition in the middle of a bit

interval. 0 forces a transition in the beginning of the bit while 1 does nothing.

9. Bipolar/Alternate mark inversion (AMI): 1 forces a positive or negative pulse

for half a bit period and they alternate while 0 does nothing.

All these schemes are shown in Figure 6.5.

6.7 Pulse Shaping

Let us think about a rectangular pulse as deﬁned in BPSK. Such a pulse is not

desirable for two fundamental reasons:

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Figure 6.6: Rectangular Pulse

(a) the spectrum of a rectangular pulse is inﬁnite in extent. Correspondingly, its

frequency content is also inﬁnite. But a wireless channel is bandlimited, means it

would introduce signal distortion to such type of pulses,

(b) a wireless channel has memory due to multipath and therefore it introduces ISI.

In order to mitigate the above two eﬀects, an eﬃcient pulse shaping funtion or

a premodulation ﬁlter is used at the Tx side so that QoS can be maintained to the

mobile users during communication. This type of technique is called pulse shaping

technique. Below, we start with the fundamental works of Nyquist on pulse shaping

and subsequently, we would look into another type of pulse shaping technique.

6.7.1 Nyquist pulse shaping

There are a number of well known pulse shaping techniques which are used to simul-

taneously to reduce the inter-symbol eﬀects and the spectral width of a modulated

digital signal. We discuss here about the fundamental works of Nyquist. As pulse

shaping is diﬃcult to directly manipulate the transmitter spectrum at RF frequen-

cies, spectral shaping is usually done through baseband or IF processing.

Let the overall frequency response of a communication system (the transmitter,

channel and receiver) be denoted as H

eff

(f) and according to Nyquist it must be

given by:

H

eff

(f) =

1

f

s

rect(

f

f

s

) (6.23)

Hence, the ideal pulse shape for zero ISI, given by h

eff

(t), such that,

H

eff

(f) ↔h

eff

(t) (6.24)

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Figure 6.7: Raised Cosine Pulse.

is given by:

h

eff

(t) =

sin(

πt

T

s

)

πt

T

s

(6.25)

(6.26)

6.7.2 Raised Cosine Roll-Oﬀ Filtering

If we take a rectangular ﬁlter with bandwidth f

0

≥

1

2T

s

and convolve it with any

arbitrary even function Z(f) with zero magnitude outside the passband of the rect-

angular ﬁlter then a zero ISI eﬀect would be achieved. Mathematically,

H

eff

(f) = rect(

f

f

0

) ∗ Z(f), (6.27)

h

eff

(t) =

sin(

πt

T

s

)

πt

T

s

z(t), (6.28)

z(t) =

cos(πρt/T

s

)

1 −(∆ρt/2T

s

)

2

. (6.29)

with ρ being the roll oﬀ factor ∈ [0, 1]. As ρ increases roll oﬀ in frequency domain

increases but that in time domain decreases.

6.7.3 Realization of Pulse Shaping Filters

Since h

eff

(t) is non-causal, pulse shaping ﬁlters are usually truncated within ±6T

s

about t = 0 for each symbol. Digital communication systems thus often store several

symbols at a time inside the modulator and then clock out a group of symbols by

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using a look up table that represents discrete time waveforms of stored symbols.

This is the way to realize the pulse shaping ﬁlters using real time processors.

Non-Nyquist pulse shaping are also useful, which would be discussed later in this

chapter while discussing GMSK.

6.8 Nonlinear Modulation Techniques

Many practical mobile radio communications use nonlinear modulation methods,where

the amplitude of the carrier is constant,regardless of the variations in the modulating

signal.The Constant envelope family of modulations has the following advantages :

1. Power eﬃcient class C ampliﬁers without introducing degradation in the spec-

tral occupancy of the transmitted signal.

2. Low out-of-band radiation of the order of -60 dB to -70dB can be achieved.

3. Limiter-discriminator detection can be used,which simpliﬁes receiver design

and provides high immunity against random FM noise and signal ﬂuctuations

due to Rayleigh fading.

However, even if constant envelope has many advantages it still uses more BW

than linear modulation schemes.

6.8.1 Angle Modulation (FM and PM)

There are a number of ways in which the phase of a carrier signal may be varied

in accordance with the baseband signal; the two most important classes of angle

modulation being frequency modulation and phase modulation.

Frequency modulation (FM) involves changing of the frequency of the carrier

signal according to message signal. As the information in frequency modulation is

in the frequency of modulated signal, it is a nonlinear modulation technique. In this

method, the amplitude of the carrier wave is kept constant (this is why FM is called

constant envelope). FM is thus part of a more general class of modulation known

as angle modulation.

Frequency modulated signals have better noise immunity and give better perfor-

mance in fading scenario as compared to amplitude modulation.Unlike AM, in an

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FM system, the modulation index, and hence bandwidth occupancy, can be varied

to obtain greater signal to noise performance.This ability of an FM system to trade

bandwidth for SNR is perhaps the most important reason for its superiority over

AM. However, AM signals are able to occupy less bandwidth as compared to FM

signals, since the transmission system is linear.

An FM signal is a constant envelope signal, due to the fact that the envelope of

the carrier does not change with changes in the modulating signal. The constant

envelope of the transmitted signal allows eﬃcient Class C power ampliﬁers to be

used for RF power ampliﬁcation of FM. In AM, however, it is critical to maintain

linearity between the applied message and the amplitude of the transmitted signal,

thus linear Class A or AB ampliﬁers, which are not as power eﬃcient, must be used.

FM systems require a wider frequency band in the transmitting media (generally

several times as large as that needed for AM) in order to obtain the advantages of

reduced noise and capture eﬀect. FM transmitter and receiver equipment is also

more complex than that used by amplitude modulation systems. Although frequency

modulation systems are tolerant to certain types of signal and circuit nonlinearities,

special attention must be given to phase characteristics. Both AM and FM may be

demodulated using inexpensive noncoherent detectors. AM is easily demodulated

using an envelope detector whereas FM is demodulated using a discriminator or

slope detector. In FM the instantaneous frequency of the carrier signal is varied

linearly with the baseband message signal m(t), as shown in following equation:

s

FM

(t) = A

c

cos[2πf

c

t +θ(t)] = A

c

cos[2πf

c

t + 2πk

f

_

m(η)dη] (6.30)

where A

c

, is the amplitude of the carrier, f

c

is the carrier frequency, and k

f

is the

frequency deviation constant (measured in units of Hz/V).

Phase modulation (PM) is a form of angle modulation in which the angle θ(t) of

the carrier signal is varied linearly with the baseband message signal m(t), as shown

in equation below.

s

PM

(t) = A

c

cos(2πf

c

t +k

θ

m(t)) (6.31)

The frequency modulation index β

f

, deﬁnes the relationship between the message

amplitude and the bandwidth of the transmitted signal, and is given by

β

f

=

k

f

A

m

W

=

∆

W

(6.32)

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where A

m

is the peak value of the modulating signal, ∆f is the peak frequency

deviation of the transmitter and W is the maximum bandwidth of the modulating

signal.

The phase modulation index β

p

is given by

β

p

= k

θ

A

m

= ∆θ (6.33)

where, ∆θ is the peak phase deviation of the transmitter.

6.8.2 BFSK

In Binary Frequency Shift keying (BFSK),the frequency of constant amplitude car-

rier signal is switched between two values according to the two possible message

states (called high and low tones) corresponding to a binary 1 or 0. Depending on

how the frequency variations are imparted into the transmitted waveform,the FSK

signal will have either a discontinuous phase or continuous phase between bits. In

general, an FSK signal may be represented as

S(t) =

_

(2E

b

/T) cos(2πf

i

t). (6.34)

where T is the symbol duration and E

b

is the energy per bit.

S

i

=

_

(E

b

)φ(t). (6.35)

φ(t) =

_

(2/T) cos(2πf

i

t). (6.36)

There are two FSK signals to represent 1 and 0, i.e.,

S

1

(t) =

_

(2E

b

/T) cos(2πf

1

t +θ(0)) →1 (6.37)

S

2

(t) =

_

(2E

b

/T) cos(2πf

2

t +θ(0)) →0 (6.38)

where θ(0) sums the phase up to t = 0. Let us now consider a continuous phase

FSK as

S(t) =

_

(2E

b

/T) cos(2πf

c

t +θ(t)). (6.39)

Expressing θ(t) in terms of θ(0) with a new unknown factor h, we get

θ(t) = θ(0) ±πht/T 0 ≤ t ≤ T (6.40)

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and therefore

S(t) =

¸

2E

b

T

cos(2πf

c

t ±πht/T +θ(0)) =

¸

2E

b

T

cos(2π(f

c

±h/2T)t +θ(0)).(6.41)

It shows that we can choose two frequencies f

1

and f

2

such that

f

1

= f

c

+h/2T (6.42)

f

2

= f

c

−h/2T (6.43)

for which the expression of FSK conforms to that of CPFSK. On the other hand, f

c

and h can be expressed in terms of f

1

and f

2

as

f

c

= [f

1

+f

2

]/2 (6.44)

h =

(f

1

−f

2

)

1/T

. (6.45)

Therefore, the unknown factor h can be treated as the diﬀerence between f

1

and f

2

,

normalized with respect to bit rate 1/T. It is called the deviation ratio. We know

that θ(t) −θ(0) = ±πht/T, 0 ≤ t ≤ T. If we substitute t = T, we have

θ(T) −θ(0) = ±πh where (6.46)

= πh →1 (6.47)

= −πh →0 (6.48)

This type of CPFSK is advantageous since by looking only at the phase, the trans-

mitted bit can be predicted. In Figure 6.8, we show a phase tree of such a CPFSK

signal with the transmitted bit stream of 1101000.

A special case of CPFSK is achieved with h = 0.5, and the resulting scheme is

called Minimum Shift Keying (MSK) which is used in mobile communications. In

this case, the phase diﬀerences reduce to only ±π/2 and the phase tree is called the

phase trellis. An MSK signal can also be thought as a special case of OQPSK where

the baseband rectangular pulses are replaced by half sinusoidal pulses. Spectral

characteristics of an MSK signal is shown in Figure 6.9 from which it is clear that

ACI is present in the spectrum. Hence a pulse shaping technique is required. In

order to have a compact signal spectrum as well as maintaining the constant envelope

property, we use a pulse shaping ﬁlter with

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Figure 6.8: Phase tree of 1101000 CPFSK sequence.

Figure 6.9: Spectrum of MSK

1. a narrow BW frequency and sharp cutoﬀ characteristics (in order to suppress

the high frequency component of the signal);

2. an impulse response with relatively low overshoot (to limit FM instant fre-

quency deviation;

3. a phase trellis with ±π/2 for odd T and 0 or π values for even T.

6.9 GMSK Scheme

GMSK is a simple modulation scheme that may be taken as a derivative of MSK.

In GMSK, the sidelobe levels of the spectrum are further reduced by passing a non-

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Figure 6.10: GMSK generation scheme.

return to zero (NRZ-L) data waveform through a premodulation Gaussian pulse

shaping ﬁlter. Baseband Gaussian pulse shaping smoothes the trajectory of the

MSK signals and hence stabilizes instantaneous frequency variations over time. This

has the eﬀect of considerably reducing the sidelobes in the transmitted spectrum.

A GMSK generation scheme with NRZ-L data is shown in Figure 6.10 and a receiver

of the same scheme with some MSI gates is shown in Figure 6.11.

6.10 GMSK Generator

The GMSK premodulation ﬁlter has characteristic equation given by

H(f) = exp(−(ln 2/2)(f/B)

2

) (6.49)

H(f) = exp(−(αf)

2

)

where,

(α)

2

= ln 2/2(1/B)

2

. (6.50)

The premodulation Gaussian ﬁltering introduces ISI in the transmitted signal, but

it can be shown that the degradation is not that great if the 3dB bandwidth-bit

duration product (BT) is greater than 0.5.

Spectrum of GMSK scheme is shown in Figure 6.12. From this ﬁgure, it is evident

that when we are decreasing BT product, the out of band response decreases but

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Figure 6.11: A simple GMSK receiver.

on the other hand irreducible error rate of the LPF for ISI increases. Therefore, a

compromise between these two is required.

Problem: Find the 3dB BW for a Gaussian LPF used to produce 0.25 GMSK

with a channel data rate Rb=270 kbps.What is the 90 percent power BW of the RF

ﬁlter?

Solution: From the problem statement it is clear that

T = 1/R

b

= 1/270 ∗ (10

3

) = 3.7µsec (6.51)

Solving for B where BT = 0.25,

B = 0.25/T = 67.567kHz (6.52)

Thus the 3 - dB bandwidth is 67.567 kHz. We use below table ﬁg 6 to ﬁnd out that

90 % power bandwidth is 0.57 R

b

.

90 % RF BW = 0.57R

b

= 153.9 kHz.

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Figure 6.12: Spectrum of GMSK scheme.

6.11 Two Practical Issues of Concern

6.11.1 Inter Channel Interference

In FDMA, subscribers are allotted frequency slots called channels in a given band

of the electromagnetic spectrum. The side lobes generated due to the transmission

of a symbol in a particular channel overlaps with the channels placed adjacently.

This is because of the fact that transmission of a time limited pulse leads to spectral

spreading in the frequency domain. During simultaneous use of adjacent channels,

when there is signiﬁcant amount of power present in the side lobes, this kind of

interference becomes so severe that the required symbol in a particular frequency

slot is completely lost.

Moreover if two terminals transmit equal power then due to wave propagation

through diﬀerent distances to the receiver, the received signal levels in the two

frequency slots will diﬀer greatly. In such a case the side lobes of the stronger signal

will severely degrade the transmitted signal in the next frequency slot having low

power level. This is known as the near far problem.

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6.11.2 Power Ampliﬁer Nonlinearity

Power ampliﬁers may be designed as class A, class B, class AB, class C and class D.

They form an essential section of mobile radio terminals. Due to power constraints

on a transmitting terminal, an eﬃcient power ampliﬁer is required which can convert

most of the input power to RF power. Class A ampliﬁer is a linear ampliﬁer but

it has a power eﬃciency of only 25 %. As we go for subsequent ampliﬁers having

greater power eﬃciency, the nonlinearity of the ampliﬁer increases.

In general, an ampliﬁer has linear input output characteristics over a range

of input signal level, that is, it has a constant gain. However, beyond an input

threshold level, the gain of the ampliﬁer starts decreasing. Thus the amplitude of

a signal applied at the input of an ampliﬁer suﬀers from amplitude distortion and

the resulting waveform obtained at the output of the ampliﬁer is of the form of

an amplitude modulated signal. Similarly, the phase characteristic of a practical

ampliﬁer is not constant over all input levels and results in phase distortion of the

form of phase modulation.

The operating point of a practical ampliﬁer is given in terms of either the input

back-oﬀ or the output back-oﬀ.

Input back −off = 10 log

1

0

_

V

in,rms

V

out,rms

_

(6.53)

Output back −off = 10 log

1

0

_

V

out,rms

V

out,rms

_

(6.54)

6.12 Receiver performance in multipath channels

For a ﬂat fading channel, the probability of error for coherent BPSK and coherent

BFSK are respectively given as,

P

e,BPSK

=

1

2

_

1 −

_

γ

1 +γ

_

(6.55)

P

e,BFSK

=

1

2

_

1 −

_

γ

2 +γ

_

(6.56)

(6.57)

where γ is given by,

γ =

E

b

N

0

E(α

2

) (6.58)

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α

2

represents the instantaneous power values of the Rayleigh fading channel and E

denotes the expectation operator.

Similarly, for diﬀerential BPSK and non coherent BFSK probability of error

expressions are

P

e,DPSK

=

1

2(1 +γ)

(6.59)

P

e,NCFSK

=

1

(2 +γ)

. (6.60)

For large values of SNR =

E

b

N

0

the error probability given above have the simpliﬁed

expression.

P

e,BPSK

=

1

4γ

(6.61)

P

e,BFSK

=

1

2γ

(6.62)

P

e,DPSK

=

1

2γ

(6.63)

P

e,NCFSK

=

1

γ

. (6.64)

From the above equations we observe that an inverse algebraic relation exists be-

tween the BER and SNR. This implies that if the required BER range is around

10

−3

to 10

−6

, then the SNR range must be around 30dB to 60dB.

6.12.1 Bit Error Rate and Symbol Error Rate

Bit error rate (P

eb

) is the same as symbol error rate (P

es

) when a symbol consists

of a single bit as in BPSK modulation. For an MPSK scheme employing gray coded

modulation, where N bits are mapped to a one of the M symbols, such that 2

N

= M,

P

eb

is given by

P

eb

≈

P

es

log

2

M

(6.65)

And for M-ary orthogonal signalling P

eb

is given by

P

eb

=

M/2

M −1

P

es

. (6.66)

6.13 Example of a Multicarrier Modulation: OFDM

Multiplexing is an important signal processing operation in which a number of sig-

nals are combined and transmitted parallelly over a common channel. In order to

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avoid interference during parallel transmission, the signals can be separated in fre-

quency and then the resulting technique is called Frequency Division Multiplexing

(FDM). In FDM, the adjacent bands are non overlapping but if overlap is allowed by

transmitting signals that are mutually orthogonal (that is, there is a precise math-

ematical relationship between the frequencies of the transmitted signals) such that

one signal has zero eﬀect on another, then the resulting transmission technique is

known as Orthogonal Frequency Division Multiplexing (OFDM).

OFDM is a technique of transmitting high bit rate data into several parallel

streams of low bit rate data. At any instant, the data transmitted simultaneously

in each of these parallel data streams is frequency modulated by carriers (called

subcarriers) which are orthogonal to each other. For high data rate communication

the bandwidth (which is limited) requirement goes on increasing as the data rate

increases or the symbol duration decreases. Thus in OFDM, instead of sending a

particular number of symbols, say P, in T seconds serially, the P symbols can be

sent in parallel with symbol duration now increased to T seconds instead of T/P

seconds as was previously.

This oﬀers many advantages in digital data transmission through a wireless time

varying channel. The primary advantage of increasing the symbol duration is that

the channel experiences ﬂat fading instead of frequency selective fading since it is

ensured that in the time domain the symbol duration is greater than the r.m.s.

delay spread of the channel. Viewed in the frequency domain this implies that the

bandwidth of the OFDM signal is less than coherent bandwidth of the channel.

Although the use of OFDM was initially limited to military applications due to

cost and complexity considerations, with the recent advances in large-scale high-

speed DSP, this is no longer a major problem. This technique is being used, in

digital audio broadcasting (DAB), high deﬁnition digital television broadcasting

(HDTV), digital video broadcasting terrestrial TV (DVB-T), WLAN systems based

on IEEE 802.11(a) or HiperLan2, asymmetric digital subscriber lines (ADSL) and

mobile communications. Very recently, the signiﬁcance of the COFDM technique for

UWA (underwater acoustic channel) has also been indicated. Moreover related or

combined technology such as CDMA-OFDM, TDMA-OFDM, MIMO-OFDM, Vec-

tor OFDM (V-OFDM), wide-band OFDM (W-OFDM), ﬂash OFDM (F-OFDM),

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OFDMA, wavelet-OFDM have presented their great advantages in certain applica-

tion areas.

6.13.1 Orthogonality of Signals

Orthogonal signals can be viewed in the same perspective as we view vectors which

are perpendicular/orthogonal to each other. The inner product of two mutually

orthogonal vectors is equal to zero. Similarly the inner product of two orthogonal

signals is also equal to zero.

Let ψ

k

(t) = e

j2πf

k

t

and ψ

n

(t) = e

j2πf

n

t

be two complex exponential signals whose

inner product, over the time duration of T

s

, is given by:

N =

_

(i+1)T

s

iT

s

ψ

k

(t).ψ

∗

n

(t)dt (6.67)

When this integral is evaluated, it is found that if f

k

and f

n

are integer multiples

of 1/T

s

then N equals zero. This implies that for two harmonics of an exponential

function having a fundamental frequency of 1/T

s

, the inner product becomes zero

.But if f

k

= f

n

then N equals T

s

which is nothing but the energy of the complex

exponential signal in the time duration of T

s

.

6.13.2 Mathematical Description of OFDM

Let us now consider the simultaneous or parallel transmission of P number of com-

plex symbols in the time slot of T

s

second (OFDM symbol time duration) and a set

of P orthogonal subcarriers, such that each subcarrier gets amplitude modulated

by a particular symbol from this set of P symbols. Let each orthogonal carrier

be of the form exp

_

j2πn

t

T

s

_

, where n varies as 0, 1, 2..(P − 1). Here the variable

‘n’ denotes the n

th

parallel path corresponding to the n

th

subcarrier. Mathemati-

cally, we can obtain the transmitted signal in T

s

seconds by summing up all the P

number of amplitude modulated subcarriers, thereby yielding the following equation:

p(t) =

P−1

n=0

c

n

g

n

(t)exp

_

j2πn

t

T

s

_

for 0 ≤ t ≤ T

s

(6.68)

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If p(t) is sampled at t = kT

s

/P, then the resulting waveform, is:

p(k) =

P−1

n=0

c

n

g

n

(kT

s

/P)exp

_

j2πn

kT

s

/P

T

s

_

=

1

√

T

s

P−1

n=0

c

n

exp

_

j2πn

k

P

_

for 0 ≤ k ≤ P −1 (6.69)

This is nothing but the IDFT on the symbol block of P symbols. This can be realized

using IFFT but the constraint is that P has to be a power of 2. So at the receiver,

FFT can be done to get back the required block of symbols. This implementation is

better than using multiple oscillators for subcarrier generation which is uneconomical

and since digital technology has greatly advanced over the past few decades, IFFTs

and FFTs can be implemented easily. The frequency spectrum, therefore consists

of a set of P partially overlapping sinc pulses during any time slot of duration T

s

.

This is due to the fact that the Fourier Transform of a rectangular pulse is a sinc

function. The receiver can be visualized as consisting of a bank of demodulators,

translating each subcarrier down to DC, then integrating the resulting signal over a

symbol period to recover the raw data.

But the OFDM symbol structure so generated at the transmitter end needs to

be modiﬁed. Since inter symbol interference (ISI) is introduced by the transmission

channel due to multipaths and also due to the fact that when the bandwidth of

OFDM signal is truncated, its eﬀect in the time domain is to cause symbol spreading

such that a part of the symbol overlaps with the adjacent symbols. In order to cope

with ISI as discussed previously the OFDM symbol duration can be increased. But

this might not be feasible from the implementation point of view speciﬁcally in terms

of FFT size and Doppler shifts.

A diﬀerent approach is to keep a guard time interval between two OFDM symbols

in which part of the symbol is copied from the end of the symbol to the front and is

popularly known as the cyclic-preﬁx. If we denote the guard time interval as T

g

and

T

s

be the useful symbol duration, then after this cyclical extension the total symbol

duration becomes T = T

g

+ T

s

. When the guard interval is longer than the length

of the channel impulse response, or the multipath delay, then ISI can be eliminated.

However the disadvantage is the reduction in data rate or throughput and greater

power requirements at the transmitting end. The OFDM transmitter and receiver

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Figure 6.13: OFDM Transmitter and Receiver Block Diagram.

sections are as given in the following diagram.

6.14 Conclusion

In this chapter, a major chunk has been devoted to digital communication systems

which obviously have certain distinction in comparison to their analog counterpart

due to their signal-space representation. The important modulation techniques for

wireless communication such as QPSK, MSK, GMSK were taken up at length. A

relatively new modulation technology, OFDM, has also been discussed. Certain

practical issues of concern are also discussed. It should be noted that albeit imple-

menting these eﬃcient modulation techniques, the channel still introduces fading in

diﬀerent ways. In order to prevent that, we need some additional signal processing

techniques mainly at the receiver side. These techniques are discussed in the next

chapter.

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6.15 References

1. B. P. Lathi and Z. Ding, Modern Digital and Analog Communication Systems,

4th ed. NY: Oxford University Press, 2009.

2. B. Sklar, Digital Communications: Fundamentals and Applications, 2nd ed.

Singapore: Pearson Education, Inc., 2005.

3. R. Blake, Electronic Communication Systems. Delmar, Singapore: Thomson

Asia Pvt Ltd, 2002.

4. J. G. Proakis and M. Salehi, Communication Systems Engineering, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

5. T. S. Rappaport, Wireless Communications: Principles and Practice, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

6. S. Haykin and M. Moher, Modern Wireless Communications. Singapore: Pear-

son Education, Inc., 2002.

7. W. H. Tranter et. al., Principles of Communication Systems Simulation. Sin-

gapore: Pearson Education, Inc., 2004.

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Chapter 7

Techniques to Mitigate Fading

Eﬀects

7.1 Introduction

Apart from the better transmitter and receiver technology, mobile communications

require signal processing techniques that improve the link performance. Equaliza-

tion, Diversity and channel coding are channel impairment improvement techniques.

Equalization compensates for Inter Symbol Interference (ISI) created by multipath

within time dispersive channels. An equalizer within a receiver compensates for

the average range of expected channel amplitude and delay characteristics. In other

words, an equalizer is a ﬁlter at the mobile receiver whose impulse response is inverse

of the channel impulse response. As such equalizers ﬁnd their use in frequency selec-

tive fading channels. Diversity is another technique used to compensate fast fading

and is usually implemented using two or more receiving antennas. It is usually em-

ployed to reduce the depths and duration of the fades experienced by a receiver in

a ﬂat fading channel. Channel coding improves mobile communication link perfor-

mance by adding redundant data bits in the transmitted message.At the baseband

portion of the transmitter, a channel coder maps a digital message sequence in to

another speciﬁc code sequence containing greater number of bits than original con-

tained in the message. Channel Coding is used to correct deep fading or spectral

null. We discuss all three of these techniques in this chapter. A general framework

of the fading eﬀects and their mitigation techniques is shown in Figure 7.1.

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Figure 7.1: A general framework of fading eﬀects and their mitigation techniques.

7.2 Equalization

ISI has been identiﬁed as one of the major obstacles to high speed data transmission

over mobile radio channels. If the modulation bandwidth exceeds the coherence

bandwidth of the radio channel (i.e., frequency selective fading), modulation pulses

are spread in time, causing ISI. An equalizer at the front end of a receiver compen-

sates for the average range of expected channel amplitude and delay characteristics.

As the mobile fading channels are random and time varying, equalizers must track

the time-varying characteristics of the mobile channel and therefore should be time-

varying or adaptive. An adaptive equalizer has two phases of operation: training

and tracking. These are as follows.

Training Mode:

• Initially a known, ﬁxed length training sequence is sent by the transmitter so

that the receiver equalizer may average to a proper setting.

• Training sequence is typically a pseudo-random binary signal or a ﬁxed, of

prescribed bit pattern.

• The training sequence is designed to permit an equalizer at the receiver to

acquire the proper ﬁlter coeﬃcient in the worst possible channel condition.

An adaptive ﬁlter at the receiver thus uses a recursive algorithm to evaluate

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the channel and estimate ﬁlter coeﬃcients to compensate for the channel.

Tracking Mode:

• When the training sequence is ﬁnished the ﬁlter coeﬃcients are near optimal.

• Immediately following the training sequence, user data is sent.

• When the data of the users are received, the adaptive algorithms of the equal-

izer tracks the changing channel.

• As a result, the adaptive equalizer continuously changes the ﬁlter characteris-

tics over time.

7.2.1 A Mathematical Framework

The signal received by the equalizer is given by

x(t) = d(t) ∗ h(t) +n

b

(t) (7.1)

where d(t) is the transmitted signal, h(t) is the combined impulse response of the

transmitter,channel and the RF/IF section of the receiver and n

b

(t) denotes the

baseband noise.

If the impulse response of the equalizer is h

eq

(t), the output of the equalizer is

ˆ y (t) = d (t) ∗ h(t) ∗ h

eq

(t) +n

b

(t) ∗ h

eq

(t) = d (t) ∗ g (t) +n

b

(t) ∗ h

eq

(t) . (7.2)

However, the desired output of the equalizer is d(t) which is the original source data.

Assuming n

b

(t)=0, we can write y(t) = d(t), which in turn stems the following

equation:

g (t) = h(t) ∗ h

eq

(t) = δ (t) (7.3)

The main goal of any equalization process is to satisfy this equation optimally. In

frequency domain it can be written as

H

eq

(f) H (f) = 1 (7.4)

which indicates that an equalizer is actually an inverse ﬁlter of the channel. If the

channel is frequency selective, the equalizer enhances the frequency components with

small amplitudes and attenuates the strong frequencies in the received frequency

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spectrum in order to provide a ﬂat, composite received frequency response and

linear phase response. For a time varying channel, the equalizer is designed to track

the channel variations so that the above equation is approximately satisﬁed.

7.2.2 Zero Forcing Equalization

In a zero forcing equalizer, the equalizer coeﬃcients c

n

are chosen to force the samples

of the combined channel and equalizer impulse response to zero. When each of the

delay elements provide a time delay equal to the symbol duration T, the frequency

response H

eq

(f) of the equalizer is periodic with a period equal to the symbol rate

1/T. The combined response of the channel with the equalizer must satisfy Nyquist’s

criterion

H

ch

(f) H

eq

(f) = 1, [f[ <

1

/

2T

(7.5)

where H

ch

(f) is the folded frequency response of the channel. Thus, an inﬁnite

length zero-forcing ISI equalizer is simply an inverse ﬁlter which inverts the folded

frequency response of the channel.

Disadvantage: Since H

eq

(f) is inverse of H

ch

(f) so inverse ﬁlter may excessively

amplify the noise at frequencies where the folded channel spectrum has high atten-

uation, so it is rarely used for wireless link except for static channels with high SNR

such as local wired telephone. The usual equalizer model follows a time varying or

adaptive structure which is given next.

7.2.3 A Generic Adaptive Equalizer

The basic structure of an adaptive ﬁlter is shown in Figure 7.2. This ﬁlter is called

the transversal ﬁlter, and in this case has N delay elements, N+1 taps and N+1

tunable complex multipliers, called weights. These weights are updated continuously

by an adaptive algorithm. In the ﬁgure the subscript k represents discrete time

index. The adaptive algorithm is controlled by the error signal e

k

. The error signal

is derived by comparing the output of the equalizer, with some signal d

k

which is

replica of transmitted signal. The adaptive algorithm uses e

k

to minimize the cost

function and uses the equalizer weights in such a manner that it minimizes the cost

function iteratively. Let us denote the received sequence vector at the receiver and

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Figure 7.2: A generic adaptive equalizer.

the input to the equalizer as

x

k

= [x

k

, x

k−1

, ....., x

k−N

]

T

, (7.6)

and the tap coeﬃcient vector as

w

k

= [w

0

k

, w

1

k

, ....., w

N

k

]

T

. (7.7)

Now, the output sequence of the equalizer y

k

is the inner product of x

k

and w

k

, i.e.,

y

k

= ¸x

k

, w

k

) = x

T

k

w

k

= w

T

k

x

k

. (7.8)

The error signal is deﬁned as

e

k

= d

k

−y

k

= d

k

−x

T

k

w

k

. (7.9)

Assuming d

k

and x

k

to be jointly stationary, the Mean Square Error (MSE) is given

as

MSE = E[e

2

k

] = E[(d

k

−y

k

)

2

]

= E[(d

k

−x

T

k

w

k

)

2

]

= E[d

2

k

] +w

T

k

E[x

k

x

T

k

]w

k

−2E[d

k

x

T

k

]w

k

(7.10)

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where w

k

is assumed to be an array of optimum values and therefore it has been

taken out of the E() operator. The MSE then can be expressed as

MSE = ξ = σ

2

k

+w

T

k

Rw

k

−2p

T

w

k

(7.11)

where the signal variance σ

2

d

= E[d

2

k

] and the cross correlation vector p between the

desired response and the input signal is deﬁned as

p = E [d

k

x

k

] = E

_

d

k

x

k

d

k

x

k−1

d

k

x

k−2

d

k

x

k−N

_

. (7.12)

The input correlation matrix R is deﬁned as an (N + 1) (N + 1) square matrix,

where

R = E

_

x

k

x

T

k

_

= E

_

¸

¸

¸

¸

¸

¸

¸

¸

¸

¸

¸

_

x

2

k

x

k

x

k−1

x

k

x

k−2

x

k

x

k−N

x

k−1

x

k

x

2

k−1

x

k−1

x

k−2

x

k−1

x

k−N

x

k−2

x

k

x

k−2

x

k−1

x

2

k−2

x

k−2

x

k−N

.

.

.

.

.

.

.

.

.

.

.

.

x

k−N

x

k

x

k−N

x

k−1

x

k−N

x

k−2

x

2

k−N

_

¸

¸

¸

¸

¸

¸

¸

¸

¸

¸

¸

_

. (7.13)

Clearly, MSE is a function of w

k

. On equating

∂ξ

∂w

k

to 0, we get the condition for

minimum MSE (MMSE) which is known as Wiener solution:

w

k

= R

−1

p. (7.14)

Hence, MMSE is given by the equation

MMSE = ξ

min

= σ

2

d

−p

T

w

k

. (7.15)

7.2.4 Choice of Algorithms for Adaptive Equalization

Since an adaptive equalizer compensates for an unknown and time varying channel,

it requires a speciﬁc algorithm to update the equalizer coeﬃcients and track the

channel variations. Factors which determine algorithm’s performance are:

Rate of convergence: Number of iterations required for an algorithm, in re-

sponse to a stationary inputs, to converge close enough to optimal solution. A fast

rate of convergence allows the algorithm to adapt rapidly to a stationary environ-

ment of unknown statistics.

Misadjustment: Provides a quantitative measure of the amount by which the

ﬁnal value of mean square error, averaged over an ensemble of adaptive ﬁlters,

deviates from an optimal mean square error.

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Computational complexity: Number of operations required to make one com-

plete iteration of the algorithm.

Numerical properties: Inaccuracies like round-oﬀ noise and representation

errors in the computer, which inﬂuence the stability of the algorithm.

Three classic equalizer algorithms are primitive for most of today’s wireless stan-

dards. These include the Zero Forcing Algorithm (ZF), the Least Mean Square Algo-

rithm (LMS), and the Recursive Least Square Algorithm (RLS). Below, we discuss

a few of the adaptive algorithms.

Least Mean Square (LMS) Algorithm

LMS algorithm is the simplest algorithm based on minimization of the MSE between

the desired equalizer output and the actual equalizer output, as discussed earlier.

Here the system error, the MSE and the optimal Wiener solution remain the same

as given the adaptive equalization framework.

In practice, the minimization of the MSE is carried out recursively, and may be

performed by use of the stochastic gradient algorithm. It is the simplest equalization

algorithm and requires only 2N+1 operations per iteration. The ﬁlter weights are

updated by the update equation. Letting the variable n denote the sequence of

iteration, LMS is computed iteratively by

w

k

(n + 1) = w

k

(n) +µe

k

(n) x(n −k) (7.16)

where the subscript k denotes the kth delay stage in the equalizer and µ is the step

size which controls the convergence rate and stability of the algorithm.

The LMS equalizer maximizes the signal to distortion ratio at its output within

the constraints of the equalizer ﬁlter length. If an input signal has a time dispersion

characteristics that is greater than the propagation delay through the equalizer, then

the equalizer will be unable to reduce distortion. The convergence rate of the LMS

algorithm is slow due to the fact that there is only one parameter, the step size, that

controls the adaptation rate. To prevent the adaptation from becoming unstable,

the value of µ is chosen from

0 < µ < 2

_

N

i=1

λ

i

(7.17)

where λ

i

is the i-th eigenvalue of the covariance matrix R.

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Normalized LMS (NLMS) Algorithm

In the LMS algorithm, the correction that is applied to w

k

(n) is proportional to

the input sample x(n −k). Therefore when x(n −k) is large, the LMS algorithm

experiences gradient noise ampliﬁcation. With the normalization of the LMS step

size by |x(n)|

2

in the NLMS algorithm, this problem is eliminated. Only when

x(n−k) becomes close to zero, the denominator term|x(n)|

2

in the NLMS equation

becomes very small and the correction factor may diverge. So, a small positive

number ε is added to the denominator term of the correction factor. Here, the step

size is time varying and is expressed as

µ(n) =

β

|x(n)|

2

+ε

. (7.18)

Therefore, the NLMS algorithm update equation takes the form of

w

k

(n + 1) = w

k

(n) +

β

|x(n)|

2

+ε

e

k

(n) x(n −k) . (7.19)

7.3 Diversity

Diversity is a method used to develop information from several signals transmitted

over independent fading paths. It exploits the random nature of radio propagation

by ﬁnding independent signal paths for communication. It is a very simple concept

where if one path undergoes a deep fade, another independent path may have a

strong signal. As there is more than one path to select from, both the instantaneous

and average SNRs at the receiver may be improved. Usually diversity decisions are

made by receiver. Unlike equalization, diversity requires no training overhead as a

training sequence is not required by transmitter. Note that if the distance between

two receivers is a multiple of λ/2, there might occur a destructive interference be-

tween the two signals. Hence receivers in diversity technique are used in such a

way that the signal received by one is independent of the other. Diversity can be of

various forms, starting from space diversity to time diversity. We take up the types

one by one in the sequel.

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Figure 7.3: Receiver selection diversity, with M receivers.

7.3.1 Diﬀerent Types of Diversity

Space Diversity

A method of transmission or reception, or both, in which the eﬀects of fading are

minimized by the simultaneous use of two or more physically separated antennas,

ideally separated by one half or more wavelengths. Signals received from spatially

separated antennas have uncorrelated envelopes.

Space diversity reception methods can be classiﬁed into four categories: selection,

feedback or scanning, maximal ratio combining and equal gain combining.

(a) Selection Diversity:

The basic principle of this type of diversity is selecting the best signal among all

the signals received from diﬀerent branches at the receiving end. Selection Diversity

is the simplest diversity technique. Figure 7.3 shows a block diagram of this method

where ’M’ demodulators are used to provide M diversity branches whose gains are

adjusted to provide the same average SNR for each branch. The receiver branches

having the highest instantaneous SNR is connected to the demodulator.

Let M independent Rayleigh fading channels are available at a receiver. Each

channel is called a diversity branch and let each branch has the same average SNR.

The signal to noise ratio is deﬁned as

SNR = Γ =

E

b

N

0

α

2

(7.20)

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where E

b

is the average carrier energy, N

0

is the noise PSD, α is a random variable

used to represent amplitude values of the fading channel.

The instantaneous SNR(γ

i

) is usually deﬁned as γ

i

= instantaneous signal power

per branch/mean noise power per branch. For Rayleigh fading channels, α has a

Rayleigh distribution and so α

2

and consequently γ

i

have a chi-square distribution

with two degrees of freedom. The probability density function for such a channel is

p (γ

i

) =

1

Γ

e

−γ

i

Γ

. (7.21)

The probability that any single branch has an instantaneous SNR less than some

deﬁned threshold γ is

Pr [γ

i

≤ γ] =

γ

_

0

p (γ

i

) dγ

i

=

γ

_

0

1

Γ

e

−γ

i

Γ

dγ

i

= 1 −e

−γ

Γ

= P(Γ). (7.22)

Similarly, the probability that all M independent diversity branches receive signals

which are simultaneously less than some speciﬁc SNR threshold γ is

Pr [γ

1

, γ

2

, . . . , γ

M

≤ γ] =

_

1 −e

−γ

Γ

_

M

= P

M

(γ) (7.23)

where P

M

(γ) is the probability of all branches failing to achieve an instantaneous

SNR = γ. Quite clearly, P

M

(Γ) < P(Γ). If a single branch achieves SNR > γ, then

the probability that SNR > γ for one or more branches is given by

Pr [γ

i

> γ] = 1 −P

M

(γ) = 1 −

_

1 −e

−γ

Γ

_

M

(7.24)

which is more than the required SNR for a single branch receiver. This expression

shows the advantage when a selection diversity is used.

To determine of average signal to noise ratio, we ﬁrst ﬁnd out the pdf of γ as

p

M

(γ) =

d

dγ

P

M

(γ) =

M

Γ

_

1 −e

−

γ

/

Γ

_

M−1

e

−

γ

/

Γ

. (7.25)

The average SNR, ¯ γ, can be then expressed as

¯ γ =

∞

_

0

γp

M

(γ) dγ = Γ

∞

_

0

Mx

_

1 −e

−x

_

M−1

e

−x

dx (7.26)

where x = γ/Γ and Γ is the average SNR for a single branch, when no diversity is

used.

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This equation shows an average improvement in the link margin without requir-

ing extra transmitter power or complex circuitry, and it is easy to implement as

it needed a monitoring station and an antenna switch at the receiver. It is not an

optimal diversity technique as it doesn’t use all the possible branches simultaneously.

(b) Feedback or Scanning Diversity:

Scanning all the signals in a ﬁxed sequence until the one with SNR more than a

predetermined threshold is identiﬁed. Feedback or scanning diversity is very similar

to selection diversity except that instead of always using the best of N signals, the N

signals are scanned in a ﬁxed sequence until one is found to be above a predetermined

threshold. This signal is then received until it falls below threshold and the scanning

process is again initiated. The resulting fading statistics are somewhat inferior, but

the advantage is that it is very simple to implement(only one receiver is required).

(c) Maximal Ratio Combining:

Signals from all of the m branches are weighted according to their individual

signal voltage to noise power ratios and then summed. Individual signals must be

cophased before being summed, which generally requires an individual receiver and

phasing circuit for each antenna element. Produces an output SNR equal to the

sum of all individual SNR. Advantage of producing an output with an acceptable

SNR even when none of the individual signals are themselves acceptable. Modern

DSP techniques and digital receivers are now making this optimal form, as it gives

the best statistical reduction of fading of any known linear diversity combiner. In

terms of voltage signal,

r

m

=

m

i=1

G

i

r

i

(7.27)

where G

i

is the gain and r

i

is the voltage signal from each branch.

(d) Equal Gain Combining:

In some cases it is not convenient to provide for the variable weighting capability

required for true maximal ratio combining. In such cases, the branch weights are

all set unity, but the signals from each branch are co-phased to provide equal gain

combining diversity. It allows the receiver to exploit signals that are simultaneously

received on each branch. Performance of this method is marginally inferior to max-

imal ratio combining and superior to Selection diversity. Assuming all the G

i

to be

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Figure 7.4: Maximal ratio combining technique.

unity, here,

r

m

=

m

i=1

r

i

. (7.28)

Polarization Diversity

Polarization Diversity relies on the decorrelation of the two receive ports to achieve

diversity gain. The two receiver ports must remain cross-polarized. Polarization

Diversity at a base station does not require antenna spacing. Polarization diversity

combines pairs of antennas with orthogonal polarizations (i.e. horizontal/vertical, ±

slant 45

o

, Left-hand/Right-hand CP etc). Reﬂected signals can undergo polarization

changes depending on the channel. Pairing two complementary polarizations, this

scheme can immunize a system from polarization mismatches that would otherwise

cause signal fade. Polarization diversity has prove valuable at radio and mobile com-

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munication base stations since it is less susceptible to the near random orientations

of transmitting antennas.

Frequency Diversity

In Frequency Diversity, the same information signal is transmitted and received

simultaneously on two or more independent fading carrier frequencies. 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. The probability of simultaneous fading will be the product of the individual

fading probabilities. This method is employed in microwave LoS links which carry

several channels in a frequency division multiplex mode (FDM). Main disadvantage

is that it requires spare bandwidth also as many receivers as there are channels used

for the frequency diversity.

Time Diversity

In time diversity, the signal representing the same information are sent over the

same channel at diﬀerent times. Time diversity repeatedly transmits information at

time spacings that exceeds the coherence time of the channel. Multiple repetition

of the signal will be received with independent fading conditions, thereby providing

for diversity. A modern implementation of time diversity involves the use of RAKE

receiver for spread spectrum CDMA, where the multipath channel provides redun-

dancy in the transmitted message. Disadvantage is that it requires spare bandwidth

also as many receivers as there are channels used for the frequency diversity. Two

important types of time diversity application is discussed below.

Application 1: RAKE Receiver

In CDMA spread spectrum systems, CDMA spreading codes are designed to provide

very low correlation between successive chips, propagation delay spread in the radio

channel provides multiple version of the transmitted signal at the receiver. Delaying

multipath components by more than a chip duration, will appear like uncorrelated

noise at a CDMA receiver. CDMA receiver may combine the time delayed versions

of the original signal to improve the signal to noise ratio at the receiver. RAKE

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Figure 7.5: RAKE receiver.

receiver collect the time shifted versions of the original signal by providing a sep-

arate correlation receiver for M strongest multipath components. Outputs of each

correlator are weighted to provide a better estimate of the transmitted signal than

provided by a single component. Demodulation and bit decisions are based on the

weighted output of the correlators. Schematic of a RAKE receiver is shown in Figure

7.5.

Application 2: Interleaver

In the encoded data bits, some source bits are more important than others, and

must be protected from errors. Many speech coder produce several important bits

in succession. Interleaver spread these bit out in time so that if there is a deep fade

or noise burst, the important bits from a block of source data are not corrupted

at the same time. Spreading source bits over time, it becomes possible to make

use of error control coding. Interleaver can be of two forms, a block structure or a

convolutional structure.

A block interleaver formats the encoded data into a rectangular array of m rows

and n columns, and interleaves nm bits at a time. Each row contains a word of

source data having n bits. an interleaver of degree m consists of m rows. source bits

are placed into the interleaver by sequentially increasing the row number for each

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successive bit, and forming the columns. The interleaved source data is then read

out row-wise and transmitted over the channel. This has the eﬀect of separating

the original source bits by m bit periods. At the receiver, de-interleaver stores the

received data by sequentially increasing the row number of each successive bit, and

then clocks out the data row-wise, one word at a time. Convolutional interleavers

are ideally suited for use with convolutional codes.

7.4 Channel Coding

In channel coding, redundant data bits are added in the transmitted message so

that if an instantaneous fade occurs in the channel, the data may still be recov-

ered at the receiver without the request of retransmission. A channel coder maps

the transmitted message into another speciﬁc code sequence containing more bits.

Coded message is then modulated for transmission in the wireless channel. Channel

Coding is used by the receiver to detect or correct errors introduced by the channel.

Codes that used to detect errors, are error detection codes. Error correction codes

can detect and correct errors.

7.4.1 Shannon’s Channel Capacity Theorem

In 1948, Shannon showed that by proper encoding of the information, errors induced

by a noise channel can be reduced to any desired level without sacriﬁcing the rate

of information transfer. Shannon’s channel capacity formula is applicable to the

AWGN channel and is given by:

C = Blog

2

_

1 +

S

N

_

= Blog

2

_

1 +

P

N

0

B

_

= Blog

2

_

1 +

E

b

R

b

N

0

B

_

(7.29)

where C is the channel capacity (bit/s), B is the channel bandwidth (Hz), P is the

received signal power (W), N

0

is the single sided noise power density (W/Hz), E

b

is

the average bit energy and R

b

is transmission bit rate.

Equation (7.29) can be normalized by the bandwidth B and is given as

C

B

= log

2

_

1 +

E

b

R

b

N

0

B

_

(7.30)

and the ratio C/B is denoted as bandwidth eﬃciency. Introduction of redundant

bits increases the transmission bit rate and hence it increases the bandwidth require-

ment, which reduces the bandwidth eﬃciency of the link in high SNR conditions, but

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provides excellent BER performance at low SNR values. This leads to the following

two inferences.

Corollary 1: While dealing within maximum channel capacity, introduction of re-

dundant bits increase the transmitter rate and hence bandwidth requirement also

increases, while decreasing the bandwidth eﬃciency, but it also decreases the BER.

Corollary 2: If data redundancy is not introduced in a wideband noisy environment,

error free performance in not possible (for example, CDMA communication in 3G

mobile phones).

A channel coder operates on digital message (or source) data by encoding the source

information into a code sequence for transmission through the channel. The error

correction and detection codes are classiﬁed into three groups based on their struc-

ture.

1. Block Code

2. Convolution Code

3. Concatenated Code.

7.4.2 Block Codes

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 n−k redundant bits are added to the k information bits for the purpose of

detecting and correcting errors. The block code is referred to as an (n, k) code, and

the rate of the code is deﬁned as R

c

= k/n and is equal to the rate of information

divided by the raw channel rate.

Parameters in Block Code

(a) Code Rate (R

c

): As deﬁned above, R

c

= k/n.

(b) Code Distance (d): Distance between two codewords is the number of ele-

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ments in which two codewords C

i

and C

j

diﬀers denoted by d (C

i

, C

j

). If the code

used is binary, the distance is known as ’Hamming distance’. For example d(10110,

11011) is 3. If the code ’C’ consists of the set of codewords, then the minimum

distance of the code is given by d

min

= min¦d (C

i

, C

j

)¦.

(c) Code Weight (w): Weight of a codeword 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. For example weight of a code 101101 is 4.

Ex 1: The block code C = 00000, 10100, 11110, 11001 can be used to represent two

bit binary numbers as:

• 00 – 00000

• 01 – 10100

• 10 – 11110

• 11 – 11001

Here number of codewords is 4, k = 2, and n = 5.

To encode a bit stream 1001010011

• First step is to break the sequence in groups of two bits, i.e., 10 01 01 00 11

• Next step is to replace each block by its corresponding codeword, i.e.,

11110 10100 10100 00000 11001

Quite clearly, here, d

min

= min¦d (C

i

, C

j

)¦ = 2.

Properties of Block Codes

(a) Linearity: Suppose C

i

and C

j

are two code words in an (n, k) block code. Let

α

1

and α

2

be any two elements selected from the alphabet. Then the code is said to

be linear if and only if α

1

C

1

+α

2

C

2

is also a code word. A linear code must contain

the all-zero code word.

(b) 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.

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(c) Cyclic: Cyclic codes are a subset of the class of linear codes which satisfy the

following cyclic shift property: If C = [C

n−1

, C

n−2

, ..., C

0

] is a code word of a cyclic

code, then [C

n−2

, C

n−3

, ..., C

0

, C

n−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.

In this context, it is important to know about Finite Field or Galois Field.

Let F be a ﬁnite set of elements on which two binary operations – addition (+) and

multiplication (.) are deﬁned. The set F together with the two binary operations is

called a ﬁeld if the following conditions are satisﬁed:

1. F is a commutative group under addition.

2. The set of nonzero elements in F is a commutative group under multiplication.

3. Multiplication is distributive over addition; that is, for any three elements a, b,

and c in F, a(b +c) = ab +ac

4. Identity elements 0 and 1 must exist in F satisfying a + 0 = a and a.1 = a.

5. For any a in F, there exists an additive inverse (−a) such that a + (−a) = 0.

6. For any a in F, there exists an multiplicative inverse a

−1

such that a.a

−1

= 1.

Depending upon the number of elements in it, a ﬁeld is called either a ﬁnite or an

inﬁnite ﬁeld. The examples of inﬁnite ﬁeld include Q (set of all rational numbers),

R (set of all real numbers), C (set of all complex numbers) etc. A ﬁeld with a ﬁnite

number of elements (say q) is called a ’Galois Field’ and is denoted by GF(q). A

ﬁnite ﬁeld entity p(x), called a polynomial, is introduced to map all symbols (with

several bits) to the element of the ﬁnite ﬁeld. A polynomial is a mathematical

expression

p (x) = p

0

+p

1

x +... +p

m

x

m

(7.31)

where the symbol x is called the indeterminate and the coeﬃcients p

0

, p

1

, ..., p

m

are

the elements of GF(q). The coeﬃcient p

m

is called the leading coeﬃcient. If p

m

is not equal to zero, then m is called the degree of the polynomial, denoted as deg

p(x). A polynomial is called monic if its leading coeﬃcient is unity. The division

algorithm states that for every pair of polynomials a(x) and b(x) in F(x), there

exists a unique pair of polynomials q(x), the quotient, and r(x), the remainder,

such that a(x) = q(x)b(x) + r(x), where deg r(x)¡deg b(x). A polynomial p(x) in

F(x) is said to be reducible if p(x)=a(x)b(x), otherwise it is called irreducible. A

monic irreducible polynomial of degree at least one is called a prime polynomial.

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An irreducible polynomial p(x) of degree ‘m’ is said to be primitive if the smallest

integer ‘n’ for which p(x) divides x

n

+1 is n = 2

m

−1. A typical primitive polynomial

is given by p(x) = x

m

+x + 1.

A speciﬁc type of code which obeys both the cyclic property as well as poly-

nomial operation is cyclic codes. Cyclic codes are a subset of the class of linear

codes which satisfy the cyclic property. These codes possess a considerable amount

of structure which can be exploited. A cyclic code can be generated by using a

generator polynomial g(p) of degree (n-k). The generator polynomial of an (n,k)

cyclic code is a factor of p

n

+ 1 and has the form

g (p) = p

n−k

+g

n−k−1

p

n−k−1

+ +g

1

p + 1. (7.32)

A message polynomial x(p) can also be deﬁned as

x(p) = x

k−1

p

k−1

+ +x

1

p +x

0

(7.33)

where (x

k−1

, . . . , x

0

) represents the k information bits. The resultant codeword c(p)

can be written as

c (p) = x(p) g (p) (7.34)

where c(p) is a polynomial of degree less than n. We would see an application of

such codes in Reed-Solomon codes.

Examples of Block Codes

(a) Single Parity Check Code: In single parity check codes (example: ASCII code),

an overall single parity check bit is appended to ’k’ information bits. Let the infor-

mation bit word be: (b

1

, b

2

, ..., b

k

), then parity check bit: p = b

1

+ b

2

+ ......... + b

k

modulo 2 is appended at the (k+1)th position, making the overall codeword: C =

(b

1

, b

2

, ..., b

k

, p). The parity bit may follow an even parity or an odd parity pattern.

All error patterns that change an odd number of bits are detectable, and all even

numbered error patterns are not detectable. However, such codes can only detect

the error, it cannot correct the error.

Ex. 2: Consider a (8,7) ASCII code with information codeword (0, 1, 0, 1, 1, 0, 0)

and encoded with overall even parity pattern. Thus the overall codeword is (0, 1, 0,

1, 1, 0, 0, 1) where the last bit is the parity bit. If there is a single error in bit 3: (0,

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1, 1, 1, 1, 0, 0, 1), then it can be easily checked by the receiver that now there are

odd number of 1’s in the codeword and hence there is an error. On the other hand,

if there are two errors, say, errors in bit 3 and 5: (0, 1, 1, 1, 0, 0, 0, 1), then error

will not be detected.

After decoding a received codeword, let p

c

be the probability that the decoder

gives correct codeword C, p

e

is the probability that the decoder gives incorrect

codeword C

,= C, and p

f

is the probability that the decoder fails to give a codeword.

In this case, we can write p

c

+p

e

+p

f

= 1.

If in an n-bit codeword, there are j errors and p is the bit error probability,

then the probability of obtaining j errors in this codeword is P

j

=

n

C

j

p

j

(1 −p)

n−j

.

Using this formula, for any (n, n −1) single parity check block code, we get

• p

c

= P

0

,

• p

e

= P

2

+P

4

+... +P

n

(n

= n if n is even, otherwise n

= n −1),

• p

f

= P

1

+P

3

+... +P

n

(n

= n −1 if n is even, otherwise n

= n).

As an example, for a (5,4) single parity check block code, p

c

= P

0

, p

e

= P

2

+ P

4

,

and p

f

= P

1

+P

3

+P

5

.

(b) Product Codes: Product codes are a class of linear block codes which pro-

vide error detection capability using product of two block codes. Consider that nine

information bits (1, 0, 1, 0, 0, 1, 1, 1, 0) are to be transmitted. These 9 bits can be

divided into groups of three information bits and (4,3) single parity check codeword

can be formed with even parity. After forming three codewords, those can be ap-

pended with a vertical parity bit which will form the fourth codeword. Thus the

following codewords are transmitted:

C1 = [1 0 1 0]

C2 = [0 0 1 1]

C3 = [1 1 0 0]

C4 = [0 1 0 1].

Now if an error occurs in the second bit of the second codeword, the received code-

words at the receiver would then be

C1 = [1 0 1 0]

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C2 = [0 1 1 1] ←

C3 = [1 1 0 0]

C4 = [0 1 0 1]

↑

and these would indicate the corresponding row and column position of the erroneous

bit with vertical and horizontal parity check. Thus the bit can be corrected. Here

we get a horizontal (4, 3) codeword and a vertical (4, 3) codeword and concatenating

them we get a (16, 9) product code. In general, a product code can be formed as

(n

1

, k

1

) & (n

2

, k

2

) →(n

1

n

2

, k

1

k

2

).

(c) Repetition Codes: In a (n,1) repetition code each information bit is repeated

n times (n should be odd) and transmitted. At the receiver, the majority decoding

principle is used to obtain the information bit. Accordingly, if in a group of n received

bit, 1 occurs a higher number of times than 0, the information bit is decoded as 1.

Such majority scheme works properly only if the noise aﬀects less than n/2 number

of bits.

Ex 3: Consider a (3,1) binary repetition code.

• For input bit 0, the codeword is (0 0 0) and for input bit 1, the codeword is

(1 1 1).

• If the received codeword is (0 0 0), i.e. no error, it is decoded as 0.

• Similarly, if the received codeword is (1 1 1), i.e. no error, it is decoded as 1.

• If the received codeword is (0 0 1) or (0 1 0) or (1 0 0), then error is detected

and it is decoded as 0 with majority decoding principle.

• If the received codeword is (0 1 1) or (1 1 0) or (1 0 1), once again error is

detected and it is decoded as 1 with majority decoding principle.

For such a (3,1) repetition code, p

c

= P

0

+P

1

, p

e

= P

2

+P

3

, and p

f

= 0.

(d) Hamming Codes: A binary Hamming code has the property that

(n, k) = (2

m

−1, 2

m

−1 −m) (7.35)

where k is the number of information bits used to form a n bit codeword, and m

is any positive integer. The number of parity symbols are n − k = m. Thus, a

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codeword is represented by C = [i

1

, ...i

n

, p

1

, ..., p

n−k

]. This is quite a useful code in

communication which is illustrated via the following example.

Ex 4: Consider a (7, 4) Hamming code. With three parity bits we can correct exactly

1 error. The parity bits may follow such a modulo 2 arithmetic:

p

1

= i

1

+i

2

+i

3

,

p

2

= i

2

+i

3

+i

4

,

p

3

= i

1

+i

3

+i

4

,

which is same as,

p

1

+i

1

+i

2

+i

3

= 0

p

2

+i

2

+i

3

+i

4

= 0

p

3

+i

1

+i

3

+i

4

= 0.

The transmitted codeword is then C = [i

1

, i

2

, ..., i

4

, p

1

, p

2

, p

3

].

Syndrome Decoding: For this Hamming code, let the received codeword be V =

[v

1

, v

2

, ..., v

4

, v

5

, v

6

, v

7

]. We deﬁne a syndrome vector S as

S = [S

1

S

2

S

3

]

S1 = v

1

+v

2

+v

3

+v

5

S2 = v

2

+v

3

+v

4

+v

6

S3 = v

1

+v

2

+v4 +v7

It is obvious that in case of no error, the syndrome vector is equal to zero. Corre-

sponding to this syndrome vector, there is an error vector e which can be obtained

from a syndrome table and ﬁnally the required codeword is taken as C = V +e. In

a nutshell, to obtain the required codeword, we perform the following steps:

1. Calculate S from decoder input V.

2. From syndrome table, obtain e corresponding to S.

3. The required codeword is then C = V +e.

A few cases are given below to illustrate the syndrome decoding.

1. Let C = [0 1 1 1 0 1 0] and V = [0 1 1 1 0 1 0]. This implies S = [0 0 0], and it

corresponds to e = [0 0 0 0 0 0 0]. Thus, C = V + e = [0 1 1 1 0 1 0].

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2. Let C = [1 1 0 0 0 1 0] and V = [1 1 0 1 0 1 0]. This means S = [0 1 1], from which

we get e = [0 0 0 1 0 0 0] which means a single bit error is there in the received bit

v

4

. This will be corrected by performing the operation C = V + e.

3. Another interesting case is, let C = [0 1 0 1 1 0 0] and V = [0 0 1 1 1 0 1] (two

errors at second and third bits). This makes S = [0 0 0] and as a result, e = [0 0

0 0 0 0 0]. However, C ,= V , and C = V + e implies the double error cannot be

corrected. Therefore a (7,4) Hamming code can correct only single bit error.

(e) Golay Codes: Golay codes are linear binary (23,12) codes with a minimum

distance of seven and a error correction capability of three bits. This is a special,

one of a kind code in that this is the only nontrivial example of a perfect code.

Every codeword lies within distance three of any codeword, thus making maximum

likelihood decoding possible.

(f) BCH Codes: BCH code is one of the most powerful known class of linear

cyclic block codes, known for their multiple error correcting ability, and the ease

of encoding and decoding. It’s block length is n = 2

m

− 1 for m ≥ 3 and number

of errors that they can correct is bounded by t < (2

m

−1)/2. Binary BCH codes

can be generalized to create classes of non binary codes which use m bits per code

symbol.

(g) Reed Solomon (RS) Codes: Reed-Solomon code is an important subset of

the BCH codes with a wide range of applications in digital communication and data

storage. Typical application areas are storage devices (CD, DVD etc.), wireless

communications, digital TV, high speed modems. It’s coding system is based on

groups of bits, such as bytes, rather than individual 0 and 1. This feature makes it

particularly good at dealing with burst of errors: six consecutive bit errors. Block

length of these codes is n = 2

m

−1, and can be extended to 2

m

or 2

m

+1. Number

of parity symbols that must be used to correct e errors is n − k = 2e. Minimum

distance d

min

= 2e + 1, and it achieves the largest possible d

min

of any linear code.

For US-CDPD, the RS code is used with m = 6. So each of the 64 ﬁeld elements

is represented by a 6 bit symbol. For this case, we get the primitive polynomial as

p(x) = x

6

+x + 1. Equating p(x) to 0 implies x

6

= x + 1.

The 6 bit representation of the ﬁnite ﬁeld elements is given in Table 7.1. The table

elements continue up to α

62

. However, to follow linearity property there should be

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Table 7.1: Finite ﬁeld elements for US-CDPD

α

5

α

4

α

3

α

2

α

1

α

0

1 0 0 0 0 0 1

α

1

0 0 0 1 0 0

α

2

0 0 1 0 0 0

. . . . . . .

. . . . . . .

α

6

= α + 1 0 0 0 0 1 1

. . . . . . .

. . . . . . .

a zero codeword, hence α

63

is assigned zero.

The encoding part of the RS polynomial is done as follows:

Information polynomial: d(x) = C

n−1

x

n−1

+C

n−2

x

n−2

+..... +C

2t

x

2t

,

Parity polynomial: p(x) = C

2t−1

x

2t−1

+... +C

0

,

Codeword polynomial: c(x) = d(x) +p(x).

Since generating an information polynomial is diﬃcult, so a generating polynomial is

used instead. Information polynomial is then the multiple of generating polynomial.

This process is given below.

Since this kind of codes are cyclic codes, we take a generating polynomial g(x)

such that d(x) = g(x)q(x) + r(x) where q(x) is the quotient polynomial and r(x)

is the remainder polynomial. The codeword polynomial would then be given as:

c(x) = g(x)q(x) + r(x) = p(x). If we assign a parity polynomial p(x) = r(x), then

the codeword polynomial c(x) = g(x)p(x) and the entire process becomes easier.

On the decoder side one has to ﬁnd a speciﬁc r(x) = p(x) or vice-versa, but due

to its complexity, it is mainly done using syndrome calculation. The details of such

a syndrome calculation can be found in [1].

7.4.3 Convolutional Codes

A continuous sequence of information bits is mapped into a continuous sequence of

encoder output bits. A convolutional code is generated by passing the information

sequence through a ﬁnite state shift register. Shift register contains ’N’ k-bit stages

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Figure 7.6: A convolutional encoder with n=2 and k=1.

and m linear algebraic function generators based on the generator polynomials.

Input data is shifted into and along the shift register, k-bits at a time. Number

of output bits for each k-bit user input data sequence is n bits, so the code rate

R

c

= k/n. The shift register of the encoder is initialized to all-zero-state before

Figure 7.7: State diagram representation of a convolutional encoder.

encoding operation starts. It is easy to verify that encoded sequence is 00 11 10

00 01 . . . for an input message sequence of 01011 . . .. Convolution codes may be

represented in various ways as given below.

State Diagram:

Since the output of the encoder is determined by the input and the current

state of the encoder, a state diagram can be used to represent the encoding process.

The state diagram is simply a graph of the possible states of the encoder and the

possible transitions from one state to another. The path information between the

states, denoted as b/c

1

c

2

, represents input information bit ’b’ and the corresponding

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Figure 7.8: Tree diagram representation of a convolutional encoder.

output bits (c

1

c

2

). Again, it is not diﬃcult to verify from the state diagram that an

input information sequence b = (1011) generates an encoded sequence c = (11, 10,

00, 01).

Tree Diagram:

The tree diagram shows the structure of the encoder in the form of a tree with the

branches representing the various states and the outputs of the coder. The encoded

bits are labeled on the branches of the tree. Given an input sequence, the encoded

sequence can be directly read from the tree. As an example, an input sequence

(1011) results in the encoded sequence (11, 10, 00, 01).

Figure 7.9: Trellis diagram of a convolutional encoder.

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Figure 7.10: Block diagram of a turbo encoder.

Trellis Diagram:

Tree reveals that the structure repeats itself once the number of stages is greater

than the constraint length. It is observed that all branches emanating from two

nodes having the same state are identical in the sense that they generate identical

output sequences. This means that the two nodes having the same label can be

merged. By doing this throughout the tree diagram, we obtain another diagram

called a Trellis Diagram which is more compact representation.

7.4.4 Concatenated Codes

Concatenated codes are basically concatenation of block and convolutional codes. It

can be of two types: serial and parallel codes. Below, we discuss a popular parallel

concatenated code, namely, turbo code.

Turbo Codes: A turbo encoder is built using two identical convolutional codes

of special type with parallel concatenation. An individual encoder is termed a com-

ponent encoder. An interleaver separates the two component encoders. The inter-

leaver is a device that permutes the data sequence in some predetermined manner.

Only one of the systematic outputs from the two component encoders is used to form

a codeword, as the systematic output from the other component encoder is only a

permuted version of the chosen systematic output. Figure 7.10 shows the block di-

agram of a turbo encoder using two identical encoders. The ﬁrst encoder outputs

the systematic V

0

and recursive convolutional V

1

sequences while the second en-

coder discards its systematic sequence and only outputs the recursive convolutional

V

2

sequence. Depending on the number of input bits to a component encoder it

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may be binary or m-binary encoder. Encoders are also categorized as systematic

or non-systematic. If the component encoders are not identical then it is called an

asymmetric turbo code.

7.5 Conclusion

Although a lot of advanced powerful techniques for mitigating the fading eﬀects

such as space diversity in MIMO systems, space-time block coding scheme, MIMO

equalization, BLAST architectures etc. have taken place in modern wireless com-

munication, nevertheless, the discussed topics in this chapter are the basic building

blocks for all such techniques and that stems the necessity for all these discussions.

The eﬀectiveness of the discussed topics would be more clear in the next chapter in

the context of diﬀerent multiple access techniques.

7.6 References

1. T. S. Rappaport, Wireless Communications: Principles and Practice, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

2. J. R. Treichler, C. R. Johnson (Jr.) and M. G. Larimore, Theory and Design

of Adaptive Filters. New Delhi: PHI, 2002.

3. S. Gravano, Introduction to Error Control Codes. NY: Oxford University

Press, 2001.

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Chapter 8

Multiple Access Techniques

Multiple access techniques are used to allow a large number of mobile users to share

the allocated spectrum in the most eﬃcient manner. As the spectrum is limited, so

the sharing is required to increase the capacity of cell or over a geographical area

by allowing the available bandwidth to be used at the same time by diﬀerent users.

And this must be done in a way such that the quality of service doesn’t degrade

within the existing users.

8.1 Multiple Access Techniques for Wireless Communi-

cation

In wireless communication systems it is often desirable to allow the subscriber to

send simultaneously information to the base station while receiving information from

the base station.

A cellular system divides any given area into cells where a mobile unit in each

cell communicates with a base station. The main aim in the cellular system design

is to be able to increase the capacity of the channel i.e. to handle as many calls

as possible in a given bandwidth with a suﬃcient level of quality of service. There

are several diﬀerent ways to allow access to the channel. These includes mainly the

following:

1) Frequency division multiple-access (FDMA)

2) Time division multiple-access (TDMA)

3) Code division multiple-access (CDMA)

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Table 8.1: MA techniques in diﬀerent wireless communication systems

Advanced Mobile Phone Systems: FDMA/FDD

Global System for Mobile: TDMA/FDD

U.S. Digital Cellular: TDMA/FDD

Japanese Digital Cellular: TDMA/FDD

CT2 Cordless Telephone: FDMA/TDD

Digital European Cordless Telephone: FDMA/TDD

U.S. Narrowband Spread Spectrum (IS-95): CDMA/FDD

4) Space Division Multiple access (SDMA)

FDMA,TDMA and CDMA are the three major multiple access techniques that

are used to share the available bandwidth in a wireless communication system.

Depending on how the available bandwidth is allocated to the users these techniques

can be classiﬁed as narrowband and wideband systems.

8.1.1 Narrowband Systems

The term narrowband is used to relate the bandwidth of the single channel to the

expected coherence bandwidth of the channel. The available spectrum is divided in

to a large number of narrowband channels. The channels are operated using FDD.

In narrow band FDMA, a user is assigned a particular channel which is not shared by

other users in the vicinity and if FDD is used then the system is called FDMA/FDD.

Narrow band TDMA allows users to use the same channel but allocated a unique

time slot to each user on the channel, thus separating a small number of users in time

on a single channel. For narrow band TDMA, there generally are a large number of

channels allocated using either FDD or TDD, each channel is shared using TDMA.

Such systems are called TDMA/FDD and TDMA/TDD access systems.

8.1.2 Wideband Systems

In wideband systems, the transmission bandwidth of a single channel is much larger

than the coherence bandwidth of the channel. Thus, multipath fading doesnt greatly

aﬀect the received signal within a wideband channel, and frequency selective fades

occur only in a small fraction of the signal bandwidth

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Figure 8.1: The basic concept of FDMA.

8.2 Frequency Division Multiple Access

This was the initial multiple-access technique for cellular systems in which each

individual user is assigned a pair of frequencies while making or receiving a call as

shown in Figure 8.1. One frequency is used for downlink and one pair for uplink.

This is called frequency division duplexing (FDD). That allocated frequency pair

is not used in the same cell or adjacent cells during the call so as to reduce the co

channel interference. Even though the user may not be talking, the spectrum cannot

be reassigned as long as a call is in place. Diﬀerent users can use the same frequency

in the same cell except that they must transmit at diﬀerent times.

The features of FDMA are as follows: The FDMA channel carries only one

phone circuit at a time. If an FDMA channel is not in use, then it sits idle and

it cannot be used by other users to increase share capacity. After the assignment

of the voice channel the BS and the MS transmit simultaneously and continuously.

The bandwidths of FDMA systems are generally narrow i.e. FDMA is usually

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implemented in a narrow band system The symbol time is large compared to the

average delay spread. The complexity of the FDMA mobile systems is lower than

that of TDMA mobile systems. FDMA requires tight ﬁltering to minimize the

adjacent channel interference.

8.2.1 FDMA/FDD in AMPS

The ﬁrst U.S. analog cellular system, AMPS (Advanced Mobile Phone System) is

based on FDMA/FDD. A single user occupies a single channel while the call is in

progress, and the single channel is actually two simplex channels which are frequency

duplexed with a 45 MHz split. When a call is completed or when a handoﬀ occurs

the channel is vacated so that another mobile subscriber may use it. Multiple or

simultaneous users are accommodated in AMPS by giving each user a unique signal.

Voice signals are sent on the forward channel from the base station to the mobile

unit, and on the reverse channel from the mobile unit to the base station. In AMPS,

analog narrowband frequency modulation (NBFM) is used to modulate the carrier.

8.2.2 FDMA/TDD in CT2

Using FDMA, CT2 system splits the available bandwidth into radio channels in the

assigned frequency domain. In the initial call setup, the handset scans the available

channels and locks on to an unoccupied channel for the duration of the call. Using

TDD(Time Division Duplexing ), the call is split into time blocks that alternate

between transmitting and receiving.

8.2.3 FDMA and Near-Far Problem

The near-far problem is one of detecting or ﬁltering out a weaker signal amongst

stronger signals. The near-far problem is particularly diﬃcult in CDMA systems

where transmitters share transmission frequencies and transmission time. In con-

trast, FDMA and TDMA systems are less vulnerable. FDMA systems oﬀer diﬀerent

kinds of solutions to near-far challenge. Here, the worst case to consider is recovery

of a weak signal in a frequency slot next to strong signal. Since both signals are

present simultaneously as a composite at the input of a gain stage, the gain is set

according to the level of the stronger signal; the weak signal could be lost in the

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noise ﬂoor. Even if subsequent stages have a low enough noise ﬂoor to provide

8.3 Time Division Multiple Access

In digital systems, continuous transmission is not required because users do not

use the allotted bandwidth all the time. In such cases, TDMA is a complimentary

access technique to FDMA. Global Systems for Mobile communications (GSM) uses

the TDMA technique. In TDMA, the entire bandwidth is available to the user but

only for a ﬁnite period of time. In most cases the available bandwidth is divided

into fewer channels compared to FDMA and the users are allotted time slots during

which they have the entire channel bandwidth at their disposal, as shown in Figure

8.2.

TDMA requires careful time synchronization since users share the bandwidth in

the frequency domain. The number of channels are less, inter channel interference

is almost negligible. TDMA uses diﬀerent time slots for transmission and reception.

This type of duplexing is referred to as Time division duplexing(TDD).

The features of TDMA includes the following: TDMA shares a single carrier fre-

quency with several users where each users makes use of non overlapping time slots.

The number of time slots per frame depends on several factors such as modulation

technique, available bandwidth etc. Data transmission in TDMA is not continuous

but occurs in bursts. This results in low battery consumption since the subscriber

transmitter can be turned OFF when not in use. Because of a discontinuous trans-

mission in TDMA the handoﬀ process is much simpler for a subscriber unit, since it

is able to listen to other base stations during idle time slots. TDMA uses diﬀerent

time slots for transmission and reception thus duplexers are not required. TDMA

has an advantage that is possible to allocate diﬀerent numbers of time slots per

frame to diﬀerent users. Thus bandwidth can be supplied on demand to diﬀerent

users by concatenating or reassigning time slot based on priority.

8.3.1 TDMA/FDD in GSM

As discussed earlier, GSM is widely used in Europe and other parts of the world.

GSM uses a variation of TDMA along with FDD. GSM digitizes and compresses

data, then sends it down a channel with two other streams of user data, each in its

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Figure 8.2: The basic concept of TDMA.

own time slot. It operates at either the 900 MHz or 1800 MHz frequency band. Since

many GSM network operators have roaming agreements with foreign operators, users

can often continue to use their mobile phones when they travel to other countries.

8.3.2 TDMA/TDD in DECT

DECT is a pan European standard for the digitally enhanced cordless telephony

using TDMA/TDD. DECT provides 10 FDM channels in the band 1880-1990 Mhz.

Each channel supports 12 users through TDMA for a total system load of 120 users.

DECT supports handover, users can roam over from cell to cell as long as they remain

within the range of the system. DECT antenna can be equipped with optional spatial

diversity to deal with multipath fading.

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8.4 Spread Spectrum Multiple Access

Spread spectrum multiple access (SSMA) uses signals which have a transmission

bandwidth whose magnitude is greater than the minimum required RF bandwidth.

A pseudo noise (PN) sequence converts a narrowband signal to a wideband noise

like signal before transmission. SSMA is not very bandwidth eﬃcient when used

by a single user. However since many users can share the same spread spectrum

bandwidth without interfering with one another, spread spectrum systems become

bandwidth eﬃcient in a multiple user environment.

There are two main types of spread spectrum multiple access techniques: Fre-

quency hopped multiple access (FHMA)

Direct sequence multiple access (DSMA) or

Code division multiple access (CDMA).

8.4.1 Frequency Hopped Multiple Access (FHMA)

This is a digital multiple access system in which the carrier frequencies of the in-

dividual users are varied in a pseudo random fashion within a wideband channel.

The digital data is broken into uniform sized bursts which is then transmitted on

diﬀerent carrier frequencies.

8.4.2 Code Division Multiple Access

In CDMA, the same bandwidth is occupied by all the users, however they are all

assigned separate codes, which diﬀerentiates them from each other (shown in Figure

8.3). CDMA utilize a spread spectrum technique in which a spreading signal (which

is uncorrelated to the signal and has a large bandwidth) is used to spread the narrow

band message signal.

Direct Sequence Spread Spectrum (DS-SS)

This is the most commonly used technology for CDMA. In DS-SS, the message signal

is multiplied by a Pseudo Random Noise Code. Each user is given his own codeword

which is orthogonal to the codes of other users and in order to detect the user, the

receiver must know the codeword used by the transmitter. There are, however, two

problems in such systems which are discussed in the sequel.

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Figure 8.3: The basic concept of CDMA.

CDMA/FDD in IS-95

In this standard, the frequency range is: 869-894 MHz (for Rx) and 824-849 MHz

(for Tx). In such a system, there are a total of 20 channels and 798 users per channel.

For each channel, the bit rate is 1.2288 Mbps. For orthogonality, it usually combines

64 Walsh-Hadamard codes and a m-sequence.

8.4.3 CDMA and Self-interference Problem

In CDMA, self-interference arises from the presence of delayed replicas of signal due

to multipath. The delays cause the spreading sequences of the diﬀerent users to

lose their orthogonality, as by design they are orthogonal only at zero phase oﬀset.

Hence in despreading a given user’s waveform, nonzero contributions to that user’s

signal arise from the transmissions of the other users in the network. This is distinct

from both TDMA and FDMA, wherein for reasonable time or frequency guardbands,

respectively, orthogonality of the received signals can be preserved.

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8.4.4 CDMA and Near-Far Problem

The near-far problem is a serious one in CDMA. This problem arises from the fact

that signals closer to the receiver of interest are received with smaller attenuation

than are signals located further away. Therefore the strong signal from the nearby

transmitter will mask the weak signal from the remote transmitter. In TDMA and

FDMA, this is not a problem since mutual interference can be ﬁltered. In CDMA,

however, the near-far eﬀect combined with imperfect orthogonality between codes

(e.g. due to diﬀerent time sifts), leads to substantial interference. Accurate and fast

power control appears essential to ensure reliable operation of multiuser DS-CDMA

systems.

8.4.5 Hybrid Spread Spectrum Techniques

The hybrid combinations of FHMA, CDMA and SSMA result in hybrid spread

spectrum techniques that provide certain advantages. These hybrid techniques are

explained below,

Hybrid FDMA/CDMA (FCDMA):

An alternative to the CDMA technique in which the available wideband spectrum

is divided into a smaller number of sub spectra with smaller bandwidths. The smaller

sub channels become narrow band CDMA systems with processing gain lower than

the original CDMA system. In this scheme the required bandwidth need not be

contiguous and diﬀerent user can be allotted diﬀerent sub spectrum bandwidths

depending on their requirements. The capacity of this hybrid FCDMA technique is

given by the sum of the capacities of a system operating in the sub spectra.

Hybrid Direct Sequence/Frequency Hopped Multiple Access Techniques (DS/FHMA):

A direct sequence modulated signal whose center frequency is made to hop pe-

riodically in a pseudo random fashion is used in this technique. One of the advan-

tages using this technique is they avoid near-far eﬀect. However, frequency hopped

CDMA systems are not adaptable to the soft handoﬀ process since it is diﬃcult

to synchronize the frequency hopped base station receiver to the multiple hopped

signals. Time and Code Division Multiple Access (TCDMA):

In this TCDMA method diﬀerent cells are allocated diﬀerent spreading codes.

In each cell, only one user per cell is allotted a particular time slot. Thus at any

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time only one user is transmitting in each cell. When a handoﬀ takes place the

spreading code of that user is changed to the code of the new cell. TCDMA also

avoids near-far eﬀect as the number of users transmitting per cell is one.

Time Division Frequency Hopping (TDFH):

This technique has been adopted for the GSM standard, where the hopping se-

quence is predeﬁned and the subscriber is allowed to hop only on certain frequencies

which are assigned to a cell. The subscriber can hop to a new frequency at the start

of a new TDMA frame, thus avoiding a severe fade or erasure event on a particu-

lar channel. This technique has the advantage in severe multipath or when severe

channel interference occurs.

8.5 Space Division Multiple Access

SDMA utilizes the spatial separation of the users in order to optimize the use of the

frequency spectrum. A primitive form of SDMA is when the same frequency is re-

used in diﬀerent cells in a cellular wireless network. The radiated power of each user

is controlled by Space division multiple access. SDMA serves diﬀerent users by using

spot beam antenna. These areas may be served by the same frequency or diﬀerent

frequencies. However for limited co-channel interference it is required that the cells

be suﬃciently separated. This limits the number of cells a region can be divided into

and hence limits the frequency re-use factor. A more advanced approach can further

increase the capacity of the network. This technique would enable frequency re-use

within the cell. In a practical cellular environment it is improbable to have just one

transmitter fall within the receiver beam width. Therefore it becomes imperative

to use other multiple access techniques in conjunction with SDMA. When diﬀerent

areas are covered by the antenna beam, frequency can be re-used, in which case

TDMA or CDMA is employed, for diﬀerent frequencies FDMA can be used.

8.6 Conclusion

In this chapter, we have mainly discussed the ﬁxed assignment type of MA tech-

niques, namely, FDMA, TDMA and CDMA. We have, however, intensionally not

covered the reservation-based MA schemes such as packet reservation MA or polling

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or token passing etc. The main idea to discuss only the basic MA techniques has

been to grow up a fair idea about the resource sharing in a wireless media when there

are many users, keeping the QoS view point in mind. The readers are encouraged

to go through the advanced topics once they ﬁnish reading the discussed 8 chapters

in this lecture notes.

8.7 References

1. T. S. Rappaport, Wireless Communications: Principles and Practice, 2nd ed.

Singapore: Pearson Education, Inc., 2002.

2. K. Feher, Wireless Digital Communications: Modulation and Spread Spectrum

Applications. Upper Saddle River, NJ: Prentice Hall, 1995.

3. J. G. Proakis, Digital Communications, 4th ed. NY: McGraw Hill, 2000.

4. G. R. Cooper and C. D. McGillem, Modern Communications and Spread Spec-

trum, NY: McGraw Hill, 1986.

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Preface

It’s been many years that I’m teaching the subject “Mobile Communication” (EC632) to the IIT Guwahati students and the current lecture notes intend to act as a supplement to that course so that our students can have an access to this course anytime. This course is mainly aimed toward senior year students of the ECE discipline, and in particular, for the ﬁnal year BTech, ﬁrst year MTech and PhD students. However, this does not necessarily imply that any other discipline students can not study this course. Rather, they also should delve deeper into this course since mobile communication is a familiar term to everyone nowadays. Although the communication aspects of this subject depends on the fundamentals of another interesting subject, communication engineering, I would strongly advocate the engineering students to go through the same in order to grow up adequate interest in this ﬁeld. In fact, the present lecture notes are designed in such a way that even a non-ECE student also would get certain basic notions of this subject. The entire lecture notes are broadly divided into 8 chapters, which, I consider to be most rudimentary topics to know the basics of this subject. The advance level topics are avoided intensionally so as to give space to the possibility of developing another lecture note in that area. In fact, this area is so vast and changing so fast over time, there is no limit of discussing the advanced level topics. The current focus has been therefore to deal with those main topics which would give a senior student suﬃcient exposure to this ﬁeld to carry out further study and/or research. Initially, after dealing with the introductory concepts (i.e., what is mobile communication, how a mobile call is made etc) and the evolution of mobile communication systems till the present day status, the cellular engineering fundamentals are discussed at length to make the students realize the importance of the practical engineering aspects of this subject. Next, the diﬀerent kinds of mobile communication channels is taken up and large scale path loss model as well as small scale fading eﬀects are dealt, both with simulation and statistical approaches. To enhance the link performance amidst the adverse channel conditions, the transmitter and receiver techniques are

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com
. It should also be mentioned that many ﬁgures in the lecture notes have been adopted from some standard text books to keep the easy ﬂow of the understanding of the topics. Once Bertrand Russell said “Science may set limits to knowledge. My sincere acknowledgment should also go to my parents and my younger brother who have nicely reciprocated my oblivion nature by their nourishing and generous attitude toward me since my childhood. It is further extended with three main signal processing techniques at the receiver. I’m fortunate to have a group of energetic students who have helped me a lot. albeit a bit late. Finally. diﬀerent kinds of multiple access techniques are covered at length with the emphasis on how several mobile communication techniques evolve via this. I have received kind helps from my friends. If the readers can visualize the continuously changing technology in this ﬁeld after reading this lecture notes and also can dream about a future career in the same. Finally. colleagues as well as my post graduate and doctoral students which I must acknowledge at the onset. diversity and channel coding. namely.jntuworld.
Abhijit Mitra November 2009
ii
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discussed next.www. I’ll consider my endeavor to be successful. My best wishes to all the readers. In general.jntuworld. During the process of developing the lecture notes. but should not set limits to imagination”. It is for them only I could ﬁnish this project. an author becomes happy if he/she sees that his/her creation could instill certain sparks in the reader’s mind. equalization. The same is true for me too.

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Contents

1 Introductory Concepts 1.1 1.2 1.3 1.4 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Evolution of Mobile Radio Communications . . . . . . . . . . . . . . Present Day Mobile Communication . . . . . . . . . . . . . . . . . . Fundamental Techniques . . . . . . . . . . . . . . . . . . . . . . . . . 1.4.1 1.5 Radio Transmission Techniques . . . . . . . . . . . . . . . . . 1 1 1 3 4 5 7 7 8 8 10 10 11 11 11 12 12 12 13 14 14 16 18 19

How a Mobile Call is Actually Made? . . . . . . . . . . . . . . . . . 1.5.1 1.5.2 1.5.3 Cellular Concept . . . . . . . . . . . . . . . . . . . . . . . . . Operational Channels . . . . . . . . . . . . . . . . . . . . . . Making a Call . . . . . . . . . . . . . . . . . . . . . . . . . .

1.6 1.7

Future Trends . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2 Modern Wireless Communication Systems 2.1 2.2 1G: First Generation Networks . . . . . . . . . . . . . . . . . . . . . 2G: Second Generation Networks . . . . . . . . . . . . . . . . . . . . 2.2.1 2.2.2 2.2.3 2.3 TDMA/FDD Standards . . . . . . . . . . . . . . . . . . . . . CDMA/FDD Standard . . . . . . . . . . . . . . . . . . . . . 2.5G Mobile Networks . . . . . . . . . . . . . . . . . . . . . .

3G: Third Generation Networks . . . . . . . . . . . . . . . . . . . . . 2.3.1 2.3.2 2.3.3 2.3.4 3G Standards and Access Technologies . . . . . . . . . . . . . 3G W-CDMA (UMTS) . . . . . . . . . . . . . . . . . . . . . 3G CDMA2000 . . . . . . . . . . . . . . . . . . . . . . . . . . 3G TD-SCDMA . . . . . . . . . . . . . . . . . . . . . . . . .

2.4

Wireless Transmission Protocols . . . . . . . . . . . . . . . . . . . . iii

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2.4.1 2.4.2 2.4.3 2.4.4 2.4.5 2.4.6 2.5 2.6

Wireless Local Loop (WLL) and LMDS . . . . . . . . . . . . Bluetooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Wireless Local Area Networks (W-LAN) . . . . . . . . . . . . WiMax . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Zigbee . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Wibree . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

19 19 20 21 21 21 22 22 23 23 23 24 27 27 27 28 29 31 33 33 34 34 37 38 38 40 43 46 47 53

Conclusion: Beyond 3G Networks . . . . . . . . . . . . . . . . . . . . References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3 The Cellular Engineering Fundamentals 3.1 3.2 3.3 3.4 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . What is a Cell? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Frequency Reuse . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Channel Assignment Strategies . . . . . . . . . . . . . . . . . . . . . 3.4.1 3.4.2 3.5 Fixed Channel Assignment (FCA) . . . . . . . . . . . . . . . Dynamic Channel Assignment (DCA) . . . . . . . . . . . . .

Handoﬀ Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3.5.1 3.5.2 3.5.3 3.5.4 Factors Inﬂuencing Handoﬀs . . . . . . . . . . . . . . . . . . Handoﬀs In Diﬀerent Generations . . . . . . . . . . . . . . . Handoﬀ Priority . . . . . . . . . . . . . . . . . . . . . . . . . A Few Practical Problems in Handoﬀ Scenario . . . . . . . .

3.6

Interference & System Capacity . . . . . . . . . . . . . . . . . . . . . 3.6.1 3.6.2 Co-channel interference (CCI) . . . . . . . . . . . . . . . . . . Adjacent Channel Interference (ACI) . . . . . . . . . . . . . .

3.7

Enhancing Capacity And Cell Coverage . . . . . . . . . . . . . . . . 3.7.1 3.7.2 3.7.3 3.7.4 The Key Trade-oﬀ . . . . . . . . . . . . . . . . . . . . . . . . Cell-Splitting . . . . . . . . . . . . . . . . . . . . . . . . . . . Sectoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Microcell Zone Concept . . . . . . . . . . . . . . . . . . . . .

3.8 3.9

Trunked Radio System . . . . . . . . . . . . . . . . . . . . . . . . . . References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

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4 Free Space Radio Wave Propagation 4.1 4.2 4.3 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Free Space Propagation Model . . . . . . . . . . . . . . . . . . . . . Basic Methods of Propagation . . . . . . . . . . . . . . . . . . . . . . 4.3.1 4.3.2 4.3.3 4.4 4.5 Reﬂection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Diﬀraction . . . . . . . . . . . . . . . . . . . . . . . . . . . .

54 54 55 57 57 58 58 59 63 64 66 68 69 69 70 70 70 71 72 72 73 73 73 73 75 75 75 76 76 76 77 77

Scattering . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

Two Ray Reﬂection Model . . . . . . . . . . . . . . . . . . . . . . . . Diﬀraction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4.5.1 4.5.2 4.5.3 Knife-Edge Diﬀraction Geometry . . . . . . . . . . . . . . . . Fresnel Zones: the Concept of Diﬀraction Loss . . . . . . . . Knife-edge diﬀraction model . . . . . . . . . . . . . . . . . . .

4.6

Link Budget Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . 4.6.1 4.6.2 Log-distance Path Loss Model . . . . . . . . . . . . . . . . . Log Normal Shadowing . . . . . . . . . . . . . . . . . . . . .

4.7

Outdoor Propagation Models . . . . . . . . . . . . . . . . . . . . . . 4.7.1 4.7.2 Okumura Model . . . . . . . . . . . . . . . . . . . . . . . . . Hata Model . . . . . . . . . . . . . . . . . . . . . . . . . . . .

4.8

Indoor Propagation Models . . . . . . . . . . . . . . . . . . . . . . . 4.8.1 4.8.2 4.8.3 Partition Losses Inside a Floor (Intra-ﬂoor) . . . . . . . . . . Partition Losses Between Floors (Inter-ﬂoor) . . . . . . . . . Log-distance Path Loss Model . . . . . . . . . . . . . . . . .

4.9

Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

4.10 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5 Multipath Wave Propagation and Fading 5.1 5.2 Multipath Propagation . . . . . . . . . . . . . . . . . . . . . . . . . . Multipath & Small-Scale Fading . . . . . . . . . . . . . . . . . . . . 5.2.1 5.2.2 5.2.3 5.3 Fading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Multipath Fading Eﬀects . . . . . . . . . . . . . . . . . . . . Factors Inﬂuencing Fading . . . . . . . . . . . . . . . . . . . .

Types of Small-Scale Fading . . . . . . . . . . . . . . . . . . . . . . . 5.3.1 Fading Eﬀects due to Multipath Time Delay Spread . . . . . v

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. . . .3. . . . .2. .www. . . . . . . . . . . . .3. . . . . . . 104
6. .1 6. .5.5. .2. . . .2. . . .3 5. . . . . . . . . . Saleh and Valenzuela Indoor Statistical Model . . . .2 5. . .4 5. . . . . . 104 Analog and Digital Modulation Techniques . . SIRCIM/SMRCIM Indoor/Outdoor Statistical Models . . . . . . . . . . . . .jntuworld. Second Order Statistics . . . . . . . .6. . . . . . . . . . . . . . . . .6. . . . . . . . . . . . . . . . . Rayleigh Simulator with Wide Range of Channel Conditions Two-Ray Rayleigh Faded Model . . . .4 NLoS Propagation: Rayleigh Fading Model . . .4
Fading Eﬀects due to Doppler Spread . . . . . . . . . . . . . . . Doppler Shift . . . . . . . . Impulse Response Model of a Multipath Channel . .5 Choice of Modulation Scheme . LoS Propagation: Rician Fading Model . . . .
6 Transmitter and Receiver Techniques 6. . . . . . . . . . . .5. . . . Generalized Model: Nakagami Distribution . . . . . .3
Signal Space Representation of Digitally Modulated Signals . . . . . . . . . .5. . . . .6
Simulation of Rayleigh Fading Models . .3 5. . Relation Between Bandwidth and Received Power . . . . . . . 103 Amplitude and Angle Modulation . . . . . . .8
Conclusion
References . . . . . . . . . . . . . . . .2 Time Dispersion Parameters . . 5. . . . . . . . .6. . .
5. . .7 5. . 102 Advantages of Modulation . . . . . . .2. . . . . . . . . . . . . . . . . . . .1 5.
78 79 80 82 84 85 87 87 89 90 91 93 94 95 96 96 96 97 97 98 98 99 99 101
Multipath Channel Parameters . . . . . .1 5. . . . .1 5.4 5.3 5. . . . . . . . . . . . . . . . . . .6 5. . . . . . . .3. . . . . . . . . . . . . . . . . . . . . . . . . . . . . .jntuworld. .
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5. . . . . . . . . .6. . . . . . . .1 6. . . . .5 5. . . . . .4 6. . . 101 Modulation .5
Statistical models for multipath propagation . . . .5 5. 104 vi
www. . . . . 5. . . . .3 6. . . . . .4.4. .3. . . .7 5. .3. . . . Linear Time Varying Channels (LTV) . . . . . . . .
5. . . .2.
Frequency Dispersion Parameters . . . Small-Scale Multipath Measurements . . . . . . . . . . . . . . . . . . . . . . . . . 5. . . . . . . . . . . . . .6. . . . . . . . . . .2
Introduction . . . . . . . . .com
.2 5. 101 6.3. . . . . Clarke and Gans’ Model: with Doppler Eﬀect . 102 Linear and Non-linear Modulation Techniques .2 5.6. . . . . . . . . . .6 Clarke’s Model: without Doppler Eﬀect . . . . . . .

. . . . . . . . . . . . . .3 Nyquist pulse shaping . .8
Nonlinear Modulation Techniques . . . . 122 6. . . . . . . . . . . . . . . . . . .jntuworld. . 119 6. . . 113
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6. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .9
GMSK Scheme . . . . . . . . . . . .4
Complex Representation of Linear Modulated Signals and Band Pass Systems . . . . . . . . . . . . . . . . . . . . . . .13 Example of a Multicarrier Modulation: OFDM . . . . . . . . . . . . . 106 BPSK . . . . . . . . . . . . . . 107 Oﬀset-QPSK . . .2 129
Introduction . . . . . . . . . . . . . . . . . . . . . 118
6. . . . . . 123 6. . . . . . . . . . . . . . . . . . . . . . . .1 Inter Channel Interference . .5. . . . . . . . . . . . . . . . . 125 6. . . . . . . . . . . 130 7. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5. . . . . . . . . . . . .com
. . 110
6. .1 6. . . . . . . . . . . . . . . . . . . . . 121 6. 122 6. . . . . . . . . . . . . . .15 References . . . . . . .12 Receiver performance in multipath channels . . . . . . . . . . . . . . . . . . .5. . . . . .www. . . . . . . . . . . . .2 Mathematical Description of OFDM . 114 6. . .6 6. . . . . . . . .8. .jntuworld. . . .7. . . . . . . . . . . .2. . . . . . . 111 6. .2 Power Ampliﬁer Nonlinearity . . . . . . . . . . . . .8. . .1 6. . . . . . . . . . . . . . . . . 127
6. . . . . . . . . . . . . . . . 106
Amplitude Modulation (DSBSC) .2 Angle Modulation (FM and PM) . . . . . . . . . . . . . . .11. .5. . . . . . . . . . . . . . . . . . . . 121 6. . . 112 Raised Cosine Roll-Oﬀ Filtering . . . . . . . . . . . . . . . . . .7
Line Coding . 110 Pulse Shaping . . 116
6. . . . 125 6. . . .12. . . . 123 6.14 Conclusion . . . 107 QPSK . . . . . . . . . . . . . . .3 6. . . . . . . . . 108 π/4 DQPSK . . . . . . . .5. . . . . . . . 128 7 Techniques to Mitigate Fading Eﬀects 7. .10 GMSK Generator . . . . . . . . . . .1 Orthogonality of Signals . .2 6.11. . . . . . . . . . . . . . . .11 Two Practical Issues of Concern . . . . .1 A Mathematical Framework . . . . . . 113 Realization of Pulse Shaping Filters . . . . . 131 vii
www. . . . . .7. . .1 Bit Error Rate and Symbol Error Rate . . . . .1 7. . . . . . . .5
. . . . . . . . . . . . . . . .1 6. . 129 Equalization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114 BFSK .13.5
Linear Modulation Techniques 6. . . . . . . . . . . . . . . . . . . . . . . . . .7.13. . . . . 105
6. . . . . . . . . .4 6. . . . . . .2 6. . . . . . . . . . . .

2.3
Zero Forcing Equalization . . 158 Wideband Systems . . . .4. . . . .www. .com
7. . . . . . . . . . . . . . . . . . . .1 7. .1 8. . . . . . . . . . . .4. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
7. . .jntuworld. . . . . . . . . . . . . . . .4.2 8. . . . . . . . .4. . . . . . . .3 8.1. . . . . . . . .3. . . . . . 136 7. . . . .5 Frequency Hopped Multiple Access (FHMA) . . . . . .1 8. . . . . . . . . . . . . . . . . . . . .2. . . . . . . . . . 156
7. . . .2. . . . . . .2 TDMA/FDD in GSM . . . . .4. . . . . . . 160 FDMA/TDD in CT2 . . . . . . . . . .4. . . . 143 Block Codes . . . . . . . . . . . . . . . . . . .2. .1 8. . . . . . . . . . .3. . . . . .3 FDMA/FDD in AMPS . . . 161 8. . . . . . . . . . . . . . . . 160 FDMA and Near-Far Problem . . . . . . . .3
Time Division Multiple Access . . . . . . . . . . . . . . . . . . .4 8. . . . . . . . . . . . .2 Narrowband Systems . . 159 8. . 156 157
8 Multiple Access Techniques 8. . . . . . . . . . . . .1
Multiple Access Techniques for Wireless Communication .jntuworld. . . .7
Space Division Multiple Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166 Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2
Frequency Division Multiple Access .1 Diﬀerent Types of Diversity . . . . . . .2 8. .5 7. 144 Convolutional Codes . . .6
Conclusion
References .4 7. . . . . . . . . . . . . .5 8. . . . . . . . . . . . . . . . 163 CDMA and Self-interference Problem . . . . . . . . . . . . . . . . . . . . . . . . . . 155 . . . . 132 Choice of Algorithms for Adaptive Equalization . 161 TDMA/TDD in DECT . . . . . . . . . . . . 163 8.com
. .4.2. 160
8. 165 Hybrid Spread Spectrum Techniques . . . . . . . . . . . .2 7. . . . . . . . . . . . . . 166
References . . . . .4 Shannon’s Channel Capacity Theorem . . . .4
Channel Coding . . . . . . . . . . . . . . . . 158
8. . . . . . . . 157 8. . . . . . . . . .3. . . . . . 164 CDMA and Near-Far Problem . . . . . . .2. . . . . . . . . . . . . . . . . . . . 134
Diversity . . 143 7.4. . . . . . . . . . . 167 viii
www. . . . 163 Code Division Multiple Access . . . . 132 A Generic Adaptive Equalizer .1 8. . .2 7. 152 Concatenated Codes . . . . . . . . . 162
8. . .1. . . . . .4. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 7. . . . . . . . . . . . . . . .6 8. . . . . . . . 165
8. . .4
Spread Spectrum Multiple Access . .3 7. . .

. . . . . . . . . . . . . Handoﬀ process associated with power levels. .com
List of Figures
1. . . . . . . . . . . . . when the user is going from i-th cell to j-th cell. showing the near and far ﬁelds. . . . . . Frequency reuse technique of a cellular system. . . . . . . . . .7 3. .1 1.10 The micro-cell zone concept. . . 3. . . . . . . . . . . . . .1 3. . . . . . . . . . . . . . . . . . Two-ray reﬂection model. . . . . . .com
. . . . . A seven-cell cluster with 60o sectors. . . . . .2 4. . .jntuworld. . . . . . . . .3 3. . . . . 3. . . . . Handoﬀ scenario at two adjacent cell boundary. .www. . . .9 Handoﬀ process with a rectangular cell inclined at an angle θ. . .4 1. . . Splitting of congested seven-cell clusters. . . . .8 3. . . . . . Basic mobile communication structure. . . . . . . . . . Phasor diagram of electric ﬁelds. . . . .1 4. .11 The buﬀerless J-channel trunked radio system. . .3 Free space propagation model. . . .2 1. . . . . . . . . . . . . 30 31 37 41 43 44 47 49 49 55 59 61 4 6 7 20 24 25 29 2 3
3. . . . . . . .4 (a) Frequency division duplexing and (b) time division duplexing. . . . . . . . . 3. . . . . . . . . . . .jntuworld. . . . . . . . . . . . . . . . . . Basic Cellular Structure. . A cell divided into three 120o sectors. The basic radio transmission techniques: (a) simplex. . .12 Discrete-time Markov chain for the M/M/J/J trunked radio system. . . Footprint of cells showing the overlaps and gaps. . . . . . . . . . .5 2. (b) half duplex and (c) full duplex. . . . . . . First tier of co-channel interfering cells . . . . . . . . . . . . . . . . . . . . . . . . . . Data transmission with Bluetooth. . . . . . . . . . . . . 4. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1. . . .1 3.
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www. . . . . . . . .5 3. .3 The worldwide mobile subscriber chart. .2 3. . . .6 3. . . . . . . . . . . . . . . . . . . . . .

109 Scematic of the line coding techniques. . .www. Diﬀraction through a sharp edge. . . . . . . . . . . . . . . . . . . . . . . .10 Schematic representation of level crossing with a Rayleigh fading envelope at 10 Hz Doppler spread. . . .13 Two-ray Rayleigh fading model. . . . . . . . . . . . Rayleigh probability density function.11 Clarke and Gan’s model for Rayleigh fading generation using quadrature amplitude modulation with (a) RF Doppler ﬁlter. . . . . . . . 5. . . . . . .6 5. . . . . . . . . . . . . . . . . 113 Phase tree of 1101000 CPFSK sequence. . . . .6 4. . . . . . . .7 4. . . . . . . . Nakagami probability density function.2 6. . . . . . . . . . . . . . . . . . . . . . . . . 111 Rectangular Pulse . . . Two ray NLoS multipath. Fresnel zones. . . . . . (b) baseband Doppler ﬁlter. . . . . . . . . . . . Frequency domain channel IR measurement. . . . . . . Relationship among diﬀerent channel functions. . . . . . . . . . . .4 5. .5 4. . . . . . . . . . . . . . .4 6. . . . . . . . . . . . . . . . . . . . . . . . . . .12 Rayleigh fading model to get both the ﬂat and frequency selective channel conditions. . . . . . . . . . . . . . . . . . Huygen’s secondary wavelets.jntuworld.9
Equivalent phasor diagram of Figure 4. . . . . . resulting in Rayleigh fading. . . . . . . . . . . . . . . . . . . . . . . . . . . Knife-edge Diﬀraction Model . . . 118 Spectrum of MSK . . . . . . . . .
61 64 65 66 68 79 83 85 86 87 91 93 93 94
5. . . . .3 5. .jntuworld. . . . 97
5. . . . . . . . .8 6.9 98 99
BPSK signal constellation. . . . . . . . . . . . .4 4.3 6. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3. . . .1 5. . . . . . . . . . . . . . . . . . . . . .2 5. . . . . . 112 Raised Cosine Pulse. . . . . . . . . . . . . . . . .8 5. . . . . Illustration of Doppler eﬀect. . . .6 6. . . . . . . . . .7 6. . . . . . . . . . . . . . . . . . . . . . Direct RF pulsed channel IR measurement. . .5 6. . . . . . . . . . . . . 118
x
www. . . . . . . .5 5. . . . . . . . . . . . . . . . . . . . . . . . . 108 QPSK transmitter. . .8 5. . . . . . . . . . . . . . . . . . . . . . . . . . . .7 5. . . . . . . . . .1 6. . . . . . . . 6. . 108 DQPSK constellation diagram. . . . . 107 QPSK signal constellation. . . . . .com
4. . . . . . . . . . .com
. . Ricean probability density function. . . . . . . . . . . . . . . . . . . 95
5. . A generic transmitted pulsed RF signal. . . . . . . . and. . . . . . . . . . . . . . . . . . . . .

. . . . . . . . . . .6 7. . . . . . . . . . . . . . . . . . . . . . . . . 142 A convolutional encoder with n=2 and k=1. . . . . . . . . . . . . . . . . . .7 7. . . .1 7. . .jntuworld. . . . . . . . . . .2 8. 140 RAKE receiver. . . . . . . . . . . . . . . . . . . .3 The basic concept of FDMA.com
6. . . 159 The basic concept of TDMA. . .13 OFDM Transmitter and Receiver Block Diagram. . . . . . . . . . . . . .www. 154 Trellis diagram of a convolutional encoder. . . . . . . . 119 6. . . . . . . . . . . . . . 164
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www. . . . .1 8.10 Block diagram of a turbo encoder. . .2 7. . . . . . . . . . . . . with M receivers. . . . . . . . .jntuworld. . . . . . . . . . . . 153 State diagram representation of a convolutional encoder. . . . . . . 162 The basic concept of CDMA.8 7. . 130 A generic adaptive equalizer. . .com
. . . . . . . . . . . . 120 6. . 133 Receiver selection diversity. 137 Maximal ratio combining technique. . . .9 A general framework of fading eﬀects and their mitigation techniques. . . . . .12 Spectrum of GMSK scheme. . .5 7. . . . . . 153 Tree diagram representation of a convolutional encoder. . . . . . . . . . . . . 154
7. . . . . . . . . . . . . . . . . . 155 8. . . . . . . . . . . 127 7. .4 7. . . . . . . .10 GMSK generation scheme. . 121 6. . . . . . . . . . . . . . . . . . . . . .3 7. . . . . . . . . . . . .11 A simple GMSK receiver. . . . . . . . . . . . . . . .

. .jntuworld. . . .1 7. . .1 Main WCDMA parameters . . . 16
Finite ﬁeld elements for US-CDPD . . . 152 MA techniques in diﬀerent wireless communication systems . 158
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www. . . . . . . . . . . . . .www. . . . .com
List of Tables
2. . .com
. . . . . . . . . . .1 8. . . .jntuworld. .

2
Evolution of Mobile Radio Communications
The ﬁrst wireline telephone system was introduced in the year 1877.com
Chapter 1
Introductory Concepts
1. In this chapter. With the demand for newer and better mobile radio communication systems during the World War II and the development of Frequency Modulation (FM) technique by Edwin Armstrong. After its discovery.1 Introduction
Communication is one of the integral parts of science that has always been a focus point for exchanging information among parties at locations physically apart.www.com
. during its initial three and a half decades it found very less market penetration owing to high
1
www. mobile communication has become the backbone of the society. the mobile radio communication systems began to witness many new changes. the evolution as well as the fundamental techniques of the mobile communication is discussed.jntuworld. Mobile communication systems as early as 1934 were based on Amplitude Modulation (AM) schemes and only certain public organizations maintained such systems.
1. telephones have replaced the telegrams and letters. However. Similarly. Its main plus point is that it has privileged a common mass of society. the term ‘mobile’ has completely revolutionized the communication by opening up innovative applications that are limited to one’s imagination. All the mobile system technologies have improved the way of living. Mobile telephone was introduced in the year 1946. Today.jntuworld.

cellular telephone system and it was deployed in 1983.S. Initially. costs and numerous technological drawbacks. The number of cellular telephone users grew from 25000 in 1984 to around 3 billion in the year 2007 and the demand rate is increasing day by day.com
.jntuworld. there has been an astronomical growth in the cellular radio and the personal communication systems. Wireless services have since then been experiencing a 50% per year growth rate. Advanced Mobile Phone System (AMPS) was the ﬁrst U. even the growth in the cellular networks was very slow.
2
www.1.www. Moreover. But with the development of the cellular concept in the 1960s at the Bell Laboratories. However. mobile communications began to be a promising ﬁeld of expanse which could serve wider populations.com
Figure 1. mobile communication was restricted to certain oﬃcial users and the cellular concept was never even dreamt of being made commercially available. A schematic of the subscribers is shown in Fig.jntuworld. 1. with the development of newer and better technologies starting from the 1970s and with the mobile users now connected to the Public Switched Telephone Network (PSTN).1: The worldwide mobile subscriber chart.

type of antenna used and the frequency of operation. PACS. Code Division Multiple Access (CDMA). the concept of cellular communication was evolved. Each cell consists of small hexagonal area with a base station located at the center of the cell which communicates with the user.3
Present Day Mobile Communication
Since the time of wireless telegraphy.
3
www. To resolve this problem. Numerous mobile radio standards have been deployed at various places such as AMPS. With the increase in the number of users. The present day cellular communication uses a basic unit called cell.
1. Initially the mobile communication was limited between one pair of users on single channel pair.com
. radio communication has been used extensively.2: Basic mobile communication structure. Our society has been looking for acquiring mobility in communication since then.com
Figure 1.jntuworld.jntuworld. The range of mobility was deﬁned by the transmitter power. accommodating them within the limited available frequency spectrum became a major problem.www. Frequency Division Multiple Access (FDMA) and their hybrids are used. To accommodate multiple users Time Division multiple Access (TDMA).

4
Fundamental Techniques
By deﬁnition. there can be three diﬀerent types of mobile communication.com
Figure 1. Depending on the radio channel. GSM. PHS and IS-95. a Mobile Station (MS) or subscriber unit communicates to a ﬁxed Base Station (BS) which in turn communicates to the desired user at the other end.com
.
1.3: The basic radio transmission techniques: (a) simplex. duplexer and an antenna while the BS consists of transceiver and channel multiplexer along with antennas mounted on the tower.jntuworld. (b) half duplex and (c) full duplex. NTT. each utilizing diﬀerent set of frequencies and allocating diﬀerent number of users and channels. mobile radio terminal means any radio terminal that could be moved during its operation. Figure 1.jntuworld.www. the MS and also 4
www. The BS are also linked to a power source for the transmission of the radio signals for communication and are connected to a ﬁxed backbone network. In general. however.2 shows a basic mobile communication with low power transmitters/receivers at the BS. control circuitry. The MS consists of transceiver.

A walkie-talkie is an example of a half duplex system which uses ‘push to talk’ and ‘release to listen’ type of switches.1
Radio Transmission Techniques
Based on the type of channels being utilized. • Half Duplex System: Half duplex radio systems that use half duplex radio channels allow for non-simultaneous bidirectional communication. The region over which the signal strength lies above such a threshold value is known as the coverage area of a BS. A minimum amount of signal strength is needed in order to be detected by the mobile stations or mobile sets which are the hand-held personal units (portables) or those installed in the vehicles (mobiles). Both the users can communicate to each other simultaneously.jntuworld. This can be done by providing two simultaneous but separate channels to both the users. One frequency channel is transmitted downstream from the BS to the MS (forward channel). One example of such a system is a pager. – Frequency Division Duplexing (FDD): FDD supports two-way radio communication by using two distinct radio channels..3: • Simplex System: Simplex systems utilize simplex channels i. the communication is unidirectional. mobile radio transmission systems may be classiﬁed as the following three categories which is also shown in Fig. However. the user can only transmit or receive information. The radio signals emitted by the BS decay as the signals travel away from it. 1. The MSC is sometimes also called Mobile Telephone Switching Oﬃce (MTSO).
5
www.com
.www.e. • Full Duplex System: Full duplex systems allow two way simultaneous communications. At a time.com
the Mobile Switching Center (MSC). The ﬁrst user can communicate with the second user. the second user cannot communicate with the ﬁrst user.
1. The ﬁxed backbone network is a wired network that links all the base stations and also the landline and other telephone networks through wires. The ﬁrst user can communicate with the second user but the second user can communicate to the ﬁrst user only after the ﬁrst user has ﬁnished his conversation.4.jntuworld. This is possible by one of the two following methods.

a minimum amount of frequency separation must be maintained between the frequency pair.4.www. Cordless telephone systems are full duplex communication systems that use radio to connect to a portable handset to a single dedicated BS.
6
www. A mobile system. is the example of the second category of a full duplex mobile system where many users connect among themselves via a single BS.com
Figure 1. This toggling takes place very rapidly and is imperceptible to the user. TDD operates by toggling transmission directions over a time interval. Because of the pairing of frequencies. simultaneous transmission in both directions is possible. A second frequency is used in the upstream direction and supports transmission from the MS to the BS (reverse channel). in general.4: (a) Frequency division duplexing and (b) time division duplexing. which is then connected to a dedicated telephone line with a speciﬁc telephone number on the Public Switched Telephone Network (PSTN). as shown in Fig. on the other hand.jntuworld. – Time Division Duplexing (TDD): TDD uses a single frequency band to transmit signals in both the downstream and upstream directions. and many MS for a single BS. A full duplex mobile system can further be subdivided into two category: a single MS for a dedicated BS. 1.jntuworld. To mitigate self-interference between upstream and downstream transmissions.com
.

we should ﬁrst look into the basics of cellular concept and main operational channels involved in making a call.5: Basic Cellular Structure. called cells.jntuworld.1
Cellular Concept
Cellular telephone systems must accommodate a large number of users over a large geographic area with limited frequency spectrum.5.com
Figure 1. i. For a large geographic coverage area. with limited number of channels.. Each cell has a low power transmitter with a coverage area equal to the area of the
7
www.e. These are given below. a high powered transmitter therefore has to be used. Each cell uses a certain number of the available channels and a group of adjacent cells together use all the available channels. then suﬃcient amount of power may not be present at a huge distance from the BS. But a high power radio transmitter causes harm to environment.
1.com
. Mobile communication thus calls for replacing the high power transmitters by low power transmitters by dividing the coverage area into small segments. If a single transmitter/ receiver is used with only a single base station.www. This cluster can repeat itself and hence the same set of channels can be used again and again.jntuworld.
1.5
How a Mobile Call is Actually Made?
In order to know how a mobile call is made. Such a group is called a cluster.

Each mobile has a mobile identiﬁcation number (MIN).com
cell.e. the mobile again searches all the FCCs for the one with the highest signal strength. He 8
www. The mobile then monitors this particular FCC. then the control channels for that particular location will be diﬀerent and hence the mobile will not work.jntuworld. These channels transmit and receive call initiation and service request messages. Control channels are usually monitored by mobiles. So all mobiles in that country or continent will search among the same set of control channels. When a user wants to make a call. However. • Reverse Control Channel (RCC): This is used for the call control purpose from the MS to the BS.jntuworld. The FCC is used for control signaling purpose from the BS to MS. they are used for setting up calls and to divert the call to unused voice channels. the control channels will be the same. These are: • Forward Voice Channel (FVC): This channel is used for the voice transmission from the BS to the MS..5.www. he sends a call request to the MSC on the reverse control channel. i. Hence these are also called setup channels. when a mobile moves to a diﬀerent country or continent.
1.com
.2
Operational Channels
In each cell. then it searches all the FCCs to determine the one with the highest signal strength. However.3
Making a Call
When a mobile is idle. • Reverse Voice Channel (RVC): This is used for the voice transmission from the MS to the BS. there are four types of channels that take active part during a mobile call. i.e. This technique of substituting a single high powered transmitter by several low powered transmitters to support many users is the backbone of the cellular concept. when the signal strength falls below a particular threshold that is insuﬃcient for a call to take place.5. it is not experiencing the process of a call.
1.. • Forward Control Channel (FCC): Control channels are generally used for controlling the activity of the call. For a particular country or continent.

e. So the call has to be transferred to the other base station.com
. In such cases.e. then the signal strength of the initial base station may not be suﬃcient to continue the call in progress. 1 mW power developed over a 100 Ω load is equivalently called 0 dBm power. This means. Thus. (justiﬁcation is left on the readers)
9
www. The MSC then sends this MIN to all the base stations. Solution: The ‘pager’ would have the longest and the ‘mobile phone’ would have the shortest battery life. which device would have the (i) shortest.. Express this power in terms of dBm. 1: Suppose a mobile unit transmits 10 W power at a certain place. that mobile sends an acknowledgment to the BS. 10 log10 (1W ) = 0dB. Ex..jntuworld.
1W = 103 mW = 30dBm = 0dB. Hence. This is called handoﬀ.jntuworld.com
also sends the MIN of the person to whom the call has to be made. 1 W is equivalent to 0 dB. a cordless phone and a mobile phone. 10W = 10 log10 (10W ) = 10dB = 40dBm. i.www. The MSC then instructs the base station to access speciﬁc unused voice channel pair. When a mobile moves from the coverage area of one base station to the coverage area of another base station i. in order to maintain the call. (ii) longest battery life? Justify. the MSC transfers the call to one of the unused voice channels of the new base station or it transfers the control of the current voice channels to the new base station. the MSC adjusts the transmitted power of the mobile which is usually expressed in dB or dBm. Solution: Usually.
Ex. xdB = (x + 30)dBm. If the MIN matches with a particular MS. The base station transmits this MIN and all the mobiles within the coverage area of that base station receive the MIN and match it with their own. from one cell to another cell. The base station then sends a message to the mobile to move to the particular channels and it also sends a signal to the mobile for ringing. 2: Among a pager. The BS then informs the MSC that the mobile is within its coverage area. In order to maintain the quality of the call. and.

Rappaport.. Proakis.www. S. NY: McGraw Hill. Digital Communications. 2002.
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www. 1995.7
References
1. Further enhancements in modulation schemes will soon increase the Internet access rates on the mobile from current 1. 3. 2000. Feher. J. G. Singapore: Pearson Education.6
Future Trends
Tremendous changes are occurring in the area of mobile radio communications. K. Wireless Communications: Principles and Practice. 2nd ed. We would review certain major features as well as standards of the mobile communication till the present day technology in the next chapter.jntuworld.com
1. so much so that the mobile phone of yesterday is rapidly turning into a sophisticated mobile device capable of more applications than PCs were capable of only a few years ago. NJ: Prentice Hall. T.
1. Rapid development of the Internet with its new services and applications has created fresh challenges for the further development of mobile communication systems.com
. Inc. 2. Bluetooth is rapidly becoming a common feature in mobiles for local connections. The mobile communication has provided global connectivity to the people at a lower cost due to advances in the technology and also because of the growing competition among the service providers. 4th ed. Upper Saddle River.8 Mbps to greater than 10 Mbps. Wireless Digital Communications: Modulation and Spread Spectrum Applications.jntuworld.

cellular telephone system. the spread of mobile communication was very fast in the 1990s when the government throughout the world provided radio spectrum licenses for Personal Communication Service (PCS) in 1. However.www. with the development of newer and better technologies starting from the 1970s and with the mobile users now connected to the PSTN.com
. Moreover. However.jntuworld. These are as
11
www. mobile communication was restricted to certain oﬃcial users and the cellular concept was never even dreamt of being made commercially available. there has been a remarkable growth in the cellular radio.2 GHz frequency band.
2.jntuworld.8 . The 2G systems introduced three popular TDMA standards and one popular CDMA standard in the market. even the growth in the cellular networks was very slow. It was the ﬁrst U.2
2G: Second Generation Networks
Digital modulation formats were introduced in this generation with the main technology as TDMA/FDD and CDMA/FDD. The main technology of this ﬁrst generation mobile system was FDMA/FDD and analog FM.
2.S.1
1G: First Generation Networks
The ﬁrst mobile phone system in the market was AMPS. deployed in Chicago in 1983.com
Chapter 2
Modern Wireless Communication Systems
At the initial phase.

com
. the new data centric standards were developed to be overlaid on 2G standards and this is known as 2. The main advantage of this standard was its low transmission bit rate which led to its better spectrum utilization. there were 3 full-rate TDMA users over each 30 KHz channel. enhanced mobile station identities etc. Certain services that have been standardized as a part of IS-95 standard are: short messaging service.
2. encryption of speech. 8 full-rate or 16 half-rate TDMA channels per carrier.2.
2.jntuworld. was aimed at designing a uniform pan-European mobile system.25 MHz channels. also popularly known as CDMAOne. The need of this system was mainly to increase the capacity over the earlier analog (AMPS) system. over-the-air activation (meaning the mobile can be activated by the service provider without any third party intervention).3
2. The initial GSM had 200 KHz radio channels.5G Mobile Networks
In an eﬀort to retroﬁt the 2G standards for compatibility with increased throughput rates to support modern Internet application.2.www. introduced by Groupe Special Mobile. the main upgradation techniques are: • supporting higher data rate transmission for web browsing
12
www. It was the ﬁrst fully digital system utilizing the 900 MHz frequency band. (c) Paciﬁc Digital Cellular (PDC): This standard was developed as the counterpart of NADC in Japan.2
CDMA/FDD Standard
Interim Standard 95 (IS-95): The IS-95 standard.jntuworld.1
TDMA/FDD Standards
(a) Global System for Mobile (GSM): The GSM standard. slotted paging. uses 64 orthogonally coded users and codewords are transmitted simultaneously on each of 1.com
follows:
2. Here.2.5G standard. low speed data services and support for SMS for which it gained quick popularity. In this system. (b) Interim Standard 136 (IS-136): It was popularly known as North American Digital Cellular (NADC) system.

all in a mobile environment. Additional features also include HSPA data transmission capabilities able to deliver speeds up to 14.to provide the framework for the deﬁnition of the 3G mobile systems.Several radio access technologies have been accepted by ITU as part of the IMT-2000 framework. Enhanced Data rates for GSM Evolution (EDGE) etc. It is based on the International Telecommunication Union (ITU) family of standards under the International Mobile Telecommunications-2000 (IMT-2000). together with the main industry and standardization bodies worldwide.
2. High Speed Circuit Switched Dada (HSCSD). General Packet Radio Service (GPRS). video calls.com
• supporting e-mail traﬃc • enabling location-based mobile service 2.4Mbit/s on the down link and 5. ubiquitous wireless communication standard for all countries. targets to implement a global frequency band that would support a single.jntuworld. 3G networks enable network operators to oﬀer users a wider range of more advanced services while achieving greater network capacity through improved spectral eﬃciency. 3G networks are wide area cellular telephone networks which evolved to incorporate high-speed internet access and video telephony. which.8Mbit/s on the uplink.www. a few of which are: Wireless Application Protocol (WAP). ITU launched IMT-2000 program. which can be summarized as follows: • high data rates: 144 kbps in all environments and 2 Mbps in low-mobility and indoor environments • symmetrical and asymmetrical data transmission 13
www. Services include wide-area wireless voice telephony.5G networks also brought into the market some popular application.5G. and broadband wireless data. superseding 2.3
3G: Third Generation Networks
3G is the third generation of mobile phone standards and technology. IMT-2000 deﬁnes a set of technical requirements for the realization of such targets.com
.jntuworld.

ITU ﬁnally approved a family of ﬁve 3G standards. This system is called Universal Mobile Telecommunications System (UMTS).com
.1
3G Standards and Access Technologies
As mentioned before. An organization called 3rd Generation Partnership Project (3GPP) has continued that work by deﬁning a mobile system that fulﬁlls the IMT-2000 standard. WCDMA will have to use another part of the radio spectrum. based on either CDMA or TDMA technology.jntuworld.
2.3. Japan.3.2
3G W-CDMA (UMTS)
WCDMA is based on DS-CDMA (direct sequencecode division multiple access) technology in which user-information bits are spread over a wide bandwidth (much larger than the information signal bandwidth) by multiplying the user data with 14
www. After trying to establish a single 3G standard.jntuworld.www.com
• circuit-switched and packet-switched-based services • speech quality comparable to wire-line quality • improved spectral eﬃciency • several simultaneous services to end users for multimedia services • seamless incorporation of second-generation cellular systems • global roaming • open architecture for the rapid introduction of new services and technology. which is WCDMA operating at 2. UMTS and WCDMA are often used as synonyms. which are part of the 3G framework known as IMT-2000: • W-CDMA • CDMA2000 • TD-SCDMA Europe.1 GHz.
2. and Asia have agreed upon a 3G standard called the Universal Mobile Telecommunications System (UMTS). there are several diﬀerent radio access technologies deﬁned within ITU. In the USA and other parts of America.

which. The wide carrier bandwidth of WCDMA allows supporting high user-data rates and also has certain performance beneﬁts. However.jntuworld. In the FDD mode. The processing gain term refers to the relationship between the signal bandwidth and the information bandwidth.enhancement methods to be used. such as transmit diversity or advanced CDMA receiver concepts. The support for handovers (HO) between GSM and WCDMA is part of the ﬁrst standard version.Table summaries the main WCDMA parameters. WCDMA supports two basic modes of operation: FDD and TDD.
15
www. all users are active at the same time on the same frequency and are separated from each other with the use of user speciﬁc spreading codes.4 and 5 MHz. depending on spectrum arrangement and the interference situation.2288 Mcps. which has a chip rate of 1.com
the spreading code. respectively. Thus. WCDMA uses coherent detection based on the pilot symbols and/or common pilot. separate 5-MHz carrier frequencies with duplex spacing are used for the uplink and downlink.jntuworld. whereas in TDD only one 5-MHz carrier is time shared between the uplink and the downlink. This means that all multi-mode WCDMA/GSM terminals will support measurements from the one system while camped on the other one. in the WCDMA system deployment is used together with the 5-MHz carrier spacing. In WCDMA each user is allocated frames of 10 ms duration. the data rate among the users can change from frame to frame. such as increased multipath diversity.com
. This fast radio capacity allocation (or the limits for variation in the uplink) is controlled and coordinated by the radio resource management (RRM) functions in the network to achieve optimum throughput for packet data services and to ensure suﬃcient quality of service (QoS) for circuit-switched users. WCDMA allows many performance. the name wideband is derived to diﬀerentiate it from the 2G CDMA (IS-95). In a CDMA system. The actual carrier spacing to be used by the operator may vary on a 200-kHz grid between approximately 4. This allows networks using both WCDMA and GSM to balance the load between the networks and base the HO on actual measurements from the terminals for diﬀerent radio conditions in addition to other criteria available.84 Mcps. during which the user-data rate is kept constant. The chip (symbol rate) rate of the spreading sequence is 3.www.

FoMA is the short name for Freedom of Mobile Multimedia Access.3
3G CDMA2000
Code division multiple access 2000 is the natural evolution of IS-95 (cdmaOne). assures backward compatibility with the second generation GSM.84 Mcps 10 ms Multiple services with diﬀerent quality of service requirements multiplexed on one connection Multi-rate concept Detection Variable spreading factor and multicode Coherent using pilot symbols or common pilot Multi-user detection. It includes additional functionality that increases its spectral eﬃciency and data rate capability.com
. Elsewhere.1: Main WCDMA parameters Multiple access method Duplexing method DS-CDMA Frequency division duplex/time division duplex Base station synchronisation Chip rate Frame length Service multiplexing Asynchronous operation 3.
2. was launched by NTT DoCoMo in Japan in 2001. CDMA permits many simultaneous transmitters on the same frequency channel.5G TDMA technologies. UMTS or W-CDMA. IS-136 and PDC TDMA technologies. FoMA.3.jntuworld. smart antennas Supported by the standard. as well as all 2. optional in the implementation
The world’s ﬁrst commercial W-CDMA service. with additional capacity and bandwidth provided by a new CDMA air interface.jntuworld. W-CDMA deployments have been exclusively UMTS based. The network structure and bit level packaging of GSM data is retained by W-CDMA.www. is the brand name for the 3G services being oﬀered by Japanese mobile phone operator NTT DoCoMo.(code division multiple access) is a mobile digital radio technology where channels are deﬁned with codes (PN sequences). Since more phones can be served by
16
www.com
Table 2.

• It has higher rates per carrier (up to 4.www.com
fewer cell sites.com
. It 17
www.jntuworld. CDMA2000 1xRTT: RTT stands for Radio Transmission Technology and the designation ”1x”.7 Mbit/s.4Mbps of instantaneous high-speed packet throughput per user on a CDMA channel.depending on the number of user. Typical deployments are expected to include 3 carriers for a peak rate of 14. This standard is being developed by Telecommunications Industry Association (TIA) of US and is is standardized by 3GPP2. a brief discussion about all these standards is given. CDMA2000 3x is (also known as EV-DO Rev B) is a multi-carrier evolution.Higher rates are possible by bundling multiple channels together. • Supports up to twice as many voice users a the 2G CDMA standard • Provides the subscriber unit with upto two times the standby time for longer lasting battery life. although the user data rates are much lower and highly dependent on other factors. • The cdma2000 1xEV-DO supports greater than 2.jntuworld. CDMA-based standards have a signiﬁcant economic advantage over TDMA. indicates the same RF bandwidth as IS-95.The main features of CDMA2000 1X are as follows: • Supports an instantaneous data rate upto 307kpbs for a user in packet mode and a typical throughput rates of 144kbps per user.9 Mbit/s on the downlink per carrier). the velociy of user and the propagating conditions. CDMA2000 EV: This is an evolutionary advancement of CDMA with the following characteristics: • Provides CDMA carriers with the option of installing radio channels with data only (CDMA2000 EV-DO) and with data and voice (CDMA2000 EV-DV) . These are the approved radio interfaces for the ITU’s IMT-2000 standard. • CDMA2000 EV-DV can oﬀer data rates upto 144kbps with about twice as many voice channels as IS-95B. In the following.or FDMA-based standards. meaning ”1 times Radio Transmission Technology”.CDMA2000 1xEV and CDMA2000 EV-DV. The main CDMA2000 standards are: CDMA2000 1xRTT.

This proposal was adopted by ITU as one of the 3G options in late 1999. which is helpful to the application of beamforming techniques.com
.www. or TD-SCDMA. especially to users at the edge of the cell. video telephony. Since it does not require paired spectrum for downlink and uplink. being pursued in the People’s Republic of China by the Chinese Academy of Telecommunications Technology (CATT). spectrum allocation ﬂexibility is also increased. and broadband multimedia content delivery. the system can more easily accommodate asymmetric traﬃc with diﬀerent data rate requirements on downlink and uplink than FDD schemes. TD-SCDMA is based on spread spectrum technology.jntuworld. Also. enhancing the experience for latency-sensitive services such as gaming.com
enhances the user experience and enables new services such as high deﬁnition video streaming. This reduces the number of users in each timeslot.4
3G TD-SCDMA
Time Division-Synchronous Code Division Multiple Access. web browsing. • It has eﬃcient support for services that have asymmetric download and upload requirements (i. • It provides increased talk-time and standby time. By dynamically adjusting the number of timeslots used for downlink and uplink.3.jntuworld. diﬀerent data rates required in each direction) such as ﬁle transfers. and the base station can deduce the downlink channel information from uplink channel estimates. • Uses statistical multiplexing across channels to further reduce latency. which reduces the implementation 18
www. is a 3G mobile telecommunications standard.e. TD-SCDMA uses TDD. in contrast to the FDD scheme used by W-CDMA. • The interference from the adjacent sectors is reduced by hybrid frequency reuse and improves the rates that can be oﬀered.
2. remote console sessions and web browsing. using the same carrier frequency for uplink and downlink means that the channel condition is the same on both directions. TD-SCDMA also uses TDMA in addition to the CDMA used in WCDMA.

www. An advantage of WLL technology is that once the wireless equipment is paid for.
2. This reduces the interference between users of the same timeslot using diﬀerent codes by improving the orthogonality between the codes.jntuworld.
2.2
Bluetooth
• Facilitates ad-hoc data transmission over short distances from ﬁxed and mobile devices as shown in Figure 2. Below. some of these applications are given. The ”S” in TD-SCDMA stands for ”synchronous”. which provides broadband telecommunication access in the local exchange. therefore increasing system capacity.1 • Uses a radio technology called frequency hopping spread spectrum.4.
19
www. It chops up the data being sent and transmits chunks of it on up to 79 diﬀerent frequencies.4. which means that uplink signals are synchronized at the base station receiver. The local loop can be thought of as the ”last mile” of the telecommunication network that resides between the central oﬃce (CO) and the individual homes and business in close proximity to the CO.com
.4
Wireless Transmission Protocols
There are several transmission protocols in wireless manner to achieve diﬀerent application oriented tasks.
2. at the cost of some hardware complexity in achieving uplink synchronization.jntuworld. there are no additional costs for transport between the CO and the customer premises equipment.1
Wireless Local Loop (WLL) and LMDS
Microwave wireless links can be used to create a wireless local loop. achieved by continuous timing adjustments.com
complexity of multiuser detection and beamforming schemes. Many new services have been proposed and this includes the concept of Local Multipoint Distribution Service (LMDS). mobility (because of lower power control frequency) and complicates radio resource management algorithms. but the non-continuous transmission also reduces coverage (because of the higher peak power needed).

Range also varies with frequency band. In its basic mode. the modulation is Gaussian frequency shift keying (GFSK).825GHz) • Uses 11Mcps DS-SS spreading and 2Mbps user data rates (will fallback to 1Mbps in noisy conditions) • IEEE 802.
20
www.1: Data transmission with Bluetooth. It can achieve a gross data rate of 1 Mb/s • Primarily designed for low power consumption. 10 meters.www.11b or 802. • IEEE 802.11a stndard provides upto 54Mbps throughput in the 5GHz band. A typical Wi-Fi home router using 802.jntuworld.4.11g with a stock antenna might have a range of 32 m (120 ft) indoors and 95 m (300 ft) outdoors. 100 meters) based on low-cost transceiver microchips in each device
2.3
Wireless Local Area Networks (W-LAN)
• IEEE 802.11g uses Complementary Code Keying Orthogonal Frequency Division Multiplexing (CCK-OFDM) standards in both 2. The DS-SS IEEE 802.11 WLAN uses ISM band (5. Wi-Fi networks have limited range.jntuworld.4GHz and 5GHz bands.com
. with a short range (powerclass-dependent: 1 meter.11b has been called Wi-Fi.com
Figure 2.275-5.

11 wireless local area network standard is limited in most cases to only 100 .6
Wibree
• Wibree is a digital radio technology (intended to become an open standard of wireless communications) designed for ultra low power consumption (button cell batteries) within a short range (10 meters / 30 ft) based around low-cost transceiver microchips in each device.4 GHz in most worldwide. such as wireless headphones connecting with cell phones via short-range radio. In contrast. 21
www. 868 MHz in Europe.300 feet (30 . scientiﬁc and medical (ISM) radio bands.4.5
Zigbee
• ZigBee is the speciﬁcation for a suite of high level communication protocols using small.jntuworld. and secure networking.10 miles (5 .15.4-2006 standard for wireless personal area networks (WPANs).16 standard (also called WirelessMAN) • WiMAX can provide broadband wireless access (BWA) up to 30 miles (50 km) for ﬁxed stations. • ZigBee operates in the industrial. 915 MHz in countries such as USA and Australia.4
WiMax
• Provides upto 70 Mb/sec symmetric broadband speed without the need for cables.15 km) for mobile stations.
2. The technology is based on the IEEE 802.16 speciﬁcation applies across a wide range of the RF spectrum. and WiMAX could function on any frequency below 66 GHz (higher frequencies would decrease the range of a Base Station to a few hundred meters in an urban environment).4. the WiFi/802. • This technology is intended to be simpler and cheaper.4.com
.100m) • The 802.com
2. and 2.
2.www. low-power digital radios based on the IEEE 802.jntuworld. long battery life. ZigBee is targeted at radio-frequency (RF) applications that require a low data rate. and 3 .

com
. 2008. S. 3. W. Singapore: Pearson Education.6
References
1.jntuworld. C. 2002. 2004. 2nd ed. there are certain objectives that are projected for 4G.. Mobile Communications Engineering. 2. Pandya. T.com
• Wibree is known as Bluetooth with low energy technology. 4th ed.
22
www. • It operates in 2.jntuworld.5
Conclusion: Beyond 3G Networks
Beyond 3G networks.There is no formal deﬁnition for 4G . 2nd ed.4 GHz ISM band with physical layer bit rate of 1 Mbps. A 4G system will be able to provide a comprehensive IP solution where voice.
2. It will be capable of providing between 100 Mbit/s and 1 Gbit/s speeds both indoors and outdoors. It would also support systems like multicarrier communication. MIMO and UWB.
2. R. represent the next complete evolution in wireless communications. Wireless Communications: Principles and Practice. data and streamed multimedia can be given to users at higher data rates than previous generations. Lee. Rappaport. New Delhi: PHI. Mobile and Personal Communication Systems and Services. or 4G (Fourth Generation).www. however. New Delhi: Tata McGraw-Hill. Inc. with premium quality and high security.

jntuworld. is called the foot print of a cell (in reality.com
Chapter 3
The Cellular Engineering Fundamentals
3. A minimum amount of signal strength (let us say. we have seen that the technique of substituting a single high power transmitter by several low power transmitters to support many users is the backbone of the cellular concept. Such a circle. Starting from the basic notion of a cell. quality of service.
3.www. the following four parameters are most important while considering the cellular issues: system capacity.jntuworld. it is amorphous). The region over which the signal strength lies above this threshold value x dB is known as the coverage area of a BS and it must be a circular region.1 Introduction
In Chapter 1.2
What is a Cell?
The power of the radio signals transmitted by the BS decay as the signals travel away from it. x dB) is needed in order to be detected by the MS or mobile sets which may the hand-held personal units or those installed in the vehicles. In practice. we would deal with these parameters in the context of cellular engineering in this chapter. spectrum eﬃciency and power management. considering the BS to be isotropic radiator. It might so happen that either there may be an overlap between any two such side by side circles or there might be a gap between the
23
www. which gives this actual radio coverage.com
.

Such a circular geometry. frequency planning. cannot serve as a regular shape to describe cells.e. which can cover the entire area without any overlap and gaps.1.. or.3
Frequency Reuse
Frequency reuse. a cell must be designed such that it is most reliable too.com
Figure 3. For any distance between the center and the farthest point in the cell from it.jntuworld.jntuworld. is a technique of reusing frequencies and channels within a communication system to improve capacity and spectral eﬃciency.www. namely. coverage areas of two adjacent circles. Hence regular hexagonal geometry is used as the cells in mobile communication. comes from the fact that the same radio frequency can be reused in a diﬀerent area for a completely diﬀerent transmission.1: Footprint of cells showing the overlaps and gaps. square and regular hexagon. it supports even the weakest mobile with occurs at the edges of the cell. Frequency reuse is one of the fundamental concepts on which commercial wireless systems are based that involve the partitioning of an RF radiating area into cells. therefore. We need a regular shape for cellular design over a territory which can be served by 3 regular polygons. Along with its regularity. Frequency reuse in mobile cellular systems means that frequencies allocated to 24
www. equilateral triangle. a regular hexagon covers the maximum area.
3. i. This is shown in Figure 3.com
. compared with a network with a single transmitter. The increased capacity in a commercial wireless network.

1)
Hence the capacity gain achieved is directly proportional to the number of times a cluster is repeated.2: Frequency reuse technique of a cellular system. The reuse of frequencies enables a cellular system to handle a huge number of calls with a limited number of channels.2) (3. in another cluster. the same frequencies can be reused in other cells not far away without interference.com
Figure 3. Consider a cellular system with S duplex channels available for use and let N be the number of cells in a cluster. as shown in (3. The repeating regular pattern of cells is called cluster. then T ∝M and.jntuworld. each covered by one base station. If each cell is allotted K duplex channels with all being allotted unique and disjoint channel groups we have S = KN under normal circumstances. if K and N remain constant.1). Since each cell is designed to use radio frequencies only within its boundaries. Clearly. or. the total number of duplex channels. if the cluster are repeated M times within the total area.jntuworld. The closest distance between the co-channel cells (in diﬀerent clusters) is determined by the choice of the cluster size and the layout of the cell cluster. small N 25
www. for a ﬁxed cell size.www. if T and K remain constant. as well as.com
.2 shows a frequency planning with cluster size of 7. then N∝ 1 . M (3. Figure 3. showing the co-channels cells in diﬀerent clusters by the same letter. Such cells are called ‘co-channel’ cells. the total number of users in the system would be T = M S = KM N . the service are reused in a regular pattern of cells. Now.

4)
which turns out to be Dn = Multiplying the actual distance √ i2 + ij + j 2 = √ N.6)
√ √ D = Dn 3R = 3N R. (3. 3
26
www.
(3. where R is the radius of any cell. The value of N is determined by calculating the amount of interference that can be tolerated for a suﬃcient quality communication. Solution: For hexagonal cells. The normalized co-channel cell distance Dn can be calculated by traveling ’i’ cells in one direction and then traveling ’j’ cells in anticlockwise 120o of the primary direction. it can be shown that the distance between two adjacent √ cell centers = 3R. where i and j are integer numbers.5)
3R between two adjacent cells with it. and hence compute the total number of cells in this large hexagon. Ex. However for small N.3)
(3.com
decreases the size of the cluster with in turn results in the increase of the number of clusters (3.www.
Ex. j ≥ 0.com
.jntuworld. we get (3. 1: Find the relationship between any two nearest co-channel cell distance D and the cluster size N. the cluster size N cannot take on any value and is given only by the following equation N = i2 + ij + j 2 . the answer must be N + 6( N ) = 3N . Hint: In general. However. the surface area of a large hexagon with radius D. co-channel cells are located much closer and hence more interference.jntuworld. Hence the smallest N having interference below the tolerated limit is used.2) and hence the capacity. Using law of vector addition.
2 Dn = j 2 cos2 (30o ) + (i + j sin(30o ))2
i ≥ 0. this large hexagon with radius D encompasses the center cluster of N cells and one-third of the cells associated with six other peripheral large hexagons. 2: Find out the surface area of a regular hexagon with radius R. Thus.

jntuworld. i. then the call is blocked and subscriber has to wait.
3.4. But this type of assignment strategy results in heavy load on switching center at heavy traﬃc condition. With increased capacity and low interference being the prime objectives. as discussed below. Later there was another approach in which the channels were borrowed from adjacent cell if all of its own designated channels were occupied.2
Dynamic Channel Assignment (DCA)
In dynamic channel assignment strategy channels are temporarily assigned for use in cells for the duration of the call. good quality of service. Suppose if all the channels are occupied. After the call is over the channel is returned and kept in a central pool.1
Fixed Channel Assignment (FCA)
In ﬁxed channel assignment strategy each cell is allocated a ﬁxed number of voice channels. In such cases the MSC supervises the borrowing process and ensures that none of the calls in progress are interrupted. This is simplest of the channel assignment strategies as it requires very simple circuitry but provides worst channel utilization.
3.4
Channel Assignment Strategies
With the rapid increase in number of mobile users.com
. Each time a call attempt is made from a cell the corresponding BS requests a channel from MSC. This was named as borrowing strategy.com
3.www. Any communication within the cell can only be made with the designated unused channels of that particular cell.jntuworld.. 27
www. The MSC then allocates a channel to the requesting the BS. a frequency reuse scheme was helpful in achieving this objectives. When compared to the FCA.4. DCA has reduced the likelihood of blocking and even increased the trunking capacity of the network as all of the channels are available to all cells. To avoid co-channel interference any channel that in use in one cell can only be reassigned simultaneously to another cell in the system if the distance between the two cells is larger than minimum reuse distance. A variety of channel assignment strategies have been followed to aid these objectives.e. the mobile service providers had to follow strategies which ensure the eﬀective utilization of the limited radio spectrum. Channel assignment strategies are classiﬁed into two types: ﬁxed and dynamic.

Once a signal
28
www. We know that for this system.. Here.com
. to keep the communication between the user pair. On the other hand. a total of 20 control channels and a total of 640 voice channels are kept. Therefore. total channels per cell = 660/4 = 165. the MSC automatically transfers the call to a new FDD channel without disturbing the conversation. This is one solution. while the conversation is still in progress. i.jntuworld. 5 cells can use 92 voice channels and the rest two can use 90 voice channels each.www. (b) For N = 7. the user channel has to be shifted from one BS to the other without interrupting the call. A schematic diagram of handoﬀ is given in Figure 3. Processing of handoﬀ is an important task in any cellular system. Among these. 5 x 92 + 2 x 90 = 640 voice channels.3.jntuworld.5
Handoﬀ Process
When a user moves from one cell to the other. Handoﬀs must be performed successfully and be imperceptible to the users. Solution: One duplex channel = 2 x 25 = 50 kHz of spectrum. total channels per cell are 660/7 ≈ 94. This process is called as handoﬀ. Among these channels. Assume 1 MHz of spectrum is allocated to control channels.
3. (ii) 7 cell. voice channels are 160 and control channels are 5 in number. 6 cells can use 3 control channels and the rest two can use 2 control channels each. there might exist other solutions too. Thus the total solution for this case is: 6 x 3 + 1 x 2 = 20 control channels. (a) For N = 4. we have to go for a more exact solution. Give a distribution of voice and control channels.com
Ex.3) by two integers i and j.e. (c) The option N = 8 is not a valid option since it cannot satisfy equation (3. 1 MHz / 50 kHz = 20 channels are kept as control channels. and. when a MS moves into another cell. Compute the number of channels available per cell if the system uses (i) 4 cell. Hence the total available duplex channels are = 33 MHz / 50 kHz = 660 in number. 3: A total of 33 MHz bandwidth is allocated to a FDD cellular system with two 25 KHz simplex channels to provide full duplex voice and control channels. and (iii) 8 cell reuse technique.

If ∆ is too small.jntuworld.com
Figure 3.com
.www.jntuworld. If ∆ is too high o the other hand. the handoﬀ threshold or the power margin varies from cell to cell. level is set as the minimum acceptable for good voice quality (Prmin ). Especially when the user is on the boundary of 29
www. then a slightly stronger level is chosen as the threshold (PrH )at which handoﬀ has to be made. called power margin.1
Factors Inﬂuencing Handoﬀs
The following factors inﬂuence the entire handoﬀ process: (a) Transmitted power: as we know that the transmission power is diﬀerent for different cells.4. deﬁned as ∆ = PrH − Prmin (3.5. Therefore ∆ should be judiciously chosen to ensure imperceptible handoﬀs and to meet other objectives. (b) Received power: the received power mostly depends on the Line of Sight (LoS) path between the user and the BS. then MSC has to be burdened with unnecessary handoﬀs.7)
is quite an important parameter during the handoﬀ process since this margin ∆ can neither be too large nor too small. as shown in Figure 3. then there may not be enough time to complete the handoﬀ and the call might be lost even if the user crosses the cell boundary. A parameter.3: Handoﬀ scenario at two adjacent cell boundary. This is because MS may not intend to enter the other cell.
3.

also ﬁxes the handoﬀ strategy of a cell.5. the LoS path plays a critical role in handoﬀs and therefore the power margin ∆ depends on the minimum received power value from cell to cell. (c) Area and shape of the cell: Apart from the power levels.4: Handoﬀ process associated with power levels.8)
30
www.jntuworld. when the user is going from i-th cell to j-th cell. the two cells. Then the handoﬀ rate λH can be written as λH = (N1 cosθ + N2 sinθ)R1 + (N1 sinθ + N2 cosθ)R2 . Assume N1 users are having handoﬀ in horizontal direction and N2 in vertical direction per unit length. let us consider a rectangular cell with sides R1 and R2 inclined at an angle θ with horizon. The number of crossings along R1 side is : (N1 cosθ + N2 sinθ)R1 and the number of crossings along R2 side is : (N1 sinθ + N2 cosθ)R2 .www. the cell structure also a plays an important role in the handoﬀ process.jntuworld. (d) Mobility of users: The number of mobile users entering or going out of a particular cell. as shown in the Figure 3.com
. To illustrate the reasons (c) and (d). (3.com
Figure 3.

given the ﬁxed area A = R1 R2 .jntuworld. i.5.9) or (3. Replacing H R1 by
A R2
and equating
dλH dR1
to zero. for R2 . Now.10)
2 2 From the above equations.5: Handoﬀ process with a rectangular cell inclined at an angle θ.
3. and.9)
Similarly. we get
2 R1 = A(
N1 sinθ + N2 cosθ ). N1 sinθ + N2 cosθ
(3.
This has two implications: (i) that handoﬀ is
minimized if rectangular cell is aligned with X-Y axis. N1 cosθ + N2 sinθ
(3.com
Figure 3. Putting the value of θ H
in (3. The above analysis has been carried out for a simple square cell and it changes in more complicated way when we consider a hexagonal cell.www.jntuworld. we have
R1 R2
=
N1 N2 .com
. we get
2 R2 = A(
N1 cosθ + N2 sinθ ). we have λH = 2 A(N1 N2 + (N1 + N2 )cosθsinθ) which √ means it it minimized at θ = 0o . The BS monitors the signal 31
www. the signal strength measurements were made by the BS and in turn supervised by the MSC. θ = 0o . (ii) that the number of users crossing the cell boundary is inversely proportional to the dimension of the other side of the cell.e. we need to ﬁnd λmin for a given θ. The handoﬀs in this generation can be termed as Network Controlled Hand-Oﬀ (NCHO)..2
Handoﬀs In Diﬀerent Generations
In 1G analog cellular systems.10). Hence λmin = 2 AN1 N2 .

The value of ∆ was in the order of 0-5 dB.com
. The Quality Of Service (QoS) has improved a lot although the complexity of the circuitry has further increased which is inevitable. In this generation. even this amount of delay could create a communication pause. Accordingly the two BS communicate and channel transfer occurs. All these types of handoﬀs are usually termed as hard handoﬀ as there is a shift in the channels involved. As compared to 1G. which started using the digital technology. This requires the value of ∆ to be in the order of 6dB to 12dB. the MSC was relieved from the entire operation. delay during handoﬀ is only 100 ms and the value of ∆ is around 20 dBm.jntuworld. However. the circuit complexity was increased here whereas the delay in handoﬀ was reduced to 1-5 s. There is also another kind of handoﬀ. Hence this can be termed as Mobile Controlled Hand-Oﬀ (MCHO).com
strengths of voice channels to determine the relative positions of the subscriber. handoﬀ decisions were mobile assisted and therefore it is called Mobile Assisted Hand-Oﬀ (MAHO).
32
www.jntuworld. These types of handoﬀs are called as soft handoﬀ as there is no change in the channel. The MSC evaluates the signal strengths received from diﬀerent BS for a single user and then shifts the user from one BS to the other without actually changing the channel. the mobile center measures the power changes received from nearby base stations and notiﬁes the two BS. In the 2G systems. The approximate time needed to make a handoﬀ successful was about 5-10 s. The special receivers located on the BS are controlled by the MSC to monitor the signal strengths of the users in the neighboring cells which appear to be in need of handoﬀ. called soft handoﬀ. In MAHO. as discussed below. In the current 3G systems. Handoﬀ in CDMA: In spread spectrum cellular systems. When compared to the other generations.www. Based on the information received from the special receivers the MSC decides whether a handoﬀ is required or not. the MS measures the power from adjacent BS and automatically upgrades the channels to its nearer BS. the mobiles share the same channels in every cell.

However.5. A good solution to avoid such a dead-lock is to use DCA with handoﬀ priority (demand based allocation). then the corresponding microcell handles the call so that there is good corner coverage. (b) Cell dragging problem: this is another practical problem in the urban area with additional microcells. This means. it is possible to provide larger and smaller cells at a same location.com
3. As illustrated in the Figure 3.3
Handoﬀ Priority
While assigning channels using either FCA or DCA strategy.4
A Few Practical Problems in Handoﬀ Scenario
(a) Diﬀerent speed of mobile users: with the increase of mobile users in urban areas.6. This technique provides large area coverage to high speed users while providing small area coverage to users traveling at low speed.5. microcells are introduced in the cells to increase the capacity (this will be discussed later in this chapter). umbrella cell is co-located with few other microcells. If the speed is less. a guard channel concept must be followed to facilitate the handoﬀs. Since there is a LOS with the BS1.www. Several schemes thus have been designed to handle the simultaneous traﬃc of high speed and low speed users while minimizing the handoﬀ intervention from the MSC. The BS can measure the speed of the user by its short term average signal strength over the RVC and decides which cell to handle that call. The users with high speed frequently crossing the micro-cells become burdened to MSC as it has to take care of handoﬀs. 33
www. a fraction of total available channels must be kept for handoﬀ requests. This approach assures that handoﬀs are minimized for high speed users and provides additional microcell channels for pedestrian users.jntuworld.jntuworld. it experiences a lot of interferences. By using diﬀerent antenna heights and diﬀerent power levels. since the user is in cell covered by BS2. the signal strength received from BS1 would be greater than that received from BS2. consider there is a LOS path between the MS and BS1 while the user is in the cell covered by BS2. This problem can be solved by judiciously choosing the handoﬀ threshold along with adjusting the coverage area. handoﬀ cannot take place and as a result. one of them being the ‘Umbrella Cell’ approach.com
. For example. But this would reduce the carried traﬃc and only fewer channels can be assigned for the residual users of a cell.
3.

6
Interference & System Capacity
Susceptibility and interference problems associated with mobile communications equipment are because of the problem of time congestion within the electromagnetic spectrum. The interference can be divided into 2 parts: co-channel interference and adjacent channel interference. The interference in the control channels leads to missed and error calls because of digital signaling.
3.1
Co-channel interference (CCI)
For the eﬃcient use of available spectrum. which decreases system capacity and service quality. This interference can occur from clash with another mobile in the same cell or because of a call in the adjacent cell. However. If there is an interference in the voice channels. cross talk is heard will appear as noise between the users. It is the limiting factor in the performance of cellular systems. Interference is more severe in urban areas because of the greater RF noise and greater density of mobiles and base stations. increasing frequency reuse also increases interference.www.com
. then the call might be lost if there is no handoﬀ in this case too.com
(c) Inter-system handoﬀ: if one user is leaving the coverage area of one MSC and is entering the area of another MSC. There can be interference between the base stations operating at same frequency band or any other non-cellular system’s energy leaking inadvertently into the frequency band of the cellular system. The reasons of CCI can be because of either adverse weather conditions or poor frequency planning or overlycrowded radio spectrum.6. Such a handoﬀ is called inter-system handoﬀ and in order to facilitate this. If D/R ratio is increased.
3. If the cell size and the power transmitted at the base stations are same then CCI will become independent of the transmitted power and will depend on radius of the cell (R) and the distance between the interfering co-channel cells (D). mobiles usually have roaming facility.jntuworld. Co-channel interference is the cross talk between two diﬀerent radio transmitters using the same radio frequency as is the case with the co-channel cells. The cells where the same set of frequencies is used are call co-channel cells.jntuworld. it is necessary to reuse frequency bandwidth over relatively small geographical areas. then the eﬀective distance between the co-channel cells will increase 34
www.

Let us calculate the SIR for this system.com
. The parameter Q is called the frequency reuse ratio and is related to the cluster size. we need to look into the signal power characteristics. the proof of which is given in the ﬁrst section.com
and interference will decrease. (3.12)
where i0 is the number of co-channel interfering cells. For hexagonal geometry √ Q = D/R = 3N . But large ‘Q’ leads to decrease in system capacity but increase in transmission quality. If Di is the distance of the i-th interferer from the mobile. In order to solve this equation from power calculations. The expression for the received power Pr at a distance d can be approximately calculated as Pr = P0 ( and in the dB expression as Pr (dB) = P0 (dB) − 10n log( d ) d0 (3.13)
where P0 is the power received at a close-in reference point in the far ﬁeld region at a small distance do from the transmitting antenna. small of ‘Q’ means small value of cluster size ‘N’ and increase in cellular capacity. the received power at a given mobile due to i-th interfering cell is proportional to (Di )−n (the value of ’n’ varies between 2 and 4 in urban cellular systems).14) d −n ) d0 (3. then SIR can be approximated as S = I R−n i0 −n i=1 Di 35 (3.11)
From the above equation.jntuworld. Choosing the options is very careful for the selection of ‘N’. The average power in the mobile radio channel decays as a power law of the distance of separation between transmitter and receiver.15)
www. The Signal to Interference Ratio (SIR) for a mobile receiver which monitors the forward channel can be calculated as S = I S
i0 i=1 Ii
(3.www. and ‘n’ is the path loss exponent. S is the desired signal power from the baseband station and Ii is the interference power caused by the i-th interfering co-channel base station. Let us take that the path loss exponent is same throughout the coverage area and the transmitted power be same.jntuworld.

e. But this reduces the capacity by 7/12 times. Therefore minimum N is 7..6. SIR can be approximately calculated as S R−4 = I 2(D − R)−4 + (D + R)−4 + (D)−4 + (D + R/2)−4 + (D − R/2)−4 which can be rewritten in terms frequency reuse ratio Q as S 1 = .17)
Using the value of N equal to 7 (this means Q = 4. The above equations are based on hexagonal geometry and the distances from the closest interfering cells can vary if diﬀerent frequency reuse plans are used. which in a way controls frequency reuse plan and the overall capacity of the cellular system.49. on a vertex as shown in the Figure 3.jntuworld. D-R/2 and D+R distance from other interfering cells in the ﬁrst tier. The mobile is at a distance of D-R from 2 closest interfering cells and approximately D+R/2.6). Subjective tests performed on AMPS cellular system which uses FM and 30 kHz channels show that suﬃcient voice quality can be obtained by SIR being greater than or equal to 18 dB.18) (3. the above expression yields that worst case SIR is 53. The eﬀect of co-channel interference can be minimized by optimizing the frequency assignments of the base stations and their transmit powers. D. Therefore N = 12 cluster size should be used.com
where the mobile is assumed to be located at R distance from the cell center. This is the example of a 7 cell reuse case. We can go for a more approximate calculation for co-channel SIR. Taking n = 4 in the above equation.70 (17. This shows that for a 7 cell reuse case the worst case SIR is slightly less than 18 dB.jntuworld. Tilting the base-station antenna to limit the spread of the signals in the system can also be done.
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www. co-channel interference controls link performance. If we take n=4 . I 2(Q − 1)−4 + (Q + 1)−4 + (Q)−4 + (Q + 1/2)−4 + (Q − 1/2)−4 (3. the value of ’N’ can be calculated as 6. The worst case is when the mobile is at the corner of the cell i.3 dB).com
. Therefore. If we consider only the ﬁrst layer of interfering cells and we assume that the interfering base stations are equidistant from the reference base station and the distance between the cell centers is ’D’ then the above equation can be converted as √ S ( 3N )n (D/R)n = (3.16) = I i0 i0 which is an approximate measure of the SIR.www.

6: First tier of co-channel interfering cells
3.2
Adjacent Channel Interference (ACI)
This is a diﬀerent type of interference which is caused by adjacent channels i.6.e.jntuworld.com
Figure 3. provided that the performance degradation can be tolerated in the system link budget. This is called near-far eﬀect. It is the signal impairment which occurs to one frequency due to presence of another signal on a nearby frequency. This problem is enhanced if the adjacent channel user is transmitting in a close range compared to the subscriber’s receiver while the receiver attempts to receive a base station on the channel.www. This problem might occur if the base station has problem in discriminating the mobile user from the ”bleed over” caused by the close adjacent channel mobile. The more adjacent channels are packed into the channel block. If the frequency separation between the channels is kept large this interference can be reduced to some extent. channels in adjacent cells. This occurs when imperfect receiver ﬁlters allow nearby frequencies to leak into the passband.com
. Adjacent channel interference occurs more frequently in small cell clusters and heavily used cells. Thus assignment of channels is given
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www. This eﬀect can also occur if a mobile close to a base station transmits on a channel close to one being used by a weak mobile.jntuworld. the higher the spectral eﬃciency.

then SIR for weak mobile is approximately S = 10−n I (3.www. It is easy to see that the cell radius must be as large as possible. then SIR is −52 dB.com
. If n = 4. Power control is also very much important for the prolonging of the battery life for the subscriber unit but also reduces reverse channel SIR in the system. The cell radius governs both the geographical area covered by a cell and also the number of subscribers who can be serviced. In designing a cellular layout. Practically.jntuworld. every cell requires an investment in a tower.com
such that they do not form a contiguous band of frequencies within a particular cell and frequency separation is maximized. and radio transmission equipment and so a large cell size minimizes the cost per subscriber. In the following.
3. Perfect base station ﬁlters are needed when close-in and distant users share the same cell. Power control is done such that each mobile transmits the lowest power required to maintain a good quality link on the reverse channel. we have seen that the frequency reuse technique in cellular systems allows for almost boundless expansion of geographical area and the number of mobile system users who could be accommodated. each base station receiver is preceded by a high Q cavity ﬁlter in order to remove adjacent channel interference.
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www. land on which the tower is placed. Eﬃcient assignment strategies are very much important in making the interference as less as possible. and we have also seen that co-channel cell distance D = 3N R. a brief description of the design trade-oﬀ is given. If a mobile is 10 times close to the base station than other mobile and has energy spill out of its passband.7
3.1
Enhancing Capacity And Cell Coverage
The Key Trade-oﬀ
Previously.jntuworld.7. given the subscriber density. in which the above two parameters play a crucial role.19)
which can be easily found from the earlier SIR expressions. the two parameters which are of great signiﬁcance are the cell radius R and the cluster size √ N. This is because. If the frequency factor is small then distance between the adjacent channels cannot put the interference level within tolerance limits.

then the number of clusters could be maximized by minimizing the size of a cluster N . shows that the cell radius should be small. the number of clusters M should be large. The other method. However. microcell zone concept can treated as enhancing the QoS in a cellular system.21). which. the geographic area covered by a cluster is √ Acluster = N Acell = N 3 3R2 /2. The history of cellular phones has been characterized by a rapid growth and expansion in cell subscribers. We discuss here two methods for dealing with an increasing subscriber density: Cell Splitting and Sectoring. We have seen earlier that the size of a cluster depends on the frequency reuse ratio Q. the cell radius is determined by the requirement that adequate signal to noise ratio be maintained over the coverage area. (3. Now. then the number of clusters M that could be accommodated is given by √ M = Atotal /Acluster = Atotal /(N 3 3R2 /2). in determining the value of N . transmitter power. Hence. are reused in every cluster. to make the maximum number of channels available to subscribers. we focus on the issues regarding system expansion. (3.com
. Now. by Equation (3.jntuworld. Though a cellular system can be expanded by simply adding cells to the geographical area.com
Eventually.
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www. The SNR is determined by several factors such as the antenna height.www. but R should be as small as possible to maximize the number of customers that the system can accommodate.21)
Note that all of the available channels N. This is because it is not always possible to counter the increasing demand for cellular systems just by increasing the geographical coverage area due to the limitations in obtaining new land with suitable requirements.20)
If the total serviced area is Atotal . if the cell radius R is ﬁxed.jntuworld. receiver noise ﬁgure etc. Given a cell radius R and a cluster size N . cell radius is determined by a trade-oﬀ: R should be as large as possible to minimize the cost of the installation per subscriber. Hence. but should be suﬃciently large so as to minimize the interference eﬀects. another trade-oﬀ is encountered in that N must be small to accommodate large number of subscribers. the way in which user density can be increased is also important to look at.

www. There are few challenges in increasing the capacity by reducing the cell radius. It might be that an increase in channel density is required only in speciﬁc parts of the system to support an increased demand in those areas.jntuworld. a system-wide reduction in cluster size may not be necessary since user density does not grow uniformly in all parts of the geographical area. if cells are small. Cell splitting involves the process of sub-dividing a congested cell into smaller cells. The key factor is to add as minimum number of smaller cells as possible
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www. As discussed previously. This increases the capacity of a cellular system since it increases the number of times that channels are reused. inserting these smaller cells. once a system has been initially deployed. there would have to be more of them and so additional base stations will be needed in the system. as new users subscribe to the cellular service.jntuworld.com
. However. Clearly. each with its own base station and a corresponding reduction in antenna size and transmitting power.com
The basic idea of adopting the cellular approach is to allow space for the growth of mobile users.
3. because the user density decreases on moving towards sub-urban areas. the demand for it is fairly low and users are assumed to be uniformly distributed over the service area.2
Cell-Splitting
Cell Splitting is based on the cell radius reduction and minimizes the need to modify the existing cell parameters.the number of channels available to customers (equivalently.7. Cell-splitting is a technique which has the capability to add new smaller cells in speciﬁc areas of the system. between the already existing cells results in an increase of capacity due to the additional number of channels per unit area. known as microcells. The other challenge is to meet the generally increasing demand that may vary quite rapidly between geographical areas of the system. the demand for channels may begin to exceed the capacity of some base stations. Since the new cells have smaller radii than the existing cells. the channel density per square kilometer) could be increased by decreasing the cluster size. When a new system is deployed. The challenge in this case is to introduce the new base stations without the need to move the already existing base station towers. a city may have highly populated areas and so the demand must be supported by cells with the smallest radius. The radius of cells will generally increase as we move from urban to sub urban areas. For instance. However.

www. However.com
. Notice that one of the new smaller cells lies in the center of each of the larger cells. the signal-to-interference ratio will be the same after cell splitting as it was before. The new smaller cells have half the cell radius of the original cells. the cellular system operates with cells of more than one size.7 shows a cellular layout with seven-cell clusters. The gradual addition of the smaller cells implies that. wherever an increase in demand occurs. Figure 3. The organization of cells into clusters is independent of the cell radius. If the entire system is 41
www. the new cells will have one-fourth of the area and will consequently need to support one-fourth the number of subscribers. Consider that the cells in the center of the diagram are becoming congested. At half the radius. Consequently.jntuworld.com
Figure 3. and cell A in the center has reached its maximum capacity. Also the signal-to-interference ratio is determined by cluster size and not by cell radius. new base stations will have to be added for new cells that do not lie in the center of the larger cells. at least for a time.jntuworld. if the cluster size is maintained. this allows the original base stations to be maintained even in the new system layout. so that the cluster size can be the same in the small-cell layout as it was in the large-cell layout.7: Splitting of congested seven-cell clusters. If we assume that base stations are located in the cell centers. Figure also shows how the smaller cells are being superimposed on the original layout.

The cell radius was chosen to guarantee a 17 dB 42
www. the number of channels per cell will be exactly as it was before. On equating the two received powers. Initially the number of users was not signiﬁcant. As the demand increases. and the cluster size is maintained. the aim of the system designers was to guarantee coverage.jntuworld. When the cell radius is reduced by a factor. to avoid co-channel interference when both large and small cell radii coexist. Then. the signal-to-interference ratio will not change. In other words. and the number of subscribers per cell will have been reduced.www.com
. Assume that PT1 and PT2 are the transmit powers of the larger and smaller base stations respectively. we have power received at old cell boundary = PT 1 /R4 and the power received at new cell boundary = PT 2 /(R/2)4 .com
replaced with new half-radius cells. channels in the old cell must be broken down into two groups. one that corresponds to larger cell reuse requirements and the other which corresponds to the smaller cell reuse requirements.jntuworld. we get PT 2 = PT 1 / 16. Ex. It turns out that the only way to avoid interference between the large-cell and small-cell systems is to assign entirely diﬀerent sets of channels to the two systems. assuming a path loss index n=4. So. The larger cell is usually dedicated to high speed users as in the umbrella cell approach so as to minimize the number of hand-oﬀs. when two sizes of cells co-exist in a system. and a 12-cell cluster size was chosen. At the beginning of this channel splitting process. more and more channels need to be accommodated and hence the splitting process continues until all the larger cells have been replaced by the smaller cells. the transmit power must be reduced by 12 dB in order to maintain the same S/I with the new system lay-out. 4: When the AMPS cellular system was ﬁrst deployed. If a cellular layout is replaced entirety by a new layout with a smaller cell radius. there would be fewer channels in the smaller power groups. it is also desirable to reduce the transmitted power. This is necessary to maintain the same frequency re-use plan in the new cell layout as well. The transmit power of the new cells with radius half that of the old cells can be found by examining the received power PR at the new and old cell boundaries and setting them equal. at which point splitting is complete within the region and the entire system is rescaled to have a smaller radius per cell. Some special care must be taken. however. provided the cluster size does not change. Consequently cells were conﬁgured with an eight-mile radius.

The channel set serving this cell has also been divided.www. a cell is shown which has been split into three 120o sectors.com
. In the Figure 3. However it has been found that the co-channel interference in a cellular system may be decreased by replacing a single omni-directional antenna at the base station by several directional antennas.3
Sectoring
Sectoring is basically a technique which can increase the SIR without necessitating an increase in the cluster size. each radiating within a speciﬁed sector. it has been assumed that the base station is located in the center of a cell and radiates uniformly in all the directions behaving as an omni-directional antenna.7. The base station feeds three 120o directional antennas.8: A cell divided into three 120o sectors.8.jntuworld. This technique for reducing co-channel interference wherein by using suit-
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www. Although a 12-cell cluster size provided more than adequate co-channel separation to meet a requirement for a 17 dB signal-to-interference ratio in an interference-limited environment. and the 17 dB signal-to-interference ratio requirement could not be met over 90 % of the coverage area. signal-to-noise ratio over 90% of the coverage area. so that each sector is assigned one-third of the available number cell of channels. It was estimated that a 7-cell cluster size should provide an adequate 18.
3. The system planners reasoned that a subsequent shift to a 7-cell cluster size would provide an adequate number of channels. each of which radiates into one of the three sectors.jntuworld.7 dB signal-to-interference ratio. is slim. Till now.com
Figure 3. however. it did not provide adequate frequency reuse to service an explosively growing customer base. The margin.

24) to be 23.4 dB which is a signiﬁcant improvement over the Omni-directional case where the worst-case S/I is found to be 17 dB (assuming a path-loss exponent. In a seven-cell-cluster layout with 120o sectored cells.com
Figure 3.jntuworld. Now. Hence the signal to interference ratio is now modiﬁed to √ S ( 3N )n = I 2
(3. able directional antennas. the signal to interference ratio for a seven-cell cluster layout using 120o sectored antennas can be found from equation (3.22)
where the denominator has been reduced from 6 to 2 to account for the reduced number of interfering sources.9.9: A seven-cell cluster with 60o sectors. a 60o sectoring of the cells has been introduced. a given cell would receive interference and transmit with a fraction of available co-channel cells is called ’sectoring’. n=4). 5: A cellular system having a seven-cell cluster layout with omni-directional antennas has been performing satisfactorily for a required signal to interference ratio of 15 dB. Likewise.jntuworld. Some cellular systems divide the cells into 60o sectors. rather than from all six.com
. the base station in the center cell will receive co-channel interference from mobile units in only two of the co-channel cells. Ex. it can be easily understood that the mobile units in a particular sector of the center cell will receive co-channel interference from only two of the ﬁrst-tier co-channel base stations.www.
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www. Similar analysis can be performed on them as well. By what percentage can the number of channels Ntotal be increased assuming a path-loss component n=4? Solution: The seven-cell cluster layout with 60o sectoring is shown in the Figure 3. However due to the need for increasing the number of available channels.

jntuworld. Moreover. additional interference will be introduced.com
It is easy to see that the shaded region in the center receives interference from just one ﬁrst-tier cell and hence the signal to interference ratio can be obtained suitably as √ S ( 3N )n ( (3)(7))4 = = = 26.
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www.07dB. For a 3-cell cluster layout. This increases the complexity of the system and also the load on the mobile switching center/base station. dividing a cell into sectors requires that a call in progress will have to be handed oﬀ (that is. practical antennas have side lobes and cannot be used to focus a transmitted beam into a perfect 120o sector or 60o sector. when the cluster size is reduced from 7 to 3.25)
This is just above the adequate S/I ratio and further reduction in cluster size is not possible. This is due to a reduction in Trunking Eﬃciency. Because sectoring involves using more than one antenna per base station. the SIR for this reduced cluster size layout can be found to be √ S ( 3N )n ( (3)(4))4 = = = 21. Thus.jntuworld.23)
Since the SIR exceeds 15 dB. Next. The calculations in the above example are actually an idealization for several reasons.www. a 3-cluster cell layout could be used for meeting the growth requirements. a term which will be explained later on.6dB. This breaks the available set of channels into smaller sets.24)
The S/I ratio is still above the requirement and so a further reduction in the cell cluster size is possible. it is also a cause of concern that a given number of channels are not able to support as many subscribers when the pool of channels is divided into small groups. Firstly. Due to this. I 1 1
(3. there are two interfering sources and hence the S/I ratio is found to be √ √ S ( 3N )n ( 33)4 = = = 16. Now. So. the available channels in the cell are divided and dedicated to a speciﬁc antenna.com
. thus reducing the trunking eﬃciency. assigned a new channel) when a mobile unit travels into a new sector.4dB. I 1 2
(3. the total number of channels increased by a factor of 7/3. I 1 1
(3. one can try reducing the cluster size from seven to four.

com
.jntuworld. use the same frequency used by that cell. Since
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www. The new microcell knows where to locate the mobile unit in a particular zone of the cell and deliver the power to that zone. 3) Size of the zone apparatus is small. that is no handovers occur between microcells. within that cell.choosing the zone with the strongest signal. It is necessary to note that all the microcell zones. The base station of that cell knows in which zone. with each of the three zone sites connected to the base station and sharing the same radio equipment. The zone site equipment being small can be mounted on the side of a building or on poles. a new microcell zone concept has been proposed. 2) Handoﬀs are reduced (also compared to decreasing the cell size) since the microcells within the cell operate at the same frequency. the system already knows the cell location of that phone. the base station transmits it to the suitable zone site.4
Microcell Zone Concept
The increased number of handoﬀs required when sectoring is employed results in an increased load on the switching and control link elements of the mobile system.com
3. Beneﬁts of the micro-cell zone concept: 1) Interference is reduced in this case as compared to the scheme in which the cell size is reduced. To overcome this problem. the BS simply switches the channel to a diﬀerent zone site and no physical re-allotment of channel takes place. The zone site receives the cellular signal from the base station and transmits that signal to the mobile phone after ampliﬁcation. within a cell. Base Station Signals: When a call is made to a cellular phone. 4) System capacity is increased.www. which in turn send a signal to the BS.7. co-channel interference is reduced between the zones and the capacity of system is increased.10. Therefore when it receives the signal. no handover occurs when the mobile unit moves between the microcells. A zone selector at the BS uses that signal to select a suitable zone to serve the mobile unit . As shown in Figure 3. this scheme has a cell divided into three microcell zones.jntuworld. Locating the mobile unit within the cell: An active mobile unit sends a signal to all zone sites. the cellular phone is located. Thus when a mobile user moves between two microcell zones of the cell. By conﬁning the power transmitted to the mobile phone.

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Figure 3.10: The micro-cell zone concept. the signal power is reduced, the microcells can be closer and result in an increased system capacity. However, in a microcellular system, the transmitted power to a mobile phone within a microcell has to be precise; too much power results in interference between microcells, while with too little power the signal might not reach the mobile phone.This is a drawback of microcellular systems, since a change in the surrounding (a new building, say, within a microcell) will require a change of the transmission power.

3.8

Trunked Radio System

In the previous sections, we have discussed the frequency reuse plan, the design trade-oﬀs and also explored certain capacity expansion techniques like cell-splitting and sectoring. Now, we look at the relation between the number of radio channels a cell contains and the number of users a cell can support. Cellular systems use the concept of trunking to accommodate a large number of users in a limited radio spectrum. It was found that a central oﬃce associated with say, 10,000 telephones

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requires about 50 million connections to connect every possible pair of users. However, a worst case maximum of 5000 connections need to be made among these telephones at any given instant of time, as against the possible 50 million connections. In fact, only a few hundreds of lines are needed owing to the relatively short duration of a call. This indicates that the resources are shared so that the number of lines is much smaller than the number of possible connections. A line that connects switching oﬃces and that is shared among users on an as-needed basis is called a trunk. The fact that the number of trunks needed to make connections between oﬃces is much smaller than the maximum number that could be used suggests that at times there might not be suﬃcient facilities to allow a call to be completed. A call that cannot be completed owing to a lack of resources is said to be blocked. So one important to be answered in mobile cellular systems is: How many channels per cell are needed in a cellular telephone system to ensure a reasonably low probability that a call will be blocked? In a trunked radio system, a channel is allotted on per call basis. The performance of a radio system can be estimated in a way by looking at how eﬃciently the calls are getting connected and also how they are being maintained at handoﬀs. Some of the important factors to take into consideration are (i) Arrival statistics, (ii)Service statistics, (iii)Number of servers/channels. Let us now consider the following assumptions for a buﬀerless system handling ’L’ users as shown in Figure 3.11: (i) The number of users L is large when compared to 1. (ii) Arrival statistics is Poisson distributed with a mean parameter λ. (iii) Duration of a call is exponentially distributed with a mean rate µ1 . (iv) Residence time of each user is exponentially distributed with a rate parameter µ2 . (v) The channel holding rate therefore is exponentially distributed with a parameter µ = µ1 + µ2 . (vi) There is a total of ’J’ number of channels (J ≤ L). To analyze such a system, let us recapitulate a queuing system in brief. Consider an M/M/m/m system which is an m-server loss system. The name M/M/m/m reﬂects

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Figure 3.11: The buﬀerless J-channel trunked radio system.

Figure 3.12: Discrete-time Markov chain for the M/M/J/J trunked radio system.

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the hour of the day during which there is the greatest demand for service.11 can be modeled as M/M/J/J system and the discrete-time Markov chain of this system is shown in Figure 3.12. The parameter λuser is also called the average arrival rate. a GoS of 2 % implies that on the average a user might not be successful in placing a call on 2 out of every 100 attempts. M stands for memoryless which here means a Poisson process). the call gets rejected and in some systems. User calling can be modeled statistically by two parameters: the average number of call requests per unit time λuser and the average holding time H. telephone systems are engineered to provide a speciﬁed grade of service during a speciﬁed busy hour.com
standard queuing theory nomenclature whereby: (i) the ﬁrst letter indicates the nature of arrival process(e. referring to the rate at which calls from a single user arrive. For example.g M stands for exponential distribution). In practice the blocking frequency varies with time. Typically.jntuworld. is not available.jntuworld. (iii) the third letter indicates the number of servers. that is. (iv) the last letter indicates that if an arrival ﬁnds all ’m’ users to be busy. Trunking mainly exploits the statistical behavior of users so that a ﬁxed number of channels can be used to accommodate a large. In the telephone system context the term Grade of Service (GoS) is used to mean the probability that a user’s request for service will be blocked because a required facility. then it will not enter the system and is lost. it becomes more likely that all channels are busy for a particular user. The average holding time is the average duration of a call. a queue may be used to hold the caller’s request until a channel becomes available. random user community. such as a trunk or a cellular channel. the buﬀerless system as shown in Figure 3.com
. In view of the above. The
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www.successive inter arrival times and service times are assumed to be statistically independent of each other.(e. As the number of telephone lines decrease. In all cases. As a result. (ii) the second letter indicates the nature of probability distribution of service times. Telephone operating companies maintain usage records and can identify a ”busy hour”.g. One would expect far more call attempts during business hours than during the middle of the night.www.

Use of the blocked-calls-cleared statistical model leads to the Erlang B formula that relates oﬀered traﬃc intensity A. The blocked calls cleared model assumes that when a channel or trunk is not available to service an arriving call. grade of service Pb .jntuworld. the call is cleared from the system. This implies that calls of long duration are much less frequent than short calls. For example. a channel that is occupied for thirty minutes during an hour carries 0.jntuworld.28)
www. Two models are widely used in traﬃc engineering to represent what happens when a call is blocked. The Erlang B formula is: Pb = AK /K! K n n=0 A /n! (3. One Erlang represents the amount of traﬃc intensity carried by a channel that is completely occupied. The purpose of the statistical model is to relate the oﬀered traﬃc intensity A.www. The second model is known as blocked calls delayed. Call arrivals or requests for service are modeled as a Poisson random process. It is based on the assumption that there is a large pool of users who do not cooperate in deciding when to place calls. and number of channels K. then the traﬃc intensity oﬀered by N users is A = NAuser . the ”grade of service” refers to the probability that a call will be delayed.26)
that is. This quantity represents the average traﬃc that a user provides to the system.27)
When the blocked-calls-delayed model is used. In this case the statistical model leads to the Erlang C formula. In this model a call that cannot be serviced is placed on a queue and will be serviced when a channel or trunk becomes available.5 Erlang of traﬃc. the product of the average arrival rate and the average holding time–is called the oﬀered traﬃc intensity or oﬀered load.com
product: Auser = λuser H (3. the grade of service Pb . If the traﬃc intensity oﬀered by a single user is Auser . Holding times are very well predicted using an exponential probability distribution. P [delay] = AK /[(K − A)(K − 1)]! .com
. AK /[(K − A)(K − 1)]! + K An /n! n=0 51 (3. and the number of channels or trunks C needed to maintain the desired grade of service. Oﬀered traﬃc intensity is a quantity that is traditionally measured in Erlangs.

32)
This explains why the sectoring of a cell into either 120o or 60o sectors reduces the trunking eﬃciency of the system. The above example indicates that the number of subscribers that can be supported by a given number of channels decreases as the pool of channels is sub-divided.255Erlangs Thus the maximum number of users per group is Ngroup = A/Auser ≈ 403. that is.1 Erlang. suppose that the channels have been divided into two groups of 50 channels each.31) (3.29)
Ex.1 ≈ 880.www. the maximum number of subscribers served is N = A/Auser = 87. Since an individual subscriber oﬀers a load of Auser = (2 calls / 60 min)3 min = 0. How many subscribers can be served by the two group cell? Solution: Using the Erlang B table with C = 50 and GOS = Pb = 2%.972/0. ξ = (1 − Pb )A/C. deﬁned as the carrier load per channel.jntuworld. Each cell has 100 channels. (3. the total oﬀered load A=87. how many subscribers can be serviced by each cell at 2 % GoS? Solution: Using Erlang B table. We can express this in terms of the trunking eﬃciency. the total oﬀered load per group is A = 40. it can be seen that for C = 100 and GoS = Pb = 2%. maximum number of users in the two group cell is 806. If the blocked calls are cleared.972 Erlangs. (3. Each subscriber is assigned to a group and can be served only by that group.com
. Thus the system growth due to sectoring is impacted by trunking eﬃciency considerations. 6: In a certain cellular system. counting both the groups.jntuworld.30)
Thus. an average subscriber places two calls per hour during a busy hour and the average holding time is 3 min.com
Ex. (3. 4: In the previous example.
52
www.

jntuworld. 2. 3. Rappaport. 2002. Singapore: Pearson Education. Inc. Modern Wireless Communications. S. New Delhi: PHI. Wireless Communications and Networking.9
References
1. 2nd ed. S. Haykin and M. Mark and W.com
.
53
www. Singapore: Pearson Education.www. 1995. T.. NJ: Prentice Hall. J. K. Wireless Communications: Principles and Practice. Moher. W. Wireless Digital Communications: Modulation and Spread Spectrum Applications. 2005..com
3. Feher. Inc. 2002.jntuworld. 4. Upper Saddle River. Zhuang.

Diﬀraction is a phenomena occurring when the wave interacts with a surface having sharp irregularities.com
. as it travels through the wireless channel. Scattering occurs when the medium through the wave is traveling contains objects which are much smaller than the wavelength of the EM wave. due to the presence of buildings. Reﬂection occurs when the EM waves impinge on objects which are much greater than the wavelength of the traveling wave. It presents challenges and conditions which are unique for this kind of transmissions.1 Introduction
There are two basic ways of transmitting an electro-magnetic (EM) signal.jntuworld. These varied phenomena’s lead to large scale and small scale propagation losses. mountains and other such obstructions. through a guided medium or through an unguided medium. A signal. These are termed so because they predict the average signal strength for large Tx-Rx separations. Guided mediums such as coaxial cables and ﬁber optic cables. typically for hundreds of kilometers.www. diﬀraction and scattering. 54
www. Models which predict the mean signal strength for arbitrary transmitter receiver distances are termed as large scale propagation models.jntuworld. Due to the inherent randomness associated with such channels they are best described with the help of statistical models. are far less hostile toward the information carrying EM signal than the wireless or the unguided medium. undergoes many kinds of propagation eﬀects such as reﬂection.com
Chapter 4
Free Space Radio Wave Propagation
4.

Pr (d) is the received power.jntuworld.com
.jntuworld.2
Free Space Propagation Model
Although EM signals when traveling through wireless channels experience fading eﬀects due to various eﬀects.
4. Gt is the transmitter antenna gain. but in some cases the transmission is with a direct line of sight such as in satellite communication. representing the attenuation suﬀered by the signal as it travels through the wireless channel is given by the diﬀerence of the transmitted and received power in dB and is expressed as: P L(dB) = 10 log Pt /Pr .1: Free space propagation model.3)
55
www. ﬁlter losses and antenna losses and not related to propagation.www. The gain of the antenna is related to the eﬀective aperture of the antenna which in turn is dependent upon the physical size of the antenna as given below G = 4πAe /λ2 .2)
The path loss. Friis free space equation is given by Pr (d) = P t Gt Gr λ 2 (4π)2 d2 L (4.com
Figure 4. (4. (4. showing the near and far ﬁelds.1)
where Pt is the transmitted power. d is the Tx-Rx separation and L is the system loss factor depended upon line attenuation. Free space model predicts that the received power decays as negative square root of the distance. Gr is the receiver antenna gain.

which is dependent upon the largest dimension of the antenna as df = 2D2 /λ.jntuworld. the far ﬁeld and the near ﬁeld. the Friis equation is used only beyond the far ﬁeld distance.com
The ﬁelds of an antenna can broadly be classiﬁed in two regions. the far ﬁeld distance is df = . (4. as a reference point. Solution: (i) Tx power = 10log(50) = 17 dB = (17+30) dBm = 47 dBm (ii) df =
2×D2 λ
2D2 2(1)2 = = 6m λ 0. It is in the far ﬁeld that the propagating waves act as plane waves and the power decays inversely with distance. 2: A unit gain antenna with a maximum dimension of 1 m produces 50 W power at 900 MHz. PT GT GR λ2 4πd2 50 × 1 × (1/3)2 = 4π1002 PR = 56
www. D=1m. Ex. do . we use a close in distance. The far ﬁeld region is also termed as Fraunhofer region and the Friis equation holds in this region. Solution: Since the operating frequency f = 900 Mhz. with the largest dimension of the antenna. At 100 m . is then given by: Pr (d) = Pr (do )(do /d)2 . Hence. (4. For this reason. Thus. Find (i) the transmit power in dBm and dB. df .4)
Also we can see that the Friis equation is not deﬁned for d=0. (ii) the received power at a free space distance of 5 m and 100 m. the wavelength λ= 3 × 108 m/s m 900 × 106 Hz
. The power received.www. Pr (d).jntuworld.33
=
2×12 1/3
= 6m
Thus the received power at 5 m can not be calculated using free space distance formula.5)
Ex.com
. 1: Find the far ﬁeld distance for a circular antenna with maximum dimension of 1 m and operating frequency of 900 MHz.

such objects can be the earth. The most important parameter.www. This would occur.com
. The amount of energy that is reﬂected back depends on the polarization of the e.3.3
Basic Methods of Propagation
Reﬂection. a part of it is transmitted into it. If the medium is a perfect conductor. it is found to be sin(θB ) =
1 1+ 2
. which has very large dimensions as compared to the wavelength of the propagating wave. when the incident angle would be such that the reﬂection coeﬃcient is equal to zero. considering perfect conductors. wave is incident is a dielectric. predicted by propagation models based on above three phenomena.
(4.8) (4. Hence all energy is reﬂected back.jntuworld. when no reﬂection occurs in the medium of origin.5 × 10−3 mW PR (dBm) = 10logPr (mW ) = −24.6)
Further. When a radio wave falls on another medium having diﬀerent electrical properties. This angle is the Brewster’s angle. By applying laws of electro-magnetics.m. Another particular case of interest arises in parallel polarization. The following subsections give an outline of these phenomena.5dBm
4. is the received power. some energy is reﬂected back and some energy is transmitted. apart from LoS communication. If the medium on which the e. the electric ﬁeld inside the conductor is always zero. diﬀraction and scattering are the three fundamental phenomena that cause signal propagation in a mobile communication system. For example. all energy is reﬂected back to the ﬁrst medium. Boundary conditions require that θi = θr and Ei = Er 57 (4. buildings and walls.7)
www.
4.m.com
= 3. Let us see some special cases. The physics of the above phenomena may also be used to describe small scale fading and multipath propagation.jntuworld.1
Reﬂection
Reﬂection occurs when an electromagnetic wave falls on an object. wave. while some energy is reﬂected back.

The propagation of secondary wavelets in the shadowed region results in diﬀraction. given by ρS = exp(−8( πσh sinθi 2 ) ) λ (4. every point on a wavefront acts as point sources for the production of secondary wavelets. the surface reﬂection coeﬃcient needs to be multiplied by a scattering loss factor ρS .12)
www. In case of rough surfaces.10)
A surface is smooth if its minimum to maximum protuberance h is less than hc . which deﬁnes a critical height hc of surface protuberances for a given angle of incidence θi .jntuworld. hc = λ . This phenomenon can be explained by the Huygen’s principle.com
for vertical polarization. The ﬁeld in the shadowed region is the vector sum of the electric ﬁeld components of all the secondary wavelets that are received by the receiver.3
Scattering
The actual received power at the receiver is somewhat stronger than claimed by the models of reﬂection and diﬀraction. The following result is a better approximation to the observed value ρS = exp(−8( πσh sinθi 2 πσh sinθi 2 ) )I0 [−8( ) ] λ λ 58 (4.com
. according to which. buildings and lampposts scatter energy in all directions. around the curved earth’s surface and obstructions like tall buildings. But the diﬀraction ﬁeld still exists an it has enough strength to yield a good signal. and rough if protuberance is greater than hc .jntuworld. given by.9)
4. the received ﬁeld strength decreases.
4.www.11)
where σh is the standard deviation of the Gaussian random variable h. As the user moves deeper into the shadowed region. The cause is that the trees. and Ei = −Er for horizontal polarization. (4.2
Diﬀraction
Diﬀraction is the phenomenon due to which an EM wave can propagate beyond the horizon.3. and they combine to produce a new wavefront in the direction of propagation. Roughness is tested by a Rayleigh criterion. 8sinθi (4. This provides extra energy at the receiver.3.

4
Two Ray Reﬂection Model
Interaction of EM waves with materials having diﬀerent electrical properties than the material through which the wave is traveling leads to transmitting of energy through the medium and reﬂection of energy back in the medium of propagation. Γrough = ρS Γ. The amount of energy reﬂected to the amount of energy incidented is represented by Fresnel reﬂection coeﬃcient Γ.jntuworld. for parallel and perpendicular polarizations.com
Figure 4. which agrees very well for large walls made of limestone. as the EM waves can not pass through conductors.2: Two-ray reﬂection model.13)
4. The equivalent reﬂection coeﬃcient is given by. which depends upon the wave polarization. (4. In general. Γ is given by: Γ|| = Er /Ei = η2 sin θt − η1 sin θi /η2 sin θt + η1 sin θi (4.com
.14)
59
www.jntuworld. angle of incidence and frequency of the wave. For example. all the energy is reﬂected back with angle of incidence equal to the angle of reﬂection and reﬂection coeﬃcient Γ = −1.www.

one LOS wave which travels a distance of d and another ground reﬂected wave. This is deduced via the following treatment. t)| = E0 d0 .20)
This means the envelop is constant with respect to time.
(4.17)
Let E0 is the free space electric ﬁeld (in V/m) at a reference distance d0 .16) (4. A two-ray model. Two propagating waves arrive at the receiver.2.com
Γ⊥ = Er /Ei = η2 sin θi − η1 sin θt /η2 sin θi + η1 sin θt . T ER OT = Eg + ELOS . d (4. The envelop of the electric ﬁeld at d meters from the transmitter at any time t is therefore |E(d. Mathematically.2. one direct path and one reﬂected wave from the ground gives a more accurate description as shown in Figure 4.jntuworld.21)
www.jntuworld.com
. that travels d . it can be expressed as: E(d . the total transmitted and received electric ﬁelds are
T ET OT = Ei + ELOS . Then E(d.19) E0 d0 cos(ωc t − φ) d (4. From Figure 4.www. t) = where φ = ωc 60 d .15)
Seldom in communication systems we encounter channels with only LOS paths and hence the Friis formula is not a very accurate description of the communication link. A simple addition of a single reﬂected wave shows that power varies inversely with the forth power of the distance between the Tx and the Rx.24) E0 d0 cos(ωc t − φ ) d (4. c (4.
(4. which consists of two overlapping waves at the receiver.22) E0 d0 cos(ωc t − φ ) d (4.18)
and d > d0 . t) = where φ = ωc and E(d .23) d c (4. t) = where φ = ωc d c (4.

3: Phasor diagram of electric ﬁelds.3. when T-R separation distance is very large compared to (ht + hr ).25)
For small values of θi . Et = Ei + Eg = Ei (1 + Γ).30).jntuworld. Γ⊥ = −1 =⇒ Et = (1 − 1)Ei = 0.
T ER OT = |ELOS + Eg |.www.4: Equivalent phasor diagram of Figure 4.29) and (4.30)
Ex 3: Prove the above two equations.com
. Assuming perfect horizontal electric ﬁeld polarization. (4.jntuworld.e. i.31)
www.
According to the law of reﬂection in a dielectric.com
Figure 4. This implies that. equation (4.27)
It can be therefore written that
T ER OT (d. i. the phase diﬀerence is θ∆ = 2π∆ ∆ωc = λ λ 61 (4.28)
In such cases.26)
the resultant electric ﬁeld is the vector sum of ELOS and Eg . the path diﬀerence is ∆=d −d = (ht + hr )2 + d2 − (ht − hr )2 + d2 . θi = θ0 and Eg = ΓEi which means the total electric ﬁeld. reﬂected wave is equal in magnitude and 180o out of phase with respect to incident wave.. then ∆≈ 2ht hr d (4.e..
Figure 4.29)
However. t) =
E0 d0 E0 d0 cos(ωc t − φ ) + (−1) cos(ωc t − φ ) d d
(4.
(4. Once the path diﬀerence is known. (4. (4.

we want to evaluate the received E-ﬁeld at any t =
T ER OT (d. d λd d
(4.43)
The corresponding approximate received electric ﬁeld is
T ER OT (d) ≈ 2
E0 d0 2πht hr ht hr =k 2 . then ∆ becomes very small and therefore ELOS and Eg are virtually identical with only phase diﬀerence.
we can then approximate that sin( θ∆ π 2πht hr )≈ ∆= < 0.3 and 4.30).38) (4. d 2
(4.4.42)
This raises the wonderful concept of ‘cross-over distance’ dc ..com
and the time diﬀerence. | E0 d0 E0 d0 E0 d0 |≈| |. deﬁned as d > dc = 4πht hr 20πht hr = .e. |≈| d d d
d c
(4. 5λ λ (4.com
. (4. Then.5rad.
Using equation (4.jntuworld.31) and further equation (4. we get
T |ER OT (d)| =
(
E0 d0 E0 d0 E0 d0 + cos(θ∆ ))2 + ( sin(θ∆ ))2 d d d (cos(θ∆ ) − 1)2 + sin2 (θ∆ ) = E0 d0 2 − 2cosθ∆ d E0 d0 θ∆ =2 sin( ).36) (4.41)
=
E0 d0 d
For
θ∆ 2
∆ < 0.jntuworld. t =
d d E0 d0 d E0 d0 d d cos(ωc − ωc ) − cos(ωc − ωc ) )= c d c c d c c = E0 d0 ∆ωc E0 d0 cos( cos(0o ) )− d c d E0 d0 E0 d0 = θ∆ − d d E0 d0 ≈ ( θ∆ − 1).40) (4.44)
62
www.39) (4. sin( θ2 ) ≈
θ∆ 2 . = c 2πfc (4.32)
When d is very large.34) (4. d
Using phasor diagram concept for vector addition as shown in Figures 4.33) .35) (4. τd = ∆ θ∆ .37)
Say.www.5rad. 2 λ λd (4.i.

jntuworld. d4 (4. the power falls oﬀ rapidly as well as two-ray model approximation gives better result than Friis equation. Path loss is independent of frequency (wavelength).45): The important observations from this equation are: 1. This equation gives fair results when the T-R separation distance crosses the cross-over distance.g the sound can be heard in a room. For e. but whenever we distance crosses the ‘cross-over distance’.com
.46)
4.1) is suﬃcient to calculate the path loss since the two-ray model does not give a good result for a short distance due to the oscillation caused by the constructive and destructive combination of the two rays.45)
The cross-over distance shows an approximation of the distance after which the received power decays with its fourth order. In that case. Received power is also proportional to h2 and h2 . Ld4 (4. The basic diﬀerence between equation (4. if height of any of the t r antennas is increased.1). we get the received power as Pr = Pt Gt Gr h2 h2 t r . The similar phenomena occurs for light also but the diﬀracted light intensity is not noticeable.1) and (4. K . using equation (4. This is because the obstacle or slit need to be of the order of the wavelength of the wave to have a signiﬁcant eﬀect.jntuworld. 2. so as to propagate behind the obstacle. It is an inherent feature of a wave be it longitudinal or transverse. 3. where the source of the sound is another room without having any line of sight. equation (4.www.45) is that when d < dc .43) in (4. the power decays as the fourth power of distance Pr (d) = with K being a constant. received power increases. 1. Thus radiation from a point source radiating in all directions can be received at any 63
www. Observations on Equation (4. under the inﬂuence of an obstacle.com
Therefore. meaning.5
Diﬀraction
Diﬀraction is the phenomena that explains the digression of a wave from a straight line path.

leading to bending of wave. The path diﬀerence between direct path and the diﬀracted path is δ= d2 + h2 + 1 d2 + h2 − (d1 + d2 ) 2 (4. Though the intensity received gets smaller as receiver is moved into the shadowed region. the sum of all wave sources is zero at a point not in the direct path of the wave and thus the wave travels in the straight line. even behind an obstacle (unless it is not completely enveloped by it).jntuworld.1
Knife-Edge Diﬀraction Geometry
As shown in Figure 4. this has a great advantage since. consider that there’s an impenetrable obstruction of hight h at a distance of d1 from the transmitter and d2 from the receiver. by diﬀraction (and scattering.5.5: Huygen’s secondary wavelets. Normally.5. But in the case of an obstacle.www. In mobile communication. in absence of an obstacle.com
. Diﬀraction is explained by Huygens-Fresnel principle which states that all points on a wavefront can be considered as the point source for secondary wavelets which form the secondary wavefront in the direction of the prorogation. the receiver is able to receive the signal even when not in line of sight of the transmitter.6.com
Figure 4.
4. as shown in Figure 4. point.47)
64
www.jntuworld. reﬂection). the eﬀect of wave source behind the obstacle cannot be felt and the sources around the obstacle contribute to the secondary wavelets in the shadowed region. This we show in the subsection given below.

52) (4.51) (4.48)
In order to normalize this. αtanα = tanβ + tanγ = h/d1 + h/d2 = h(d1 + d2 )/d1 d2 . we usually use a Fresnel-Kirchoﬀ diﬀraction parameter v.com
Figure 4.53)
65
www.50) (4. which can be further simpliﬁed as δ = d1 (1 + h2 /2d2 ) + d2 (1 + h2 /2d2 ) − (d1 + d2 ) 1 2 = h2 /(2d1 ) + h2 /(2d2 ) = h2 (d1 + d2 )/(2d1 d2 ).www.6: Diﬀraction through a sharp edge.49) (4.com
. expressed as v = h 2(d1 + d2 )/(λd1 d2 ) = α (2d1 d2 )/(λ(d1 + d2 )) (4.jntuworld.jntuworld. (4. Thus the phase diﬀerence equals φ = 2πδ/λ = 2πh2 (d1 + d2 )/λ2(d1 d2 ). With the following considerations that α=β+γ and α ≈ tanα we can write.

com
. the more is the object in the shadowed region greater is the diﬀraction loss of the signal. and therefore the phase diﬀerence becomes φ = πv 2 /2. we can observe that: (i) phase diﬀerence is a function of the height of the obstruction. Thus successive Fresnel zones have phase diﬀerence of π which means they alternatively 66
www. The eﬀect of diﬀraction loss is explained by Fresnel zones as a function of the path diﬀerence.
4. (ii) phase diﬀerence is a function of the position of the obstruction from transmitter and receiver.www. and also.5.7: Fresnel zones.jntuworld.2
Fresnel Zones: the Concept of Diﬀraction Loss
As mentioned before.54)
From this.7.com
Figure 4. as shown in Figure 4. (4. The successive Fresnel zones are limited by the circular periphery through which the path diﬀerence of the secondary waves is nλ/2 greater than total length of the LOS path.jntuworld.

Compute the diﬀraction loss.6 = 15. Also ﬁnd out in which Fresnel zone the tip of the obstruction lies. where symbols have usual meaning. Ex 4: Calculate the ﬁrst Fresnel zone obstruction height maximum for f = 800 MHz. depending on its geometry. Now. The radius of the each Fresnel zone is maximum at middle of transmitter and receiver (i.www. The diﬀraction losses are minimum as long as obstruction doesn’t block volume of the 1st Fresnel zone.com
. it obstructs contribution from some of the secondary wavelets.com
provide constructive and destructive interference to the received the signal. h = 25m.89 = 16. if there’s no obstruction. resulting in diﬀraction and also the loss of energy.e. which is the vector sum of energy from unobstructed sources. But if an obstruction is present.48m 500
Ex 5: Given f=900 MHz.
3 8
× 100 × 400 = 10 (0. To have good power strength. As a rule of thumb.jntuworld.89m
Thus H1 = 10 + 6. diﬀraction eﬀects are negligible beyond 55% of 1st Fresnel zone. d1 = d2 = 1 km.89m
(b) H2 = Thus H2 = 10 + 5. It is seen that the loci of a Fresnel zone varied over d1 and d2 forms an ellipsoid with the transmitter and receiver at its focii.3) = 5.
67
www. when d1 = d2 ) and decreases as moved to either side.48m .jntuworld. then all Fresnel zones result in only the direct LOS prorogation and no diﬀraction eﬀects are observed. Solution: λ= H= H1 =
λ(d1 +d2 ) d1 +d2
3 250×250 8
c 3 × 108 3 = = m 2 × 106 f 8 × 10 8
500
= 6. obstacle should be within the 60% of the ﬁrst fresnel zone. please note that height of the obstruction can be positive zero and negative also.

7dB v
Since loss = -Gd (dB) = 21.8: Knife-edge Diﬀraction Model Given.5. It considers the object like hill or mountain as a knife edge sharp
68
www.com
.225/v) − 1 < v <= 0 v > 2.74 1 3 10
Gd (dB) = 20 log( 225 ) = −21.com
Figure 4.5 − 0.www.5 2
4.7 dB
n= Thus n=4.
(2.74)2 = 3. Gd (dB) = 20 log(0.24
Solution: v=h 2(d1 + d2 ) = 25 λd1 d2 2 × 2000 = 2.3
Knife-edge diﬀraction model
Knife-edge diﬀraction model is one of the simplest diﬀraction model to estimate the diﬀraction loss.jntuworld.jntuworld.62v) Gd (dB) = 20 log(0.

62) − 1 <= v <= 0 0 <= v <= 1 1 <= v <= 2. The important factor is to select the correct reference distance d0 .8. The electric ﬁeld strength.jntuworld.55)
v
The diﬀraction gain due to presence of knife edge can be given as Gd (db) = 20log|F (v)| Gd (db) = 0v <= −1 Gd (db) = 20log(0.com
.
69
www. P L(d)( d n d ) =⇒ P L(dB) = P L(d0 ) + 10nlog( ) d0 d0 (4.60) (4.5exp(−0.62)
The value of n varies with propagation environments.61)
Gd (db) = 20log(0. while for micro-cell system it varies from 10m-1m. The value of n varies from 4 to 6 for obstruction of building.
(4.4
When there are more than one obstruction.4 (4.1
Link Budget Analysis
Log-distance Path Loss Model
According to this model the received power at distance d is given by. then the equivalent model can be found by one knife-edge diﬀraction model as shown in Figure 4.225/v)
v > 2.38 − 0.jntuworld. Moreover it does not account for the shadowing eﬀects. and 3 to 5 for urban scenarios.4 − sqrt(0.
4. For large cell area it is 1 Km.com
object.57) (4.www.5 − 0.58) (4. Limitations: Surrounding environmental clutter may be diﬀerent for two locations having the same transmitter to receiver separation.59) (4. The value of n is 2 for free space.1v 2 ))) Gd (db) = 20log(0.95v))
Gd (db) = 20log(0. Ed of a knife-edge diﬀracted wave is given by
∞
Ed /Eo = F (v) = (1 + j)/2
(exp((−jπt2 )/2)dt.6
4.1184 − (0.56) (4.6.

com
. Okumura model. we discuss some of the outdoor models. Durkin’s model.jntuworld.7. In practice n and σ values are computed from measured data.6.2
Log Normal Shadowing
The equation for the log normal shadowing is given by. P L(dB) = P L(dB) + Xσ = P L(d0 ) + 10nlog( d ) + Xσ d0 (4.com
4.63)
where Xσ is a zero mean Gaussian distributed random variable in dB with standard deviation σ also in dB.It can be applicable for base station eﬀective antenna heights (ht ) ranging from 30 m to 1000 m.65) (4.66)
4.5(1 − erf ( √ )) 2 and Q(z) = 1 − Q(−z) (4.64)
So the probability that the received signal level (in dB) will exceed a certain value γ is P (Pd > γ) = Q( γ − Pr ).7
Outdoor Propagation Models
There are many empirical outdoor propagation models such as Longley-Rice model.www. Certain modiﬁcations over the rudimentary model like an extra urban factor (UF) due to urban clutter near the reciever is also included in this model. Hata model etc. followed by a few indoor models too. Average received power The ‘Q’ function is given by. 70
www.jntuworld.The frequency coverage of this model is in the range of 200 MHz to 1900 MHz and distances of 1 Km to 100 Km. Below. z Q(z) = 0. σ (4.1
Okumura Model
The Okumura model is used for Urban Areas is a Radio propagation model that is used for signal prediction.
4. Longley-Rice model is the most commonly used model within a frequency band of 40 MHz to 100 GHz over diﬀerent terrains.

G(hr ) is the mobile antenna height gain factor.jntuworld.71)
71
www. G(hr ) = 20 log10 (hr /3). The values of Amu (fc .urban (d)dB = 69.2
Hata Model
The Hata model is an empirical formulation of the graphical path-loss data provided by the Okumura and is valid over roughly the same range of frequencies.www.82 log10 (ht )−a(hr )+(44.69) (4. d) + Amu (fc . d) and GAREA are obtained from Okumura’s empirical plots. d) − G(ht ) − G(hr ) − GAREA (4.55 log10 (ht )) log10 (d) (4.com
Okumura used extensive measurements of base station-to-mobile signal attenuation throughout Tokyo to develop a set of curves giving median attenuation relative to free space (Amu ) of signal propogation in irregular terrain.jntuworld.com
. Amu (fc .55+26. The empirical pathloss formula of Okumura at distance d parameterized by the carrier frequency fc is given by PL (d)dB = L(fc . d) is the median attenuation in addition to free-space path loss across all environments.G(ht ) is the base station antenna height gain factor.GAREA is the gain due to type of environment.70)
G(hr ) = 10 log10 (hr /3).68) (4.16 log10 (fc )−13. 30m < ht < 1000m hr ≤ 3m 3m < hr < 10m (4. Okumura’s model has a 10-14 dB empirical standard deviation between the path loss predicted by the model and the path loss associated with one of the measurements used to develop the model.7.
4. 150-1500 MHz. Okumura derived empirical formulas for G(ht ) and G(hr ) as follows: G(ht ) = 20 log10 (ht /200). d) is free space path loss at distance d and carrier frequency fc . This empirical formula simpliﬁes the calculation of path loss because it is closed form formula and it is not based on empirical curves for the diﬀerent parameters.67)
where L(fc .9−6. The standard formula for empirical path loss in urban areas under the Hata model is PL.
Correlation factors related to terrain are also developed in order to improve the models accuracy.

www.suburban (d)dB = PL. and is given by PL.4 (4.urban (d)dB − 2(log10 (fc /28))2 − 5. The Hata model well approximates the Okumura model for distances d > 1 Km.and the variability of the environment is much greater for smaller range of Tx-Rx separation distances. but it does not model propogation well in current cellular systems with smaller cell sizes and higher frequencies.jntuworld.1dB Corrections to the urban model are made for the suburban.97dB else it is a(hr ) = 8.
4.56 log10 (fc ) − 0.com
. Hence it is a good model for ﬁrst generation cellular systems.8
Indoor Propagation Models
The indoor radio channel diﬀers from the traditional mobile radio channel in ways .7)hr − (1.72)
Unlike the Okumura model.the Hata model does not provide for any speciﬁc pathcorrelation factors. Indoor environments are also not captured by the Hata model.8.Partitions that are formed as a part of building structure are called 72
www.com
The parameters in this model are same as in the Okumura model.2(log10 (11.Features such as lay-out of the building.and the building type strongly inﬂuence the propagation within the building.11 log10 (fc ) − 0.54hr ))2 − 1.75hr ))2 − 4. diﬀraction and scattering with variable conditions.the distances covered are much smaller .the construction materials. In general.Indoor radio propagation is dominated by the same mechanisms as outdoor: reﬂection.indoor channels may be classiﬁed as either line-of-sight or obstructed.
4.1
Partition Losses Inside a Floor (Intra-ﬂoor)
The internal and external structure of a building formed by partitions and obstacles vary widely.jntuworld.29(log10 (1.For small to medium sized cities this factor is given by a(hr ) = (1.and a(hr ) is a correction factor for the mobile antenna height based on the size of coverage area.8)dB and for larger cities at a frequencies fc > 300 MHz by a(hr ) = 3.

may be insigniﬁcant when we consider the small-scale rapid path losses.3
Log-distance Path Loss Model
It has been observed that indoor path loss obeys the distance power law given by P L(dB) = P L(d0 ) + 10n log10 (d/d0 ) + Xσ (4. which are common terrestrial models and these mainly explains the large scale path loss.making it diﬃcult to apply general models to speciﬁc indoor installations.jntuworld.www.9
Summary
In this chapter. Inc. Moher.
4. Partitions vary widely in their physical and electrical characteristics.as well as the type of construction used to create the ﬂoors and the external surroundings.com
hard partitions . These.2
Partition Losses Between Floors (Inter-ﬂoor)
The losses between ﬂoors of a building are determined by the external dimensions and materials of the building. S.10
References
1. Singapore: Pearson Education. Even the number of windows in a building and the presence of tinting can impact the loss between ﬂoors. 73
www. 2002. and Xσ represents a normal random variable in dB having standard deviation of σ dB. S. Inc. Wireless Communications: Principles and Practice. one important factor introduced in this chapter is log-distance path loss model. and partitions that may be moved and which do not span to the ceiling are called soft partitions. 2002.8.
4.8. three principal propagation models have been identiﬁed: free-space propagation. Singapore: Pearson Education..com
.. 2.jntuworld. T. however. reﬂection and diﬀraction. Modern Wireless Communications.
4. This is discussed in the next chapter.
4. Rappaport. Regarding path-loss. Haykin and M.73)
where n depends on the building and surrounding type. 2nd ed.

Zhuang.
74
www.com
. Wireless Communications and Networking. Mark and W.www. J. New Delhi: PHI. W.com
3.jntuworld.jntuworld. 2005.

Adding the eﬀect of movement of either Tx or Rx or the surrounding clutter to it. the received overall signal amplitude or phase changes over a small amount of time.2
Multipath & Small-Scale Fading
Multipath signals are received in a terrestrial environment. Therefore there would be multipath interference.com
.com
Chapter 5
Multipath Wave Propagation and Fading
5.. Causes of multipath include atmospheric ducting. i.jntuworld.1 Multipath Propagation
In wireless telecommunications.e. In digital radio communications (such as GSM) multipath can cause errors and aﬀect the quality of communications.www.
75
www. ionospheric reﬂection and refraction. causing multipath fading. The eﬀects of multipath include constructive and destructive interference. where diﬀerent forms of propagation are present and the signals arrive at the receiver from transmitter via a variety of paths. We discuss all the related issues in this chapter. and phase shifting of the signal.jntuworld. Mainly this causes the fading. multipath is the propagation phenomenon that results in radio signals reaching the receiving antenna by two or more paths. and reﬂection from water bodies and terrestrial objects such as mountains and buildings.
5.

then this eﬀect dominates fading. or multipath delays of a radio signal over a short period or short travel distance. (2) 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. (3) Speed of surrounding objects – If objects in the radio channel are in motion. Time dispersion or echoes caused by multipath propagation delays.1
Fading
The term fading.2. 2. This might be so severe that large scale radio propagation loss eﬀects might be ignored. and phase shifting of the signal.com
5.2.
5. they induce a time varying Doppler shift on multipath components.jntuworld.3
Factors Inﬂuencing Fading
The following physical factors inﬂuence small-scale fading in the radio propagation channel: (1) Multipath propagation – Multipath is the propagation phenomenon that results in radio signals reaching the receiving antenna by two or more paths.com
. phases. If the surrounding objects move at a greater rate than the mobile. the following are the main multipath eﬀects: 1. small-scale fading. the received signal will be distorted. 3. The eﬀects of multipath include constructive and destructive interference. means rapid ﬂuctuations of the amplitudes.
5.2
Multipath Fading Eﬀects
In principle.2. 76
www. (4) Transmission Bandwidth of the signal – If the transmitted radio signal bandwidth is greater than the “bandwidth” of the multipath channel (quantiﬁed by coherence bandwidth).www. or. Random frequency modulation due to varying Doppler shifts on diﬀerent multipath signals. Rapid changes in signal strength over a small travel distance or time interval.jntuworld.

jntuworld.3. BS and TS στ (5. And in such a case. Frequency Selective Fading Frequency selective fading occurs when the signal bandwidth is more than the coherence bandwidth of the mobile radio channel or equivalently the symbols duration of the signal is less than the rms delay spread. then the fading is ﬂat fading.com
. Equivalently if the symbol period of the signal is more than the rms delay spread of the channel. There are two types of fading due to the time dispersive nature of the channel. Also TS στ (5.
5.www.jntuworld.2)
where TS is the symbol period and στ is the rms delay spread. symbol period) and the channel parameters (rms delay spread and Doppler spread).
So we can say that ﬂat fading occurs when BS BC (5.4) BC (5. Hence we have four diﬀerent types of fading. mobile channel has a constant gain and linear phase response over its bandwidth.3)
77
www.com
5.1
Fading Eﬀects due to Multipath Time Delay Spread
Flat Fading Such types of fading occurs when the bandwidth of the transmitted signal is less than the coherence bandwidth of the channel.1)
where BS is the signal bandwidth and BC is the coherence bandwidth.3
Types of Small-Scale Fading
The type of fading experienced by the signal through a mobile channel depends on the relation between the signal parameters (bandwidth.

www.9) TC (5. Therefore a signal undergoes fast fading if TS where TC is the coherence time and BS BD (5.
78
www.6)
where BD is the Doppler spread. Slow Fading In such a channel. A rule of thumb for a channel to have ﬂat fading is if στ ≤ 0.7) TC (5. signal undergoes frequency dispersion leading to distortion.3. Due to Doppler spreading. the rate of the change of the channel impulse response is much less than the transmitted signal.1 TS (5. Transmission involving very low data rates suﬀer from fast fading.2
Fading Eﬀects due to Doppler Spread
Fast Fading In a fast fading channel. We can consider a slow faded channel a channel in which channel is almost constant over atleast one symbol duration. Hence TS and BS BD (5.8)
We observe that the velocity of the user plays an important role in deciding whether the signal experiences fast or slow fading. all attenuated and delayed in time.com
.5)
5.jntuworld.com
At the receiver. the channel impulse response changes rapidly within the symbol duration of the signal.jntuworld. we obtain multiple copies of the transmitted signal. The channel introduces inter symbol interference.

As the BS is located at an elevated place.3
Doppler Shift
The Doppler eﬀect (or Doppler shift) is the change in frequency of a wave for an observer moving relative to the source of the wave.jntuworld. The expected Doppler shift of the EM wave then comes out to be ± vcr fo or. where ∆t is the time required for the mobile to travel from A to B. is illustrated below. while it receives signals from a remote BS source S.www. along a path segment length d between points A and B. In mobile communication. The exact scenario. In classical physics (waves in a medium). the relationship between the observed frequency f and the emitted frequency fo is given by: f= v ± vr v ± vs f0 (5. vs is the velocity of the source relative to the medium and vr is the velocity of the receiver relative to the medium.1: Illustration of Doppler eﬀect.10)
where v is the velocity of waves in the medium.1. ± vr . Consider a mobile moving at a constant velocity v. The diﬀerence in path lengths traveled by the wave from source S to the mobile at points A and B is ∆l = d cos θ = v∆t cos θ. and θ is assumed to be the same at points A and B since the
79
www.
5.jntuworld.com
Figure 5. a cos φ factor would λ also be multiplied with this.3. the above equation can be slightly changed according to our convenience since the source (BS) is ﬁxed and located at a remote elevated level from ground.com
. as given in Figure 5.

t) =
−∞
x(τ ) h(d. at an elevation of 20◦ .www. Find out the expected Doppler shift. t)|
For a causal system: h(d. t − τ ) dτ
(5. t) and x(t).11)
and hence the apparent change in frequency. t) = x(t) ∗ h(d.
5. consider time variation due to receiver motion and time varying impulse response h(d. v = 500 kmph the horizontal component of the velocity is v = v cos θ = 500 × cos 20◦ = 130 m/s Hence. Solution As given here.4
Impulse Response Model of a Multipath Channel
Mobile radio channel may be modeled as a linear ﬁlter with time varying impulse response in continuous time. or Doppler shift (fd ) is fd = Example 1 An aircraft is heading towards a control tower with 500 kmph. The phase change in the received signal due to the diﬀerence in path lengths is therefore ∆ϕ = 2π∆l 2πv∆t = cos θ λ λ (5.13)
∞ −∞ |h(d. = .3. the Doppler shift change will be 390 Hz to −390 Hz.com
source is assumed to be very far away. the transmitted signal.jntuworld.com
. The received signal y(d.jntuworld. it can be written that λ= 900 × 106 1 = m 8 3 × 10 3 130 = 390 Hz 1/ 3 v 1 ∆ϕ . t) at any position d would be
∞
y(d. 2π ∆t λ (5. for t < 0 and for a stable system ∞ 80
dt <
www. Communication between aircraft and control tower occurs at 900 MHz.12)
fd =
If the plane banks suddenly and heads for other direction. cos θ. To show this. t) = 0.

t − τ ) dτ = x(t) ∗ h(vt.e. t)
(5.N − 1}.
Since v is a constant. each bin having a time delay width equal to ( τi+1 − τi ) = ∆τ and τi = i∆τ for i ∈ {0. 1. {y(t) = x(t) ∗ h(t. t) =
−∞
x(τ ) h(vt. t) = x(t) ∗ h(d. t − τ ) dτ.com
Applying causality condition in the above equation.16) (5. y(vt.e. . The model may be used to analyze transmitted RF signals having bandwidth less than 2/ .www.15)
81
www. t) is just a function of t. τ )ejωc t → y(t) = Re{r(t)ejωc t } Baseband equivalent channel impulse response model is given by 1 1 c(t) → hb (t. the integral limits are changed to
t
y(d.N − 1} Bandpass channel impulse response model is x(t) → h(t... ∆τ If there are N multipaths. τ ) 2 2 Average power is 1 x2 (t) = |c(t)|2 2 (5.17) (5.jntuworld. h(d. the position of the receiver is d = vt. τ ) = Re{hb (t. τ ) → r(t) = c(t) ∗ hb (t. including the ﬁrst arriving component. 2..
Since the receiver moves along the ground at a constant velocity v.. 1.
t
y(vt. .com
.jntuworld.18) (5. t) =
−∞
x(τ ) h(d. i. where N represents the total number of possible equally-spaced multipath components. τi )|i = 0. maximum excess delay is given by N ∆τ . The useful frequency span of the model is 2/∆τ . i. Therefore the above equation can be expressed as
t
y(t) =
−∞
x(τ ) h(vt.14)
It is useful to discretize the multipath delay axis τ of the impulse response into equal time delay segments called excess delay bins. t − τ ) = 0 for t − τ < 0 ⇒ τ > t. t − τ ) dτ..

The low pass channel output
1 N −1 ai exp[jθi ]p(t − τi ) 2 i=0
N −1
ai exp[jθi ]. transmitted RF signal x(t) = Re{p(t)ej2πfc t } where p(t) = is r(t) = =
i=0 4τmax Tbb
(5. τ ) exp[j(2πfc τi (t) + ϕi (t. p(t) ≈ δ(t − τ ) (5.com
.com
The baseband impulse response of a multipath channel can be expressed as
N −1
hb (t.19)
where ai (t.23)
for 0 ≤ t ≤ Tbb and 0 elsewhere. Let us consider two extreme channel sounding cases.20)
For measuring hb (τ ).
τmax Tb rect(t − − τi ).e.jntuworld. Tbb 2
82
www.5
Relation Between Bandwidth and Received Power
In actual wireless communications. τ )|2 . The received power delay proﬁle in a local area is given by p(τ ) ≈ k|hb (t. we use a probing pulse to approximate δ(t) i.jntuworld. The phase term 2πfc τi (t) + ϕi (t. respectively. τ ) =
i=0
ai (t. τ ))]δ(τ − τi (t))
(5. impulse response of a multipath channel is measured using channel sounding techniques. then
N −1
hb (τ ) =
i=0
ai exp[jθi ]δ(τ − τi )
(5. Consider a pulsed. τ )|2 over a local area. plus any additional phase shifts which are encountered in the channel. If the channel impulse response is wide sense stationary over a small-scale time or distance interval.3.21)
Power delay proﬁle is taken by spatial average of |hb (t. τ ) and τi (t) are the real amplitudes and excess delays. of the ith multipath component at time t.www.22)
5. (5. τ ) in the above equation represents the phase shift due to free space propagation of the ith multipath component..

then average small-scale receiver power is simply sum of average powers received from each multipath components.
N −1 N −1
Ea. let the envelope be c(t) = 2. transmitted into the same channel and for simplicity.www.com
. consider a CW signal. The received power at any time t0 is |r(t0 )|
2
=
1 τmax 1 τmax 1 τmax
N −1
τmax
r(t)r∗ (t)dt
0 τmax
=
1 4
N −1
a2 (t0 )p2 (t − τk ) dt k
k=0 τmax
0 N −1
=
a2 (t0 ) k
k=0 0
τmax Tb rect(t − − τi ) Tbb 2
2
dt
=
k=0
a2 (t0 ).jntuworld.jntuworld.θ [PW B ] = Ea. τ )]
(5.θ [
i=0
|ai exp(jθi )|2 ] ≈
i=0
a2 i
(5.2: A generic transmitted pulsed RF signal. k
Interpretation: If the transmitted signal is able to resolve the multipaths.25)
83
www.24)
Now instead of a pulse. Then
N −1
r(t) =
i=0
ai exp[jθi (t.com
Figure 5.

θ [
i=0 N −1
|ai exp(jθi )|2 ]
N −1 N
≈
i=0
a2 + 2 i
i=0 i. rij = cos(θi − θj ) = 0. the received signal is
∞
αn (t)e−j2πfc τn (t) . t)x(t − τ ) dτ = Z(t)x(t − τi ) 84
(5. t) = Z(t)δ(τ − τi ) where Z(t) = this case.d over [0. h(τ.www. In
r(t) =
−∞
h(τ.i. t) = F T [h(τ. t)ej2πf τ df
(5. ai varies little but θi varies greatly resulting in large ﬂuctuations.6
Linear Time Varying Channels (LTV)
The time variant transfer function(TF) of an LTV channel is FT of h(t. t)] =
−∞
h(τ.
N −1
Ea. t)] = The received signal
∞
∞ −∞
r(t) =
−∞
R(f.θ [PCW ] = Ea.
∞
H(f.27) (5.30)
www. For ﬂat fading channel. This occurs if multipath components are uncorrelated or if multipath phases are i.t.com
.r.jntuworld. t)e−j2πf τ dτ H(f. τ .jntuworld. τ ) w.28)
h(τ.29)
where R(f. t) = H(f. large fading occurs due to phase shift of unresolved paths. t)X(f ).j=i
rij cos(θi − θj )
where rij = Ea [ai aj ].com
and the instantaneous power is
N −1
|r(t)|2 = |
i=0
ai exp[jθi (t. If the signal bandwidth is greater than multipath channel bandwidth then fading eﬀects are negligible 2.3.
5. then Ea.θ [PW B ].
If. Bottomline: 1.26)
Over local areas. τ )]|2
(5.θ [PCW ] = Ea. t) = F T −1 [H(f. t)ej2πf t df
(5. 2π]. If the signal bandwidth is less than the multipath channel bandwidth.

Immediate measurements of the square of the channel impulse response convolved with the probing pulse can be taken. 85
www. t)e−j2πνt dt
(5. where the channel becomes multiplicative. envelope detected.32)
H(τ. t)] =
−∞
H(f. ν)] = Delay Doppler spread:
∞ ∞ −∞
H(f.com
Figure 5.33)
5.jntuworld.7
Small-Scale Multipath Measurements
Direct RF Pulse System A wideband pulsed bistatic radar usually transmits a repetitive pulse of width Tbb s. ν) = F T [h(τ. Doppler spread functions:
∞
H(f. t)] =
−∞
h(τ. If the oscilloscope is set on averaging mode.www.31)
and H(f. The signal is then
ampliﬁed. t)e−j2πνt dt
(5. ν) = F T [H(f. then this system provides a local average power delay proﬁle.com
.jntuworld. t) = F T −1 [H(f. and displayed and stored on a high speed oscilloscope.3. and uses a receiver with a wide bandpass ﬁlter (BW =
2 Tbb
Hz). ν)ej2πνt dν
(5.3: Relationship among diﬀerent channel functions.

The sweeper scans a particular frequency band.jntuworld. the S-parameter test set transmits a known signal level at port 1 and monitors the received signal at port 2. These signals allow the analyzer to measure the complex response.com
. If the ﬁrst arriving signal is blocked or fades. A vector network analyzer controls a swept frequency synthesizer. severe fading occurs. S21 (ω).4: Direct RF pulsed channel IR measurement.com
Figure 5.jntuworld.www. centered on the carrier. An Sparameter test set is used to monitor the frequency response of the channel. Frequency Domain Channel Sounding In this case we measure the channel in the frequency domain and then convert it into time domain impulse response by taking its inverse discrete Fourier transform (IDFT). The number and spacing of the frequency step impacts the time resolution of the impulse response measurement. The S21 (ω) measure is the measure of the signal ﬂow from transmitter antenna to receiver
86
www. of the channel over the measured frequency range. and it is possible the system may not trigger properly. For each frequency step. This system is subject to interference noise. by stepping through discrete frequencies.

4.1
Time Dispersion Parameters
These parameters include the mean excess delay. This system is also non real-time.jntuworld. They all can be determined from the power delay proﬁle.5: Frequency domain channel IR measurement.
5.
5. we deﬁne some parameters.com
. it is not suitable for time-varying channels unless the sweep times are fast enough.e. antenna (i. This system is suitable only for indoor channel measurements. These parameters can be broadly divided in to two types.com
Figure 5.. Hence. the channel).rms delay spread and excess delay spread.www.4
Multipath Channel Parameters
To compare the diﬀerent multipath channels and to quantify them.jntuworld. The mean excess delay is the ﬁrst moment of the power delay proﬁle and is
87
www.

P (0) = 1 watt. If BPSK modulation is used. it can be written as στ = τ¯2 − (¯)2 τ (5.5µs 88
www. Example 2 1. P (1) = 1 watt (1)(0) + (1)(1) = 0.34)
where ak is the amplitude. no equalizer is required at the receiver.5µs 1+1
τ=
τ 2 = 0.35)
Since the rms delay spread is the square root of the second central moment of the power delay proﬁle.5µs2 στ = 0. Sketch the power delay proﬁle and compute RMS delay spread for the following:
1
P (τ ) =
n=0
δ(τ − n × 10−6 ) (in watts)
2.com
.jntuworld.36)
As a rule of thumb. τk is the excess delay and P (τk ) is the power of the individual multipath signals. what is the maximum bit rate that can be sent through the channel without needing an equalizer? Solution
1. for a channel to be ﬂat fading the following condition must be satisﬁed στ ≤ 0. For this case.com
deﬁned as a2 τk k = a2 k P (τk )τk P (τk )
τ= ¯
(5.www.37)
where TS is the symbol duration.1 TS (5.jntuworld. The mean square excess delay spread is deﬁned as τ¯2 =
2 P (τk )τk P (τk )
(5.

. considering omni-directional unity gain antennas λ PT 1 )( )2 4π PRmin
dmax = ( τmax =
dmax λ PT 1 =( )( )2 c 4πc PRmin 1 PT 1 )( )2 4πf PRmin
τmax = (
5. For ﬂat fading channel.
Here dmin = d0 and dmax = di + dr Transmitted power = PT .2 × 104 = 200 kbps
For BPSK we need Rb = Rs = 200 kbps Example 3 A simple delay spread bound: Feher’s upper bound Consider a simple worst-case delay spread scenario as shown in ﬁgure below.e. we have the following parameters.jntuworld.jntuworld.com
2.4.2
Frequency Dispersion Parameters
To characterize the channel in the frequency domain. Minimum received power = PRmin = PT hreshold PRmin λ = GT G R ( )2 PT 4πdmax Put GT = GR = 1 i. we need
στ Ts 0.com
.www.1
⇒ Rs =
1 Ts
= 0.
89
www.

com
(1) Coherence bandwidth: it is a statistical measure of the range of frequencies over which the channel can be considered to pass all the frequency components with almost equal gain and linear phase.38)
However. Practically. if the coherence bandwidth is considered to be the bandwidth over which the frequency correlation function is above 0. This variation may be due to the relative motion between the mobile and the base station or the motion of the objects in the channel.39)
The coherence bandwidth describes the time dispersive nature of the channel in the local area. if the coherence time is considered to be the bandwidth over which the time correlation is above 0. When channel behaves like this. then it is approximated as BC ≈ 1 .5
Statistical models for multipath propagation
Many multipath models have been proposed to explain the observed statistical nature of a practical mobile channel.9.
Another parameter is the Doppler spread (BD ) which is the range of frequencies over which the received Doppler spectrum is non zero. If the coherence bandwidth is considered to be the bandwidth over which the frequency correlation function is above 0.
5.jntuworld.com
.40)
ν where fm is the maximum doppler spread given be fm = λ . coherence bandwidth is the minimum separation over which the two frequency components are aﬀected diﬀerently.www.jntuworld. (2) Coherence time: this is a statistical measure of the time duration over which the channel impulse response is almost invariant. Both the ﬁrst order and second order statistics
90
www. 5στ (5. then it is deﬁned as BC ≈ 1 . When this condition is satisﬁed then we say the channel to be ﬂat. Essentially it is the minimum time duration over which two received signals are aﬀected diﬀerently. it is said to be slow faded.5. A more convenient parameter to study the time variation of the channel is the coherence time.5. For an example. 50στ (5. then it can be approximated as TC ≈ 9 16πfm (5.

The Rayleigh model is used to model the statistical time varying nature of the received envelope of a ﬂat fading envelope. resulting in Rayleigh fading.43)
= Zr + jZi = Rejφ
91
www. then by Central Limit Theorem we have.So we have
N
˜ E=
n=1
En ejθn
(5. Let En be the electric ﬁeld and Θn be the relative phase of the various multipath signals.6: Two ray NLoS multipath.com
.jntuworld. Below. The most popular of these models are Rayleigh model.1
NLoS Propagation: Rayleigh Fading Model
Let there be two multipath signals S1 and S2 received at two diﬀerent time instants due to the presence of obstacles as shown in Figure 5. have been examined in order to ﬁnd out the eﬀective way to model and combat the channel eﬀects.jntuworld.
5.e. are suﬃciently large number of multipaths) and all the En are IID distributed.com
Figure 5. we discuss about the main ﬁrst order and second order statistical models.
N N →∞
˜ lim E = lim
N →∞
En ejθn
n=1
(5.41)
Now if N→ ∞(i.6.42) (5. Now there can either be constructive or destructive interference between the two signals. which describes the NLoS propagation.www.5.

www.jntuworld.com

**where Zr and Zi are Gaussian Random variables. For the above case R= and φ = tan−1 Zi Zr (5.45)
**

2 Zr + Zi2

(5.44)

**For all practical purposes we assume that the relative phase Θn is uniformaly distributed. E[e
**

jθn

1 ]= 2π

2π

ejθ dθ = 0

0

(5.46)

**It can be seen that En and Θn are independent. So, ˜ E[E] = E[ ˜ E[ E ] = E[
**

2

En ejθn ] = 0

N

(5.47)

2 En = P0 n=1

En ejθn

∗ En e−jθn ] = E[ m n

En Em ej(θn −θm ) ] =

**(5.48) where P0 is the total power obtained. To ﬁnd the Cumulative Distribution Function(CDF) of R, we proceed as follows. FR (r) = Pr (R ≤ r) =
**

A

fZi ,Zr (zi , zr )dzi dzr

(5.49)

where A is determined by the values taken by the dummy variable r. Let Zi and Zr be zero mean Gaussian RVs. Hence the CDF can be written as FR (r) =

A

√

−(Zr +Z ) i 1 e 2σ2 dZi dZr 2 2πσ

2

2

(5.50)

**Let Zr = p cos(Θ) and Zi = p sin(Θ) So we have
**

2π 2π

FR (r) =

0 0

√

−p2 1 e 2σ2 pdpdθ 2πσ 2

(5.51)

= 1 − e 2σ2

−r 2

(5.52)

Above equation is valid for all r ≥ 0. The pdf can be written as fR (r) = r − r22 e 2σ σ2 (5.53)

and is shown in Figure 5.7 with diﬀerent σ values. This equation too is valid for all r ≥ 0. Above distribution is known as Rayleigh distribution and it has been derived 92

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Figure 5.7: Rayleigh probability density function. for slow fading. However, if fD fading. We observe the following: E[R] = π σ 2 (5.54) (5.55) (5.56) (5.57) 1 Hz, we call it as Quasi-stationary Rayleigh

E[R2 ] = 2σ 2 var[R] = (2 − π 2 )σ 2

median[R] = 1.77σ.

5.5.2

LoS Propagation: Rician Fading Model

Rician Fading is the addition to all the normal multipaths a direct LOS path.

Figure 5.8: Ricean probability density function.

93

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fR (r) =

r −(r2 +A2 ) Ar e 2σ2 I0 ( 2 ) σ2 σ

(5.58)

for all A ≥ 0 and r ≥ 0. Here A is the peak amplitude of the dominant signal and I0 (.) is the modiﬁed Bessel function of the ﬁrst kind and zeroth order. A factor K is deﬁned as KdB = 10 log As A → 0 then KdB → ∞. A2 2σ 2 (5.59)

5.5.3

Generalized Model: Nakagami Distribution

A generalization of the Rayleigh and Rician fading is the Nakagami distribution.

Figure 5.9: Nakagami probability density function. Its pdf is given as, fR (r) = where, Γ(m) is the gamma function Ω is the average signal power and m is the fading factor.It is always greater than or equal to 0.5. 2rm−1 mm −mr2 ( )e Ω Γ(m) Ωm (5.60)

When m=1, Nakagami model is the Rayleigh model. When m= (M + 1)2 2M + 1

94

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Figure 5.10: Schematic representation of level crossing with a Rayleigh fading envelope at 10 Hz Doppler spread.

where M= Nakagami fading is the Rician fading. As m → ∞ Nakagami fading is the impulse channel and no fading occurs. A 2σ

5.5.4

Second Order Statistics

To design better error control codes, we have two important second order parameters of fading model, namely the level crossing rate (LCR) and average fade duration (AFD). These parameters can be utilized to assess the speed of the user by measuring them through the reverse channel. The LCR is the expected rate at which the Rayleigh fading envelope normalized to the local rms amplitude crosses a speciﬁc level ’R’ in a positive going direction.

∞

NR =

0

rp(R, r)dr = ˙ ˙ ˙

√

2πfD ρe−ρ

2

(5.61)

where r is the time derivative of r(t), fD is the maximum Doppler shift and ρ is the ˙ value of the speciﬁed level R, normalized to the local rms amplitude of the fading envelope. The other important parameter, AFD, is the average period time for which the

95

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2
(5.65)
Apart from LCR.
96
www. 2πfD ρ
2
2
(5.jntuworld.62)
p(r)dr = 1 − e−ρ .6. The average rate is expressed as N (rm ) = 2v .com
receiver power is below a speciﬁed level R.com
. sometimes used in the mobile communication. the obtained Rayleigh output is ﬂat faded signal but not frequency selective.6
5. the output of the Clarke’s model is passed through Doppler ﬁlter in the RF or through two initial baseband Doppler ﬁlters for baseband processing as shown in Figure 5.
1 − e−ρ τ=√ ¯ 2πfD ρe−ρ2 e−ρ − 1 =√ . Also the Doppler eﬀect is not accounted for.66)
Another statistical parameter. τ= ¯ As Pr (r ≤ R) =
0
1 Pr (r ≤ R) Nr
R
(5. λ (5. It is deﬁned as the ratio between the minimum value and the mean square value of the faded signal. is called as depth of fading. two independent Gaussian low pass noise sources are used to produce in-phase and quadrature fading branches.
5.63)
therefore. Usually. This is the basic model and is useful for slow fading channel. which is deﬁned as the number of times the signal envelope crosses the middle value (rm ) in a positive going direction per unit time.www.
5. an average value of 10% as depth of fading gives a marginal fading scenario.2
Clarke and Gans’ Model: with Doppler Eﬀect
In this model.jntuworld.11. another parameter is fading rate. Here.6.64) (5.1
Simulation of Rayleigh Fading Models
Clarke’s Model: without Doppler Eﬀect
In it.

and. (b) baseband Doppler ﬁlter.6. very complex and diﬃcult to implement.6. however. we have the two ray Rayleigh fading model which can be easily implemented in software as shown in Figure 5.www.com
. hb (t) = α1 ejφ1 δ(t) + α2 ejφ2 δ(t − τ ) (5. So.4
Two-Ray Rayleigh Faded Model
The above model is.11: Clarke and Gan’s model for Rayleigh fading generation using quadrature amplitude modulation with (a) RF Doppler ﬁlter.13. This is achieved through varying the parameters ai and τi .3
Rayleigh Simulator with Wide Range of Channel Conditions
To get a frequency selective output we have the following simulator through which both the frequency selective and ﬂat faded Rayleigh signal may be obtained.67)
where α1 and α2 are independent Rayleigh distributed and φ1 and φ2 are independent and uniformaly distributed over 0 to 2π. 97
www.
5.12.
5.jntuworld.jntuworld. as given in Figure 5.com
Figure 5. By varying τ it is possible to create a wide range of frequency selective fading eﬀects.

www.12: Rayleigh fading model to get both the ﬂat and frequency selective channel conditions.
5.com
.6
SIRCIM/SMRCIM Indoor/Outdoor Statistical Models
SIRCIM (Simulation of Indoor Radio Channel Impulse-response Model) generates realistic samples of small-scale indoor channel impulse response measurements.com
Figure 5.5
Saleh and Valenzuela Indoor Statistical Model
This method involved averaging the square law detected pulse response while sweeping the frequency of the transmitted pulse.6. Sub98
www.jntuworld. The amplitudes of the received components are independent Rayleigh random variables with variances that decay exponentially with cluster delay as well as excess delay within a cluster.6. The model assumes that the multipath components arrive in clusters.jntuworld.
5. The clusters and multipath components within a cluster form Poisson arrival processes with diﬀerent rates.

Haykin and M.www.
5. W.com
. Zhuang. Mainly the frequency selective fading.13: Two-ray Rayleigh fading model. Wireless Communications and Networking.com
Figure 5. 2002. Mark and W. Inc. Wireless Communications: Principles and Practice.. J.e. 2nd ed. fast fading and deep fading can be considered the major obstruction from the channel severity view point..8
References
1. the main channel impairment. T. a similar program that generates small-scale urban cellular and micro-cellular channel impulse responses. Singapore: Pearson Education. Singapore: Pearson Education.
5. 2005. S.. 3. Inc. New Delhi: PHI. Rappaport. Certain eﬃcient signal processing techniques to mitigate these eﬀects will be discussed in Chapter 7. 2002. Moher. has been introduced which becomes so severe sometimes that even the large scale path loss becomes insigniﬁcant in comparison to it. fading.jntuworld.jntuworld. sequent work by Huang produced SMRCIM (Simulation of Mobile Radio Channel Impulse-response Model). i.7
Conclusion
In this chapter. Modern Wireless Communications.
99
www. Some statistical propagation models have been presented based on the fading characteristics. 2. S.

K. Upper Saddle River. P. 5. Feher. Introduction to Wireless and Mobile Systems. Blake. Wireless Digital Communications: Modulation and Spread Spectrum Applications.com
4. 1995. 6. D. Nelson. Singapore: Thomson Asia Pvt Ltd. Zeng.jntuworld. 2004.www. India: Thomson Learning. Wireless Communications Technology.jntuworld. NJ: Prentice Hall. Delmar. Agarwal and Q-A. 2007.
100
www.com
. R.

with maximum utilization of the channel resources. To counteract these eﬀects is one of the requirements while designing a transmitter and receiver end technique.2
Modulation
Modulation is a process of encoding information from a message source in a manner suitable for transmission. The information transmitted by the source is received by the destination via a physical medium called a channel.jntuworld. Modulation can be done by varying certain characteristics of carrier waves according to the message signal.
6. noise and interference in the transmitted information bearing signal.www. introduces distortion. which may be wired or wireless.jntuworld. The other requirements are power and bandwidth eﬃciency at a low implementation complexity. The baseband signal is called the modulating signal and the passband signal is called the modulated signal.com
Chapter 6
Transmitter and Receiver Techniques
6. It involves translating a baseband message signal to a passband signal. This physical medium. Demodulation is the reciprocal process of modulation which involves extraction of original baseband signal from the modulated passband signal.
101
www.1 Introduction
Electrical communication transmitter and receiver techniques strive toward obtaining reliable communication at a low cost.com
.

2. it is necessary to increase the signal power. Power eﬃciency describes the ability of a modulation technique to preserve the ﬁdelity of the digital message at low power levels.2. occupies a minimum of bandwidth. A desirable modulation scheme provides low bit error rates at low received signal to noise ratios.jntuworld. Reduction in antenna size: The antenna height and aperture is inversely proportional to the radiated signal frequency and hence high frequency signal radiation result in smaller antenna size.com
. The performance of a modulation scheme is often measured in terms of its power eﬃciency and bandwidth eﬃciency. 3. In a digital communication system. The system capacity of a digital mobile communication system is directly related to the bandwidth eﬃciency of the modulation scheme. and is easy and cost-eﬀective to implement.1
Choice of Modulation Scheme
Several factors inﬂuence the choice of a digital modulation scheme. in order to increase noise immunity. Increases the range of communication: Low frequency baseband signals suﬀer from attenuation and hence cannot be transmitted over long distances.jntuworld.www.2
Advantages of Modulation
1. and is given by the channel capacity formula ηBmax = C S = log2 (1 + ) B N (6.
102
www.com
6. the maximum possible bandwidth eﬃciency is limited by the noise in the channel. Shannon’s channel coding theorem states that for an arbitrarily small probability of error. multiple users can be accommodated within a band of the electromagnetic spectrum. performs well in multipath and fading conditions.2. since a modulation with a greater value of ηb (=
R B)
will transmit more data in a given spectrum allocation. Bandwidth eﬃciency describes the ability of a modulation scheme to accommodate data within a limited bandwidth.1)
6. So translation to a higher frequency band results in long distance transmission. Facilitates multiple access: By translating the baseband spectrum of signals from various users to diﬀerent frequency bands.
There is a fundamental upper bound on achievable bandwidth eﬃciency.

4) (6. For example an amplitude modulated wave consists of the addition two terms: the message signal multiplied with the carrier and the carrier itself. From the principle of homogeneity: Let us scale the input by a factor a.2. From the principle of superposition: Let m(t) = m1 (t) + m2 (t) be applied simultaneously at the input of the modulator. If m(t) is the message signal and sAM (t) is the modulated signal given by: sAM (t) = Ac [1 + km(t)] cos(2πfc t) Then. If this input-output relation satisﬁes the principle of homogeneity and superposition then the modulation technique is said to be linear.2)
www. So m(t) = am1 (t) and the corresponding output becomes : sAM 1 (t) = Ac [1 + am1 (t)] cos(2πfc t) = asAM 1 (t) 2.jntuworld. 1. The principle of homogeneity states that if the input signal to a system (in our case the system is a modulator) is scaled by a factor then the output must be scaled by the same factor. Hence AM is a nonlinear technique but DSBSC modulation is a linear technique since it satisﬁes both the above mentioned principles. sAM 1 (t) and sAM 2 (t) are the outputs obtained when m1 (t) and m2 (t) are applied one at a time.com
6.3) (6. The resulting output is: sAM (t) = Ac [1 + m1 (t) + m2 (t)] cos(2πfc t) = sAM 1 (t) + sAM 2 (t) = Ac [2 + m1 (t) + m2 (t)] cos(2πfc t) Here. 103 (6.com
.3
Linear and Non-linear Modulation Techniques
The mathematical relation between the message signal (applied at the modulator input) and the modulated signal (obtained at the modulator output) decides whether a modulation technique can be classiﬁed as linear or non-linear.www. The principle of superposition states that the output of a linear system due to many simultaneously applied input signals is equal to the summation of outputs obtained when each input is applied one at a time.jntuworld.

s(t) = s1 φ1 (t) + s2 φ2 (t) where φ1 (t) and φ2 (t) are given by. having a symbol duration T. the resulting amplitude or angle modulation technique is called analog modulation.6) (6. a modulation scheme can be classiﬁed as an amplitude or angle modulation.5)
www. constituting the two orthogonal axis in this two dimensional signal space and is expressed mathematically as.2.
6.5
Analog and Digital Modulation Techniques
The nature of the information generating source classiﬁes a modulation technique as an analog or digital modulation technique. These orthonormal signals are independent of each other and form a basis set of the signal space.
6.4
Amplitude and Angle Modulation
Depending on the parameter of the carrier (amplitude or angle) that is changed in accordance with the message signal. is expressed as a linear combination of two orthonormal signals φ1 (t) and φ2 (t).jntuworld.3
Signal Space Representation of Digitally Modulated Signals
Any arbitrary signal can be expressed as the linear combination of a set of orthogonal signals or equivalently as a point in an M dimensional signal space. When analog messages generated from a source passe through a modulator.jntuworld. φ1 (t) = 2 cos(2πfc t) T 104 (6.www. When digital messages undergo modulation the resulting modulation technique is called digital modulation. where M denotes the cardinality of the set of orthogonal signals.2. Generally a digitally modulated signal s(t).com
6.com
. Angle modulation varies a sinusoidal carrier signal in such a way that the angle of the carrier is varied according to the amplitude of the modulating baseband signal. These orthogonal signals are normalized with respect to their energy content to yield an orthonormal signal set having unit energy. Amplitude modulation involves variation of amplitude of the carrier wave with changes in the message signal.

10) (6.11) (6.13) (6.
6. When sI (t) and sQ (t) are incorporated in the formation of the following complex signal. ˜ Analogously.8)
Hence sI (t) and sQ (t) are known as the in-phase and quadrature-phase components respectively. (6.7)
The coeﬃcients s1 and s2 form the coordinates of the signal s(t) in the two dimensional signal space.www.com
.com
φ2 (t) =
2 cos(2πfc t) T
(6.9)
When s(t) passes through h(t). band-pass systems characterized by an impulse response h(t) can be expressed in terms of its in-phase and quadrature-phase components as: h(t) = hI (t)cos(2πfc t) − hQ (t)sin(2πfc t) The complex baseband model for the impulse response therefore becomes. s(t) = sI (t) + sQ (t) ˜ then s(t) can be expressed in a more compact form as: s(t) = Re{˜(t)e(j2πfc t) } s where s(t) is called the complex envelope of s(t).jntuworld.4
Complex Representation of Linear Modulated Signals and Band Pass Systems
A band-pass signal s(t) can be resolved in terms of two sinusoids in phase quadrature as follows: s(t) = sI (t)cos(2πfc t) − sQ (t)sin(2πfc t) (6. ˜ h(t) = hI (t) + hQ (t) h(t) can therefore be expressed in terms of its complex envelope as ˜ h(t) = Re{h(t)ej2πfc t }. the output r(t) of the bandpass system is given by the following convolution ˜ 1 ˜ r(t) = s(t) ⊗ h(t) ˜ ˜ 2 105 (6.14)
www.12) (6.jntuworld. then in the complex baseband domain.

(6. Thus DSBSC is a linear modulation technique. cos(2πfc t).5. The transmitted signal DSBSC signal s(t) can be expressed as: s(t) = Am(t)exp(j2πfc t). varies linearly with the modulating digital signal. Ac (6.17)
If m(t) is scaled by a factor of a. the modulation index is given by k= Am . Linear modulation techniques are bandwidth eﬃcient and hence are very attractive for use in wireless communication systems where there is an increasing demand to accommodate more and more users within a limited spectrum.18)
www. Coherent demodulation requires knowledge 106 (6. The resulting modulated carrier or AM signal can be represented as: sAM (t) = Ac [1 + km(t)] cos(2πfc t).com
6. is varied in accordance to the instantaneous amplitude of the modulating message signal m(t). s12 (t) = A[m1 (t) + m2 (t)]cos(2πfc t) = Am1 (t)cos(2πfc t) + Am2 (t)cos(2πfc t) = s1 (t) + s2 (t) where A is the carrier amplitude and fc is the carrier frequency.1
Linear Modulation Techniques
Amplitude Modulation (DSBSC)
Generally. AM demodulation techniques may be broadly divided into two categories: coherent and non-coherent demodulation. In DSBSC the amplitude of the transmitted signal.com
. m(t). Hence the principle of superposition is also satisﬁed. in amplitude modulation. s(t).16)
This is a nonlinear technique and can be made linear by multiplying the carrier with the message signal.5
6.www.jntuworld. Hence the principle of homogeneity is satisﬁed. (6.15)
The modulation index k of an AM signal is deﬁned as the ratio of the peak message signal amplitude to the peak carrier amplitude.The resulting modulation scheme is known as DSBSC modulation. For a sinusoidal modulating signal m(t) =
Am Ac
cos(2πfm t). is also scaled by the same factor as seen from the above equation.jntuworld. then s(t). the amplitude of a high frequency carrier signal. the output of the modulator. Moreover.

19)
A typical BPSK signal constellation diagram is shown in Figure 6.com
Figure 6. it is easier to see it as two independently modulated quadrature carriers.
6. Normally. N0 (6.1. and energy per bit Eo = 1 A2 Tb then the transmitted BP SK signal is 2 c sBP SK (t) = m(t) 2Eb cos(2πfc t + θc ). The probability of bit error for many modulation schemes in an AW GN channel is found using the Q-function of the distance between the signal points.20)
6. In case of BP SK. With this interpretation. The phase of the carrier in the QPSK takes 1 of 4 equally spaced shifts. PeBP SK = Q( 2Eb ).2
BPSK
In binary phase shift keying (BPSK).1: BPSK signal constellation.com
.5. If the sinusoidal carrier has an amplitude A. the phase of a constant amplitude carrier signal is switched between two values according to the two possible signals m1 and m2 corresponding to binary 1 and 0. the even (or odd) bits are 107
www.5. the two phases are separated by 180o . of the transmitted carrier frequency and phase at the receiver.jntuworld.www. respectively. Tb (6.jntuworld.3
QPSK
The Quadrature Phase Shift Keying (QPSK) is a 4-ary PSK signal. whereas non-coherent detection requires no phase information. Although QPSK can be viewed as a quaternary modulation.

4) and the constellation disgram is shown in Figure 6.2.
108
www. 3. The QPSK transmitted signal is deﬁned by: si (t) = A cos(ωt + (i − 1)π/2).2: QPSK signal constellation.www.5. 2.com
.
Figure 6.jntuworld. as shown in Figure 6. used to modulate the in-phase component of the carrier.21)
6.3.4
Oﬀset-QPSK
As in QPSK. i = (1. while the odd (or even) bits are used to modulate the quadrature-phase component of the carrier. (6.3: QPSK transmitter.jntuworld. Each bit in these streams has a duration of twice the bit duration. the NRZ data is split into two streams of odd and even bits.com
Figure 6.

com
. of the original data stream.
109
www.jntuworld. whose non-linearity increases as the eﬃciency goes high. are used then due to distortion. The constellation diagram of OﬀsetQPSK is the same as QPSK. When high eﬃciency power ampliﬁers. Tb . From the constellation diagram it is observed that a signal point in any quadrant can take a value in the diagonally opposite quadrant only when two pulses change their polarities together leading to an abrupt 180 degree phase shift between adjacent symbol slots. Abrupt phase changes leading to sudden changes in the signal amplitude in the time domain corresponds to signiﬁcant out of band high frequency components in the frequency domain. Oﬀset-QPSK diﬀers from QPSK in that the d1 (t) and d2 (t) are aligned such that the timing of the pulse streams are oﬀset with respect to each other by Tb seconds.com
Figure 6. This is prevented in O-QPSK and the allowed phase transitions are ± 90 degree. These odd (d1 (t)) and even bit streams (d2 (t)) are then used to modulate two sinusoidals in phase quadrature.www. After modulation they are added up and transmitted.jntuworld.and hence these data streams are also called the in-phase and and quadrature phase components. Thus to reduce these sidelobes spectral shaping is done at baseband. harmonics are generated and this leads to what is known as spectral regrowth. Since sudden 180 degree phase changes cannot occur in OQPSK.4: DQPSK constellation diagram. this problem is reduced to a certain extent.

1 forces a transition in the middle of the bit while 0 does nothing.22)
Here. 3.6
Line Coding
Speciﬁc waveforms are required to represent a zero and a one uniquely so that a sequence of bits is coded into electrical pulses. 6. Biphase-L: Manchester 1 forces positive transition while 0 forces negative transition. Biphase-M: There is always a transition in the beginning of a bit interval. The modulated signal during the time slot of kT < t < (k + 1)T given by: s(t) = cos(2πfc t + ψk+1 ) (6.jntuworld.jntuworld.5
π/4 DQPSK
The data for π/4 DQPSK like QPSK can be thought to be carried in the phase of a single modulated carrier or on the amplitudes of a pair of quadrature carriers.com
. −3π/4 for 11 and −π/4 for 10.com
6. There are various ways to accomplish this and the diﬀerent forms are summarized below. 2. ψk+1 = ψk + ∆ψk and ∆ψk can take values π/4 for 00. 3π/4 for 01. 1.
6. Non-return to zero mark (NRZ-M): 1 forces negative and positive transitions while 0 causes no transitions. Non-return to zero space (NRZ-S): 0 forces negative and positive transitions while 1 causes no transitions.
110
www. In case of consecutive bits of same type a transition occurs in the beginning of the bit period. 4.4. Return to zero (RZ): 1 goes high for half a period while 0 remains at zero state. This is known as line coding.www. Non-return to zero level (NRZ-L): 1 forces a a high while 0 forces a low.5. The constellation diagram along with possible transitions are shown in Figure 6. This corresponds to eight points in the signal constellation but at any instant of time only one of the four points are possible: the four points on axis or the four points oﬀ axis. 5.

5: Scematic of the line coding techniques.com
Figure 6. Such a pulse is not desirable for two fundamental reasons: 111
www.jntuworld. 0 forces a transition in the middle of the bit while 1 does nothing.jntuworld.com
. 0 forces a transition in the beginning of the bit while 1 does nothing. 7. Biphase-S: There is always a transition in the beginning of a bit interval.www.5. 8. Diﬀerential Manchester: There is always a transition in the middle of a bit interval.7
Pulse Shaping
Let us think about a rectangular pulse as deﬁned in BPSK. Bipolar/Alternate mark inversion (AMI): 1 forces a positive or negative pulse for half a bit period and they alternate while 0 does nothing.
6. 9. All these schemes are shown in Figure 6.

Let the overall frequency response of a communication system (the transmitter. This type of technique is called pulse shaping technique. (b) a wireless channel has memory due to multipath and therefore it introduces ISI. channel and receiver) be denoted as Hef f (f ) and according to Nyquist it must be given by: Hef f (f ) = 1 f rect( ) fs fs (6. But a wireless channel is bandlimited. an eﬃcient pulse shaping funtion or a premodulation ﬁlter is used at the Tx side so that QoS can be maintained to the mobile users during communication. We discuss here about the fundamental works of Nyquist.
6. such that.www. spectral shaping is usually done through baseband or IF processing. Correspondingly.24)
www.jntuworld.1
Nyquist pulse shaping
There are a number of well known pulse shaping techniques which are used to simultaneously to reduce the inter-symbol eﬀects and the spectral width of a modulated digital signal.com
. Hef f (f ) ↔ hef f (t) 112 (6.jntuworld.6: Rectangular Pulse (a) the spectrum of a rectangular pulse is inﬁnite in extent. the ideal pulse shape for zero ISI. given by hef f (t). we start with the fundamental works of Nyquist on pulse shaping and subsequently. Below.7. In order to mitigate the above two eﬀects. its frequency content is also inﬁnite.com
Figure 6. we would look into another type of pulse shaping technique. means it would introduce signal distortion to such type of pulses. As pulse shaping is diﬃcult to directly manipulate the transmitter spectrum at RF frequencies.23)
Hence.

is given by: hef f (t) =
πt sin( Ts ) πt Ts
(6.jntuworld.28) (6.2
Raised Cosine Roll-Oﬀ Filtering
1 2Ts
If we take a rectangular ﬁlter with bandwidth f0 ≥
and convolve it with any
arbitrary even function Z(f) with zero magnitude outside the passband of the rectangular ﬁlter then a zero ISI eﬀect would be achieved.26)
6. As ρ increases roll oﬀ in frequency domain increases but that in time domain decreases.7. Hef f (f ) = rect( hef f (t) = z(t) = f ) ∗ Z(f ).www.com
.jntuworld.3
Realization of Pulse Shaping Filters
Since hef f (t) is non-causal.27) (6.
6. 1 − (∆ρt/2Ts )2
with ρ being the roll oﬀ factor ∈ [0. f0 πt sin( Ts ) z(t).7: Raised Cosine Pulse. Digital communication systems thus often store several symbols at a time inside the modulator and then clock out a group of symbols by
113
www. πt
Ts
(6.7. pulse shaping ﬁlters are usually truncated within ±6Ts about t = 0 for each symbol.29)
cos(πρt/Ts ) .com
Figure 6. 1]. Mathematically.25) (6.

However.which simpliﬁes receiver design and provides high immunity against random FM noise and signal ﬂuctuations due to Rayleigh fading. In this method.
6. even if constant envelope has many advantages it still uses more BW than linear modulation schemes. FM is thus part of a more general class of modulation known as angle modulation. which would be discussed later in this chapter while discussing GMSK. it is a nonlinear modulation technique. This is the way to realize the pulse shaping ﬁlters using real time processors.8. in an 114
www.jntuworld.www.where the amplitude of the carrier is constant.8
Nonlinear Modulation Techniques
Many practical mobile radio communications use nonlinear modulation methods. the two most important classes of angle modulation being frequency modulation and phase modulation. As the information in frequency modulation is in the frequency of modulated signal. Limiter-discriminator detection can be used.The Constant envelope family of modulations has the following advantages : 1. the amplitude of the carrier wave is kept constant (this is why FM is called constant envelope).com
.1
Angle Modulation (FM and PM)
There are a number of ways in which the phase of a carrier signal may be varied in accordance with the baseband signal.regardless of the variations in the modulating signal. 3.Unlike AM.
6. 2. Non-Nyquist pulse shaping are also useful. Frequency modulated signals have better noise immunity and give better performance in fading scenario as compared to amplitude modulation.jntuworld.com
using a look up table that represents discrete time waveforms of stored symbols. Power eﬃcient class C ampliﬁers without introducing degradation in the spectral occupancy of the transmitted signal. Frequency modulation (FM) involves changing of the frequency of the carrier signal according to message signal. Low out-of-band radiation of the order of -60 dB to -70dB can be achieved.

Although frequency modulation systems are tolerant to certain types of signal and circuit nonlinearities. and hence bandwidth occupancy.jntuworld. due to the fact that the envelope of the carrier does not change with changes in the modulating signal. Both AM and FM may be demodulated using inexpensive noncoherent detectors. as shown in equation below.This ability of an FM system to trade bandwidth for SNR is perhaps the most important reason for its superiority over AM. thus linear Class A or AB ampliﬁers. sP M (t) = Ac cos(2πfc t + kθ m(t)) (6. it is critical to maintain linearity between the applied message and the amplitude of the transmitted signal. which are not as power eﬃcient. However. can be varied to obtain greater signal to noise performance. The constant envelope of the transmitted signal allows eﬃcient Class C power ampliﬁers to be used for RF power ampliﬁcation of FM.jntuworld. An FM signal is a constant envelope signal.31)
The frequency modulation index βf . AM signals are able to occupy less bandwidth as compared to FM signals. is the amplitude of the carrier.com
FM system.32)
www. must be used. the modulation index. special attention must be given to phase characteristics. Phase modulation (PM) is a form of angle modulation in which the angle θ(t) of the carrier signal is varied linearly with the baseband message signal m(t). deﬁnes the relationship between the message amplitude and the bandwidth of the transmitted signal.com
. FM systems require a wider frequency band in the transmitting media (generally several times as large as that needed for AM) in order to obtain the advantages of reduced noise and capture eﬀect. and is given by βf = kf Am ∆ = W W 115 (6.30)
where Ac . In FM the instantaneous frequency of the carrier signal is varied linearly with the baseband message signal m(t). as shown in following equation: sF M (t) = Ac cos[2πfc t + θ(t)] = Ac cos[2πfc t + 2πkf m(η)dη] (6. since the transmission system is linear. fc is the carrier frequency. and kf is the frequency deviation constant (measured in units of Hz/V). FM transmitter and receiver equipment is also more complex than that used by amplitude modulation systems.www. In AM. however. AM is easily demodulated using an envelope detector whereas FM is demodulated using a discriminator or slope detector.

Depending on how the frequency variations are imparted into the transmitted waveform.39)
Expressing θ(t) in terms of θ(0) with a new unknown factor h.35) (6. (6. In general.2
BFSK
In Binary Frequency Shift keying (BFSK).e. The phase modulation index βp is given by βp = kθ Am = ∆θ where.34)
where T is the symbol duration and Eb is the energy per bit. S1 (t) = S2 (t) = (2Eb /T ) cos(2πf1 t + θ(0)) (2Eb /T ) cos(2πf2 t + θ(0)) →1 →0 (6.
There are two FSK signals to represent 1 and 0.com
. an FSK signal may be represented as S(t) = (2Eb /T ) cos(2πfi t). (6.com
where Am is the peak value of the modulating signal.33)
6.8. i. (6.38)
where θ(0) sums the phase up to t = 0.the FSK signal will have either a discontinuous phase or continuous phase between bits.www.jntuworld.jntuworld. Si = φ(t) = (Eb )φ(t). we get θ(t) = θ(0) ± πht/T 0≤t≤T (6. Let us now consider a continuous phase FSK as S(t) = (2Eb /T ) cos(2πfc t + θ(t)).36)
(2/T ) cos(2πfi t). ∆f is the peak frequency deviation of the transmitter and W is the maximum bandwidth of the modulating signal..40)
116
www.the frequency of constant amplitude carrier signal is switched between two values according to the two possible message states (called high and low tones) corresponding to a binary 1 or 0. ∆θ is the peak phase deviation of the transmitter.37) (6. (6.

A special case of CPFSK is achieved with h = 0.46) (6.8.com
. If we substitute t = T .jntuworld. Hence a pulse shaping technique is required. In Figure 6. 1/T (6.47) (6. and the resulting scheme is called Minimum Shift Keying (MSK) which is used in mobile communications. We know that θ(t) − θ(0) = ±πht/T . we show a phase tree of such a CPFSK signal with the transmitted bit stream of 1101000. the unknown factor h can be treated as the diﬀerence between f1 and f2 . On the other hand. the transmitted bit can be predicted.41) T
It shows that we can choose two frequencies f1 and f2 such that f1 = fc + h/2T f2 = fc − h/2T (6.com
and therefore S(t) = 2Eb cos(2πfc t ± πht/T + θ(0)) = T 2Eb cos(2π(fc ± h/2T )t + θ(0)).42) (6. Spectral characteristics of an MSK signal is shown in Figure 6. In this case. 0 ≤ t ≤ T .(6. we have θ(T ) − θ(0) = ±πh = πh = −πh where →1 →0 (6. normalized with respect to bit rate 1/T . fc and h can be expressed in terms of f1 and f2 as fc = [f1 + f2 ]/2 h= (f1 − f2 ) . we use a pulse shaping ﬁlter with
117
www.9 from which it is clear that ACI is present in the spectrum.45)
Therefore.44) (6. An MSK signal can also be thought as a special case of OQPSK where the baseband rectangular pulses are replaced by half sinusoidal pulses.www.5.48)
This type of CPFSK is advantageous since by looking only at the phase. the phase diﬀerences reduce to only ±π/2 and the phase tree is called the phase trellis.43)
for which the expression of FSK conforms to that of CPFSK. It is called the deviation ratio.jntuworld. In order to have a compact signal spectrum as well as maintaining the constant envelope property.

a narrow BW frequency and sharp cutoﬀ characteristics (in order to suppress the high frequency component of the signal).jntuworld. 3. an impulse response with relatively low overshoot (to limit FM instant frequency deviation.9: Spectrum of MSK 1.com
.jntuworld.
Figure 6.8: Phase tree of 1101000 CPFSK sequence.
6. In GMSK.www.com
Figure 6. a phase trellis with ±π/2 for odd T and 0 or π values for even T.9
GMSK Scheme
GMSK is a simple modulation scheme that may be taken as a derivative of MSK. 2. the sidelobe levels of the spectrum are further reduced by passing a non-
118
www.

jntuworld.49)
The premodulation Gaussian ﬁltering introduces ISI in the transmitted signal. (6. but it can be shown that the degradation is not that great if the 3dB bandwidth-bit duration product (BT) is greater than 0. return to zero (NRZ-L) data waveform through a premodulation Gaussian pulse shaping ﬁlter.www. it is evident that when we are decreasing BT product.12.com
. Baseband Gaussian pulse shaping smoothes the trajectory of the MSK signals and hence stabilizes instantaneous frequency variations over time.
6.50) (6.10
GMSK Generator
The GMSK premodulation ﬁlter has characteristic equation given by H(f ) = exp(−(ln 2/2)(f /B)2 ) H(f ) = exp(−(αf )2 ) where. A GMSK generation scheme with NRZ-L data is shown in Figure 6. the out of band response decreases but 119
www. (α)2 = ln 2/2(1/B)2 .jntuworld.10 and a receiver of the same scheme with some MSI gates is shown in Figure 6.com
Figure 6. From this ﬁgure.11.10: GMSK generation scheme.5. This has the eﬀect of considerably reducing the sidelobes in the transmitted spectrum. Spectrum of GMSK scheme is shown in Figure 6.

11: A simple GMSK receiver. on the other hand irreducible error rate of the LPF for ISI increases.52) (6. B = 0.567kHz (6.9 kHz.25 GMSK with a channel data rate Rb=270 kbps. 90 % RF BW = 0.jntuworld.51)
Thus the 3 .7µsec Solving for B where BT = 0.25.57Rb = 153.25/T = 67.com
Figure 6.
120
www.jntuworld.567 kHz.www. Therefore.dB bandwidth is 67.com
. a compromise between these two is required. Problem: Find the 3dB BW for a Gaussian LPF used to produce 0.57 Rb .What is the 90 percent power BW of the RF ﬁlter? Solution: From the problem statement it is clear that T = 1/Rb = 1/270 ∗ (103 ) = 3. We use below table ﬁg 6 to ﬁnd out that 90 % power bandwidth is 0.

121
www.1
Two Practical Issues of Concern
Inter Channel Interference
In FDMA.11. This is because of the fact that transmission of a time limited pulse leads to spectral spreading in the frequency domain. In such a case the side lobes of the stronger signal will severely degrade the transmitted signal in the next frequency slot having low power level. this kind of interference becomes so severe that the required symbol in a particular frequency slot is completely lost. The side lobes generated due to the transmission of a symbol in a particular channel overlaps with the channels placed adjacently.jntuworld. During simultaneous use of adjacent channels. Moreover if two terminals transmit equal power then due to wave propagation through diﬀerent distances to the receiver. when there is signiﬁcant amount of power present in the side lobes. This is known as the near far problem.
6.com
.12: Spectrum of GMSK scheme.www. the received signal levels in the two frequency slots will diﬀer greatly. subscribers are allotted frequency slots called channels in a given band of the electromagnetic spectrum.jntuworld.11
6.com
Figure 6.

www.rms Vout. Thus the amplitude of a signal applied at the input of an ampliﬁer suﬀers from amplitude distortion and the resulting waveform obtained at the output of the ampliﬁer is of the form of an amplitude modulated signal.58)
www.BP SK = Pe.rms Vout. class B.54)
6. Due to power constraints on a transmitting terminal.jntuworld. Class A ampliﬁer is a linear ampliﬁer but it has a power eﬃciency of only 25 %.BF SK 1 1− 2 1 = 1− 2 γ 1+γ γ 2+γ (6.2
Power Ampliﬁer Nonlinearity
Power ampliﬁers may be designed as class A.jntuworld. class C and class D. γ= Eb E(α2 ) N0 122 (6. beyond an input threshold level. Input back − of f = 10 log1 0 Vin. Similarly. The operating point of a practical ampliﬁer is given in terms of either the input back-oﬀ or the output back-oﬀ. an ampliﬁer has linear input output characteristics over a range of input signal level.55) (6. an eﬃcient power ampliﬁer is required which can convert most of the input power to RF power.57) where γ is given by.rms (6. the nonlinearity of the ampliﬁer increases. that is. class AB.com
6.53)
Output
back − of f = 10 log1 0
(6.11. the probability of error for coherent BPSK and coherent BFSK are respectively given as. In general.56) (6. As we go for subsequent ampliﬁers having greater power eﬃciency. However.com
. the phase characteristic of a practical ampliﬁer is not constant over all input levels and results in phase distortion of the form of phase modulation. They form an essential section of mobile radio terminals. it has a constant gain.12
Receiver performance in multipath channels
For a ﬂat fading channel. the gain of the ampliﬁer starts decreasing.rms Vout. Pe.

1
Bit Error Rate and Symbol Error Rate
Bit error rate (Peb ) is the same as symbol error rate (Pes ) when a symbol consists of a single bit as in BPSK modulation.13
Example of a Multicarrier Modulation: OFDM
Multiplexing is an important signal processing operation in which a number of signals are combined and transmitted parallelly over a common channel.12.61) (6. then the SNR range must be around 30dB to 60dB. 1 4γ 1 Pe.com
α2 represents the instantaneous power values of the Rayleigh fading channel and E denotes the expectation operator.60)
the error probability given above have the simpliﬁed
From the above equations we observe that an inverse algebraic relation exists between the BER and SNR.63) (6.DP SK = Pe. Similarly.65)
And for M-ary orthogonal signalling Peb is given by Peb = M/2 Pes .62) (6. M −1 (6.BP SK = (6.59) (6. γ Pe.jntuworld.jntuworld.66)
6. This implies that if the required BER range is around 10−3 to 10−6 . Peb is given by Peb ≈ Pes log2 M (6.DP SK = 2γ 1 Pe. In order to 123
www.com
. such that 2N = M . = (2 + γ)
(6.64)
Eb N0
1 2(1 + γ) 1 .BF SK = 2γ 1 Pe.N CF SK For large values of SN R = expression.
6. For an MPSK scheme employing gray coded modulation. for diﬀerential BPSK and non coherent BFSK probability of error expressions are Pe. where N bits are mapped to a one of the M symbols.www.N CF SK = .

124
www.m. Although the use of OFDM was initially limited to military applications due to cost and complexity considerations. this is no longer a major problem. This technique is being used. there is a precise mathematical relationship between the frequencies of the transmitted signals) such that one signal has zero eﬀect on another. OFDM is a technique of transmitting high bit rate data into several parallel streams of low bit rate data. In FDM. Thus in OFDM. At any instant. MIMO-OFDM.com
avoid interference during parallel transmission.www.com
. Moreover related or combined technology such as CDMA-OFDM. the adjacent bands are non overlapping but if overlap is allowed by transmitting signals that are mutually orthogonal (that is.s. For high data rate communication the bandwidth (which is limited) requirement goes on increasing as the data rate increases or the symbol duration decreases. wide-band OFDM (W-OFDM). Very recently. instead of sending a particular number of symbols. delay spread of the channel. in digital audio broadcasting (DAB). asymmetric digital subscriber lines (ADSL) and mobile communications. digital video broadcasting terrestrial TV (DVB-T). then the resulting transmission technique is known as Orthogonal Frequency Division Multiplexing (OFDM). in T seconds serially. Vector OFDM (V-OFDM). the data transmitted simultaneously in each of these parallel data streams is frequency modulated by carriers (called subcarriers) which are orthogonal to each other. say P. Viewed in the frequency domain this implies that the bandwidth of the OFDM signal is less than coherent bandwidth of the channel. the signals can be separated in frequency and then the resulting technique is called Frequency Division Multiplexing (FDM). the P symbols can be sent in parallel with symbol duration now increased to T seconds instead of T/P seconds as was previously. high deﬁnition digital television broadcasting (HDTV). ﬂash OFDM (F-OFDM). with the recent advances in large-scale highspeed DSP.jntuworld. TDMA-OFDM. WLAN systems based on IEEE 802.11(a) or HiperLan2. This oﬀers many advantages in digital data transmission through a wireless time varying channel. The primary advantage of increasing the symbol duration is that the channel experiences ﬂat fading instead of frequency selective fading since it is ensured that in the time domain the symbol duration is greater than the r. the signiﬁcance of the COFDM technique for UWA (underwater acoustic channel) has also been indicated.jntuworld.

This implies that for two harmonics of an exponential function having a fundamental frequency of 1/Ts . where n varies as 0.But if fk = fn then N equals Ts which is nothing but the energy of the complex exponential signal in the time duration of Ts . wavelet-OFDM have presented their great advantages in certain application areas.
6.com
OFDMA. we can obtain the transmitted signal in Ts seconds by summing up all the P number of amplitude modulated subcarriers.13. the inner product becomes zero .13.jntuworld. Let ψk (t) = ej2πfk t and ψn (t) = ej2πfn t be two complex exponential signals whose inner product.68)
125
www. over the time duration of Ts . is given by:
(i+1)Ts ∗ ψk (t). thereby yielding the following equation:
P −1
p(t) =
n=0
cn gn (t)exp j2πn
t Ts
f or
0 ≤ t ≤ Ts
(6. 1.com
. Here the variable ‘n’ denotes the nth parallel path corresponding to the nth subcarrier.2
Mathematical Description of OFDM
Let us now consider the simultaneous or parallel transmission of P number of complex symbols in the time slot of Ts second (OFDM symbol time duration) and a set of P orthogonal subcarriers. Let each orthogonal carrier be of the form exp j2πn Tts .
6. such that each subcarrier gets amplitude modulated by a particular symbol from this set of P symbols.1
Orthogonality of Signals
Orthogonal signals can be viewed in the same perspective as we view vectors which are perpendicular/orthogonal to each other.www.67)
When this integral is evaluated. 2.jntuworld. it is found that if fk and fn are integer multiples of 1/Ts then N equals zero.. The inner product of two mutually orthogonal vectors is equal to zero.ψn (t)dt
N=
iTs
(6. Similarly the inner product of two orthogonal signals is also equal to zero.(P − 1). Mathematically.

then the resulting waveform. This can be realized using IFFT but the constraint is that P has to be a power of 2. The OFDM transmitter and receiver 126
www. But this might not be feasible from the implementation point of view speciﬁcally in terms of FFT size and Doppler shifts.69)
=
cn exp j2πn
n=0
k P
This is nothing but the IDFT on the symbol block of P symbols. In order to cope with ISI as discussed previously the OFDM symbol duration can be increased. therefore consists of a set of P partially overlapping sinc pulses during any time slot of duration Ts . The frequency spectrum. IFFTs and FFTs can be implemented easily.jntuworld.com
. If we denote the guard time interval as Tg and Ts be the useful symbol duration. So at the receiver. its eﬀect in the time domain is to cause symbol spreading such that a part of the symbol overlaps with the adjacent symbols. translating each subcarrier down to DC. then after this cyclical extension the total symbol duration becomes T = Tg + Ts . A diﬀerent approach is to keep a guard time interval between two OFDM symbols in which part of the symbol is copied from the end of the symbol to the front and is popularly known as the cyclic-preﬁx.com
If p(t) is sampled at t = kTs /P . or the multipath delay.www. then ISI can be eliminated. When the guard interval is longer than the length of the channel impulse response. then integrating the resulting signal over a symbol period to recover the raw data. The receiver can be visualized as consisting of a bank of demodulators. is:
P −1
p(k) =
n=0
cn gn (kTs /P )exp j2πn 1 √ Ts
P −1
kTs /P Ts f or 0 ≤ k ≤ P − 1 (6. This is due to the fact that the Fourier Transform of a rectangular pulse is a sinc function. This implementation is better than using multiple oscillators for subcarrier generation which is uneconomical and since digital technology has greatly advanced over the past few decades. But the OFDM symbol structure so generated at the transmitter end needs to be modiﬁed. Since inter symbol interference (ISI) is introduced by the transmission channel due to multipaths and also due to the fact that when the bandwidth of OFDM signal is truncated. However the disadvantage is the reduction in data rate or throughput and greater power requirements at the transmitting end. FFT can be done to get back the required block of symbols.jntuworld.

com
.
6. A relatively new modulation technology. The important modulation techniques for wireless communication such as QPSK. the channel still introduces fading in diﬀerent ways. OFDM.com
Figure 6. MSK.13: OFDM Transmitter and Receiver Block Diagram. In order to prevent that. Certain practical issues of concern are also discussed.jntuworld.jntuworld. GMSK were taken up at length.
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www.14
Conclusion
In this chapter. sections are as given in the following diagram. a major chunk has been devoted to digital communication systems which obviously have certain distinction in comparison to their analog counterpart due to their signal-space representation. we need some additional signal processing techniques mainly at the receiver side. These techniques are discussed in the next chapter. has also been discussed.www. It should be noted that albeit implementing these eﬃcient modulation techniques.

. 2004. NY: Oxford University Press. Singapore: Pearson Education.jntuworld. Haykin and M. Modern Wireless Communications. Singapore: Thomson Asia Pvt Ltd. Wireless Communications: Principles and Practice. Blake. Modern Digital and Analog Communication Systems. G. 2nd ed. Inc. Singapore: Pearson Education. 2005. Digital Communications: Fundamentals and Applications. 3. Singapore: Pearson Education. Salehi. 2002. Tranter et. 6. W. S. Singapore: Pearson Education. 2nd ed. 4.15
References
1. Inc. 5. Ding. Inc.. B. Moher. Proakis and M. Inc. R.com
6. J. 2002. H. Sklar.. Lathi and Z. Delmar. 2009. Principles of Communication Systems Simulation.. Rappaport. Inc. 2002. 4th ed.. 2. Electronic Communication Systems. Singapore: Pearson Education.
128
www. P.. 2nd ed. T. B. S. 2002.com
.jntuworld. 7. al.www. Communication Systems Engineering.

An equalizer within a receiver compensates for the average range of expected channel amplitude and delay characteristics.com
Chapter 7
Techniques to Mitigate Fading Eﬀects
7.1 Introduction
Apart from the better transmitter and receiver technology. As such equalizers ﬁnd their use in frequency selective fading channels. In other words.com
. Equalization compensates for Inter Symbol Interference (ISI) created by multipath within time dispersive channels. We discuss all three of these techniques in this chapter. an equalizer is a ﬁlter at the mobile receiver whose impulse response is inverse of the channel impulse response.jntuworld.jntuworld.1. a channel coder maps a digital message sequence in to another speciﬁc code sequence containing greater number of bits than original contained in the message. It is usually employed to reduce the depths and duration of the fades experienced by a receiver in a ﬂat fading channel. Channel Coding is used to correct deep fading or spectral null.At the baseband portion of the transmitter. A general framework of the fading eﬀects and their mitigation techniques is shown in Figure 7. 129
www. Diversity and channel coding are channel impairment improvement techniques. Channel coding improves mobile communication link performance by adding redundant data bits in the transmitted message. Diversity is another technique used to compensate fast fading and is usually implemented using two or more receiving antennas.www. mobile communications require signal processing techniques that improve the link performance. Equalization.

• The training sequence is designed to permit an equalizer at the receiver to acquire the proper ﬁlter coeﬃcient in the worst possible channel condition. • Training sequence is typically a pseudo-random binary signal or a ﬁxed. equalizers must track the time-varying characteristics of the mobile channel and therefore should be timevarying or adaptive. An adaptive ﬁlter at the receiver thus uses a recursive algorithm to evaluate 130
www. An equalizer at the front end of a receiver compensates for the average range of expected channel amplitude and delay characteristics.1: A general framework of fading eﬀects and their mitigation techniques. Training Mode: • Initially a known.www. These are as follows. frequency selective fading).jntuworld.2
Equalization
ISI has been identiﬁed as one of the major obstacles to high speed data transmission over mobile radio channels. of prescribed bit pattern. An adaptive equalizer has two phases of operation: training and tracking.jntuworld.e. If the modulation bandwidth exceeds the coherence bandwidth of the radio channel (i.. ﬁxed length training sequence is sent by the transmitter so that the receiver equalizer may average to a proper setting.com
.
7. As the mobile fading channels are random and time varying.com
Figure 7. modulation pulses are spread in time. causing ISI.

2. ˆ (7. the adaptive equalizer continuously changes the ﬁlter characteristics over time.channel and the RF/IF section of the receiver and nb (t) denotes the baseband noise.jntuworld. If the channel is frequency selective. we can write y(t) = d(t). the adaptive algorithms of the equalizer tracks the changing channel.2)
However. Assuming nb (t)=0. the equalizer enhances the frequency components with small amplitudes and attenuates the strong frequencies in the received frequency 131
www. h(t) is the combined impulse response of the transmitter.1
A Mathematical Framework
The signal received by the equalizer is given by x(t) = d(t) ∗ h (t) + nb (t) (7. • When the data of the users are received.jntuworld. the desired output of the equalizer is d(t) which is the original source data.
7. Tracking Mode: • When the training sequence is ﬁnished the ﬁlter coeﬃcients are near optimal. the output of the equalizer is y (t) = d (t) ∗ h (t) ∗ heq (t) + nb (t) ∗ heq (t) = d (t) ∗ g (t) + nb (t) ∗ heq (t) .4)
which indicates that an equalizer is actually an inverse ﬁlter of the channel.1)
where d(t) is the transmitted signal. If the impulse response of the equalizer is heq (t). • As a result. • Immediately following the training sequence. In frequency domain it can be written as Heq (f ) H (f ) = 1 (7.3)
The main goal of any equalization process is to satisfy this equation optimally.com
the channel and estimate ﬁlter coeﬃcients to compensate for the channel.com
.www. user data is sent. which in turn stems the following equation: g (t) = h (t) ∗ heq (t) = δ (t) (7.

The adaptive algorithm is controlled by the error signal ek .www. When each of the delay elements provide a time delay equal to the symbol duration T. composite received frequency response and linear phase response. In the ﬁgure the subscript k represents discrete time index. the equalizer coeﬃcients cn are chosen to force the samples of the combined channel and equalizer impulse response to zero.com
.jntuworld. The combined response of the channel with the equalizer must satisfy Nyquist’s criterion Hch (f ) Heq (f ) = 1. The adaptive algorithm uses ek to minimize the cost function and uses the equalizer weights in such a manner that it minimizes the cost function iteratively. For a time varying channel.2.2
Zero Forcing Equalization
In a zero forcing equalizer.
7.5)
where Hch (f ) is the folded frequency response of the channel. The usual equalizer model follows a time varying or adaptive structure which is given next. Thus.2. The error signal is derived by comparing the output of the equalizer. the frequency response Heq (f ) of the equalizer is periodic with a period equal to the symbol rate 1/T. so it is rarely used for wireless link except for static channels with high SNR such as local wired telephone.3
A Generic Adaptive Equalizer
The basic structure of an adaptive ﬁlter is shown in Figure 7.com
spectrum in order to provide a ﬂat. N+1 taps and N+1 tunable complex multipliers. and in this case has N delay elements. an inﬁnite length zero-forcing ISI equalizer is simply an inverse ﬁlter which inverts the folded frequency response of the channel. These weights are updated continuously by an adaptive algorithm. |f | < 1/2T (7. This ﬁlter is called the transversal ﬁlter.jntuworld. Let us denote the received sequence vector at the receiver and
132
www. the equalizer is designed to track the channel variations so that the above equation is approximately satisﬁed. Disadvantage: Since Heq (f ) is inverse of Hch (f ) so inverse ﬁlter may excessively amplify the noise at frequencies where the folded channel spectrum has high attenuation. with some signal dk which is replica of transmitted signal.
7.2. called weights.

T yk = xk .10)
133
www.www..... xk−N ]T .com
Figure 7...jntuworld.8)
The error signal is deﬁned as ek = dk − yk = dk − xT wk . and the tap coeﬃcient vector as
0 1 N wk = [wk .
(7.com
.. the input to the equalizer as xk = [xk .e. .9)
Assuming dk and xk to be jointly stationary.. the output sequence of the equalizer yk is the inner product of xk and wk .2: A generic adaptive equalizer. wk = xT wk = wk xk .jntuworld.7)
Now.. .. k (7. i.6)
(7. xk−1 .. the Mean Square Error (MSE) is given as M SE = E[e2 ] = E[(dk − yk )2 ] k = E[(dk − xT wk )2 ] k
T = E[d2 ] + wk E[xk xT ]wk − 2E[dk xT ]wk k k k
(7. k
(7. wk ]T . wk .

.14)
(7.11)
2 where the signal variance σd = E[d2 ] and the cross correlation vector p between the k
desired response and the input signal is deﬁned as p = E [dk xk ] = E dk xk dk xk−1 dk xk−2 · · · dk xk−N .
x2 k−N
Clearly. . (7. Misadjustment: Provides a quantitative measure of the amount by which the ﬁnal value of mean square error. . The MSE then can be expressed as
2 T M SE = ξ = σk + wk Rwk − 2pT wk
(7. it requires a speciﬁc algorithm to update the equalizer coeﬃcients and track the channel variations. MSE is a function of wk .12)
The input correlation matrix R is deﬁned as an (N + 1) × (N + 1) square matrix.
x2 k
xk xk−1 x2 k−1 xk−2 xk−1 . Hence. 134
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.jntuworld.15)
7. A fast rate of convergence allows the algorithm to adapt rapidly to a stationary environment of unknown statistics. On equating
to 0.www.jntuworld. .com
where wk is assumed to be an array of optimum values and therefore it has been taken out of the E() operator. . in response to a stationary inputs. (7. .
(7. averaged over an ensemble of adaptive ﬁlters.13) . to converge close enough to optimal solution.
xk xk−2 xk−1 xk−2 x2 k−2 . Factors which determine algorithm’s performance are: Rate of convergence: Number of iterations required for an algorithm.
··· ··· ··· ···
xk xk−N
xk−N xk xk−N xk−1 xk−N xk−2 · · ·
∂ξ ∂wk
xk−1 xk−N xk−2 xk−N .4
Choice of Algorithms for Adaptive Equalization
Since an adaptive equalizer compensates for an unknown and time varying channel. . . where
x k−1 xk T R = E xk xk = E xk−2 xk . MMSE is given by the equation
2 M M SE = ξmin = σd − pT wk . deviates from an optimal mean square error.2. we get the condition for
minimum MSE (MMSE) which is known as Wiener solution: wk = R−1 p.

If an input signal has a time dispersion characteristics that is greater than the propagation delay through the equalizer.jntuworld. and may be performed by use of the stochastic gradient algorithm. the MSE and the optimal Wiener solution remain the same as given the adaptive equalization framework. as discussed earlier.www. These include the Zero Forcing Algorithm (ZF). the value of µ is chosen from
N
0<µ<2
i=1
λi
(7. The convergence rate of the LMS algorithm is slow due to the fact that there is only one parameter. LMS is computed iteratively by wk (n + 1) = wk (n) + µek (n) x (n − k) (7. Below. Numerical properties: Inaccuracies like round-oﬀ noise and representation errors in the computer. then the equalizer will be unable to reduce distortion.16)
where the subscript k denotes the kth delay stage in the equalizer and µ is the step size which controls the convergence rate and stability of the algorithm. Three classic equalizer algorithms are primitive for most of today’s wireless standards.com
. The ﬁlter weights are updated by the update equation. we discuss a few of the adaptive algorithms.17)
where λi is the i-th eigenvalue of the covariance matrix R. Here the system error. 135
www. which inﬂuence the stability of the algorithm. Least Mean Square (LMS) Algorithm LMS algorithm is the simplest algorithm based on minimization of the MSE between the desired equalizer output and the actual equalizer output. Letting the variable n denote the sequence of iteration. that controls the adaptation rate.jntuworld.com
Computational complexity: Number of operations required to make one complete iteration of the algorithm. and the Recursive Least Square Algorithm (RLS). It is the simplest equalization algorithm and requires only 2N+1 operations per iteration. the step size. the minimization of the MSE is carried out recursively. The LMS equalizer maximizes the signal to distortion ratio at its output within the constraints of the equalizer ﬁlter length. In practice. To prevent the adaptation from becoming unstable. the Least Mean Square Algorithm (LMS).

136
www. a small positive number ε is added to the denominator term of the correction factor. Therefore when x (n − k) is large. both the instantaneous and average SNRs at the receiver may be improved. Hence receivers in diversity technique are used in such a way that the signal received by one is independent of the other. the denominator term x (n)
in the NLMS equation
becomes very small and the correction factor may diverge. Only when
2
x(n−k) becomes close to zero.19)
7. Diversity can be of various forms. this problem is eliminated. the correction that is applied to wk (n) is proportional to the input sample x (n − k). Usually diversity decisions are made by receiver. diversity requires no training overhead as a training sequence is not required by transmitter. Note that if the distance between two receivers is a multiple of λ/2.www. starting from space diversity to time diversity. another independent path may have a strong signal. It is a very simple concept where if one path undergoes a deep fade.jntuworld.com
Normalized LMS (NLMS) Algorithm In the LMS algorithm. there might occur a destructive interference between the two signals. With the normalization of the LMS step size by x (n)
2
in the NLMS algorithm. the step size is time varying and is expressed as µ (n) = β . x (n) 2 + ε (7. the LMS algorithm experiences gradient noise ampliﬁcation. the NLMS algorithm update equation takes the form of wk (n + 1) = wk (n) + β ek (n) x (n − k) . It exploits the random nature of radio propagation by ﬁnding independent signal paths for communication. Here.jntuworld. As there is more than one path to select from.com
. Unlike equalization. x (n) 2 + ε (7.18)
Therefore. We take up the types one by one in the sequel. So.3
Diversity
Diversity is a method used to develop information from several signals transmitted over independent fading paths.

Selection Diversity is the simplest diversity technique. Space diversity reception methods can be classiﬁed into four categories: selection. maximal ratio combining and equal gain combining.jntuworld.jntuworld. with M receivers.
7.3: Receiver selection diversity. The signal to noise ratio is deﬁned as SN R = Γ = 137 Eb 2 α N0 (7. (a) Selection Diversity: The basic principle of this type of diversity is selecting the best signal among all the signals received from diﬀerent branches at the receiving end.com
Figure 7.com
. feedback or scanning. Signals received from spatially separated antennas have uncorrelated envelopes. in which the eﬀects of fading are minimized by the simultaneous use of two or more physically separated antennas.3 shows a block diagram of this method where ’M’ demodulators are used to provide M diversity branches whose gains are adjusted to provide the same average SNR for each branch. ideally separated by one half or more wavelengths.www. The receiver branches having the highest instantaneous SNR is connected to the demodulator. or both.1
Diﬀerent Types of Diversity
Space Diversity A method of transmission or reception. Each channel is called a diversity branch and let each branch has the same average SNR.3. Let M independent Rayleigh fading channels are available at a receiver.20)
www. Figure 7.

γM ≤ γ] = 1 − e
−γ Γ
M
= PM (γ)
(7.
(7. .jntuworld.www. PM (Γ) < P (Γ). when no diversity is used.25)
The average SNR.com
. N0 is the noise PSD. If a single branch achieves SNR > γ.22)
Similarly.com
where Eb is the average carrier energy.26)
where x = γ/Γ and Γ is the average SNR for a single branch. .jntuworld.24)
which is more than the required SNR for a single branch receiver. . The instantaneous SNR(γi ) is usually deﬁned as γi = instantaneous signal power per branch/mean noise power per branch.
138
www. . we ﬁrst ﬁnd out the pdf of γ as pM (γ) = γ M d PM (γ) = 1 − e− /Γ dγ Γ
M −1
γ e− /Γ . can be then expressed as ¯
∞ ∞ M −1
γ= ¯
0
γpM (γ) dγ = Γ
0
M x 1 − e−x
e−x dx
(7. This expression shows the advantage when a selection diversity is used. Γ (7. The probability density function for such a channel is p (γi ) = 1 −γi e Γ . Quite clearly. γ2 .23)
where PM (γ) is the probability of all branches failing to achieve an instantaneous SNR = γ. Γ
(7. For Rayleigh fading channels. the probability that all M independent diversity branches receive signals which are simultaneously less than some speciﬁc SNR threshold γ is Pr [γ1 . To determine of average signal to noise ratio.21)
The probability that any single branch has an instantaneous SNR less than some deﬁned threshold γ is
γ γ
Pr [γi ≤ γ] =
0
p (γi ) dγi =
0
−γ 1 −γi e Γ dγi = 1 − e Γ = P (Γ). γ . then the probability that SNR > γ for one or more branches is given by Pr [γi > γ] = 1 − PM (γ) = 1 − 1 − e
−γ Γ
M
(7. α has a Rayleigh distribution and so α2 and consequently γi have a chi-square distribution with two degrees of freedom. α is a random variable used to represent amplitude values of the fading channel.

(d) Equal Gain Combining: In some cases it is not convenient to provide for the variable weighting capability required for true maximal ratio combining.com
This equation shows an average improvement in the link margin without requiring extra transmitter power or complex circuitry. (b) Feedback or Scanning Diversity: Scanning all the signals in a ﬁxed sequence until the one with SNR more than a predetermined threshold is identiﬁed. but the advantage is that it is very simple to implement(only one receiver is required). and it is easy to implement as it needed a monitoring station and an antenna switch at the receiver. Advantage of producing an output with an acceptable SNR even when none of the individual signals are themselves acceptable. This signal is then received until it falls below threshold and the scanning process is again initiated.com
.27)
where Gi is the gain and ri is the voltage signal from each branch. rm =
i=1 m
Gi ri
(7. It allows the receiver to exploit signals that are simultaneously received on each branch.jntuworld. The resulting fading statistics are somewhat inferior. as it gives the best statistical reduction of fading of any known linear diversity combiner. Assuming all the Gi to be
139
www. but the signals from each branch are co-phased to provide equal gain combining diversity. Performance of this method is marginally inferior to maximal ratio combining and superior to Selection diversity. Feedback or scanning diversity is very similar to selection diversity except that instead of always using the best of N signals.jntuworld. the branch weights are all set unity. Produces an output SNR equal to the sum of all individual SNR. which generally requires an individual receiver and phasing circuit for each antenna element. In such cases. (c) Maximal Ratio Combining: Signals from all of the m branches are weighted according to their individual signal voltage to noise power ratios and then summed. Modern DSP techniques and digital receivers are now making this optimal form. It is not an optimal diversity technique as it doesn’t use all the possible branches simultaneously. In terms of voltage signal.www. the N signals are scanned in a ﬁxed sequence until one is found to be above a predetermined threshold. Individual signals must be cophased before being summed.

jntuworld.28)
Polarization Diversity Polarization Diversity relies on the decorrelation of the two receive ports to achieve diversity gain. Polarization diversity has prove valuable at radio and mobile com-
140
www. Left-hand/Right-hand CP etc). here. rm =
i=1
m
ri .e. unity. Pairing two complementary polarizations.jntuworld.4: Maximal ratio combining technique. Polarization Diversity at a base station does not require antenna spacing. this scheme can immunize a system from polarization mismatches that would otherwise cause signal fade. The two receiver ports must remain cross-polarized.com
Figure 7.www. Reﬂected signals can undergo polarization changes depending on the channel.
(7. ± slant 45o . Polarization diversity combines pairs of antennas with orthogonal polarizations (i.com
. horizontal/vertical.

Main disadvantage is that it requires spare bandwidth also as many receivers as there are channels used for the frequency diversity. Time diversity repeatedly transmits information at time spacings that exceeds the coherence time of the channel. Disadvantage is that it requires spare bandwidth also as many receivers as there are channels used for the frequency diversity. the same information signal is transmitted and received simultaneously on two or more independent fading carrier frequencies. propagation delay spread in the radio channel provides multiple version of the transmitted signal at the receiver.com
. Delaying multipath components by more than a chip duration. RAKE
141
www. Time Diversity In time diversity. Frequency Diversity In Frequency Diversity. A modern implementation of time diversity involves the use of RAKE receiver for spread spectrum CDMA.com
munication base stations since it is less susceptible to the near random orientations of transmitting antennas. Multiple repetition of the signal will be received with independent fading conditions. Application 1: RAKE Receiver In CDMA spread spectrum systems.www. the signal representing the same information are sent over the same channel at diﬀerent times. thereby providing for diversity. Two important types of time diversity application is discussed below. CDMA spreading codes are designed to provide very low correlation between successive chips. The probability of simultaneous fading will be the product of the individual fading probabilities. CDMA receiver may combine the time delayed versions of the original signal to improve the signal to noise ratio at the receiver.jntuworld. where the multipath channel provides redundancy in the transmitted message.jntuworld. will appear like uncorrelated noise at a CDMA receiver. 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. This method is employed in microwave LoS links which carry several channels in a frequency division multiplex mode (FDM).

jntuworld. Spreading source bits over time. Interleaver can be of two forms. Schematic of a RAKE receiver is shown in Figure 7.com
.5. Many speech coder produce several important bits in succession. Each row contains a word of source data having n bits. the important bits from a block of source data are not corrupted at the same time. and interleaves nm bits at a time. it becomes possible to make use of error control coding. a block structure or a convolutional structure. and must be protected from errors. source bits are placed into the interleaver by sequentially increasing the row number for each 142
www. an interleaver of degree m consists of m rows. Application 2: Interleaver In the encoded data bits.jntuworld. Demodulation and bit decisions are based on the weighted output of the correlators. some source bits are more important than others.com
Figure 7. Interleaver spread these bit out in time so that if there is a deep fade or noise burst.www. Outputs of each correlator are weighted to provide a better estimate of the transmitted signal than provided by a single component. receiver collect the time shifted versions of the original signal by providing a separate correlation receiver for M strongest multipath components.5: RAKE receiver. A block interleaver formats the encoded data into a rectangular array of m rows and n columns.

jntuworld. de-interleaver stores the received data by sequentially increasing the row number of each successive bit.
7. Shannon’s channel capacity formula is applicable to the AWGN channel and is given by: C = B log2 1 + S N = B log2 1 + P N0 B = B log2 1 + Eb Rb N0 B (7. are error detection codes. At the receiver. and then clocks out the data row-wise. This has the eﬀect of separating the original source bits by m bit periods. Error correction codes can detect and correct errors.30)
and the ratio C/B is denoted as bandwidth eﬃciency.
7. Shannon showed that by proper encoding of the information.jntuworld.4
Channel Coding
In channel coding. which reduces the bandwidth eﬃciency of the link in high SNR conditions. the data may still be recovered at the receiver without the request of retransmission.com
.29) can be normalized by the bandwidth B and is given as Eb Rb C = log2 1 + B N0 B (7. Equation (7. N0 is the single sided noise power density (W/Hz). Introduction of redundant bits increases the transmission bit rate and hence it increases the bandwidth requirement.1
Shannon’s Channel Capacity Theorem
In 1948. Codes that used to detect errors.29)
where C is the channel capacity (bit/s).www. one word at a time. Eb is the average bit energy and Rb is transmission bit rate. but 143
www. Convolutional interleavers are ideally suited for use with convolutional codes. A channel coder maps the transmitted message into another speciﬁc code sequence containing more bits. redundant data bits are added in the transmitted message so that if an instantaneous fade occurs in the channel. Channel Coding is used by the receiver to detect or correct errors introduced by the channel. Coded message is then modulated for transmission in the wireless channel. P is the received signal power (W). B is the channel bandwidth (Hz).4. The interleaved source data is then read out row-wise and transmitted over the channel. and forming the columns.com
successive bit. errors induced by a noise channel can be reduced to any desired level without sacriﬁcing the rate of information transfer.

Corollary 1 : While dealing within maximum channel capacity. error free performance in not possible (for example.jntuworld. Rc = k/n. while decreasing the bandwidth eﬃciency. 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. introduction of redundant bits increase the transmitter rate and hence bandwidth requirement also increases.com
provides excellent BER performance at low SNR values. Corollary 2 : If data redundancy is not introduced in a wideband noisy environment.www.com
. CDMA communication in 3G mobile phones). A total of n−k redundant bits are added to the k information bits for the purpose of detecting and correcting errors. Block Code 2. The block code is referred to as an (n.4. Concatenated Code. 1. (b) Code Distance (d): Distance between two codewords is the number of ele-
144
www. parity bits are added to blocks of message bits to make codewords or code blocks. The error correction and detection codes are classiﬁed into three groups based on their structure.
7. Parameters in Block Code (a) Code Rate (Rc ): As deﬁned above.jntuworld.2
Block Codes
Block codes are forward error correction (FEC) codes that enable a limited number of errors to be detected and corrected without retransmission. but it also decreases the BER. k information bits are encoded into n code bits. A channel coder operates on digital message (or source) data by encoding the source information into a code sequence for transmission through the channel. 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. In a block encoder. k) code. Convolution Code 3. This leads to the following two inferences.

Cj )} = 2. 11110. (c) Code Weight (w): Weight of a codeword is given by the number of nonzero elements in the codeword. and n = 5.e. 11001 can be used to represent two bit binary numbers as: • 00 – 00000 • 01 – 10100 • 10 – 11110 • 11 – 11001 Here number of codewords is 4.jntuworld.. 11011) is 3. the weight is basically the number of 1s in the codeword. the ﬁrst k bits are identical to the information bits.com
. 10 01 01 00 11 • Next step is to replace each block by its corresponding codeword. here.e. To encode a bit stream 1001010011 • First step is to break the sequence in groups of two bits. Ex 1: The block code C = 00000. (b) Systematic: A systematic code is one in which the parity bits are appended to the end of the information bits. 11110 10100 10100 00000 11001 Quite clearly. the distance is known as ’Hamming distance’.jntuworld. If the code used is binary. Let α1 and α2 be any two elements selected from the alphabet. Cj ).. A linear code must contain the all-zero code word. 10100. i. k = 2.
145
www. then the minimum distance of the code is given by dmin = min {d (Ci . Then the code is said to be linear if and only if α1 C1 + α2 C2 is also a code word. k) block code. Properties of Block Codes (a) Linearity: Suppose Ci and Cj are two code words in an (n. For example weight of a code 101101 is 4. dmin = min {d (Ci .com
ments in which two codewords Ci and Cj diﬀers denoted by d (Ci . If the code ’C’ consists of the set of codewords. Cj )}. For example d(10110. For an (n. i.www. and the remaining n − k bits of each code word are linear combinations of the k information bits. For a binary code. k) code.

all cyclic shifts of C are code words. a ﬁeld is called either a ﬁnite or an inﬁnite ﬁeld. 6.www.. there exists an multiplicative inverse a−1 such that a.. where deg r(x)¡deg b(x). That is.. is introduced to map all symbols (with several bits) to the element of the ﬁnite ﬁeld. pm are the elements of GF(q). In this context.. A polynomial p(x) in F(x) is said to be reducible if p(x)=a(x)b(x). then m is called the degree of the polynomial. 5. p1 .com
(c) Cyclic: Cyclic codes are a subset of the class of linear codes which satisfy the following cyclic shift property: If C = [Cn−1 . it is important to know about Finite Field or Galois Field. that is. + pm xm (7. C0 ] is a code word of a cyclic code.a−1 = 1. b. .. otherwise it is called irreducible. a(b + c) = ab + ac 4.. Let F be a ﬁnite set of elements on which two binary operations – addition (+) and multiplication (. Cn−1 ]. The set of nonzero elements in F is a commutative group under multiplication. For any a in F. The coeﬃcient pm is called the leading coeﬃcient. is also a code word.. Identity elements 0 and 1 must exist in F satisfying a + 0 = a and a. called a polynomial. the remainder. there exists a unique pair of polynomials q(x). Multiplication is distributive over addition. 2. The division algorithm states that for every pair of polynomials a(x) and b(x) in F(x). A ﬁeld with a ﬁnite number of elements (say q) is called a ’Galois Field’ and is denoted by GF(q). F is a commutative group under addition. Depending upon the number of elements in it. such that a(x) = q(x)b(x) + r(x). C0 .1 = a. The examples of inﬁnite ﬁeld include Q (set of all rational numbers). Cn−2 . and c in F. The set F together with the two binary operations is called a ﬁeld if the following conditions are satisﬁed: 1.. C (set of all complex numbers) etc. 3.. A ﬁnite ﬁeld entity p(x).jntuworld. Cn−3 .jntuworld. and r(x). obtained by a cyclic shift of the elements of C.) are deﬁned. If pm is not equal to zero.. R (set of all real numbers). for any three elements a. denoted as deg p(x).. A polynomial is called monic if its leading coeﬃcient is unity.
146
www. the quotient. For any a in F.31)
where the symbol x is called the indeterminate and the coeﬃcients p0 .com
. then [Cn−2 . there exists an additive inverse (−a) such that a + (−a) = 0. A monic irreducible polynomial of degree at least one is called a prime polynomial. A polynomial is a mathematical expression p (x) = p0 + p1 x + . . .

.34)
where c(p) is a polynomial of degree less than n. 0.. Thus the overall codeword is (0. 1. Examples of Block Codes (a) Single Parity Check Code: In single parity check codes (example: ASCII code). . 1. an overall single parity check bit is appended to ’k’ information bits. . . bk ). 0. 0.com
. A typical primitive polynomial is given by p(x) = xm + x + 1.. making the overall codeword: C = (b1 . However. and all even numbered error patterns are not detectable. 147
www. 1... such codes can only detect the error. 0. The parity bit may follow an even parity or an odd parity pattern. 0.k) cyclic code is a factor of pn + 1 and has the form g (p) = pn−k + gn−k−1 pn−k−1 + · · · + g1 p + 1. 0) and encoded with overall even parity pattern.. .jntuworld.. b2 .. it cannot correct the error. The resultant codeword c(p) can be written as c (p) = x (p) g (p) (7.33) (7. then parity check bit: p = b1 + b2 + . 1) where the last bit is the parity bit. 1. Let the information bit word be: (b1 . 1..com
An irreducible polynomial p(x) of degree ‘m’ is said to be primitive if the smallest integer ‘n’ for which p(x) divides xn +1 is n = 2m −1. We would see an application of such codes in Reed-Solomon codes. These codes possess a considerable amount of structure which can be exploited. b2 . All error patterns that change an odd number of bits are detectable. 1....www. A speciﬁc type of code which obeys both the cyclic property as well as polynomial operation is cyclic codes.. Ex. If there is a single error in bit 3: (0. . bk . .jntuworld. x0 ) represents the k information bits.32)
where (xk−1 . A cyclic code can be generated by using a generator polynomial g(p) of degree (n-k). 2: Consider a (8. p). A message polynomial x(p) can also be deﬁned as x (p) = xk−1 pk−1 + · · · + x1 p + x0 (7.7) ASCII code with information codeword (0. + bk modulo 2 is appended at the (k+1)th position. The generator polynomial of an (n. Cyclic codes are a subset of the class of linear codes which satisfy the cyclic property..

After decoding a received codeword. otherwise n = n). pc = P0 . 1). 1. • pe = P2 + P4 + .4) single parity check block code.com
1. n − 1) single parity check block code.jntuworld. say. 1. 0. we can write pc + pe + pf = 1. 1. 0. + Pn (n = n if n is even. 1. 0. for any (n.. pe = P2 + P4 . 1). then the probability of obtaining j errors in this codeword is Pj = n Cj pj (1 − p)n−j .. we get • pc = P0 . (b) Product Codes: Product codes are a class of linear block codes which provide error detection capability using product of two block codes. errors in bit 3 and 5: (0. If in an n-bit codeword. the received codewords at the receiver would then be C1 = [1 0 1 0] 148
www. Now if an error occurs in the second bit of the second codeword. 0. there are j errors and p is the bit error probability. otherwise n = n − 1). then error will not be detected. let pc be the probability that the decoder gives correct codeword C. As an example. These 9 bits can be divided into groups of three information bits and (4. 0. Using this formula. 0. then it can be easily checked by the receiver that now there are odd number of 1’s in the codeword and hence there is an error. 1. 1. • pf = P1 + P3 + . if there are two errors.www. 1. In this case. 1.jntuworld.com
. 0. those can be appended with a vertical parity bit which will form the fourth codeword. and pf is the probability that the decoder fails to give a codeword. for a (5. and pf = P1 + P3 + P5 .3) single parity check codeword can be formed with even parity. On the other hand.. + Pn (n = n − 1 if n is even. 0. After forming three codewords. 0) are to be transmitted. Thus the following codewords are transmitted: C1 = [1 0 1 0] C2 = [0 0 1 1] C3 = [1 1 0 0] C4 = [0 1 0 1].. pe is the probability that the decoder gives incorrect codeword C = C. 1. Consider that nine information bits (1. 1.

once again error is detected and it is decoded as 1 with majority decoding principle. the majority decoding principle is used to obtain the information bit. then error is detected and it is decoded as 0 with majority decoding principle. Such majority scheme works properly only if the noise aﬀects less than n/2 number of bits.1) repetition code each information bit is repeated n times (n should be odd) and transmitted.com
C2 = [0 1 1 1] ← C3 = [1 1 0 0] C4 = [0 1 0 1] ↑ and these would indicate the corresponding row and column position of the erroneous bit with vertical and horizontal parity check. k) = (2m − 1.35)
where k is the number of information bits used to form a n bit codeword. 1 occurs a higher number of times than 0. • If the received codeword is (0 0 0). 3) codeword and a vertical (4. 3) codeword and concatenating them we get a (16. the codeword is (1 1 1). • If the received codeword is (0 1 1) or (1 1 0) or (1 0 1). no error. • For input bit 0. pe = P2 + P3 .1) binary repetition code. if in a group of n received bit.jntuworld. (d) Hamming Codes: A binary Hamming code has the property that (n. a product code can be formed as (n1 . the codeword is (0 0 0) and for input bit 1. it is decoded as 1. Ex 3: Consider a (3.www. Accordingly. (c) Repetition Codes: In a (n. k1 ) & (n2 . and pf = 0.e. • If the received codeword is (0 0 1) or (0 1 0) or (1 0 0). k1 k2 ). k2 ) → (n1 n2 . For such a (3. i. if the received codeword is (1 1 1). 2m − 1 − m) (7. the information bit is decoded as 1. At the receiver. i. Thus. a 149
www. The number of parity symbols are n − k = m. pc = P0 + P1 . • Similarly.1) repetition code. Here we get a horizontal (4. In general. it is decoded as 0. and m is any positive integer. 9) product code. Thus the bit can be corrected.com
.jntuworld. no error.e.

. and it corresponds to e = [0 0 0 0 0 0 0]. p1 + i1 + i2 + i3 = 0 p2 + i2 + i3 + i4 = 0 p3 + i1 + i3 + i4 = 0.in . p1 . to obtain the required codeword. 150
www. 1. The parity bits may follow such a modulo 2 arithmetic: p1 = i1 + i2 + i3 .www.. 4) Hamming code. v7 ]. This is quite a useful code in communication which is illustrated via the following example...com
. Calculate S from decoder input V.com
codeword is represented by C = [i1 . Corresponding to this syndrome vector. v2 . .
Ex 4: Consider a (7. In a nutshell. p2 .. . p3 = i1 + i3 + i4 . A few cases are given below to illustrate the syndrome decoding. we perform the following steps: 1. i2 . Let C = [0 1 1 1 0 1 0] and V = [0 1 1 1 0 1 0]. which is same as. pn−k ]. p2 = i2 + i3 + i4 . p3 ]. 3. v6 . Syndrome Decoding: For this Hamming code. obtain e corresponding to S. Thus. C = V + e = [0 1 1 1 0 1 0]..jntuworld.. let the received codeword be V = [v1 . From syndrome table. The required codeword is then C = V + e.. i4 . .. This implies S = [0 0 0]. there is an error vector e which can be obtained from a syndrome table and ﬁnally the required codeword is taken as C = V + e.. v5 . 2.jntuworld. v4 . p1 .. the syndrome vector is equal to zero. We deﬁne a syndrome vector S as S = [S1 S2 S3 ] S1 = v1 + v2 + v3 + v5 S2 = v2 + v3 + v4 + v6 S3 = v1 + v2 + v4 + v7 It is obvious that in case of no error. The transmitted codeword is then C = [i1 . With three parity bits we can correct exactly 1 error..

we get the primitive polynomial as p(x) = x6 + x + 1. This will be corrected by performing the operation C = V + e.com
2. However. DVD etc. Every codeword lies within distance three of any codeword. rather than individual 0 and 1. For US-CDPD. wireless communications.4) Hamming code can correct only single bit error. This makes S = [0 0 0] and as a result. the RS code is used with m = 6. and the ease of encoding and decoding.). (e) Golay Codes: Golay codes are linear binary (23.com
. For this case. Equating p(x) to 0 implies x6 = x + 1. Block length of these codes is n = 2m − 1.www. one of a kind code in that this is the only nontrivial example of a perfect code. known for their multiple error correcting ability. high speed modems. The table elements continue up to α62 . thus making maximum likelihood decoding possible. and C = V + e implies the double error cannot be corrected. However.jntuworld. from which we get e = [0 0 0 1 0 0 0] which means a single bit error is there in the received bit v4 . It’s coding system is based on groups of bits. e = [0 0 0 0 0 0 0]. Minimum distance dmin = 2e + 1. let C = [0 1 0 1 1 0 0] and V = [0 0 1 1 1 0 1] (two errors at second and third bits). 3. So each of the 64 ﬁeld elements is represented by a 6 bit symbol. Let C = [1 1 0 0 0 1 0] and V = [1 1 0 1 0 1 0]. digital TV. C = V . Therefore a (7. and can be extended to 2m or 2m + 1. such as bytes.1. It’s block length is n = 2m − 1 for m ≥ 3 and number of errors that they can correct is bounded by t < (2m − 1)/2.jntuworld. This feature makes it particularly good at dealing with burst of errors: six consecutive bit errors. Typical application areas are storage devices (CD. (f) BCH Codes: BCH code is one of the most powerful known class of linear cyclic block codes. The 6 bit representation of the ﬁnite ﬁeld elements is given in Table 7. (g) Reed Solomon (RS) Codes: Reed-Solomon code is an important subset of the BCH codes with a wide range of applications in digital communication and data storage. and it achieves the largest possible dmin of any linear code. Another interesting case is. to follow linearity property there should be
151
www. Binary BCH codes can be generalized to create classes of non binary codes which use m bits per code symbol. This is a special. Number of parity symbols that must be used to correct e errors is n − k = 2e. This means S = [0 1 1].12) codes with a minimum distance of seven and a error correction capability of three bits.

The details of such a syndrome calculation can be found in [1]. 0 .. + C2t x2t . .. On the decoder side one has to ﬁnd a speciﬁc r(x) = p(x) or vice-versa. Shift register contains ’N’ k-bit stages
152
www..jntuworld.
a zero codeword. + C0 . then the codeword polynomial c(x) = g(x)p(x) and the entire process becomes easier. 1 . A convolutional code is generated by passing the information sequence through a ﬁnite state shift register. α1 0 0 0 . This process is given below. 0 0 0 . we take a generating polynomial g(x) such that d(x) = g(x)q(x) + r(x) where q(x) is the quotient polynomial and r(x) is the remainder polynomial. . hence α63 is assigned zero. . 0 . . .jntuworld. but due to its complexity. α4 0 0 0 .. α6 = α + 1 .3
Convolutional Codes
A continuous sequence of information bits is mapped into a continuous sequence of encoder output bits. . . The encoding part of the RS polynomial is done as follows: Information polynomial: d(x) = Cn−1 xn−1 + Cn−2 xn−2 + . it is mainly done using syndrome calculation. 0 .
7. . so a generating polynomial is used instead. Since this kind of codes are cyclic codes. Parity polynomial: p(x) = C2t−1 x2t−1 + . . . Information polynomial is then the multiple of generating polynomial. . α3 0 0 1 . α2 0 1 0 .com
Table 7.. 0 .1: Finite ﬁeld elements for US-CDPD α5 1 α1 α2 . Since generating an information polynomial is diﬃcult.. α0 1 0 0 . The codeword polynomial would then be given as: c(x) = g(x)q(x) + r(x) = p(x). . Codeword polynomial: c(x) = d(x) + p(x).com
.www. If we assign a parity polynomial p(x) = r(x). .4. . 1 .

. The path information between the states. for an input message sequence of 01011 . The shift register of the encoder is initialized to all-zero-state before
Figure 7. Input data is shifted into and along the shift register.6: A convolutional encoder with n=2 and k=1. denoted as b/c1 c2 .7: State diagram representation of a convolutional encoder. . k-bits at a time. and m linear algebraic function generators based on the generator polynomials. The state diagram is simply a graph of the possible states of the encoder and the possible transitions from one state to another.. Convolution codes may be represented in various ways as given below. . so the code rate Rc = k/n.com
. It is easy to verify that encoded sequence is 00 11 10 00 01 . State Diagram: Since the output of the encoder is determined by the input and the current state of the encoder. encoding operation starts.com
Figure 7. represents input information bit ’b’ and the corresponding 153
www.jntuworld. . Number of output bits for each k-bit user input data sequence is n bits.www.jntuworld. a state diagram can be used to represent the encoding process.

Again. Tree Diagram: The tree diagram shows the structure of the encoder in the form of a tree with the branches representing the various states and the outputs of the coder. the encoded sequence can be directly read from the tree. 10.com
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Figure 7. 00.9: Trellis diagram of a convolutional encoder.8: Tree diagram representation of a convolutional encoder. Given an input sequence. 01). The encoded bits are labeled on the branches of the tree.www.jntuworld.
Figure 7. 00. 10. 01).
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www. As an example. it is not diﬃcult to verify from the state diagram that an input information sequence b = (1011) generates an encoded sequence c = (11.jntuworld. output bits (c1 c2 ). an input sequence (1011) results in the encoded sequence (11.

we discuss a popular parallel concatenated code. namely.jntuworld.
7. The interleaver is a device that permutes the data sequence in some predetermined manner. This means that the two nodes having the same label can be merged.com
Figure 7. An interleaver separates the two component encoders. turbo code.www. we obtain another diagram called a Trellis Diagram which is more compact representation. The ﬁrst encoder outputs the systematic V0 and recursive convolutional V1 sequences while the second encoder discards its systematic sequence and only outputs the recursive convolutional V2 sequence.4. Below. Trellis Diagram: Tree reveals that the structure repeats itself once the number of stages is greater than the constraint length. It can be of two types: serial and parallel codes.10: Block diagram of a turbo encoder. Depending on the number of input bits to a component encoder it 155
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. as the systematic output from the other component encoder is only a permuted version of the chosen systematic output.10 shows the block diagram of a turbo encoder using two identical encoders. Only one of the systematic outputs from the two component encoders is used to form a codeword. It is observed that all branches emanating from two nodes having the same state are identical in the sense that they generate identical output sequences. By doing this throughout the tree diagram. Turbo Codes: A turbo encoder is built using two identical convolutional codes of special type with parallel concatenation.4
Concatenated Codes
Concatenated codes are basically concatenation of block and convolutional codes. An individual encoder is termed a component encoder. Figure 7.jntuworld.

) and M. nevertheless. Singapore: Pearson Education.jntuworld. If the component encoders are not identical then it is called an asymmetric turbo code. Inc. Encoders are also categorized as systematic or non-systematic. have taken place in modern wireless communication. S. The eﬀectiveness of the discussed topics would be more clear in the next chapter in the context of diﬀerent multiple access techniques.www.jntuworld. 2002. NY: Oxford University Press. Johnson (Jr. J. 2002. Gravano. 3.com
. G. BLAST architectures etc. space-time block coding scheme. Rappaport. 2001. Introduction to Error Control Codes.
7. Wireless Communications: Principles and Practice. R. 2nd ed. Treichler.5
Conclusion
Although a lot of advanced powerful techniques for mitigating the fading eﬀects such as space diversity in MIMO systems. R. C. 2. T. Larimore.
7. MIMO equalization. the discussed topics in this chapter are the basic building blocks for all such techniques and that stems the necessity for all these discussions.. New Delhi: PHI.com
may be binary or m-binary encoder.
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www. S.6
References
1. Theory and Design of Adaptive Filters.

And this must be done in a way such that the quality of service doesn’t degrade within the existing users.com
Chapter 8
Multiple Access Techniques
Multiple access techniques are used to allow a large number of mobile users to share the allocated spectrum in the most eﬃcient manner. There are several diﬀerent ways to allow access to the channel.1
Multiple Access Techniques for Wireless Communication
In wireless communication systems it is often desirable to allow the subscriber to send simultaneously information to the base station while receiving information from the base station.jntuworld. to handle as many calls as possible in a given bandwidth with a suﬃcient level of quality of service. so the sharing is required to increase the capacity of cell or over a geographical area by allowing the available bandwidth to be used at the same time by diﬀerent users.jntuworld. The main aim in the cellular system design is to be able to increase the capacity of the channel i.com
.e.
8. A cellular system divides any given area into cells where a mobile unit in each cell communicates with a base station. These includes mainly the following: 1) Frequency division multiple-access (FDMA) 2) Time division multiple-access (TDMA) 3) Code division multiple-access (CDMA)
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www.www. As the spectrum is limited.

each channel is shared using TDMA.
8. and frequency selective fades occur only in a small fraction of the signal bandwidth
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www. Digital Cellular: Japanese Digital Cellular: CT2 Cordless Telephone: Digital European Cordless Telephone: U.1.jntuworld.S. there generally are a large number of channels allocated using either FDD or TDD.TDMA and CDMA are the three major multiple access techniques that are used to share the available bandwidth in a wireless communication system. Depending on how the available bandwidth is allocated to the users these techniques can be classiﬁed as narrowband and wideband systems. In narrow band FDMA. For narrow band TDMA. The channels are operated using FDD.1.www.S. a user is assigned a particular channel which is not shared by other users in the vicinity and if FDD is used then the system is called FDMA/FDD. Thus. thus separating a small number of users in time on a single channel. The available spectrum is divided in to a large number of narrowband channels.com
Table 8. Narrow band TDMA allows users to use the same channel but allocated a unique time slot to each user on the channel.com
.1
Narrowband Systems
The term narrowband is used to relate the bandwidth of the single channel to the expected coherence bandwidth of the channel. the transmission bandwidth of a single channel is much larger than the coherence bandwidth of the channel. multipath fading doesnt greatly aﬀect the received signal within a wideband channel. Such systems are called TDMA/FDD and TDMA/TDD access systems. Narrowband Spread Spectrum (IS-95): FDMA/FDD TDMA/FDD TDMA/FDD TDMA/FDD FDMA/TDD FDMA/TDD CDMA/FDD
4) Space Division Multiple access (SDMA) FDMA.
8.2
Wideband Systems
In wideband systems.jntuworld.1: MA techniques in diﬀerent wireless communication systems Advanced Mobile Phone Systems: Global System for Mobile: U.

8.1: The basic concept of FDMA.2
Frequency Division Multiple Access
This was the initial multiple-access technique for cellular systems in which each individual user is assigned a pair of frequencies while making or receiving a call as shown in Figure 8. the spectrum cannot be reassigned as long as a call is in place. One frequency is used for downlink and one pair for uplink. The FDMA channel carries only one
If an FDMA channel is not in use.com
. Even though the user may not be talking. This is called frequency division duplexing (FDD). FDMA is usually 159
www. After the assignment of the voice channel the BS and the MS transmit simultaneously and continuously.jntuworld. That allocated frequency pair is not used in the same cell or adjacent cells during the call so as to reduce the co channel interference.jntuworld.com
Figure 8. The bandwidths of FDMA systems are generally narrow i.1.www.e. Diﬀerent users can use the same frequency in the same cell except that they must transmit at diﬀerent times. then it sits idle and
it cannot be used by other users to increase share capacity. The features of FDMA are as follows: phone circuit at a time.

www.
8. the handset scans the available channels and locks on to an unoccupied channel for the duration of the call. FDMA requires tight ﬁltering to minimize the
8. Multiple or simultaneous users are accommodated in AMPS by giving each user a unique signal.2. and on the reverse channel from the mobile unit to the base station. adjacent channel interference.
8. FDMA and TDMA systems are less vulnerable. Voice signals are sent on the forward channel from the base station to the mobile unit.com
implemented in a narrow band system The symbol time is large compared to the average delay spread. A single user occupies a single channel while the call is in progress. and the single channel is actually two simplex channels which are frequency duplexed with a 45 MHz split.3
FDMA and Near-Far Problem
The near-far problem is one of detecting or ﬁltering out a weaker signal amongst stronger signals. the worst case to consider is recovery of a weak signal in a frequency slot next to strong signal. the call is split into time blocks that alternate between transmitting and receiving.2.S. analog cellular system.2.com
. CT2 system splits the available bandwidth into radio channels in the assigned frequency domain. In contrast. the weak signal could be lost in the
160
www. FDMA systems oﬀer diﬀerent kinds of solutions to near-far challenge. In AMPS. Using TDD(Time Division Duplexing ). Since both signals are present simultaneously as a composite at the input of a gain stage.2
FDMA/TDD in CT2
Using FDMA. the gain is set according to the level of the stronger signal. AMPS (Advanced Mobile Phone System) is based on FDMA/FDD.jntuworld.1
FDMA/FDD in AMPS
The ﬁrst U. The near-far problem is particularly diﬃcult in CDMA systems where transmitters share transmission frequencies and transmission time.jntuworld. The complexity of the FDMA mobile systems is lower than that of TDMA mobile systems. When a call is completed or when a handoﬀ occurs the channel is vacated so that another mobile subscriber may use it. In the initial call setup. analog narrowband frequency modulation (NBFM) is used to modulate the carrier. Here.

TDMA is a complimentary access technique to FDMA. TDMA has an advantage that is possible to allocate diﬀerent numbers of time slots per frame to diﬀerent users.www.
8.2.3
Time Division Multiple Access
In digital systems. each in its 161
www. Data transmission in TDMA is not continuous but occurs in bursts.3. Because of a discontinuous transmission in TDMA the handoﬀ process is much simpler for a subscriber unit. The number of channels are less. GSM uses a variation of TDMA along with FDD. Thus bandwidth can be supplied on demand to diﬀerent users by concatenating or reassigning time slot based on priority. since it is able to listen to other base stations during idle time slots. inter channel interference is almost negligible.jntuworld. GSM digitizes and compresses data. GSM is widely used in Europe and other parts of the world. the entire bandwidth is available to the user but only for a ﬁnite period of time.1
TDMA/FDD in GSM
As discussed earlier. Global Systems for Mobile communications (GSM) uses the TDMA technique.jntuworld.com
noise ﬂoor. TDMA uses diﬀerent time slots for transmission and reception. This results in low battery consumption since the subscriber transmitter can be turned OFF when not in use. The features of TDMA includes the following: TDMA shares a single carrier frequency with several users where each users makes use of non overlapping time slots. available bandwidth etc. continuous transmission is not required because users do not use the allotted bandwidth all the time. This type of duplexing is referred to as Time division duplexing(TDD). TDMA uses diﬀerent time slots for transmission and reception thus duplexers are not required. TDMA requires careful time synchronization since users share the bandwidth in the frequency domain. Even if subsequent stages have a low enough noise ﬂoor to provide
8. as shown in Figure 8. In TDMA. then sends it down a channel with two other streams of user data. In most cases the available bandwidth is divided into fewer channels compared to FDMA and the users are allotted time slots during which they have the entire channel bandwidth at their disposal.com
. In such cases. The number of time slots per frame depends on several factors such as modulation technique.

jntuworld.www. It operates at either the 900 MHz or 1800 MHz frequency band. DECT provides 10 FDM channels in the band 1880-1990 Mhz.com
.2: The basic concept of TDMA. Each channel supports 12 users through TDMA for a total system load of 120 users. Since many GSM network operators have roaming agreements with foreign operators.com
Figure 8.
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www. own time slot.2
TDMA/TDD in DECT
DECT is a pan European standard for the digitally enhanced cordless telephony using TDMA/TDD. users can roam over from cell to cell as long as they remain within the range of the system. users can often continue to use their mobile phones when they travel to other countries. DECT antenna can be equipped with optional spatial diversity to deal with multipath fading.jntuworld.3. DECT supports handover.
8.

In DS-SS. the receiver must know the codeword used by the transmitter. which diﬀerentiates them from each other (shown in Figure 8. There are.com
. However since many users can share the same spread spectrum bandwidth without interfering with one another.www. the same bandwidth is occupied by all the users.jntuworld.2
Code Division Multiple Access
In CDMA.
8. CDMA utilize a spread spectrum technique in which a spreading signal (which is uncorrelated to the signal and has a large bandwidth) is used to spread the narrow band message signal.1
Frequency Hopped Multiple Access (FHMA)
This is a digital multiple access system in which the carrier frequencies of the individual users are varied in a pseudo random fashion within a wideband channel.4
Spread Spectrum Multiple Access
Spread spectrum multiple access (SSMA) uses signals which have a transmission bandwidth whose magnitude is greater than the minimum required RF bandwidth. There are two main types of spread spectrum multiple access techniques: Frequency hopped multiple access (FHMA) Direct sequence multiple access (DSMA) or Code division multiple access (CDMA).
8. 163
www. however they are all assigned separate codes.jntuworld. Each user is given his own codeword which is orthogonal to the codes of other users and in order to detect the user. SSMA is not very bandwidth eﬃcient when used by a single user. however.4.com
8. The digital data is broken into uniform sized bursts which is then transmitted on diﬀerent carrier frequencies. spread spectrum systems become bandwidth eﬃcient in a multiple user environment. A pseudo noise (PN) sequence converts a narrowband signal to a wideband noise like signal before transmission. Direct Sequence Spread Spectrum (DS-SS) This is the most commonly used technology for CDMA.3). the message signal is multiplied by a Pseudo Random Noise Code.4. two problems in such systems which are discussed in the sequel.

3: The basic concept of CDMA.2288 Mbps.www. the frequency range is: 869-894 MHz (for Rx) and 824-849 MHz (for Tx). the bit rate is 1. For each channel. CDMA/FDD in IS-95 In this standard. wherein for reasonable time or frequency guardbands. In such a system. respectively. The delays cause the spreading sequences of the diﬀerent users to lose their orthogonality. For orthogonality.3
CDMA and Self-interference Problem
In CDMA. nonzero contributions to that user’s signal arise from the transmissions of the other users in the network. it usually combines 64 Walsh-Hadamard codes and a m-sequence. self-interference arises from the presence of delayed replicas of signal due to multipath.com
.
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Figure 8.jntuworld. 164
www.jntuworld. This is distinct from both TDMA and FDMA. as by design they are orthogonal only at zero phase oﬀset. orthogonality of the received signals can be preserved. Hence in despreading a given user’s waveform.4. there are a total of 20 channels and 798 users per channel.

In CDMA. the near-far eﬀect combined with imperfect orthogonality between codes (e.com
8. this is not a problem since mutual interference can be ﬁltered. One of the advantages using this technique is they avoid near-far eﬀect. Therefore the strong signal from the nearby transmitter will mask the weak signal from the remote transmitter.jntuworld. Hybrid FDMA/CDMA (FCDMA): An alternative to the CDMA technique in which the available wideband spectrum is divided into a smaller number of sub spectra with smaller bandwidths.5
Hybrid Spread Spectrum Techniques
The hybrid combinations of FHMA.
8. In this scheme the required bandwidth need not be contiguous and diﬀerent user can be allotted diﬀerent sub spectrum bandwidths depending on their requirements.4
CDMA and Near-Far Problem
The near-far problem is a serious one in CDMA. This problem arises from the fact that signals closer to the receiver of interest are received with smaller attenuation than are signals located further away. Accurate and fast power control appears essential to ensure reliable operation of multiuser DS-CDMA systems. Hybrid Direct Sequence/Frequency Hopped Multiple Access Techniques (DS/FHMA): A direct sequence modulated signal whose center frequency is made to hop periodically in a pseudo random fashion is used in this technique. However.jntuworld.com
. The smaller sub channels become narrow band CDMA systems with processing gain lower than the original CDMA system.4. CDMA and SSMA result in hybrid spread spectrum techniques that provide certain advantages. frequency hopped CDMA systems are not adaptable to the soft handoﬀ process since it is diﬃcult to synchronize the frequency hopped base station receiver to the multiple hopped signals.4. due to diﬀerent time sifts).www. In each cell. The capacity of this hybrid FCDMA technique is given by the sum of the capacities of a system operating in the sub spectra. only one user per cell is allotted a particular time slot. These hybrid techniques are explained below.g. In TDMA and FDMA. Thus at any 165
www. however. Time and Code Division Multiple Access (TCDMA): In this TCDMA method diﬀerent cells are allocated diﬀerent spreading codes. leads to substantial interference.

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time only one user is transmitting in each cell. This technique has the advantage in severe multipath or when severe channel interference occurs. TDMA and CDMA. namely.
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.jntuworld. in which case TDMA or CDMA is employed. When diﬀerent areas are covered by the antenna beam.
8. We have. where the hopping sequence is predeﬁned and the subscriber is allowed to hop only on certain frequencies which are assigned to a cell. we have mainly discussed the ﬁxed assignment type of MA techniques. thus avoiding a severe fade or erasure event on a particular channel. TCDMA also avoids near-far eﬀect as the number of users transmitting per cell is one. for diﬀerent frequencies FDMA can be used. A more advanced approach can further increase the capacity of the network. When a handoﬀ takes place the spreading code of that user is changed to the code of the new cell. The radiated power of each user is controlled by Space division multiple access. SDMA serves diﬀerent users by using spot beam antenna.5
Space Division Multiple Access
SDMA utilizes the spatial separation of the users in order to optimize the use of the frequency spectrum. frequency can be re-used.6
Conclusion
In this chapter. however. FDMA. This limits the number of cells a region can be divided into and hence limits the frequency re-use factor. Therefore it becomes imperative to use other multiple access techniques in conjunction with SDMA.jntuworld. In a practical cellular environment it is improbable to have just one transmitter fall within the receiver beam width. The subscriber can hop to a new frequency at the start of a new TDMA frame. This technique would enable frequency re-use within the cell.www. These areas may be served by the same frequency or diﬀerent frequencies. However for limited co-channel interference it is required that the cells be suﬃciently separated. intensionally not covered the reservation-based MA schemes such as packet reservation MA or polling
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www. Time Division Frequency Hopping (TDFH): This technique has been adopted for the GSM standard. A primitive form of SDMA is when the same frequency is reused in diﬀerent cells in a cellular wireless network.

The main idea to discuss only the basic MA techniques has been to grow up a fair idea about the resource sharing in a wireless media when there are many users.www. Rappaport. NY: McGraw Hill.
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. Singapore: Pearson Education.. S. D. Inc. NJ: Prentice Hall. K. 4. Upper Saddle River.
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www. McGillem. 3. Modern Communications and Spread Spectrum. 2000. NY: McGraw Hill. Cooper and C. 1995. 1986.jntuworld. keeping the QoS view point in mind. 2nd ed.jntuworld. Wireless Digital Communications: Modulation and Spread Spectrum Applications. 4th ed. J. G. 2. R. Digital Communications. Wireless Communications: Principles and Practice. Feher.7
References
1. 2002.com
or token passing etc. G. Proakis. T. The readers are encouraged to go through the advanced topics once they ﬁnish reading the discussed 8 chapters in this lecture notes.