design and development of a hybrid tdma/cdma mac protocol

Design and Development of a Hybrid TDMA/CDMA

MAC Protocol for Multimedia Wireless Networks

A Thesis

Submitted For the Degree of

Doctor of Philosophy

in the Faculty of Engineering

by

D. Rajaveerappa

Department of Electrical Communication Engineering

INDIAN INSTITUTE OF SCIENCEBangalore – 560 012

April 2004

Acknowledgements

I would like to express my deep sense of gratitude to my research guide and teacher Prof.

(Dr.) Pallapa Venkataram, Professor, Dept. of ECE, I. I. Sc., Bangalore, for his continued

support and valuable guidance. I want to thank him for giving me the opportunity to work

on this research topic with more interest. I cannot thank him enough for his meticulous

and painstaking efforts to guide me and given me a lot of suggestions throughout my

research period.

I thank Prof. (Dr.) Anurag Kumar, Chairman, Dept. of ECE, I. I. Sc., Bangalore,

for providing me the facilities to work in his Department and his helps to me for the

completion of this work. I also thank Prof. (Dr.) Vinod Sharma, Asso. Chairman, ECE,

for his support in this regard.

I thank Prof. (Dr.) A. Selvarajan, Emiritus Professor (formerly Chairman of ECE

Department, I. I. Sc., Bangalore), for his valuable advice, support, and help during my

research period. I thank Prof. (Dr.) G. V. Anand, formerly Chairman, Dept. of ECE, I.

I. Sc., Bangalore, for providing me the facilities to work in his Department and his helps

to me for the completion of this work.

I thank my teachers, Prof. (Dr.) V. U. Reddy, Dept. of ECE, Prof. (Dr.) Kumar

N. Sivarajan, and Prof. (Dr.) M. K. Ghosh, Dept. of Mathematics for having given me

the knowledge in the field of Wireless Communications and Probability Theory. I thank

Prof. (Dr.) Utpal Mukherjee for his guidance in my course works at the beginning of my

research works.

I thank my teachers and examiners, Prof. (Dr.) Anurag Kumar, Chairman, Dept.

of ECE, Prof. (Dr.) Lawrence Jenkins, Dept. of EE, Prof. (Dr.) Jamadhagni, Dept.

i

Acknowledgements ii

of CEDT, for helping me to continue my research works further. I thank Prof. (Dr.)

T. S. Vedavathy and Prof. (Dr.) A.P. ShivaPrasad, Dept. of ECE, for their constant

encouragements to complete my research works. I thank all the teachers of the Department

of ECE. I also thank the administrative staffs of Dept. of ECE in this regard.

I thank Prof. (Dr.) Balakrishnan, Divisional Chairman for Information Sciences for

providing me the computing facility at SERC. I also thank Prof. (Dr.) S. M. Rao, Chair-

man of SERC, in this regard. I thank Prof. (Dr.) Pandian, Dept. of Civil Engg., formerly

CCE Chairman, for providing me the financial assistance and the hostel accommodation

for a better study at this campus. I thank Prof. (Dr.) P. Venkataram, QIP Coordinator

and CCE Chairman, Prof. (Dr.) Y. V. Venkatesh, Dean of Engg., Prof. (Dr.) V. M. H.

Govinda Rao, (formerly Dean of Engg., and QIP Coordinator), and Prof. (Dr.) D. K.

Subramaniam, Dept. of CSA, (formerly Dean of Engg.), for their academic support in

my research.

I thank our Honourable Director of I. I. Sc., Prof. (Dr.) Govardhana Mehta, Divisional

Chairmen, Deans, Council of Wardens, Registrars and other administrative staffs for their

support in my research. I thank our former Honourable Director of I.I.Sc., Prof. (Dr.)

Padmanabhan, for granting me admission to this research. I thank Dr. Panneer Selvam,

Deputy Registrar (Academic) for his valuable advice and help to finish this research work.

I thank Dr. Bharathi Venkataram, M.B.B.S., M.D., Mr. Gautham, and Mr. Manu for

their concern about my health during my stay at I.I.Sc. I thank my friend and colleagues

Mr. Abdul Gaffur, Mr. Mathi and my teachers Prof. (Dr.) Abdul Sattar and Prof. (Dr.)

T.G. Palanivelu for the constant support in my research works.

I thank Mr. Chetan Kumar, Engineer, WIPRO, Dr. Mainak Chatterjee, Mr. Sonal

Ranjan, Mr. Babu, Mr. Gopi Garge, ERNET LAB., Mr. Avinash Shenoy, TIFR, Mr.

Kiran, SERC, Mr. Madan, SERC, Mr. Gundu Rao, SERC, Ms. Sarala, SERC, and other

staffs of SERC for their helps in providing computing facility for me at SERC. I thank

Dr. RadhaKrishna Rao, my QIP colleague, and my friends Dr. Jimson Mathew, Mr.

ArunKumar M.C., Samsung, Dr. Muthuvel, Dr. Pinto George, Mr. Sridharan, Rutgers

University, and Mr. Alavi (CUSAT) for their moral support in this regard.

Acknowledgements iii

I thank my friends Mr. Anindya, Mr. Gowri, Mr. Mahanty, Mr. Salil, Mr. Nelson,

Dr. Mani, Mr. Appa Rao, Mr. Rayala Ravi, Mr. Srinivas (and his parents), Mr. Prabhu,

Mr. Nazar, Mr. Elango, Mr. Anoop Sharma, Mr. Rajesh Reddy, Mr. K.N.S. Reddy, Dr.

Saji Varghese, Mr. Jayanta Madhu, Mr. Sushil, Mr. Saju, Mr. Benson, Mr. Partha, Mr.

Ajay Kumar Naik, Mr. Shantanu, Mr. Joseph, Mr. Joe Mathai, Mr. Dennis, and Mr.

Srinivas for keeping me in a family environment during my research period.

I thank Prof. (Dr.) C.S. Sridhar (formerly HOD of DOE, CUSAT, Cochin), Prof.

(Dr.) K. Vasudevan, Prof. (Dr.) P.R.S. Pillai, Prof. (Dr.) K.T.Mathew, Prof. (Dr.)

K.G. Balakrishnan, HOD, DOE, CUSAT, Cochin, Dr. Tessamma Thomas, Prof. (Dr.)

Paulose Jacob, and Dr. Wilson for their encouragements to complete my research.

I thank former Honourable Vice Chancellors of Cochin University, Prof. (Dr.) K.G.

Adiyodi (Late) and Prof. (Dr.) Babu Joseph for providing me the required study leave

for the purpose of this research. I thank the present Honourable Vice Chancellor, Prof.

(Dr.) UnniKrishnan Nair for his continued support and help to finish my research. I

thank the other administrative staffs of Cochin University, Cochin, in this regard.

I thank AICTE, New Delhi, for providing me the necessary financial assistance for my

research at I.I.Sc., Bangalore.

I thank ETRI, Seoul, South Korea, for providing me financial assistance to visit their

place for the purpose of presentation of my research paper.

I thank all my teachers and professors who reviewed my research papers and this thesis

work.

I thank Prof. (Dr.) Samuelson, Prof. (Dr.) Jaya Bhaskar, Prof. (Dr.) Prabhakaran,

Dr. James Jacob, Dr. Santhosh Abraham Paul, Dr. Joy Kuri, Prof. K.L. Sebastian,

Prof. (Dr.) Lawrence Jenkins, Prof. (Dr.) Joy Thomas, Prof. (Dr.) Mohan Vasu, and

the Pastors and Brothers who paryed for my success in this research.

Finally, I thank all the members of Protocol Engineering and Technology Lab., Dept.

of ECE, I.I.Sc., and other friends who have helped me directly or indirectly in connection

with this research.

Abstract

A wireless local area network (WLAN) provides high bandwidth to users in a limited

geographical area. This network faces certain challenges and constraints that are not

imposed on their wired counterparts. They are: frequency allocation, interference and

reliability, security, power consumption, human safety, mobility, connection to wired LAN,

service area, handoff and roaming, dynamic configuration and the throughput. But the

wireless medium relies heavily on the features of MAC protocol and the MAC protocol

is the core of medium access control for WLANs. The available MAC protocols all have

their own merits and demerits.

In our research works, we propose a hybrid MAC protocol for WLAN. In the design, we

have combined the merits of the TDMA and CDMA systems to improve the throughput

of the WLAN in a picocellular environment. We have used the reservation and polling

methods of MAC protocols to handle both the low and high data traffics of the mobile

users. We have strictly followed the standards specified by IEEE 802.11 for WLANs to

implement the designed MAC protocol.

We have simulated the hybrid TDMA/CDMA based MAC protocols combined with

RAP (Randomly Addressed Polling) for Wireless Local Area Networks. We have de-

veloped a closed form mathematical expressions analytically for this protocol. We have

also studied the power control aspects in this environment and we derived a closed form

mathematical expressions analytically for this power control technique.

This hybrid protocol is capable of integrating different types of traffic (like CBR,

VBR and ABR services) and compiles with the requirements of next-generation systems.

The lower traffic arrival is dealt with the Random Access and the higher traffic arrival

iv

Abstract v

is with the Polling methods. This enables us to obtain higher throughput and low-

mean delay performance compared to the contention-reservation-based MAC schemes.

The protocol offers the ability to integrate different types of services in a flexible way

by the use of multiple slots per frame, while CDMA allows multiple users to transmit

simultaneously using their own codes. The RAP uses an efficient ”back-off” algorithm to

improve throughput at higher arrival rates of user’s data. The performance is evaluated

in terms of throughput, delay, and rejection rate using computer simulation.

A detailed simulation is carried out regarding the maximum number of users that

each base station can support on a lossy channel. This work has analyzed the desired

user’s signal quality in a single cell CDMA (Code Division Multiple Access) system in

the presence of MAI (Multiple Access Interference). Earlier power control techniques

were designed to assure that all signals are received with equal power levels. Since these

algorithms are designed for a imperfect control of power, the capacity of the system is

reduced for a given BER (Bit-Error Rate). We proposed an EPCM (Efficient Power

Control Mechanism) based system capacity which is designed for the reverse link (mobile

to base station) considering the path loss, log-normal shadowing and Rayleigh fading.

We have simulated the following applications for the further improvement of the per-

formance of the designed MAC protocol:

• Designed protocol is tested under different traffic conditions.

• The protocol is tested for multimedia traffic under application oriented QoS require-

ments.

• Buffer Management and resource allocation.

• Call Admission Control (hand-offs, arrival of new users).

• The adaptability to the variable nature of traffic.

• The propagation aspects in the wireless medium.

The proposed MAC protocol has been simulated and analysed by using C++/MATLAB

Programming in IBM/SUN-SOLARIS UNIX environment. The results were plotted using

Abstract vi

MATLAB software.

All the functions of the protocol have been tested by an analysis and also by simulation.

• Call admission control function of the protocol has been tested by simulation and

analysis in a multimedia wireless network topology and from analysis we found that

at low traffic the throughput is high and at high traffic the throughput is kept

constant at a reasonable high value. The simulation results also justify/coordinate

the analysis results.

• Dynamic channel allocation function of the protocol was tested and analysed and

the coordinated results show that at low traffic, high throughput and at high traffic

the throughput is constant.

• Buffer management function of the protocol simulation shows the results that the

packet loss can be controlled to a minimum by adjusting the buffer threshold level

at any traffic conditions.

• Maintenance of data transfer during the hand-offs function was simulated and the

results show that the blocked calls are less during low traffic and at high traffic the

blocked calls can be kept constant at low value.

Thus, the proposed model aimed at having high throughput, high spectral efficiency, low

delay, moderate BER and moderate blocking probability.

We have considered a pico cell with a maximum of several users and studied the

power efficiency of combined channel coding and modulation with perfect power controlled

CDMA system. Thus our simulation of the ”software radio” has flexibility in choosing the

proper channel coders dynamically depending upon the variations of AWGN channel.

Contents

Acknowledgements i

Abstract iv

1 Introduction 1

1.1 Multimedia Wireless Networks (MWNs) . . . . . . . . . . . . . . . . . . . 5

1.1.1 Multimedia Service Requirements . . . . . . . . . . . . . . . . . . . 8

1.2 Wireless Problems in MWNs . . . . . . . . . . . . . . . . . . . . . . . . . . 9

1.3 MAC in Wireless Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . 10

1.3.1 OSI Layered Architecture . . . . . . . . . . . . . . . . . . . . . . . 11

1.3.2 The MAC Sublayer . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

1.4 Proposed MAC Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15

1.5 Organization of the rest of the Thesis . . . . . . . . . . . . . . . . . . . . . 15

2 MAC Protocols for Multimedia Wireless Networks 16

2.1 MAC Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16

2.1.1 Carrier Sense Multiple Access (CSMA) - Random Access (RA) . . . 17

2.1.2 Polling Based MAC Protocols . . . . . . . . . . . . . . . . . . . . . 20

2.1.3 FDMA Based MAC Scheme . . . . . . . . . . . . . . . . . . . . . . 23

2.1.4 TDMA Based MAC Scheme . . . . . . . . . . . . . . . . . . . . . . 24

2.1.5 DS-SS (CDMA) Based MAC Scheme . . . . . . . . . . . . . . . . . 25

2.2 Requirements of MWN’s MAC Protocol . . . . . . . . . . . . . . . . . . . 28

2.3 Some of the Existing Works on MWN’s MAC . . . . . . . . . . . . . . . . 33

vii

CONTENTS viii

3 Design of the Proposed MAC Protocol 36

3.1 System Model for the Proposed MAC Protocol . . . . . . . . . . . . . . . . 37

3.2 Frame Structure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

3.2.1 The Random Access (RA): TDMA/CDMA . . . . . . . . . . . . . . 39

3.2.2 The Random Access Polling : RAP . . . . . . . . . . . . . . . . . . 41

3.3 Calculation of Throughput and Delay . . . . . . . . . . . . . . . . . . . . . 42

3.4 Analytical Model for the Proposed MAC Protocol . . . . . . . . . . . . . . 44

3.4.1 Random Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44

3.4.2 Random Access Polling . . . . . . . . . . . . . . . . . . . . . . . . . 46

3.5 Power Control for the Proposed MAC Protocol . . . . . . . . . . . . . . . 48

3.5.1 Definitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50

3.5.2 The Correlation Receiver for CDMA System . . . . . . . . . . . . . 51

3.5.3 Analytical equation and performance analysis for EPCM . . . . . . 52

3.5.4 Analytical Model for the Power Control . . . . . . . . . . . . . . . . 55

4 Functions of the Proposed MAC Protocol 57

4.1 Multiuser and Multistream Admission Control . . . . . . . . . . . . . . . . 57

4.1.1 CAC Algorithm . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

4.2 Channel Allocation based on the QoS Requirements of the Multimedia

Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

4.3 Buffer Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70

4.4 Maintenance of Data Transfer during the Hand-offs . . . . . . . . . . . . . 75

5 Simulation 82

5.1 Simulation and Results of the Power Control Algorithm . . . . . . . . . . . 87

5.2 Results of the Admission Control of the Protocol . . . . . . . . . . . . . . 91

5.3 Results of Channel Allocation Scheme of the Protocol . . . . . . . . . . . . 93

5.4 Results of Buffer Management Scheme . . . . . . . . . . . . . . . . . . . . 94

5.5 Results of Maintenance of Data Transfer during the Hand-offs . . . . . . . 95

CONTENTS ix

6 Testing of Protocol in Software Radio Environment 96

6.1 Software Radio and its Layered Architecture . . . . . . . . . . . . . . . . . 97

6.2 Source Encoding and Decoding Methods . . . . . . . . . . . . . . . . . . . 99

6.3 Channel Encoding and Decoding Techniques . . . . . . . . . . . . . . . . . 100

6.4 TDMA/CDMA System and Results . . . . . . . . . . . . . . . . . . . . . . 107

7 Conclusions 118

7.1 Future Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

Bibliography 120

List of Tables

1.1 Main parameters of different existing and emerging access technologies . . 4

4.1 Downlink Structure Matrix . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

4.2 Uplink Resource Management Matrix . . . . . . . . . . . . . . . . . . . . . 67

5.1 Simulation parameters used in MAC Power Control . . . . . . . . . . . . . 87

x

List of Figures

1.1 Bi-directional radio access system as a function of the data rate, range and

mobility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

1.2 Seamless system network including a variety of access technologies . . . . . 5

1.3 Comparison of FDMA, TDMA and CDMA technologies . . . . . . . . . . . 7

1.4 The OSI Reference Model . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

1.5 Classification of Multiple Access Protocols . . . . . . . . . . . . . . . . . . 14

2.1 Performance of MAC Protocols: ALOHA, CSMA/CA, CSMA/CD and

TOKEN PASSING . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18

3.1 The infrastructure WLAN network used for the simulation of the proposed

MAC protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

3.2 The TDMA/CDMA Frame Structure . . . . . . . . . . . . . . . . . . . . . 40

3.3 The RAP Frame Structure . . . . . . . . . . . . . . . . . . . . . . . . . . . 41

3.4 The analytical model (Markov Model) for the performance analysis to cal-

culate the throughput and the delay. . . . . . . . . . . . . . . . . . . . . . 46

3.5 The Correlation Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52

3.6 Multipath fading due to reflection . . . . . . . . . . . . . . . . . . . . . . . 54

4.1 Illustration of Dynamic Resource Allocation for Multiplexing VBR Streams 62

4.2 Intra and Inter-bunch Links . . . . . . . . . . . . . . . . . . . . . . . . . . 63

4.3 The Zone Concept . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64

4.4 Downlink Co-channel Interference Case . . . . . . . . . . . . . . . . . . . . 66

xi

LIST OF FIGURES xii

4.5 MS Uplink Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68

4.6 Dynamic Threshold Vs. Total Queue Length . . . . . . . . . . . . . . . . . 71

4.7 Simulation Model of BS Switch for Buffer Management . . . . . . . . . . . 73

4.8 A Cellular Hand-off System Model . . . . . . . . . . . . . . . . . . . . . . 76

4.9 Hand-off Management Scheme . . . . . . . . . . . . . . . . . . . . . . . . . 78

5.1 Performance Curves for the Combined Protocol . . . . . . . . . . . . . . . 84

5.2 Performance Curves for the Combined Protocol . . . . . . . . . . . . . . . 85

5.3 Performance Curves for the Combined Protocol obtained from the mathe-

matical analysis. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86

5.4 Analytical results for 16 Users in 0 to 1.0 Watt Tx. Power Range . . . . . 88

5.5 Analytical results for 16 Users in 0 to 1.0 µW Tx. Power Range . . . . . . 89

5.6 Power control analysis under no fading conditions . . . . . . . . . . . . . . 89

5.7 Power control analysis under fading conditions . . . . . . . . . . . . . . . . 90

5.8 BER analysis under no fading conditions . . . . . . . . . . . . . . . . . . . 90

5.9 BER analysis under fading conditions . . . . . . . . . . . . . . . . . . . . 91

5.10 New Call Blocking Probability Performance with fading of channel . . . . . 92

5.11 New Call Blocking Probability Performance without fading of channel . . . 92

5.12 Channel Blocking Probability without Power Control . . . . . . . . . . . . 93

5.13 Channel Blocking Probability with Power Control . . . . . . . . . . . . . . 93

5.14 Packet Loss Probability Vs. Buffer Threshold Optimization Parameter . . 94

5.15 Packets Loss Vs. Buffer Threshold . . . . . . . . . . . . . . . . . . . . . . 94

5.16 Hand-off Call Blocking Probability Performance with fading of channel . . 95

5.17 Hand-off Call Blocking Probability Performance without fading of channel 95

6.1 Software Radio Block Diagram . . . . . . . . . . . . . . . . . . . . . . . . . 98

6.2 Software Radio Layering Model . . . . . . . . . . . . . . . . . . . . . . . . 98

6.3 A/D Converter in Software Radio . . . . . . . . . . . . . . . . . . . . . . . 99

6.4 Encoder for a (2,1) convolution code . . . . . . . . . . . . . . . . . . . . . 106

6.5 Syndrome calculation and error-correction for the convolution code . . . . 106

LIST OF FIGURES xiii

6.6 Speech Signal of user 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

6.7 Power Control for uncoded BPSK modulation with different noise variance

σN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

6.8 Power Control for block coded BPSK modulation with different noise vari-

ance σN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109

6.9 Power Control for cyclic coded BPSK modulation with different noise vari-

ance σN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109

6.10 Power Control for convolution coded BPSK modulation with different noise

variance σN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110

6.11 Power Control for different channel coding shemes with BPSK modulation

with noise variance σN = 1.0 . . . . . . . . . . . . . . . . . . . . . . . . . . 110









6.16 BER for user 1 with uncoded BPSK modulation when only one user is in

the cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

6.17 BER for user 1 with uncoded BPSK modulation when seven users are in

the cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

6.18 BER for user 1 with block coded BPSK modulation when only one user is

in the cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114

6.19 BER for user 1 with block coded BPSK modulation when seven users are

in the cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114

6.20 BER for user 1 with cyclic coded BPSK modulation when only one user is

in the cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115

LIST OF FIGURES xiv

6.21 BER for user 1 with cyclic coded BPSK modulation when seven users are

in the cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115

6.22 BER for user 1 with convolution coded BPSK modulation when only one

user is in the cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116

6.23 BER for user 1 with convolution coded BPSK modulation when seven users

are in the cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116

Chapter 1

Introduction

After the first launch of analog mobile communication systems, better known as ”first-

generation” system, there has been an explosive increase in the number of mobile users

in the last two decades. The first generation systems are based on analog technology and

provide voice services. ”Second generation” systems (like the TDMA - based systems

GSM, IS-54, ANSI-136 as well as the CDMA - based system IS-95) use digital technology

for similar services as analog systems. These systems have been extended to provide

higher data rate services and packet transmissions. To this end, GSM will incorporate

advanced schemes as High Speed Circuit Switched Data (HSCSD), General Packet Radio

Service (GPRS), and Enhanced Data Rates for GSM Evolution (EDGE) to increase the

data rates that are available for the users [1].

The first version of ”third generation” mobile radio systems called International Mobile

Telecommunications - 2000 (IMT-2000) has been standardized world wide (IMT-2000 in

ITU and UMTS in Europe). Their main goals are to support broadband data services

of upto 2 Mbps via a wideband radio interface and to enable international roaming for

circuit-switched and packet-oriented services. IMT-2000 supports time division duplex

(TDD) and frequency division duplex (FDD) to enable symmetric and asymmetric data

services in a spectrum efficient way. Third generation systems address a mass market for

mobile multimedia communications and enhanced multimedia services (please refer to fig.

1.1).

1

Chapter 1. Introduction 2

Mobile multimedia communication will be the concept towards a fourth generation

wireless communication in which users will want to use a wide range of different inter-

active multimedia services in personal and professional applications such as teleworking,

teleshopping, emergency services, telemaintenance, telemedicine, virtual reality, identifi-

cation, support for the handicapped, interactive training or personal television. These

services must be available to the users at anytime, from anywhere and in anyform.

Fourth Generation mobile communications (4G mobile) involve a mix of concepts and

technologies in the making. Some can be recognized as derived from 3G and are called

”evolutionary” (e.g., evolutions of WCDMA (Wideband CDMA) and cdma2000), while

others involve new approaches to wireless mobile and are sometimes labeled ”revolution-

ary” (e.g., OFDM (Orthogonal Frequency Division Multiplexing)/WCDMA or converged

broadband wireless core). The converged broadband wireless system is nothing but the to-

tal convergence of wireless mobile and wireless access communications. The future issues

in this aspect are on spectrum efficiency, dynamic bandwidth allocation, secured wireless

applications, improved quality of service, revolutionary digital transceiver technologies,

and so on.

Due to the dominating role of mobile radio access, it is expected that more handsets

than PCs (Personal Computers) will be connected to the Internet around 2004. Therefore,

mobile terminals will be the major man-machine interface instead of the PC. Given the

dominating role of IP-based data traffic, future wireless communication networks have

to be designed for economic packet data transfer. The expected data services lead to

high data rates requirements for future systems. Therefore, research activities on wireless

communication systems beyond the third generation have already started in many parts

of the world. Please refer to table 1.1.

Future wireless communication systems beyond third generation will be characterised

by a horizontal communications model between different access technologies to the user

terminals such as cellular, cordless, WLAN type systems, short range connectivity, broad-

cast systems, and wired systems. They will be integrated on a common, flexible, and


GSM

DECT

UMTS

IEEE 802.11BRANHiperlan 2

BRANFixed Urban

Hiper AccessBlue Tooth

Total data rate per cell

Mobility & Range

0.5 Mbps 2 Mbps 20 Mbps 155 Mbps

High SpeedVehicularRural

VehicularUrban

PedestrianIndoor

Personal Area

Figure 1.1: Bi-directional radio access system as a function of the data rate, range andmobility

expandable platform to complement each other in an optimum way and to satisfy differ-

ent service requirements in a variety of radio environments. These access systems will be

connected to a seamless IP-based core network. Please refer to fig. 1.2.

Users will have a single number, e.g., a telephone number, a PIN (Personal Identi-

fication Number) number, or an IP address, for all access technologies. A new media

access system (generalized access network) connects the core network to the appropriate

access technology. It also contain mobility management. Global roaming is required for

all access technologies. Other key requirements include the internetworking between those

different access systems in terms of horizontal (intra-sytem) and vertical (inter-system)

handover as well as seamless services with service negotiations including mobility, secu-

rity, and quality of service. They will be handled in the newly developed media access

system and in the core network. The media access system connects each access system to

the common core network.


Table 1.1: Main parameters of different existing and emerging access technologies

Systems Data rates Technology Range Mobility Frequency Application

GSM,GPRS, 9.6 Kbps to TDMA, upto high 900,1800 public,HSCSD,EDGE 384 Kbps FDD 35 Km 1900 MHz privateIMT-2000, Max. WCDMA, 30 m to high 2 GHz -do-UMTS 2 Mbps CDMA 20 KmDECT Max. TDMA upto low 1880 to office,

1.694 Mbps 500 m 1900 MHz residenceBluetooth Max. TDMA, 0.1 to low 2.4 to cable,

721 Kbps FH 100 m 2.4835 GHz small officeHiperlan 2 Max. OFDM, 50 to low 5.15 to corporate,

25 Mbps TDMA 300 m 5.35 GHZ publicIEEE802.11 Max. OFDM, 50 to low 5.47 to -do-

25 Mbps CSMA 300 m 5.725 GHzDAB 1.5 OFDM ≤ 100 high 176 to audio

Mbps Km 230 MHz broadcastDVB 5 to OFDM ≤ 100 medium below video

31 Mbps Km to high 860 MHz broadcastCable Modem ≤ 40 FDD 5 to No 60 to residential

Mbps 20 Km 860 MHzADSL ≤ 6.144 DMT 2 to No Base Band home,

Mbps no carrier 6 Km office

In addition to existing emerging radio access technologies, new radio interfaces can also

be integrated into such a seamless network architecture. Research on new radio interfaces

has already started to increase the data rates of third generation mobile radio systems

by an order of magnitude in order to reach at least 10 - 20 Mbps in a cellular environ-

ment and 2 Mbps for moving vehicles. Moreover, the cost per transmitted bit should be

reduced significantly. Candidate multiple access techniques include CDMA (Code Divi-

sion Multilple Access). This is capable of MAI (Multiple Access Interference) elimination

and ISI (Inter Symbol Interference) suppression, irrespective of the encountered wireless

frequency selective channels [2, 3].


connect

IP Based Core Network

Media Access System

Laptop

Cell phone

Desktop PC

Fixed Telephone

ApplicationsServices &

IMT−2000UMTS

WLAN Type

To other entities

Short

range

Cellular GSM

DABDVB

Wireline DSL

New Radio Interfaces

−ions

Figure 1.2: Seamless system network including a variety of access technologies

1.1 Multimedia Wireless Networks (MWNs)

The desire to free communications from physical connection together with increasing de-

mand for new wireless services, such as internet connections, has motivated tremendous

efforts to develop Multimedia Wireless Networks. The term ”mutimedia” is referred to

the information expressed in multiple media such as: text, voice, video, graphics, im-

age and audio with different source characteristics (data rates, activity ratio, burstiness)

and different quality requirements (BER) in application such as teleconferencing, med-

ical imaging, entertainment and educational video, and advertising. It also reflects the

fact that the transmission of these informations may occur in several different channel

types (indoor, outdoor, cellular, microcellular) with different transmission characteristics.

Broadband networks based on ATM and IP will provide the integration of telecommu-

nications and computer communications for the provision of multimedia services to the

user. The third generation mobile systems should enable these multimedia services also


to be offered to users. These mobile systems include Universal Mobile Telecommunication

Systems (UMTS) and International Mobile Telecommunications - 2000 (IMT-2000). The

personal and terminal roaming are supported by terminals and networks in future mobile

systems from which Service Providers can create a wide range of mobile multimedia ap-

plications.

A number of proposals are currently under development for future generation mobile and

personal communication systems. These systems will bring with them the potential to

provide a wide range of services operating in a wide range of wireless environments. It

is widely acknowledged that different radio interfaces for the diffferent transmission en-

vironments that are expected to be encountered. Indeed the pan-European program on

Research and Development in Advanced Communications in Europe (RACE) suggests

four different radio interfaces for the portable terminals for home, office, vehicular and

public environments. However, within each of these radio interfaces, will be the need to ac-

commodate sources with different data rates, burstiness and quality requirements. There

are wireless access options like (FDMA (Frequency Division Multiple Access), CDMA

(Code Division Multiple Access), and TDMA (Time Division Multiple Access)) available

to meet these requirements [4, 5].

For multimedia applications, it is important to know the maximum bit rate achievable for

the multiple access techniques in various channel types. One common feature for various

mobile channels is the multipath effect. It can cause severe ISI (Inter Symbol Interfer-

ence) and hence restricts the maximum data transmission rate. With the use of equalizers,

symbol rates of several times the coherence bandwidth are possible for FDMA, TDMA

systems to reduce ISI. For CDMA systems, a correlation receiver will be used which com-

bines signals from several paths. The time varying property of the multipath channel can

be estimated in a better way in a CDMA receiver than in FDMA and TDMA due to the

wider bandwidth (more information) used.

When the source is bursty it is desirable to make the channel resource available to other

users during silence periods, or at the very least to ensure that interference generated by

the silence is minimized. With FDMA and TDMA systems, it is easy to detect silence


periods and efficiently switch off the transmitter at these times. However, in general, it is

not possible to reallocate the channel to another user until the transmission is completely

terminated. One important advantage of CDMA is that it allows the channel resource to

be reallocated to other users. In this case, this is achieved indirectly by the fact that the

reduced interference allows other users (with different codes) to access the systems and

still maintain an acceptable inteference level (this is sometimes referred to as statistical

multiplexing). But there are overloads in the case of CDMA. So, a minimum transmission

bit rate greater than zero is required in CDMA to maintain synchronization during silence

periods.

In all multiple access schemes transmission quality in any given channel and transmitter

quality can be traded (until the error floor is reached). Similarly, channel coding and in

case of CDMA, processing gain can be used to adjust bit error rate (BER) performance.

In case of FDMA and TDMA, some bandwidth adjustment will be required [6].

N: number of mobile users

. . . .

Time

Frequency

Code

FB

N1 2 3 . . . . .

.

Time

Frequency

Code

12

NTS

FDMAFB: Frequency Bands

TDMATS: Time Slots Time

Frequency

Code

12

N

DSCodes

CDMADS: Direct Sequence

Figure 1.3: Comparison of FDMA, TDMA and CDMA technologies

Adjustment of transmitter power to achieve the desired transmission quality has serious

implications for frequency reuse (cellular) schemes with a pre-assigned frequency plan

(most FDMA, TDMA cellular systems) because interference levels are changed and the


given reuse distances may no longer be sufficient. In the case of CDMA, adjustment of

transmitter power for a given user does impact on overall system capacity, but since this

capacity has a soft limit, significant disruption is less likely. Please refer to fig. 1.3.

1.1.1 Multimedia Service Requirements

A typical distributed multimedia environment in which multiple, remotely located users

participate in a joint work or design project demand for:

• resource sharing to integrate information on a more global basis and ensure the use

of system’s available information

• multimedia data integration such as images, graphics, sound, text, and structural

data, to present information in a more immediate and understandable form

• local intelligence and autonomy to perform tasks independently of centralised system

• graphical interfaces to reduce training costs and assist occasional and inexperienced

users, and

• vendor independence to achieve free from any specific hardware vendor.

A multimedia environment, presently consists of applications accessing pages, facsim-

ile, answering machines, telephone lines, speech synthesis, and digital recording and play

back. Its key contributions are the integration of multimedia into a cohesive nomadic

information infrastructure and a graceful transition from desktop to nomadic locals. This

integration is at the service and user interface levels [7, 8].

It is important to develop new and more highly integrated nomadic computing platforms

capable of employing emerging digital wireless communication networks, but it is also

important to learn from experiments how these devices might be used. QoS (Quality of

Service) is a set of user-perceived attributes of which makes a service what it is. It is

expressed in user understandable language and manifests itself as a number of parame-

ters, all of which either subjective or objective values. Subjective QoS just determines the

degree of satisfaction of the users from the specific service. On the other hand, objective


QoS can be the present parameters whose values are definite and can be measured by the

providers as well as by the service user.

With respect to communications, we identify two objects, ”remote operation” and ”mul-

timedia transport system”. The concept of remote operation is already defined in the OSI

application layer and also well understood in the internet world. The QoS parameters

play an important role and the multimedia transport system is one of the main actors in

the system.

1.2 Wireless Problems in MWNs

The salient features of multimedia wireless network environment may be described as the

”3M”s environment: ”real-time Multimedia, Multihop, and Mobile”. In the past, there

have been studies and implementations of systems that combined any two of these ”M”s,

but not all three. For example, real- time Multimedia and Multihop have been studied

in certain satellite system (e.g., Iridium); real-time Multimedia and Mobile are pervasive

in cellular radio system and Berkeley’s dynamic hand-off with ”hints” for multimedia,

mobile, cellular networks; and Multihop and Mobile were well studied in the 1970s Ad-

vanced Research Projects Agency (ARPA) packet radio project. A well known wireless

communication problem is the reception of the same transmissions at multiple bases. The

problem is apparent with the infrared networks, where a zone may have a number of bases.

Transmission for both types of badge contain a sequence number, allowing duplicates to

be filtered at any level in the system. Now add to this the fact that the mobile users

would typically be moving in a lower bandwidth, error-prone environment, where fading

problems are quite common.

Research in the area of wireless communications, as reported, has tended to focus either

on physical layer issues such as CDMA versus TDMA and multiple access protocols for

the wireless environment or on higher layer control isssues such as call hand-off, hierar-

chical cell design, and Dynamic Channel Allocation (DCA) among cells for the voice-call


environment. Relatively, little has been reported on control issues arising when the wire-

less networks must carry multimedia traffic, or interface with much higher speed wired

networks designed to carry such traffic.

The solution for these problems in MWNs is the evolution towards future communica-

tion systems is strongly influenced by increasing transmission capacities, changing service

requirements and emerging ATM technologies. The next generation high performance lo-

cal and metropolitan area networks have to be designed including control schemes which

provide high throughput while simultaneously supporting real-time services. The slotted

ring architecture enchanced by spatial slot reuse is one potential approach for data trans-

mission and admission control in distributed Giga bit networks. The efficient medium

access control protocols are proposed to maintain fairness and bounded access delays and

are designed for networks operating as Gbits LAN as well as for access control to interface

users of ATM systems on a shared medium.

1.3 MAC in Wireless Networks

More recently there has been a growing interest to provide wireless access to applica-

tions that are typically of Local Area Networks giving rise to the concept of Wireless

Local Area Networks (WLANs). In general, wireless systems as a whole is divided into

distinct coverage areas called cells. Individual radio carrier frequencies are assigned to

each of the cells from a set of available frequencies in a manner such that the adjoining

cells are not on the same frequency. This minimizes the interference between cells while

allowing the same frequency to be used many times in a given system giving rise to the

concept of frequency reuse. Each cell is identified as by a base station that serves all

the mobile users in that cell. It is not possible to guarantee the users of a cell to have

good propagation conditions among themselves all the time. Hence, it is the function of

the base station to establish connectivity by allocating resources, maintain connectivity,

route data packets and monitor all kinds of activity in the cell. In wireless networks,

sharing of the spectrum is not only limited but also inherently expensive. So, a need for a


multiple access protocol (MAC) arises whenever a resource is shared by a number of inde-

pendent users. So, one of the main consideration in the design of the wireless networks is

to incorporate multiple access schemes that make efficient use of the allocated bandwidth.

1.3.1 OSI Layered Architecture

Layering, or layered protocol architecture, is a form of hierarchical modularity that is

central to data network design. The most popular of these is the Open Systems In-

terconnection (OSI) which was developed as an international standard by International

Standard Organization (ISO). The OSI model has seven layers. The first layer is the

physical layer which provides virtual link for transmitting a sequence of bits between any

pair of nodes joined by a physical communication channel. The second layer is the data

link control (DLC) layer. The purpose of this layer is to convert unreliable bit pipe in the

physical layer to error-free packets and communicate asynchronously in both directions.

In some networks, particularly local area networks, some or all the communications take

place over multiple access links. For these links, the signals received at one node is a

function of the signals from a multiplicity of transmitting nodes, and the signal from one

node might be heard at a multiplicity of other nodes. The appropriate layers for mul-

tiaccess communication are somewhat different from those in networks of point-to-point

links. There is still a need for a DLC layer to provide a virtual error-free packet link to

higher layers, and there is still a need for a physical layer to provide a bit-pipe. However,

there is also a need for intermediate layer to manage the multiaccess link so that frames

can be sent by each node without constant interference from the other nodes. This is

called Medium Access Control (MAC). Please refer to fig. 1.4.

1.3.2 The MAC Sublayer

The MAC sublayer is usually considered as the lower sublayer of DLC layer. The ser-

vice provided by the MAC to the DLC is that of an intermittent synchronous bit-pipe.

The function of the MAC is to allocate the multiaccess channel so that each node can


Medium Access Control

Application Layer

Presentation Layer

Session Layer

Transport Layer

Network Layer

Data Link Layer

Physical Layer

Logical Link Control

Figure 1.4: The OSI Reference Model

successfully transmit its frames without undue interference from other nodes. We can

view multiaccess communication in queuing terms. Each node has a queue of packets

to be transmitted and the multiaccess channel is a common server. Ideally, the server

should view all the waiting packets as one combined queue to be served by the appropriate

queueing discipline. Unfortunately, the server does not know which node contain packets

and which do not. Thus, the interesting part of the problem is that knowledge about the

state of the queue is distributed [9-19].

Need for MAC Protocols:

Whenever there are more than one independent user, trying to access the same resources

at the same time, conflicts can occur resulting in corruption of data packets of all the con-

tending users. It is not always possible to allocate resources to individual users because

the resource is scarce and expensive. So, a need for multiple access (MA) protocol arises.

One could avoid the need for a multiple access protocol by letting each user have its own

resources. But this is not generally done because the resources are not only scarce but

also expensive.

Properties of MAC Protocols:


The design of a MAC protocol depends upon the environment and the kind of require-

ment. In spite of the specific properties of the MAC protocol for a particular environment,

all good MAC protocols should posses the following properties:

1. The protocol must control the way in which the nodes access the channel by requiring

that the nodes confirm to certain rules.

2. The protocol must allocate the channel efficiently. The efficiency is measured in

terms of channel throughput and delay of transmissions.

3. The allocation should be fair towards individual nodes, i.e., not being partial to any

node.

4. The protocol should be capable of supporting various kinds of traffic which will have

different bit rates.

5. The protocol should be stable, i.e., it should maintain equilibrium.

6. The protocol should be robust with respect to equipment failure and changing con-

ditions.

Classification of MAC Protocols:

The MAC protocols are classified into three main groups. These are contentionless proto-

cols, contention protocols and the class of CDMA protocols. The contentionless protocols

are those in which all the nodes are scheduled in a fixed fashion and each node is allocated

a part of the resource. It can so happen that the channel is allocated to the nodes which

have something to transmit. In contention protocols the nodes cannot be sure that a

transmission will collide or not because other nodes might be transmitting at that time.

So, these protocols must resolve the conflicts. The CDMA protocols do not belong to

either the contentionless or the contention protocols. As long as the number of interfering

users is not too large, the signal-to-noise ratio will be large enough to extract the desired

signal without error. Then in this case the protocol behaves as a contentionless protocol.

However if the number of users rises above a certain limit, the interference becomes too


large for a desired signal to be extracted and contention occurs, thus making the protocol

interference limited. Therefore, the protocol is basically contentionless unless too many

users access the same channel at the same time. This is why we place the CDMA protocols

between the contentionless and contention protocols. Please refer to fig. 1.5.

CSMA...

Polling, RAP,Token Passing,....

ALOHAs−ALOHA

Contentionless (S) Contention

CDMA

(RA)

MULTIPLE ACCESS PROTOCOLS

RARepeated

RAReservedFixed

AssignedDemandAssigned

FDMATDMA ...

ImplicitExplicit .....

S: Scheduling

RA: Random Access

Figure 1.5: Classification of Multiple Access Protocols

Performance Evaluation:

Multiple access schemes are evaluated according to various criteria. The characteristics

that are desirable are high bandwidth utilization and low message delays. But there are

other attributes which need to be considered. The ability of the protocol to simultaneously

support traffic of different priorities, variable message lengths, variable bit rates, different

delay constraints, is essential as higher bandwidth utilization is achieved by multiplexing

all traffic types. The MAC protocols are of different types, which differ by the static or

dynamic nature of bandwidth allocation algorithm, centralized or distributed nature of

decision making and the degree of adaptivity of the algorithm to the changing needs.


1.4 Proposed MAC Protocol

In our work, we have considered a wireless local area network (WLAN) to provide high

bandwidth to multimedia users in a limited geographical area. This network faces certain

challenges and constraints that are not imposed on their wired counterparts. They are:

frequency allocation, interference and reliability, security, power consumption, human

safety, mobility, connection to wired LAN, service area, handoff and roaming, dynamic

configuration and the throughput. But the wireless medium relies heavily on the features

of MAC protocol and the MAC protocol is the core of medium access control for WLANs.

The available MAC protocols all have their own merits and demerits.

This work proposes a hybrid MAC protocol for MWN. In the design, we have combined

the merits of the TDMA and CDMA systems to improve the throughput of the WLAN

in a picocellular environment. We are using the reservation and polling methods of MAC

protocols to handle both the low and high data traffics of the mobile users. We have

strictly followed the standards specified by IEEE 802.11 for WLANs to implement this

MAC protocol. We have simulated the Hybrid TDMA/CDMA based MAC protocols

combined with RAP (Randomly Addressed Polling) for Wireless Local Area Networks.

We have developed a closed form mathematical expressions analytically for this protocol.

We have also studied the power control aspects in this environment and derived a closed

form mathematical expressions analytically for this power control technique.

1.5 Organization of the rest of the Thesis

The rest of the thesis is organized as follows: chapter 2 describes about the MAC proto-

cols for Multimedia Wireless Networks, chapter 3 gives the complete details about the

design of our proposed MAC protocol, chapter 4 discusses about the various applica-

tions related to the proposed protocol, chapter 5 describes the simulation of the proposed

protocol, performance analysis of the proposed protocol, and the results of both simula-

tion and performance analysis, chapter 6 gives a detail about the testing of protocol in

software radio environment. Finally, chapter 7 concludes this thesis.

Chapter 2

MAC Protocols for Multimedia

Wireless Networks

Since our work is aimed at developing a MAC protocol, in this chapter we briefly discuss

some of the existing MAC protocols.

2.1 MAC Protocols

The medium access control relies heavily on the features of the multiple access protocol

and hence that the protocol is the core of the medium access control for wireless networks.

Although there are many proposed multiple access protocols for wireless networks, those

meeting the features can generally be classified as CSMA, polling, TDMA and CDMA.

The general characteristics of these multiple access protocols are summarized as below

and the uplink multiple access, taking mobility into performance consideration are dis-

cussed [20, 21].

16

Chapter 2. MAC Protocols for Multimedia Wireless Networks 17

2.1.1 Carrier Sense Multiple Access (CSMA) - Random Access

(RA)

Due to the popularity of Ethernet, CSMA is attractive to vendors and researchers. CSMA

is a member of the ALOHA (it is a name given by the researchers of Hawaii University,

USA) family protocols. ALOHA is applied to large-scale wireless networks. The original

version called pure ALOHA, has an unstable peak throughput around 18 percent. An im-

proved version, called slotted ALOHA, requires transmission to fall into synchronous slots

and so reduces the possible collision duration. Slot synchronization increases throughput

to 36 percent. Again, this is an unstable peak value. CSMA, which senses the status of

channel before transmitting, is the simplest way to improve ALOHA. This gives increased

throughput and therefore, the IEEE 802.3 committee chose 1-persistent CSMA with col-

lision detection (CSMA/CD) as the MAC for wired LANs.

The success of CSMA/CD in Ethernet relies on the ease of sensing the carrier by mea-

suring the current or voltage in the cable. This is the primary reason that CSMA has

been successfully applied in wired networks even though it was originally designed for

radio networks. Despite advances in technology, carrier sensing is still a major problem

for radio networks due to the hidden terminal problem. Reliable carrier (or transmission)

sensing is extremely difficult due to severe channel fading in indoor environments and the

use of directional antennas.

A second concern about CSMA is its instability. As shown in many research works,

ALOHA and CSMA protocols are not stable. This problem is seen by looking at the

delay curves. Delay increases roughly in proportion to the offered load. However, near

the peak throughput, the delay suddenly increases with a tremendous slope whose exact

value depends on the total number of users. This situation makes peak throughput a

meaningless number and requires networks to operate at low offered load. This is the rea-

son why IEEE 802.3 picked 1-persistent rather than non-persistent CSMA. Furthermore,

measurements on practical Ethernets show operating throughput around only 30 to 35

percent rather the 80 percent peak performance. Please refer fig. 2.1.


0 1 2 3 4 5 6 7 8 9 100

0.2

0.4

0.6

0.8

1

G(Offered Load)

S(T

hro

ughput)

MAC Protocol Performance

Pure ALOHA Slotted ALOHACSMA

0 1 2 3 4 50

0.2

0.4

0.6

0.8

1

G(Offered Load)

S(T

hro

ughput)

Pure ALOHA Slotted ALOHA CSMA NP−CSMA/CA 1P−CSMA/CA Slotted NP−CSMA/CASlotted 1P−CSMA/CASlotted NP−CSMA/CD

10−1

100

101

0

0.2

0.4

0.6

0.8

1

a (Propagation Time/Frame Transmission Time)

U (

Util

izatio

n)

MAC Protocol Performance

TOKEN(N=2) TOKEN(N=5) TOKEN(N=10) CSMA/CD(N=2) CSMA/CD(N=5) CSMA/CD(N=10)

0 5 10 15 20 250.2

0.4

0.6

0.8

1

N (Number of Active Stations)

U (

Util

izatio

n)

TOKEN(a=0.1) TOKEN(a=0.5) TOKEN(a=1.0) CSMA/CD(a=0.1)CSMA/CD(a=0.5)CSMA/CD(a=1.0)

Figure 2.1: Performance of MAC Protocols: ALOHA, CSMA/CA, CSMA/CD and TO-KEN PASSING

Finally, the another concern about these protocols are spatial domain factors. Infras-

tructure LANs typically have a cellular structure, with base station (or an access point)

taking charge of one cell so that seamless service to mobile nodes is supported. Conse-

quently, interference from other cells (most likely neighboring cells) may influence MAC

protocol operation, and this intruduces spatial domain effects. The possibility that this

situation occurs is not small in a severe, unpredictable fading channel such as indoor radio

channels. The probability can be estimated from the system design and fading statistics.

A good example to demonstrate this factor is slotted ALOHA [22, 23].

CSMA is better than slotted ALOHA, but the slotted non-persistent CSMA is better

than CSMA. In this, each mobile node has a probability ’p’ to ”hear” two base stations

(in a two-cell interference model) when it is in an overlapped region. Since the strength

of a received signal is not a simple function of geographical distance, especially in indoor

fading channels, the fading and wave propagation effects can be considered. The nodes

in different cells may sense each others’ carrier. Assume a mobile node in a cell has a

probability ’q’ to sense the carriers (transmissions) in another cell. Finally, the hidden

terminal must be considered. This effect is stochastically modelled by the probability

’Ph’ that a node becomes a hidden terminal of another. The reason to use stochastic


modelling is to best represent the mobility and channel dynamics. The analytical results

based on the above parametric model yield numerical results shown in fig. 2.1. It is

observed that the throughput is reduced to 40 to 45 percent even with small ’Ph’ and ’p’.

But in practical wireless LANs, interference may come from several cells. Because of this

instability, CSMA may not be attractive for wireless networks.

To alleviate the problem and to increase reliability, a modification of CSMA - CSMA

with collision avoidance (CSMA/CA) - is possibly adopted. However, in general, unless

associated with polling or constrained token passing or handshaking, it is believed useless

since fading usually lasts for a time equal to a few symbols at the data rate of LANs. The

protocol that represents the current focus of IEEE 802.11 committee, uses CSMA/CA

with a four way handshake to combat indoor channels [24].

The basic concept is that every ready user sends a request-to-send (RTS) to the access

point using CSMA/CA. When the access point gets an RTS from a user, the access point

polls that user by sending clear-to-send (CTS). The ready user then sends its packets to

the access point. After correctly receiving the packet, the access point sends a positive

acknowledgement (PACK) to the user. The performance analysis of this four way hand-

shake CSMA/CA protocol is done as follows: assume that the (RTS)-(CTS) transmission

cycle occupies a normalized (to packet transmission) time ’b’ and that the polling is per-

fectly reliable and contains no overhead. A non-persistent CSMA is adopted in the (RTS)

transmission while the random backoff time to achieve the upper bound on throughput.

Therefore, collision of RTS packets is the only possible loss for the protocol. It is also

stochastically modelled as: the hidden terminal with probability ’Ph’, joint region frac-

tion ’p’, and the interference from a node in another cells with probability ’q’. Then, the

throughput is around 0.42 when Ph = 0.2, higher than that of the original non-persistent

CSMA with the same hidden terminal probability. Although this is not a significant in-

crease (10 percent), it is still a success to combat the hidden terminal problem caused by

fading channels. However, it pays a price in the case where there is no hidden terminal.

The throughput reduces to 63 percent, a substantial reduction from the original CSMA,

the cause is of course the (RTS)-(CTS) overhead. Another problem associated with this


protocol is opertaion in huge multicell wireless LANs since the CSMA can be influenced

by the interference [25-29].

2.1.2 Polling Based MAC Protocols

As previously discussed, the distributed random access protocols do not perform well for

multiple access in wireless networks, a some kind of centralized protocols can be consid-

ered here. Token passing among the mobile nodes is obviously not feasible since loss of

token is very likely in fading channels, especially in wireless networks that do not guar-

antee a fully connected or well connected network topology.

The next possibility is to use polling, since typical wireless networks have an infrastructure

that connects base station to serve mobile nodes under their coverage. Such centralized

multiple access control has robustness in the presence of fading and dynamic channel

characteristics. Therefore, it has been proposed as a frame structure for medium access

control of wireless LANs. However, it is hard for polling to achieve high efficiency in

spite of its good reliability. Base stations using polling typically require a hand-shaking

procedure and a strict registration process for mobile nodes which implies a restricted

roaming service. Also, even short packets for hand-shaking require an overhead for car-

rier and timing recovery. Such an overhead never occupies a short duration especially in

the severe fading channels.

Randomly Addressed Polling (RAP):

It seems that a multiple access protocol for wireless LANs must be a compromise between

centralized and decentralized protocols to enjoy the advantages of both without suffering

from their primary disadvantages. Therefore, a family of new multiple access control pro-

tocols, called Randomly Addressed Polling (RAP) are proposed, which can be considered

as a decentralized version of polling. RAP is based on the following ideas:

• A centralized protocol (such as polling) is reliable, especially for wireless networks

that usually operate in very unpredictable channels, though it requires constant

monitoring of all nodes.


• Constant monitoring of all mobile nodes is not feasible in wireless networks. RAP

requires the knowledge of the ”random address(es)” of only active users (rather than

all users).

The operation of RAP, designed for the infrastructure wireless networks, is summerized

as follows:

Step 1: When a base station is ready to collect uplink packets, it broadcasts a ”READY”

message to all mobile nodes in its coverage area (the ”READY” indication may be piggy-

backed on a previous message, thereby avoiding an extra transmission).

Step 2: Each active mobile node that intends to transmit generates a random number

from the set IR = {0, 1, 2, ..., P − 1}. This random number may be generated in advance

before the reception of ”READY”.

Step 3: All active mobile nodes simultaneously transmit their random numbers. Trans-

mission must be simultaneous and orthogonal; for example, by orthogonal codes such as

those for Code Division Multiple Access (CDMA), or by different frequencies. Further-

more, random number exchange can be divided into ’L’ identical ”stages” where each

stage involves the generation and exchange of random numbers as described above.

Step 4: In general, a mobile node may transmit its random number ’Q’ times at each

stage. The base station may use some sort of voting policy to decide the correctly trans-

mitted random number(s). With the error-free transmission assumed, Q=1 is enough. In

case the base station cannot recognize certain number(s), it assumes no reception.

Step 5: The base station listens to all random numbers (”addresses”) at each stage, based

on the device with the centralized architecture. Suppose there are ’N’ active mobile nodes.

At the lth(1 ≤ l ≤ L) stage, there are ’N’ random numbers represented so as to be dis-

tinguishable by the base stations. Let these ’N’ random numbers be r11, r2

2, ..., rlN which

may not be distinct at the lth stage. If there is no response from the mobile nodes, stop

this polling cycle.

Step 6: Among these ’L’ stages, suppose that the base station received the largest number

of distinct random addresses at the l∗th stage. Denote these numbers as R1 < ... < RN∗.

Then the base station polls according to the mobile nodes’ l∗th random numbers. When


the base station polls mobile nodes with Rr(1 ≤ r ≤ N∗) at the l∗th stage, the mobile

nodes who sent the random number at the l∗th stage transmit packet(s) to the base sta-

tion. Collision is possible since there might be two mobile nodes sending the same random

numbers. If N = N∗, no collision exists.

Step 7: If the base station successfully/unsuccessfully receives the packet from any mo-

bile node, it sends a positive/negative acknowledge (”PACK”/”NACK”) right away before

polling the next one(s). If the mobile node receives ”PACK”, it removes the packet from

its buffer. Otherwise, the mobile node(s) keep the packet(s) for future polling. After

all scheduled transmissions, the base station re-polls again (i.e., repeats steps 1 to 6).

Although re-polling may allow new active mobile nodes to join, it is assumed that no

new active mobile node is allowed to join re-polling for the simplicity of analysis and

congestion control.

As the downlink (from the network to mobile node) transmission is typically done by

broadcasting, but only on the uplink multiple access protocol. Note that RAP requires

no handoff for users roaming from one cell to another cell. A mobile node can send

packet(s) to all reachable base stations. If any form of handoff is needed for the purpose

of network management, handoff can be initiated completely by mobile users.

To illustrate the procedures of RAP, consider an example. Suppose there are mobile nodes

A,B,C,D,E,F,G, and H, under the coverage of a base station (or an access point). We

choose P=5 and L=2 as the parameters of RAP. At the beginning of the polling cycle,

only A,D,E,G, and H, have packets to transmit. When ”READY” is received by these

active nodes, random numbers are generated by them according to step 2.

A:3,0 D:2,3 E:2,1 G:1,4 H:1,1

The base station collects the random addresses by proper orthogonal signalling and de-

tection. At the first stage, the base station recognizes random addresses 3,2, and 1. At

the second stage, the base station recognizes addresses 0,3,1, and 4. Therefore, the base

station polls mobile nodes according to the random addresses of the second stage. When

the base station polls ’0’, the mobile node ’A’ sends its packet. Under error-free assump-

tion, ’A’ will receive ”PACK” from base station to complete its uplink data transmission,


as will ’D’ and ’G’. However, when the base station polls ’1’ of the second stage, packets

from ’E’ and ’H’ collide. Not considering the capture effect (since a good power control is

assumed) both ’E’ and ’H’ will receive ”NACK” and go to the re-polling cycle. Of course,

considering the possibility of channel errors, A,D, and G may also receive ”NACK” and

join the re-polling.

2.1.3 FDMA Based MAC Scheme

In FDMA, signals from various users are assigned different frequencies, just as in the

analog systems. Guard bands are maintained between adjacent signal spectra to minimize

cross talk between channels.

The advantages of FDMA are:

1. A capacity increase can be obtained by reducing the information bit rate and using

efficient digital codes.

2. Technological advances required for implementation are simple.

3. A system can be configured so that improvements in terms of speech coder bit rate

reduction could be readily incorporated.

The disadvantages of FDMA are:

1. Since the system arhitecture, based on FDMA, does not differ significantly from

the analog system, and the improvement available in capacity depends on operation

at a reduced S/I (Signal Power-to-Interference Power) ratio. But the narrowband

digital approach give only limited advantages in this regard so that modest capacity

improvements could be expected from a given spectrum allocation.

2. Narrowband technology involves narrowband filters, and because these are not real-

ized in Very Large-Scale Integrated (VLSI) digital circuits, this may set a high cost

”floor” for terminals even under volume production conditions.


3. The maximum bit rate per channel is fixed and small, inhibiting the flexibility in

bit rate capability that may be a requirement for computer file transfer in some

applications in future.

2.1.4 TDMA Based MAC Scheme

In a TDMA system, data from each user is configured in a time intervals called slots.

Several slots make up a frame. Each slot is made up of a preamble plus information bits

addressed to various stations. The function of the preamble is to provide identification and

incidental information and to allow synchronization of the slot at the intended receiver.

Gaurd times are used between each users’ transmissions to minimize cross talk between

channels. The data is transmitted via a radio carrier from a base station to several active

mobiles in the downlink. In the reverse direction (uplink), transmissions from mobile to

base station is time sequenced and synchronized on a common frequency for TDMA.

A TDMA system using multiple slots can support a wide range of user bit rates by

selecting the lowest multiplexing rate or a multiple of it. This enables supporting a

variety of voice-coding techniques at efficient bit rates with different voice qualities. Data

communication customers could make the same kinds of decisions, choosing and paging

for digital data rates as required. This would allow customers to request and pay for

bandwidth on demand.

The advantages of TDMA are:

1. TDMA permits a flexible bit rates not only for multiples of basic single channel rate

but also submultiples for low bit rate broadcast type traffic.

2. TDMA potentially integrates into VLSI without narrowband filters, giving a low

cost floor in volume production.

3. TDMA offers the oppurtunity for frame by frame monitoring of signal strength/bit

error rates to enable either mobiles or base stations to initiate and execute handoffs.

4. TDMA utilizes bandwidth more efficiently because no frequency guard band is re-

quired between channels.


5. TDMA transmits each signal with sufficient guard time between slots to accommo-

date:

(a) Time inaccuracies because of clock instability,

(b) Delay spread,

(c) Transmission time delay because of propagation distance, and

(d) The ”tails” of signal pulses in TDMA because of transient responses.

The disadvantages of TDMA are:

1. For mobiles particularly handsets, TDMA on the uplink demands high peak power

in transmit mode, which shortens battery life,

2. TDMA requires a substantial amount of signal processing for matched filtering and

correlation detection for synchronizing with a time slot.

2.1.5 DS-SS (CDMA) Based MAC Scheme

In wideband system, the entire system bandwidth is made available to each user and is

many times larger than the bandwidth required to transmit informations. Such systems

are known as ”Spread Spectrum” (SS) systems. There are two fundamental types of

SS: Direct Sequence Spread Spectrum (DS-SS) and Frequency Hopping Spread Spectrum

(FH-SS).

In DS-SS, the user’s data is spreaded using a Pseudo-Random sequence code (PN code)

and modulated and then transmitted. The received signal after despreading, resolves into

multiple signals with different time delays. A RAKE receiver can recover the multiple

time-delayed signals and can combine them into one signal, providing an inherent time

diversity receiver with lower frequency of deep fades. When many mobile users transmit

data to a common base station, then each user will use a unique PN code to spread his

data for transmission. The PN codes of all users are orthogonal. It is this correlation

property of the codes that makes it possible for the extraction of the desired signal at the

receiver. By multiplying the information-bearing signal b(t) by the spreading code c(t),


each information bit is chopped into a number of small time increments commonly called

as chips. The transmitted signal m(t), thus may be expressed as:

m(t) = c(t).b(t) (2.1)

which is a wideband signal. The received signal r(t) contains of the transmitted signal

m(t) plus the additive interference i(t). The interference signal contains MAI (Multiple

Access Interference) and fading and any other external interference signals. Therefore,

r(t) = m(t) + i(t) + n(t) = c(t).b(t) + i(t) + n(t) (2.2)

where, n(t) is Additive White Gaussian Noise (AWGN) in the receiver. To receive the

original signal b(t), the received signal r(t) is multiplied by the code which was used in

the transmitter. Therefore, the demodulated output z(t) at the receiver is given by

z(t) = c(t).r(t) = c2(t).b(t) + c(t).i(t) + c(t).n(t) (2.3)

Since, c2(t) = 1 (the autocorrelation property of the PN code),

z(t) = b(t) + c(t).i(t) + c(t).n(t) (2.4)

Thus, the information-bearing signal b(t) is reproduced at the receiver along with the

interference plus noise components. It is to be noticed that the information-bearing

signal b(t) is narrow band and the spurious component [c(t).i(t)] is wideband. Hence, by

applying the multiplier output to a base-band LPF filter with a bandwidth just enough

to accommodate the signal b(t), the spurious component [c(t).i(t)] is made narrowband,

thus removing most of its power. The effect of i(t) is then significantly reduced at the

receiver output. But the orthogonal properties of the PN codes must be maintained to

eliminate the MAI problem in a CDMA system, i.e., ci(t).cj(t) = 0; i �= j where ci(t) is

the PN code of ith user and cj(t) is the PN code of jth user. The advantages of CDMA

are:


1. The transmission bandwidth exceeds the coherence bandwidth.

2. Provides an inherent robustness against mobile channel degradations.

3. It has greater resistance to interference effects in a frequency reuse situation.

4. There is no hard limit on the number of mobile users who can simultaneously gain

access.

5. The interleaving and error-correcting codes are effective with CDMA system.

6. It tolerates a fair amount of interfering signals compared to FDMA and TDMA,

which typically cannot tolerate any such interference. Because of this, the problem

of frequency band assignment and adjacent cell interference are greatly simplified.

7. Since the interference with others is not a problem, the flexibility in system design

and deployment are significantly improved.

8. Capacity improvements with DS-CDMA also result from voice activity patterns dur-

ing two-way conversation (i.e., times when a party is not talking) that cannot be

cost-effectively exploited in FDMA or TDMA systems. So, CDMA can accommo-

date more mobile users than FDMA and TDMA radios on the same bandwidth.

9. DS-CDMA share the same frequencies in adjacent microcells whereas with TDMA

and FDMA it is not feasible.

10. Further capacity gains can also result from antenna technology advancement by

using directional antennas, which allow the microcell area to be divided into sectors.

11. CDMA signals are secured in wireless medium because of the unique PN code used

by a user.

The disadvantages of CDMA are:

1. There is a graceful degradation of the performance of CDMA system when the

number of mobile users increases i.e., under heavy load.


2. The proper generation of orthogonal PN codes requires a complicated hardware

system.

3. Because of the mobility of users in a multimedia wireless mobile networks, there is

a problem called ”near-far” problem (i.e., the received power at the base station is

different for different users).

In a cell containing ’N’ mobile users, the number of effective interferers is (N-1) regardless

of how they are distributed within the cell. Because of this, the quality of a particular

user is degraded. To remove the ”near-far” problem, we use ”Automatic Power Control

(APC)”. The APC operates such that the incident power at the center of the cell from

each mobile is the same as for every other mobile in the cell, regardless of the distance from

the center of the cell. APC conserves battery power in the mobiles, minimizes interference

to other users, and helps to overcome fading.

Frequency Hopping (FH) is the periodic changing of frequency or frequency set associated

with transmission. If the modulation is multiple Frequency Shift Keying (FSK), two or

more frequencies are in the set that change at each hop. For other modulations, a single

center or carrier frequency is changed at each hop. So, the FH signals may be considered

as a sequences of modulated pulses with pseudo-random carrier frequencies. The set of

possible carrier frequencies is called the hop set. FH systems work best when a limited

number of signals are sent in the presence of non-hopped signals where mutual interference

can be avoided. The major advantage of FH system is that it can operate very well in

frequency selective fading channel environment. The major disadvantage with FH system

is with increasing hopping rate, the cost of a frequency synthesizer increasing and its

reliability decreases, synchronization becomes more difficult.

2.2 Requirements of MWN’s MAC Protocol

The requirements for a decent MAC protocols are described below, may generally be

considered as the expected features of any wireless networks [30-35].


1. Throughput: Since the spectrum is a scarce resource, throughput is definitely one

of the most critical consideration in the design of a MAC protocol. For example,

with 10 Mbps physical transmission and over 80 percent throughput for CSMA/CD,

Ethernet can deliver over 8 Mbps performance in principle. But practically, it is

only 3 to 3.5 Mbps performance more important than the theoretical throughput.

2. Delay: Delay characteristics are important for every application, but especially for

time- bounded services and multimedia applications such as voice and video.

3. Transparent to Different PHY Layer: The physical transmission layer includes direct

sequence spread spectrum, frquency-hopped spread spectrum and diffused infrared.

These physical transmission layers are different not only in system design but also in

propagation characteristics; however, one MAC must handle all of them. One way

to achieve this goal is to have a physical dependent layer, a physical convergence

layer, and an appropriate MAC-PHY interface in each station. Based on such an

architecture, a single MAC can exchange data with different PHYs transparently

via MAC-PHY interface.

4. Ability to Serve Data, Voice and Video: With the increasing popularity of multi-

media applications, desirable wireless network must be able to provide some time-

bounded services such as voice and video in addition to the mandatory data services.

5. Fairness of Access: The fading characteristics of indoor channels may cause unequal

received power at the base station even when power control is enforced. Such a

situation may result in unfair access to network. That is one mobile node may have

much less power received at the base station than another mobile node. When the

MAC protocol is operating in the contention mode (necessary for initial registration

and often used for uplink traffic), the disadvantaged mobile node may not have the

chance to access the channel for a while. A MAC protocol should be able to resolve

this situation since it is possible that capture can happen with as small as 6 to 9 dB

power difference while the dynamic range of fading can be as large as several tens

of dB.


6. Battery Power Consumption: Since mobile nodes typically rely on battery power,

efficient utilization of transmit and receive power is another important considera-

tion for a MAC protocol. Since, we need power to constantly monitor access points

or handshake with stations for the purpose of synchronization, power control or

exchanging state information, very limited power should be used for packet trans-

mission. Sleep mode should be possible at the receiver front end. Active receive

mode may consume more battery power than transmission mode operation since

modern commercial digital communication systems may typically have transmis-

sion power like 100 mW but need 100 mA of current to support the digital signal

processor operation at the receiver.

7. Maximum Number of Nodes: According to market studies, a single cell wireless

network may need to support hundreds of nodes. Therefore, a MAC should not

limit the maximum number of nodes in order to maintain satisfactory performance.

8. Robustness in Collocated Networks: It is quite likely for two or more wireless net-

works to operate in the same region or in same region where interference between

different networks exists. Some protocols cannot function normally in this situation.

Consider two wireless LANs that operate in two nearby buildings. For certain parts

of these LANs, it may be more difficult to communicate with other parts of their

own LANs than to communicate with the other LAN. Serious trouble can result

from this situation if the MAC uses token passing. It is possible to wrongly pass the

token to a node in the other network. Generally speaking, there are two concerns

for collocated networks. Other users may illegally break into the network, causing

seurity crisis. This can be solved by an appropriate authentication procedure for

new users. Another issue is the interference from collocated networks. For example,

if we apply traditional CSMA protocols in wireless LANs, interference from another

network can cause disastrous hidden terminals.

9. Ability to Support Handoff/Roaming Between Service Areas: A MAC protocol has to

support a handoff function to serve nodes moving from one cell to another. In indoor


environments, due to the fast fading, handoff is not a straight forward problem.

For time-bounded handoff services, the ability of a MAC to support handoff in

real-time is not an easy task either, especially if we take power consumption into

considerations. In second-generation cellular networks, a central switching office

controls handoff; this is not practical for wireless networks. This makes handoffs in

wireless networks more difficult and most likely to be done in distributed fashion.

10. Establish Peer-to-Peer Connectivity without a priori Knowledge: A MAC of a wire-

less network should support adhoc networking. Therefore, there should be no re-

quirement for a priori information about network topology (e.g., whether there is

communication among all nodes).

11. Unauthorized Network Access Impact on Throughput: Since neither the MAC nor the

network management function can identify any unauthorized access before receiv-

ing its transmission, such access inevitably has impact on the network throughput

and delay. A successful MAC and network security scheme should reject such an

unauthorized access and minimize its impact.

12. Ability to Support Broadcast (Multicast): Although broadcasting is the natural form

of communication for wireless networks, the MAC should support multicast.

13. Critical Delays Limit Large Area Coverage: A typical coverage area for wireless

LANs may be from 500 ft2 to 1000 ft2 which introduces roughly 500 to 1000

nanoseconds (ns) propagation delay. Wireless LANs are likely to operate at more

than 10 to 20 Mchips per second (Mc/s) for DS-SS and more than 1 Msymbols per

second for other PHYs. Delay in the range of 500 to 1000 ns can cause big troubles

for some MACs. For instance, if the MAC has to operate precisely synchronously

among each pair of communication nodes, propagation delay can destroy signal

quality and thus limits the coverage area. A more exact example is a synchronous

CDMA system with 20 Mc/s transmission and a contention mode. Two nodes with

500 ns (i.e., 500 feet) difference in propagation delay can ruin the synchronization

scheme.


14. Insensitivity to Capture Effects: Although the capture effect can increase through-

put, it can also prohibit fair access. One solution is to enforce insensitivity at the

receiver end. A MAC is expected to keep receiver sensitivity to enhance physical

transmission and avoid any potential problem from capture.

15. Support Priority Traffic: In addition to the time-bounded services, the MAC is

expected to support traffic with different priorities.

16. Ability to Support Non-reciprocal Traffic: A special feature of wireless LAN traffic

is that the downlink traffic is often much greater than the uplink traffic. A good

MAC should definitely support this feature.

17. Preservation of Packet Order: This is important for multimedia services including

voice, audio and video.

18. Ability to Work in a Wide Range of Systems: As wireless LANs may cover many

areas and may serve a wide range of number of nodes, the MAC should be robust

to geographical size and number of nodes on the LAN.

19. Limit the Complexity of PHY: A critically important concern is to keep the complex-

ity of physical layer (medium dependent layer, PHY convergence layer and MAC-

PHY interface) to a minimum. To design a wireless network is an integrated prob-

lem, from physical layer upto network management. A MAC design that pushes

difficulty into other parts/layers of the system is undesirable.

20. Ability to Market and Complexity: The final step for success of wireless network is to

deliver high-quality products in time to users, which usually means that simplicity.

In addition to above requirements, there are more concerns with infrastructure wireless

networks. Multicell coverage is governed by an access point which is typically a base

station or a repeater. The coverage of each cell should overlap neighboring cell(s) properly,

that is, the overlapping region is intended to be minimized to increase system capacity

but also kept to certain portion so that seamless service is possible. This joint region

among cells introduces extra problems. They are listed as follows:


1. Self-Interference: There exists a possibility that more than two access points (such

as two repeaters) intend to transmit a packet to a node in the joint region simulta-

neously. This causes interference and the packet is likely to be lost.

2. Self-Collision: Another possible situation is that a node in the joint region transmits

a packet that is received by more than one access points. Collision or bandwidth

waste while routing this packet to its destination is likely without appropriate man-

agement.

3. Up-Down Collision: If node A in one cell transmitting uplink while node B in

another cell is receiving downlink, it is possible that node B may be able to hear

(receive) node A’s transmission and this situation results in collision, unless we can

perfectly schedule all transmissions. Fortunately, this situation which is similar to

the hidden terminal problem and it happens only with a very small probability if

the cells are well separated. Since up-down collisions are very destructive, any MAC

should carefully take it into account.

The first two problems are eliminated by careful coordination among access points. A

possible collision window always exists, although it can be kept small if coordination

among access points and uplink/downlink has taken care of this problem.

2.3 Some of the Existing Works on MWN’s MAC

The existing recent research works done by eminent researchers in the areas of multimedia

wireless networks (MWNs) are listed below.

1. Design of QoS Algorithms for Multimedia Communication

The technical growth in high speed networks, wireless networks, mobile computing and

powerful end-systems are enabling new types of presentational and conversational multi-

media applications, such as video conferencing, video-on-demand and news-on-demand.

The media involved in these applications are temporally continuous in nature. So, such

applications require real time multimedia communication and demand guaranteed Quality


of Service (QoS). QoS of an application gives an intuition of how valuable the services are.

The QoS parameters generally considered in multimedia communication are: end-to-end

delay, packet error rate, delay jitter and bandwidth which are to be negotiated before

launching a presentational or conversational multimedia application.

A QoS based call admission control scheme for mobile networks is designed. This work

gives a design model of a neural network based linear programming model which enables

to use the resources optimally and advises optimal admission of new applications based

on readjusting the bandwidth allocation.

A mobile agent based QoS management system is designed to satisfy the five functional

principles of QoS architecture, i.e., integration, separation, transparency, asynchronous

resource management and performance. A mobile agent is a software module that mi-

grates sequentially through a set of nodes, remain stationary and interact with resources

remotely or perform any combination of these three extremes for reservation of resources.

It has been proved that mobile paradigm saves considerable amount of bandwidth, reduces

network traffic and enables quicker QoS readjustments.

A QoS adaptation and mixer algorithms for wireless communication is also designed.

This work proposes new QoS adaptation procedures to provide desired QoS in wired/wireless

networks which allow the system to recover automatically, if possible from QoS violations

by identifying a new configuration of system units that might support the initially agreed

QoS and by performing a user transparent transition from the original configuration to

the new one. The Mixer Algorithms are proposed and that might be used in the case of

very bad atmospheric conditions at the time of unable to support the desired QoS but

requires a timely delivery of data [36, 37].

2. Design of Multiaccess Protocols for Multimedia Wireless Networks

Optimal sharing of a common transmission medium by multiple users in a computer

networking environment has been an area of active research in the recent years. Many

multiaccess protocols have been designed, investigated and implemented for a wide range

of networks. Most of these protocols have been used in Wireless LANs offer throughput

of the order of 0.7 maximum when the propagation delay is less compared to the packet


transmission time.

An Optimal Channel Utilization Multiaccess (OCUM) protocol is designed and demon-

strated the analytical and simulated results of the protocol throughput performance.

Linnenbank, Venkataram and Mullender have designed a MAC protocol for limited

bandwidth wireless networks. The protocol is combination of autonomous algorithms for

dynamic channel assignment in a TDMA wireless multimedia system. The throughput per

mobile station is higher compared to other MAC protocols, it offers low latency for both

real-time and non-real time communication. Furthermore, the throughput and latency

remain stable under high loads [38].

Chapter 3

Design of the Proposed MAC

Protocol

In this work we investigate the combination of TDMA with direct sequence (DS), a hybrid

protocol called TDMA/CDMA with a hand-shaking technique, request-to-send/clear-to-

send (RTS/CTS) method. The TDMA/CDMA uses multiple slots per frame while the

CDMA allows multiple users in a slot using their own DS codes. The RTS/CTS mini-

packets are exchanged between source and destination stations prior to sending the DATA

packets. This method is used when the mobile users are less in number in the cell, i.e., if

there are ’M’ number of terminals, then upto a maximum of ’M/2’ terminals are considered

in the picocellular infrastructure wireless local area network (WLAN). We also investigate

the polling schemes proposed for use in WLANs as per IEEE 802.11 standards for MAC

protocols. This protocol supports when the number of users are greater than ’M/2’ and

upto ’M’ users in the cell.

In the conflict-free polling MAC protocol, the base station may invite a portable to

transmit by broadcasting a short mini-packet with its identification. The portable will

send back a data packet, if it has any. Otherwise, it refrains from transmitting or sends

back a short ”keep alive” mini-packet. There are two major short comings of this protocol

: first, the average data packet delays are larger when the number of portables is larger

but the overall traffic across the cell is light and second, the operation of polling scheme by

36

Chapter 3. Design of the Proposed MAC Protocol 37

itself does not specify how new portables join a cell. A low-delay centralized environment

is also ideal for polling, since it provides better control. But it is impractical for bursty

data users.

Therefore, the central theme of this work is that a combination of random access and

polling may prove right for bursty data as well as the bulky data in a cellular system.

Such a centralized control over multiple access provides pretty reliable medium access

control and robustness in the presence of fading and dynamic channel characteristics. In

this work, we are confining our attention in WLANs with infrastructure. The coverage

of a base station is known as a cell in this work. To provide seamless data services,

the adjacent cells should appropriately overlap. Within a cell, many protocols based on

token passing, carrier sensing, ALOHA may be possible. But, they are all facing some

difficulties to be a perfect solution for wireless LANs. At the sametime, a more general

MAC protocol should combine multiple access and handoff into considerations. Thus the

new combined protocol proposed in this work meets the above mentioned requirements

and it is found that this is an attractive protocol for wireless networks [39].

3.1 System Model for the Proposed MAC Protocol

To describe the access system we consider only one mobile cell in which there are M active

nodes (or users) which are generating messages to be transmitted to another node. This

network is called as an infrastructure network as per the IEEE 802.11 standards where

the base station (BS) controls all the nodes within the cell. There are two kinds of links

possible in this model. One is the uplink that is concerned about the data transmission

from mobile station (MS) to BS and the another one is the downlink that is about the

data transmission from BS to MS. Topologically, the base station is, normally, positioned

to have good propagation conditions with each portable most of the time. On the other

hand, the locations of the portables are uncertain, and in some cases time varying. The

wireless propagation conditions have a strong impact on the choice of a suitable multiple

access protocol (see figure 3.1).


MS

MSBS

MS

MS

MS : Mobile Station

BS : Base Station

Figure 3.1: The infrastructure WLAN network used for the simulation of the proposedMAC protocol

3.2 Frame Structure

For the proposed protocol, the time is divided into fixed size frames. A frame contains S

time slots for communication. There are two special slots in each frame, the Request-To-

Send (RTS) and the Clear-To-Send (CTS) slots. Apart from these slots, we have DATA

slots and the RTS/CTS slots divided into minislots [40].

• Slot Descriptions and CDMA Codes:

– RTS Slots : These slots are divided into minislots that allow the transmission

of single byte messages preceded by two synchronisation bytes. Minislots are


used to inform the BS about the message length in the next frame and the BS

allocates data slots as per these informations available in the minislots.

– CTS Slots : The BS uses these slots to send CTS messages. It informs the

mobile stations that the MS can establish a connection set up and the BS

broadcasts a table containing the slot allocation for the current frame to the

active MSs that are within the BS’s cell.

– DATA Slots : These slots are used to send packets containing application data

and these packets may be subjected to errors due to the various propagation

conditions of the wireless medium.

– minislots : These slots are the slots within the RTS and CTS slots. The RTS

minislots are used in the uplink and the CTS minislots are used in the down-

link. The CTS minislots in TDMA/CDMA MAC protocol are arranged in the

form of a matrix containing allocation of data slots in rows and allocation of

CDMA codes in columns.

– CDMA Codes : These are the Pseudo Random Codes (PN Codes) which can

be generated by using shift registers and EX-OR logic gates. These codes are

orthogonal codes and are useful when we talk about MAI among MSs. There

are other codes like Gold and Walsh Codes for better MAI behaviour among

all MSs. The information about the allocation of CDMA codes is available in

the CTS minislots.

3.2.1 The Random Access (RA): TDMA/CDMA

In random access(RA), the uplink (UL) slots are divided into RTS minislots and data slots

and the downlink (DL) slots are divided into CTS minislots (see figure 3.2). A user who


A, B, C, D, X

............

D

2::::

M1 M2 M3 .................... MN

: CDMA Codes

M1, M2, M3, ...., MN : Mini Slots

D1 D2 D3 .................... DL

C1, C2, C3, ....., CK

D1, D2, D3, ..., DL : DATA Slots

A

B

D

X

D

C

AC

DB X

:::

. . . . . . . . .

. . . . . . . . .

. . . . . . . . .

. . . . . . . . .

C

RTS : Request to Send

CTS : Clear to Send

RTS CTS

DATA

RTSRTSRTS RTS

DATA

DATA CTS

. . . . . . . .

C1

CK

FrameRTS CTS

Figure 3.2: The TDMA/CDMA Frame Structure

is ready to transmit will send a RTS signal with some control informations to BS. The

BS has a scheduling algorithm which enables the user to get a CTS signal regarding in

which data slot and with which CDMA code, the user should be in a position to transmit

the data. In the hybrid protocol, each terminal transmits during the time slot(s) that is

allowed to transmit by using its own code sequence(s). Therefore, upto certain number of

terminals are allowed to transmit simultaneously during a time-slot with acceptable BER

(Bit Error Rate). The processing of the requests of the nodes and the scheduling are done

during the CTS slot. The CTS is divided into minislots, each holding information of the

corresponding data slot in the next frame. Each minislot is further divided into grids,

where grids equal to the maximum number of nodes that can transmit data simultaneously

in a data slot. Each of these grid is initialized with a code which the scheduler allocates


to the node which succeeded in getting a reservation for that slot. We assume that each

node generates messages with an arrival rate λ which is Poisson distributed. The message

length of each node is exponentially distributed. A node cannot generate a new message

until all packets of the current message are transmitted completely. A node which has

generated a message in the current frame cannot try to access the data slots in the same

frame.

3.2.2 The Random Access Polling : RAP

B C

B

C

Frame

Code 1

Code 2

.........

RTS(Polling)

Frame

RTS(Polling) DATA

CTSDataDataDataA BC

D X

A

XD

Unused

DATA

A

XDCTS

Figure 3.3: The RAP Frame Structure

In RTS(polling), the UL slots are grouped into access periods comprising of ’a’ slots,

’b’ codes and ’a×b’ polling periods (see figure 3.3). Here, ’p’ is the number of terminals

that are polled during one frame (p≥a). The DL slots are grouped into control and data

slots. Each active MT (Mobile Terminal) will generate a random number which will be

used for polling in the control slot. For example, if MTs A, B, C, D and X (as shown in

figure 3.3) generate the random numbers 0, 2, 2, 3 and 4 which shows the collision of MTs

B and C. The CTS shows the slot allocation for all these MTs except B and C. There

are two advantages of having access and polling periods occur cyclically. First, once an

MT synchronizes with the cycle, the possibility of incorrect channel sensing is eliminated.


Second, the discontinuous channel sensing used here also leads to extended battery life.

When an MT generates data, to transmit, it sends a reservation burst randomly in one

of the ’a’ slots during the next access period. The reservation burst contains its identity

number among other information (for example, message length). If the access burst is

transmitted successfully (without a collision), the MT-id is put at the top of the polling

list maintained by the BS. A collision occurs when two or more MTs transmit their

reservation bursts in the same slot and none of them is included in the polling list. When

the access period is over, polling period starts and the BS polls from the top of the polling

list. After polling the MT, the BS inserts its MT-id at the bottom of the polling list if

it has still data to be transmitted, otherwise BS removes that id from the polling list

because that MT has completed its data transmission. When an MT does not get access

during an access period, it backs off with a pre-determined back-off algorithm and tries

randomly to access the next available access slots.

3.3 Calculation of Throughput and Delay

When communication requirements of the users in the system are all different, the channel

allocation and access control schemes of this proposed protocol is too complex to be

modelled. Therefore, we use a model where each connection of the same type generates

Poisson traffic with equal intensity. In this section, we evaluate the performance of the

above system model in terms of throughput, delay and rejection rate. We assume that

the users in the cell arrive at base station with a Poisson process and generates messages

over a period which is exponentially distributed.

For random access protocol :

Throughput of RAP, Sr = Nt ÷Ns (3.1)

where Nt=Number of packets transmitted successfully and Ns=Number of packets put

into the system.

Mean delay of RAP, τ r = τa + τt (3.2)


where τa=mean access delay (in RTS slots) and τt=mean transmission time (in DATA

slots). This is calculated over all frames.

Rejection rate of RAP, Rr = Cbr ÷ Ct

r (3.3)

where Cbr=total blocked calls and Ct

r=total number of calls. A blocked call occurs when

the call is not able to get access after certain number of frames.

For polling protocol (PP):

Throughput of PP, T p = P ÷ [a+ p× k] (3.4)

where p = number of polling periods,

k = number of slots in one polling period,

a = number of access periods (in slots),

and P = number of slots polled during ’p’ number of polling periods.

Mean delay of PP, πp = πa + πt (3.5)

where πa=mean access delay (in ’a’ slots) and πt=mean transmission time (in ’p’ periods).

This is calculated over all frames.

Rejection rate of PP, Rp = Cbp ÷ Ct

p (3.6)

where Cbp=total blocked calls and Ct

p=total number of calls. A blocked call occurs when

the call is not able to get access even after certain number of frames with the maximum

number of attempts (AMAX) using a good back-off algorithm. The back-off algorithm

used in this study is given by

xb = rand(2nb − 1), nb ≤ AMAX (3.7)

where rand(n) is a function returning a random integer between zero and n, and

xb = number of access slots to backoff;


nb = number of access attempts;

AMAX = maximum permissible attempts (depends on the total time for communication).

In both the protocols, the terminals produce messages with 1% activity of the users and

the cumulative distribution function of the message length (in packets) is given by

F (x) = 1− e−µx, 0 ≤ x ≤ 1000 (3.8)

Each message is coded in RLP (Radio Link Protocol) blocks where each block consists of

five slots and µ is the mean message length.

3.4 Analytical Model for the Proposed MAC Proto-

col

3.4.1 Random Access

For the random access protocol, we use the M/M/n/n/K Markov model as shown in the

figure 3.4. We obtain the steady state equation as:

+πA = O (3.9)

where +π is a steady state probability vector and it is equal to

+π = (π0 π1 π2 ... πn) (3.10)

and A is the generator matrix and O is a null matrix. For this Markov chain, the

recurrent non-null and the absorbing properties are satisfied. ’K’ is the number of users

and the number of data slots are ’n’. The average number of packets served by the system

is calculated as:

L =(Kρ)

∑n−1i=0

(K−1i

)ρi∑n

i=0

(Ki

)ρi

(3.11)

where ρ = λµis the offered traffic to the system with the arrival rate, λ as per Poisson


distribution and the service rate, µ as per the exponential distribution. The packet success

probability is calculated as:

PS =c∑k=0

n∑j=0

(1− πj)(1− Pb(k)) (3.12)

where ’c’ is the number of CDMA codes allocated to the active users in a data slot and

the steady state probabilities are given as follows:

π0 =1∑n

i=0

(Ki

)ρi

(3.13)

and

πj =

(K

j

)ρjπ0; j = 1, 2, 3, ..., n (3.14)

We have assumed the channel is an AWGN and the modulation employed is DS-SS BPSK.

The BER is given by the relationship as follows:

Pb(k) =1

2erfc

√√√√ Eb

N0 +23Eb(

k−1Gp

)

(3.15)

where k=number of active users

Gp=the processing gain which is equal to RF bandwidth divided by the baseband band-

width.

Eb=the energy per bit in Joules

and N0=the two-sided psd in Watts/Hz.

The throughput is given by

T = PSL (3.16)

and the delay is given by

W =K

T− 1

λ(3.17)


. . . . . . . . nn-1210

(K-2) (K-n+2) (K-n+1)(K-1)K

2 3 (K-n-1) (K-n)u uuuu

Figure 3.4: The analytical model (Markov Model) for the performance analysis to calcu-late the throughput and the delay.

3.4.2 Random Access Polling

For RAP protocol the analytical equations are given as follows: the probabilities that there

are ’i’ successful packets and ’j’ collision packets out of ’n’ active users in ’p’ random access

slots are given as:

qni,j =1

pn

(p

i

)(p− i

j

)∑j

n!∏jk=1 ak!

(3.18)

where ak is the number of collision packets in slots which is ≥ 2:

j∑k=1

ak = n− i (3.19)

The mean time to transmit ’n’ active users is:

T (n) =∑i,j

qni,j [i(Tp + Ts + Ta) + j(Tp + Tc + Ta + Tb) + T (n− i)] (3.20)

where Ts=packet transmission time

Tc=packet collision period

and usually these two parameters are considered to be equal in our analysis.


Tp=time to poll an active user.

Ta=additional delays like propagation delay, overhead time to transmit and detect the

user’s random access slot number.

Tb=back-off time interval.

The busy period is calculated by the arrivals in the polling cycle is Bernoulli’s distribution

with parameter (1− e−λT (n)) and this is given as:

T =N∑n=0

(N

n

)(1− e−λT (n))

n(e−λT (n))

N−nT (n) (3.21)

where N is the total number of users in the system. The expected value of successful

packets transmission in a polling cycle is given as:

U = NTsN∑n=0

(1− e−λT (n))

(N

n

)(1− e−λT (n))

n(e−λT (n))

N−n(3.22)

The system throughput is calculated as:

S =U

T(3.23)

The average delay is:

D = Db +Dp (3.24)

where Db=time that packet arrives in the BS until the current polling cycle is over

and Dp=time that the system begins the new polling cycle until the packet is successfully

transmitted. These two time parameters are given as follows:

Db = T − 1

λ(1− e−λT ) (3.25)

Dp =

∑Nn=0 Dp(n)

N(3.26)


where Dp(n) is given as:

Dp(n) =∑i,j

qni,j[i(i− 1)

2(Tp + Ts + Ta) +

ij

2(Tp + Tc + Ta + Tb) + (n− i)(i(Tp + Ts + Ta) +

j(Tp + Tc + Ta + Tb)) +Dp(n− i)] (3.27)

3.5 Power Control for the Proposed MAC Protocol

In this work, a detailed study is carried out regarding the maximum number of users that

each base station can support on a lossy channel. The author has analyzed the desired

user’s signal quality in a single cell CDMA (Code Division Multiple Access) system in

the presence of MAI (Multiple Access Interference). Earlier power control techniques

were designed to assure that all signals are received with equal power levels. Since these

algorithms are designed for a imperfect control of power, the capacity of the system is

reduced for a given BER (Bit-Error Rate). But the EPCM (Efficient Power Control

Mechanism) proposed in this work assures a better system capacity and the EPCM is

designed for the reverse link (mobile to base station) considering the path loss, log-normal

shadowing and Rayleigh fading. The simulation results show that the proposed method

satisfies the required quality of service for multimedia wireless LAN application. The

EPCM is also analyzed mathematically and it is found that both the simulation and the

analytical methods are coincident within the limits of approximation [41].

An important design choice in cellular wireless communication is the multiple access

capability that allows a number of users to share spectrum resources. The selected mul-

tiple access techniques strongly influence modulation, coding and handoff methods, all of

which finally determine the capacity, quality and coverage performance of the air interface.

CDMA is an emerging multiple access technique in which each user data stream is spread

using an orthogonal random sequence. The spread signal occupies a much large band-

width than the original signal making this a DS-SS (Direct Sequence- Spread Spectrum)


modulation technique. The received signals are extracted by correlating with the corre-

sponding spreading code. This results in a coding gain for the desired user signal. But the

standard correlation receiver for CDMA system is susceptible to the near-far problem. So,

the power control technique is required in the system to maintain the trade-off between

capacity of the system and the user’s BER.

There are many power control techniques have been suggested for cellular communi-

cations. In CDMA, the Eb

N0(where Eb is the energy contained in a data bit and N0 is the

two-sided power spectral density of AWGN (Additive White Gaussian Noise) added to the

communication channel) at the correlation receiver output equals the signal power mul-

tiplied by coding gain, divided by the sum of receiver noise and other users’ interference

power. If one user has a high power at the base station, while its own Eb

N0will improve,

other users will suffer a lower Eb

N0. Therefore, the power of each user at the base station

must be controlled to be the same, since all users are equally important. This requires

a dynamic control of user power to compensate for spreading, shadowing and multipath

fading. This is done through open, closed and outer loop control. In open loop power

control, the mobile estimates the average propagation loss by estimating the forward link

(base station to mobile) received power and adjusts its transmit power accordingly. Open

loop compensates for spreading loss and shadowing. The variations due to multipath

fading are adjusted using a closed loop control. For this, the base station checks the Eb

N0at

the receiver correlator output against a set point and sends the mobile a step up or step

down power command. The closed loop power control is fast enough to follow Rayleigh

fading for low mobile speeds. However, at high mobile speeds closed loop control cannot

fully track fast fading. Since power control errors and its dynamics depend on the mobile

speed, these are compensated by outer loop that adjusts the set point to achieve a frame

error rate of about 1%. The set point is initially high at low speeds, then reduces with

mobile speed but increases again at higher speeds. Finally, just as reverse link power

control is used to maintain satisfactory Eb

N0at the base station, forward link control is also

necessary to compensate for variable outer cell interference and maintain the required Eb

N0

at the mobile. Hence, the cuts in interference to other users and directly translates to


improved capacity [42, 43].

3.5.1 Definitions

i) Transmitted Power level : This is the power level at the output of the transmitting

antenna of the mobile user towards the base station.

ii) Received Power level : This is the power level at the input of the receiving antenna

of the base station, received from the mobile user.

iii) Perfect power control : The respective received power levels of mobiles at the base

station are made identical for estimating the constant BER.

iv) Imperfect power control : The level of imperfections in the power control mechanism

depends on several causes (e.g. the power control algorithms, the speed of the mobile,

the dynamic range of the transmitted power level, the distribution of mobile users in a

particular cell, the propagation statistics, etc.).

v) Path-loss : It is a large scale attenuation of the received power level at the base

station and it varies with the distance between the transmitter (mobile user) and the

receiver (base station).

vi) Short-term fading : It is defined as the amplitude variation in the received signal

level, termed signal fading, are due to the time-variant multipath characteristics of the

channel and this is also called as Rayleigh fading.

vii) Long-term fading : This is defined as the variation in the received signal level

when the surrounding environmental clutter (e.g. trees) may be vastly different at two

different locations having the same transmitter-receiver separation but have different levels

of clutter (unwanted signals) on the propagation path and this is also called as log-normal

shadowing.

viii) Reverse-Link (Up-Link) power control : It is for reducing near-end to far-end

interference. The interference occurs when a mobile unit close to the cell site can mask

the received signal at the cell site so that the signal from a far-end mobile unit is unable to

be received by the cell site at the same time. It is a unique type of interference occurring

in the mobile radio environment. This link is from mobile to base station.


ix) Forward-Link (Down-Link) power control : It is used to reduce the necessary

interference outside its own cell boundary. This link is from base station to mobile.

3.5.2 The Correlation Receiver for CDMA System

We consider a DS-SS multiple access system with BPSK signalling and a correlation

receiver. There are K users in the system. User k’s power received at the base station is

sk(t) :

sk(t) =√2Pkbk(t− τk)ak(t− τk)

cos [ωc(t− τk) + θk] (3.28)

where, Pk is transmitted power of the signal, bk(t−τk) represents the data signal, ak(t−τk)

represents PN (pseudo-random) sequence, and cos[ωc(t− τk) + θk] represents the modu-

lating waveform. θk is the random phase which is uniformly distributed over [0, 2π], while

τk is the random delay which is uniformly distributed over [0, Tb]. Tb is the duration of the

data bit and Tc is the duration of the each chip in the PN sequence. The total number of

chips per bit, N is given by N = Tb

Tc= fcfbwhere fc is the chip rate (in chips/sec) and fb is

the data bit rate (in bits/sec).

We assume that Pk is independent for each user and independent of θk and τk . As

shown in figure 3.5, the data signal is given by bk(t) =∑∞i=−∞ bk,i pTb

(t − iTb) where

bk,i ∈ {−1,+1} is an infinite sequence of data bits and pTb(t) is a rectangular pulse with

unity amplitude and duration Tb. The spreading signal is ak(t) =∑∞j=−∞ ak,j pTc(t− jTc)

where ak,j ∈ {−1,+1} and pTc(t) is a rectangular pulse with unity amplitude and duration

Tc. Assume that we receive signal from user 1 and that θ1 = 0 and τ1 = 0. When a single

propagation path exists, the received signal for K users is given by :

r1(t) = n(t) + s1(t) +K∑k=2

sk(t− τk) (3.29)


where n(t) is AWGN with 2-sided psd N0

2and s1(t) is the desired signal. Ak =

√Pk

P′k

where

P′k is the desired power set point of the power control algorithm.

1

2

k

b (t)

b (t)

b (t)

A a (t) cos[w t +O ]

A a (t) cos[w t +O ]

1

k

1

k

A a (t) cos[w t +O ]2 2c

c

c

:

:

:

1

2

k

A

A

A

1

2

k

n(t)

ReceiverOutput , r(t)

w : carrier angular frequency.c

AWGN

Delay

Figure 3.5: The Correlation Receiver

3.5.3 Analytical equation and performance analysis for EPCM

With perfect power control, all users assigned to a base station should have the same

received power at the base station. We consider here a single cell system with K users. In

a CDMA system (see figure 3.1), the number of users that the cell can support is limited

by the total received interference at the base station and it will vary with time. All users

share the same frequency spectrum and experience an interference from all other active

users throughout. Given a quality of service requirement, the Eb

N0ratio for user 1 is given

as :EbN0

=P1∑Kk=2 Pk

.

(fcfb

)(3.30)


and the probability of error (BER) for user 1 in CDMA environment with BPSK modu-

lation is given as :

Pe = Q

√P1

2Tb

2√(NTc

2

6

∑Kk=2 Pk

)+ N0Tb

4

(3.31)

where P1 is the received power level of user 1 at the base station and Pk is that of kth

user.

The received power Pk of the kth user at the base station may vary due to LOS (line- of-

sight) path loss (for free space propagation α = 2 and α can vary from 1.6 to 8 depending

upon the morphology of the building) which is computed as :

Pk =Ptkλk

αGmkGbk

(4π)α[dk(t)]α (3.32)

where Ptk= transmitted power of kth user. λk =3.108(m/sec)fk(Hz)

meters, wavelength of kth user

and fk = operating carrier frequency. Gmk= kth user transmitting antenna gain, Gbk =

base station receiver antenna gain, dk(t) = distance between the kth user and the base

station and it varies with time because the user is moving and that is equal to dk ± t.vk

where vk = speed of the kth user and +ve sign indicates the user is going away from base

station and -ve sign indicates the user is coming towards the base station.

The mobile user’s radio signal R(t) can be artificially characterized by three compo-

nents LT(t), ST(t) and PL(t) based on natural physical phenomena.

R(t) = LT (t).ST (t).PL(t) (3.33)

where LT(t) = long-term fading or log-normal fading and its variation is due to the terrain

contour between the base station and the mobile unit. ST(t) = multipath fading, short-

term fading or Rayleigh fading and its variation is due to the wave reflected from the

surrounding buildings and other structures (see figure 3.6) and PL(t) = path loss and its

variation is due to the change of distance between the mobile user and the base station.


The multipath propagation is modelled using the Bernett-Vignant Reliability Equation

FMdB = 30.log(D) + 10.log(6.A.B.f)

−10.log(1− R)− 70 (3.34)

where

R = Reliability factor

A = roughness factor for the terrain

B = climate conditioning factor of the propagation medium

D = distance between the mobile user and the base station in Km

and f = RF carrier frequency in GHz.

BSMS

~ 3 m

30 to 100 m

Direct path

Reflected path

2 Km or further

O

O 2

1

Figure 3.6: Multipath fading due to reflection

The user k’s received signal power level is :

Pk ∝ D−4 (3.35)


But for the free space propagation, Pk is given as :

Pk ∝ D−2 (3.36)

In this work, consider initially the path loss, that is, when the user is moving with a

constant velocity and starting with some initial transmitting power, the received power

at the base station for that user is varying with time. This happens for all the users

in the particular cell served by single base station. But in order to eliminate the near-

far problem with CDMA technique, it is to maintain a constant received power for the

user with a desired quality of service (BER). Therefore, the transmitted power has to be

stepped up or down accordingly with the mobility of the user. So, first study is made

on the power control technique without the multiple access interference (MAI) assuming

that all PN codes of the users are orthogonal ideally. But in practice, it is very difficult

to maintain the orthogonality among the PN sequences and therefore include the MAI in

the consecutive analysis for calculating the BER tolerances.

3.5.4 Analytical Model for the Power Control

The distance variation of each user is given by the pdf (probability density function) as :

fD(d) =

1R2−R1

if R1 ≤ d ≤ R2

0 otherwise(3.37)

where D is a random variable which is uniformly distributed from distance R1 to R2 and

the received power at the base station is given by the pdf as :

fT (t) =

√p

2.t32 .(R2−R1)

if pR2

2≤ d ≤ p

R21

0 otherwise(3.38)

where T is a random variable defined as :

T =p

D2(3.39)


where p is given as :

p =Pt.Gb.Gm.λ

2

(4.π)2(3.40)

The Rayleigh model for multipath fading is given as a pdf :

fR(r) =

rσ2

R. exp

(−r22.σ2

R

)if r ≥ 0

0 otherwise(3.41)

where R is the random variable represents the received signal envelope at the base station.

The log-normal fading for shadowing effects is modeled as a pdf :

fL(l) =

20. log e√2.π.σL.l

. exp(− (10. log l)2

2.σ2L

)if l ≥ 0

0 otherwise(3.42)

where L is the random variable for the received power at the base station. Now, we cal-

culate the expected received power W at the base station for each mobile user as : Since,

W = f(T,R,L)

E(W ) =p

R1.R2

+ 2.σ2R + exp

(σ2L

)(3.43)

where σR is the amount of deviation in received power level due to Rayleigh fading and

σL is the amount of deviation in received power level due to log-normal fading.

Then, we calculate the probability of bit-error from the equation :

Pe =1

2erfc

(√EbN0

)(3.44)

whereEbN0

=

[E(W )

I(W )

].

(fcfb

)(3.45)

The interference for the kth user is given as :

I(Wk) =K∑i=1

E(Wi)− E(Wk) (3.46)

where the details about the parameters are given in equation (3.30).

Chapter 4

Functions of the Proposed MAC

Protocol

In this chapter we discuss the functions of the designed MAC protocol in a multimedia

wireless network in particular to admission control, channel allocation, buffer management

and data transfer during the hand-offs.

4.1 Multiuser and Multistream Admission Control

The challenge in internetworking the wireless and wired networks is to guarantee quality

of service (QoS) requirements while taking into account the radio frequency spectrum

limitations and radio propagation impairments. Call Admission Control (CAC) is one

method to manage network resources in order to adapt to traffic loading variations. In

particular, the admission control is necessary when there are various types of traffic with

various QoS requirements and when the system operates in the vicinity of its full capacity.

In this work, we propose a CAC algorithm for the TDMA/CDMA protocol. The objectives

of the CAC algorithm are to maximize the utilization of network resources and at the same

time, to provide the users who are transmitting different types of traffic with acceptable

QoS. The CAC decisions are also based on the availability of the radio resources (e.g.,

radio frequency channels and error correcting codes).

57

Chapter 4. Functions of the Proposed MAC Protocol 58

The QoS requirements under considerations are indicated by BER and transmission

delay. Transmission errors are due to radio channel impairments and buffer overflows

at base stations. To maintain the same QoS, the higher the data rate, the more the

transmitted power is needed, and the interference to the other users is larger. When the

system resources are limited, admission of a high-rate data service user will degrade the

whole system’s performance. It is thus essential that a base station identifies each user’s

data rate as part of its call admission policy, with the objective of efficiently managing

system’s resources without degrading QoS for both voice and data users. However, the

wireless networks inherently accompanies handoff, that has a significant impact on the

traffic performance of the networks such as call blocking probability.

4.1.1 CAC Algorithm

The CAC in multimedia wireless networks is also affected by handoff. The CAC should

consider that an accepted call can be forced-terminated before completion of service,

which is more unbearable to users than blocking of service. The traffic performance

metrics, considered by CAC, are as follows:

• Carried Load: This represents the ratio of the bandwidth used by completely ser-

viced calls to the total capacity of mobile network. If a call is forced-terminated,

the bandwidth used by the call is not taken into account.

• Call Dropping Probability: It is the probability that an accepted call is forced-

terminated before it completes the service.

• Call Blocking Probability: It means the likelihood that a new arriving call is blocked.

Actually, it depends on the CAC scheme.

• Call Completion Probability: It means the probability that a newly arriving call

completes its service without forced-termination.

• Handoff Failure Probability: It is the probability that a handoff fails because the

target call has no free channel. This is also called as forced-termination probability.


Admitting a new user will degrade the link quality for existing users. In order to

have high capacity and an acceptable service quality, a trade-off between channel quality,

dropping or outage, and blocking probability has to be made. So, the CAC algorithm is

used to make this trade-off [44, 45].

In the downlink, algorithm using output power levels at BSs are studied. Since the

codes are fairly orthogonal within one cell, it may be too difficult to make an admission

decision based on information from just one BS. Thus, a new call is admitted as long as

the output power levels, at the consulted BSs, stay below a certain predefined threshold.

When the maximum output power constraint is limited on a BS basis, the admission

decision is based on this algorithm:

Admit a new user if

Pb < Pbt (4.1)

where Pb is transmitted output power from BSb before the new user is admitted, i.e.,

Pb = pi at BSb, and Pbt is a predefined threshold. Here, pi is the output power from the

ith transmitter.

On the other hand, when the output power is limited on a channel basis, the admission

decision follows the following rule:

Admit a new user if

Pj < Pcht (4.2)

where Pj is the output power used by channel j and Pcht is a predefined threshold.

Consider the TDMA/CDMA system. Each frame of the TDMA/CDMA system is

divided into ’m’ slots and every packet occupies exactly one slot. Multiple users can

transmit their packets in the same slot. Therefore, for every user, the data rate is given

as ri′= mri, where ri is the data rate of type i users and the spreading factor, Fi

′=

W/ri′= Fi/m, where W is the spread bandwidth and Fi is the spreading factor of type i

users which is equal to W/ri.

Let kij denotes the number of type i users in slot j and Kj = (k1j , k2j, ..., kLj), for L

types of users (e.g., out of 50 users, 20 voice users type and 30 data users type).


For Kj to be admissible in slot j, it must hold that

L∑i=1

kij

1 + Fi′

Qi

≤ 1

f + 1(4.3)

where f =∑Li=1

MAIiPri

, Qi = the required Eb/N0 for any type i user, MAIi = Multiple

Access Interference for type i users, and Pri = received power level of type i users.

Therefore,L∑i=1

kij

1 + Fi

mQi

≤ 1

f + 1(4.4)

Since kij = 0 if m > Fi/(fQi), to avoid trivial cases, we assume that m ≤ 1/(fQi) for

all i.

Since K =∑mj=1 Kj = (k1, k2, ..., kL), where ki denotes number of i type users in one

frame, we say K is admissible for the TDMA/CDMA system iff there existsKj , 1 ≤ j ≤ m,

such that Kj is admissible in slot j.

The basic parameters and the pseudo-code of the CAC algorithm are given as follows:

Basic Parameters:

Let C = number of channels, Th = threshold (from 0 to C), Bc = occupied bandwidth,

Br = requested bandwidth, PD = call dropping probability, and PD.tar = target call

dropping probability.

CAC Algorithm:

1. initialize Th = C

2. set timer

3. wait for call request arrival

4. if new call request arrives

5. if (Bc + Br) ≤ (C-Th)

6. admit new call with rate Br

7. else


8. reject new call request

9. if handoff call request arrives

10. if (Bc + Br) ≤ C

11. admit handoff call with rate Br

12. else

13. reject handoff call request

14. update PD

15. if PD ≥ PD.tar

16. decrease Th

17. reset timer and go back to step (3)

18. if timer expires

19. increase Th

20. reset timer and go back to step (3)

21. else go back to step (3)

22. STOP

23. END

4.2 Channel Allocation based on the QoS Require-

ments of the Multimedia Applications

The reliable and efficient transmission of high quality variable bit rate (VBR) video

through the wireless networks generally requires network resources be allocated in a dy-

namic fashion.


CONTROL]

Source 1

::::

::::

Source 2

Source N

RESOURCEALLOCATION

NETWORK

Buffer/Scheduler

[and CALL ADMISSION

Figure 4.1: Illustration of Dynamic Resource Allocation for Multiplexing VBR Streams

Bandwidth allocation and management for individual stream generally must be done

at the edges of the network, in order to conserve computational resources on network

switches, as illustrated in figure 4.1. Such systems will likely not have complete knowl-

edge of the network state, and must therefore make their use of network resources as

minimal as possible to maintain a given QoS. If a source requests more bandwidth than it

actually uses, the overall network utilization drops. Conversely, if the source exceeds its

bandwidth request, packet loss and delay will become significant. While off-line systems

could compute the exact dynamic bandwidth requirements for a stream before transmit-

ting it, but on-line processing is desirable in many applications. Systems such as video

conferencing and live news-on-demand absolutely require on-line processing. In addi-

tion, on-line processing is needed in any system that dynamically transcodes video, or

that splices and combines segments in a computational requirements low, the information

used to make bandwidth decision should be directly available from the compressed video

stream. The overall goal is to have resource management systems that can accurately


estimate the required bandwidth in real time [46].

We use in this work the concept of ”bunch” of cells as a group of cells strongly inter-

connected; all resources available to a bunch are dynamically allocated by the BSC. Such

a system architecture is well suited especially for TDMA/CDMA system.

BSCBS

BS

BS

Bunch of BSs

BS

BS

BS

Bunch of BSsZone

BSC

Figure 4.2: Intra and Inter-bunch Links

So, we propose a DCA (Dynamic Channel Allocation) algorithm based on an ”Inter-

ference Matrix” based scheme. We consider a cellular system with a limited number of

BSs that are connected to a BSC. Such a bunch of cells can be used for instance to cover

a floor of a building or a group of streets. BS can use a simple antenna. In this case,

the BS-BSC link can be an optical cable carrying the radio-frequency signal and the real

transceivers may be located at the BSC level (Please refer to figure 4.2).

Between BSs, within a bunch, high speed signaling can be used for channel allocation.

The BSs in a bunch are assumed to be synchronized. The BSC has complete knowledge

of all allocated resources, actual transmission powers for any channel at any time. The

centralization at the BSC level permits the use of near optimal algorithms for resource


allocation and management (e.g., power control, bit rate, DCA, etc.). [47, 48].

The bunch size can be adapted to the type of environment. In the indoor case, a

bunch can be a WLAN, for example. In rural environment the bunch may include only

several macro-cells (e.g., 3 cells of a sectorized site).

The Zone Concept:

In classical systems, the cell is seen as an entity. The limit between two cells generally

consists of small areas where a mobile could be received by both BSs. The pico-cells in

urban areas or in building, which we study here, do not easily integrate a regular pattern.

The given environment results in a complex propagation situation. Base stations are

likely to be located as close as necessary to guarantee a continuous coverage so that

optimal planning of base station location becomes impracticable. This leads to important

overlapping where the mobile station is received by more than one base station.

Z1

BS1BS2

BS3

Z7

Z6

Z5Z4

Z2

Z3

Figure 4.3: The Zone Concept

We account for the cells overlapping by using the concept of coverage zones. A zone

is an area that is covered homogeneously by one or several BSs. By homogeneously we


mean that all the zone is covered by the same number of BSs (Please refer to figure 4.3).

DCA Algorithm:

Channel allocation in a bunch is made by the BSC for incoming and outgoing calls.

Resources within the bunch are allocated according to a ”Interference-Matrix” based

scheme.

A compatibility matrix (called as ”structure matrix”) of all cells within the bunch is

progressively built (please refer to table 4.1). The ”Interference-Matrix” scheme is then

used on the basis of this structure matrix. The structure matrix is defined as a record of

relation between zones and BSs. A zone could be either covered by specific BS, interfered

or non-interfered. The category of each pair (zone, BS) is related to a path loss range

(PL in dB) between the zones and BS. Two thresholds are to be defined: Lc (in dB) for

the coverage and LI (in dB) for the interference avoidance.

Table 4.1: Downlink Structure Matrix

BS

Zone C I NI

In table 4.1, C: covered ⇒ PL < Lc, I: Interfered ⇒ Lc < PL < LI , and NI: Non-

interfered ⇒ LI < PL. This matrix is an apriori known by the BSC. All BSs of the same

bunch broadcast a pilot tone (beacon channel) and the system configuration information

on the same frequency channel. Every mobile sets a list of the best received BSs and

transmits it to the BSC during the communication. This information is used to build the

structure matrix. This matrix is quasi-static. It should be updated periodically in order

to reflect the current topographic and propagation conditions.

Let us assume the simple situation where a MS in communication with BS1 is interfered

by BS2 using the same downlink channel (Please refer to figure 4.4).

In figrure 4.4, P1 (dBm) = transmitted power for BS1, P2 (dBm) = transmitted

power for BS2, PL1 = path loss for BS1, and PL2 = path loss for BS2. If the MS is


MS

P1P2

BS1

BS2

PL1

PL2

Figure 4.4: Downlink Co-channel Interference Case

located in the zone covered by BS1, we have PL1 < Lc. According to the ”Interference-

Matrix” scheme, the situation in above case is possible only if we have PL2 > LI . If we

are not considering the fading effects, the CIR (Carrier to Interference Ratio) in dB is

(CIR)dB > (P1− Lc)− (P2− LI).

The current status of resources at the receiver site (zone on the downlink, BS on the

uplink) is recorded in a ”resource management matrix” (Please refer to table 4.2). The

matrix describes the state of each resource (i.e., free, used or blocked due to interference by

nearby users). The resource management matrix is manipulated by the DCA algorithm.

These uplink and downlink matrices are dynamically updated at call set-up, handovers and

call releases. But the CIR and handoff constraints can reduce the number of assignments

to a computationally feasible range.

In the above table 4.2, F (free): resource not used in current BS and not by neigh-

bouring BSs interfering the current zone, B (blocked): resource used by neighbouring BS

interfering the current zone, and U (used): resource used in the current BS.


Table 4.2: Uplink Resource Management Matrix

BS

Resource F B U

The request for uplink transmission (from MS) and the zone localization of the MS is

made on a random access channel. This random access channel is the same for all BSs

for a given bunch (Please refer to figure 4.5).

When a mobile is sending a request, this request may be received by more than one

BS. The BSC deduces the zone where the mobile is from the BSs that have received the

request.

The ”allocation algorithm” operates using the structure matrix and a dynamic resource

management matrix.

Let us assume a demand for nγ resources in a location (BS on the uplink, zone on the

downlink) is received:

1. The BSC pre-selects all apriori free resources. The apriori free resources are those

with a free state in the resource management matrix.

2. The BSC sets a list of candidate resource. A candidate resource is an apriori free

resource of which allocation would not perturb other users. Let nc be the number

of candidate resources. If nc < nγ, the request is rejected. If nc > nγ , the score of

the candidate is evaluated. The BSC determines which zones would be blocked by

allocating a specific resource. If for such a zone this resource is already blocked, we

increase the resource score.

3. The nγ resources with the highest score are selected. The resources are allocated and

the resource management matrix is updated, accounting for the additional blocking

caused by the new allocation.

After de-allocation, the resource management matrix is updated as well. In circuit


MS

BS1

BS2

BS3

Figure 4.5: MS Uplink Access

reservation, when not enough resources are available at request, the call is blocked or

dropped. For packet reservation, when there are not enough available resources, the

request is queued. We consider only one queue for the whole bunch. A released resource

is attributed to the first user in queue First-In-First-Out (FIFO), to whom the resource

becomes a candidate resource.

The initial power settings is an important aspect to consider. At each call set-up a

target CIR value is requested according to the service type. The DCA should determine

the physical channel that is available and that satisfies the target CIR value (CIRtarget).

We propose to classify the initial CIR value request in two categories, ”low CIR”

with the threshold, CIRL and respectively ”high CIR” with the threshold, CIRH . If

0 < CIRtarget < CIRL, then the system will return as an initial CIRtarget, CIRL. If

CIRL < CIRtarget < CIRH , then CIRH is considered as initial CIRtarget value. If

CIRtarget > CIRH , the call is not accepted.

The fine tuning of the CIRtarget value is initially proposed to the requested CIR target


value is performed after the call set-up by the Power Control Algorithm. The system

performance was evaluated through simulation performed in a single cell. The QoS is

defined by considering the new call blocking probability for a certain CIR value. The

propagation model is used to calculate the path loss as a parameter with fading (both

Rayleigh and Log-normal).

The traffic model includes multimedia speech and video calls which are generated

according to Poisson process assuming a mean call duration for both. The mobiles are

uniformly distributed in distance over the cell area.

Refering to equ. (4.3), the spread bandwidth vector, Wk = (W1,W2, ...,WN) consists

a N number of channels available for k requests of i type users in any of the slot in a single

cell TDMA/CDMA system.

Therefore, the total bandwidth available is: W =∑Nk=1 Wk. Then the spreading factor

is: Fik =Wk

ri.

For Kj to be admissible in slot j, it must hold that (please refer to equ. (4.4))

L∑i=1

kij

1 + Fik

mQi

≤ 1

f + 1(4.5)

When a requested channel Wk is not available for a new call/hand-off call, then that

call is assumed to be blocked.

The basic parameters and the pseudo-code of the DCA algorithm are given as follows:

Basic Parameters:

Let CIRtarget = target CIR, CIRL = low CIR, and CIRH = high CIR.

Pseudo-Code:

1. begin

2. compute ’structure matrix’

3. compute ’interference matrix’

4. if 0 < CIRtarget < CIRL then CIRtarget = CIRL

5. if CIRL < CIRtarget < CIRH then CIRtarget = CIRH


6. if CIRtarget > CIRH then call is not accepted

7. update ’structure matrix’

8. update ’interference matrix’

9. end

4.3 Buffer Management

The BS normally has a buffer to avoid packet discard due to competition for a destination

port among several packets. To improve buffer utilization, shared buffer switches have

been proposed. In such a switch, each incoming packet is lined up in the logical queue

for the appropriate output port in order to prevent an out of order packet delivery.

With the employment of a shared buffer switch, when the traffic destined to a certain

output port becomes heavy, the packets belonging to that traffic may occupy the buffer.

This prevents the packets destined to the other destination from entering the switch, and

eventually provides a large packet loss probability. To overcome this problem, a restriction

on the maximum queue length (threshold) has been introduced. When the queue length

exceeds the threshold value, the switch discards the arriving packets to be queued. But

the packet loss probability strongly depends on the traffic conditions. Consequently, one

should adapt the threshold dynamically to the traffic conditions to reduce the packet loss

probability. So, the packet loss probability can be kept below a certain value under any

traffic conditions by changing the threshold dynamically. But there can be a scheme which

decides the threshold for each logical queue according to the establishment and release

of the packet. If the threshold is small, when many packets are stored in buffer, arriving

packets may be discarded even if several buffer spaces are available. These unused spaces

result in inefficient buffer utilization. In our work, we set the thresholds to a higher value.

Thus the buffer may be used more efficiently when the switch becomes congested. To

calculate the threshold dynamically, our method employs a simple linear function which

is suitable for high-speed switching. To evaluate the performance of the proposed protocol,


we consider two classes of traffic, a delay-sensitive class such as voice and a loss-sensitive

class such as text data and investigate methods of calculation of appropriate thresholds

for the individual traffic classes [49, 50].

Calculation of the Dynamic Threshold:

The threshold T(t) at time t is given as follows:

T (t) = c.[B −Q(t)] (4.6)

where B = the shared buffer size, Q(t) = the number of output ports, and the total

queue length at time t and c = a constant (please refer to figure 4.6).

P=1

0

B

BQ(t)

T(t)

P < 1

P > 1

Figure 4.6: Dynamic Threshold Vs. Total Queue Length

An arriving packet destined for output port i is discarded when the queue length Qi(t)

for output port i is larger than or equal to threshold T(t), that is,

Qi(t) ≥ T (t) (4.7)


In equ. (4.6), the threshold is linearly decreased from B to 0 as the number of packets

Q(t) in the buffer increases. Thus equ. (4.6) produces a larger threshold value when Q(t)

is small, especially when the parameter c is set to a value larger than 1. In such cases, equ.

(4.6) does not work. Furthermore, the threshold decreases as Q(t) becomes large, and

eventually all of the logical queue lengths satisfy equ. (4.7). This causes packet discard

even though several buffer spaces are available. To overcome this problem, the threshold

T(t) should be set to a larger value than the logical queue length Qi(t) when the total

queue length is almost equal to the buffer size Q(t) � B. From the above discussion, in

this work, T(t) is calculated as

T (t) = B − P.Q(t) (4.8)

where P is a positive constant.

Figure 4.6 shows the dynamic threshold value as a function of the total queue length.

The simple linear function used in this work can calculate the threshold instantaneously,

which may be suitable for base station switches in which rapid processing is essential.

For simplicity, we assume a discrete time system divided into slots of fixed length

equivalent to the packet time. In such a time system, any events, such as arrivals or

departure of packets, occur at the boundaries of time slots.

Switch Model:

We consider (N x N) switch with a shared buffer which can store up to B packets.

The switch can store N arriving packets and read out N packets for departure within a

time slot. Furthermore, the time spent in the circuits composing the switch is negligibly

small. From the above assumptions, the switch can be modeled as N queues as shown

in figure 4.7. Note that due to the restriction of the shared buffer capacity, the total

queue length is maximum of B. The arriving packets are discarded when the length of the

corresponding logical queue is larger than the threshold or the buffer is full. Otherwise

they are served in FIFO (First-In-First-Out) order.

The basic parameters and the pseudo-code of the buffer management algorithm are

given as follows:


��

��

��

��

��

:

��

0

1

N−1

0

1

N−1

Buffer Size, B

Output PortsInput Ports

Dynamic Threshold, T(t)

:::

:::

Q(t)

Q(t)

Q(t)

1

0

N−1

::

Figure 4.7: Simulation Model of BS Switch for Buffer Management

Basic Parameters:

Let T = upper threshold for the queue length, L = lower threshold for the queue

length, L=bT where b is a constant, and ∆T = threshold control step size = B/K where

B = buffer size and K = number of bands. p1 = actual packet loss rate, θmax = pre-

specified value for packet loss rate by controlling T (vary from L to T), δ = elapsed time

used to check the loss rate mismatch, and ε = tolerance value to eliminate unnecessary

updates.

Pseudo-Code:

1. if |p1 − θmax| > ε for δ secs, then T = [T + ∆T |p1 − θmax|]

2. let ε = 0.01 and δ = 1 sec

3. begin: get new p1 value

4. start = time-now


5. while |p1 − θmax| > ε

6. stop = time-new

7. T = [T + ∆T |p1 − θmax|]

8. break out of the while loop

9. endif

10. get new p1 value

11. endwhile

12. go to begin

Traffic Model:

As the traffic model for each input port, we consider a two-state bursty source. The

source alternates between an on state, in which a packet having the same destination is

generated at every slot, and an off state, in which no packet is generated. The duration

of the on and off periods are both assumed to follow geometric distribution.

The traffic sources are assumed to be identical and independent of each other. We

also assume that at least one packet is generated in each on period, but the duration of

an off period may be zero. Let α denote the probability of transition from on to off; then

the probability p(i) that the duration of the on period is equal to i is

p(i) = α(1− α)i−1; i ≥ 1 (4.9)

Therefore, the mean duration of the on period, that is, the mean burst length Lb is

Lb =∞∑i=1

ip(i) =1

α(4.10)

The probability q(i) that the duration of off period is equal to i is

q(i) = β(1− β)i; i ≥ 0 (4.11)


where β denotes the probability of transitions from off to on. Thus, the mean duration

of the off period Loff is

Loff =∞∑i=0

iq(i) =1− β

β(4.12)

From equations (4.10) and (4.12), the average load on each input port, ρn is

ρn =Lb

Lb + Loff=

β

α + β − αβ(4.13)

From equ. (4.13), one can vary the average load ρn at a given burst length Lb by

adjusting the off duration Loff . In a similar way, Lb can be varied for a given ρn. The

probability γnm that an arriving packet at the input port n (0 ≤ n ≤ N − 1) is destined

to the output port m (0 ≤ m ≤ N − 1) is assumed to be given, where

N−1∑m=0

γnm = 1 (4.14)

4.4 Maintenance of Data Transfer during the Hand-

offs

In wireless networks, we allow user to move around different areas hence causing ”hand-

offs” in the network. When a user arrives for the first time to the network and the

user is assigned a channel, then he/she is considered as a ”new call”. Two of the most

important performance measures in wireless networks are the new call and the hand-off

blocking probabilities. The hand-offs are a fundamental feature in cellular systems, their

performance and efficiency strongly depend on the use of adequate hand-off algorithms.

For cellular communication systems, in order to ensure the mobility and capacity, to

maintain the desired coverage areas, to maintain the data transfer and to avoid problems

of interference, it is necessary to correctly assign the calls to the corresponding service

areas in the whole cell and in the entire network.


The previous studies, however were based primarily on hard hand-off in TDMA sys-

tems. Unlike TDMA systems, CDMA systems can provide soft hand-off. This is a ”make-

before-break” method. When the signal from a new base station is stronger than the

threshold value PADD, a new link to the BS is established while maintaining the existing

link. In this case the call is said to be in soft hand-off. It is assumed that a mobile station

can be in soft hand-off with two BSs with strong signals. If the signal from a third BS

becomes stronger than either of the two strong signals, another hand-off occurs and the

network drops the weakest link. When the signal from either the old BS or the new BS

weakens to below PDROP , the bad connection is released and only a single good connection

is maintained after that time. Soft hand-off region may vary according to hand-off related

parameters such as PADD and PDROP , and hand-off is also affected by radio propagation

characteristics and the required Eb/N0 value. The hand-off mechanism should support

the multi-rate traffic. But high rate traffic suffers from more blockings than the low rate

traffic in a mutimedia traffic environment. In the case of high rate traffic, it is desirable

that hand-off traffic may be controlled not to suffer from severe hand-off failures compared

with low rate traffic [51, 52].

Group II

R

0.775R

CELL A CELL B

Group I

Figure 4.8: A Cellular Hand-off System Model


For our proposed MAC protocol, soft hand-off has been modeled. In order to model

the soft hand-off, we have the following assumptions: calls are uniformly generated within

a cell. The new call arrival to a cell is a Poisson process with rate λ. The time between

decision for the mobility transitions or call termination of a user with a call in progress

will be exponentially distributed with mean 1/µ for the cell. This time is refered as ”dwell

time”. ’R’ denotes the radius of the hexagonal cell. A soft hand-off call occurs when a

mobile station either crosses the boundary or undergoes a transition event as follows:

from the normal area to the hand-off area (Please refer to figure 4.8). If the hand-off area

is larger, then it can support fast moving calls, but for slow moving calls which experience

weak fast fading and do not require receiver diversity, it can be beneficial to perform soft

hand-off, considering both the mobility of calls and the received signal strength or the bit

error rate (BER).

The Connection Admission Control (CAC) for the wireless networks must execute

certain functions for an execution of a handover (hand-off). Therefore, the implications

and requirements imposed on the CAC by the handover procedures are:

• utilisation of the radio link

• protocol

• execution time

• observance of the traffic QoS parameters

The different options and their charactreistics are as follows:

• Bandwidth Reservation: If a single user can occupy the entire bandwidth of a radio

cell, part of the capacity in neighbouring radio cells is not available for allocation

to new users or new connections. This may be the optimum solutions in terms of

execution time and QoS observance, but the utilisation of the radio link is very poor.

• Blocking of Handover Request: Only a part of the current connections are handed

over, the others have to be dropped (if the MS had two separate receivers it might be


possible to temporarily retain the connections which could not be handed over in the

old radio cell). This option is fast and simple if the BSC knows which connections

might be dropped, but in general this is difficult to decide. For example, imagine a

mutimedia session comprising a video and an audio connections. It does not make

sense to retain one of them while dropping the other. Nevertheless, if the coherence

time of radio channel is short (which depends on the radio frequency and the user’s

speed), the handover has to be performed fast. In this case, partial blocking of the

handover request may occur.

::::

(C )

::

::::

hmThe hand−off callsarrival rate as perPoisson Process

Reservation−On−DemandQueue

Radio Resource (C)

ReservedChannel

h

Figure 4.9: Hand-off Management Scheme

We have tackled the hand-off requests by the following method: channels are reserved

for hand-off calls in a typical hand-off scheme because hand-off success is more important

than a new call acceptance. When all channels, except the reserved channels, are used,

new calls are blocked and only hand-off requests are accepted. If all channels are used,

hand-off requests are placed into a queue waiting for a free channel (please refer to figure

4.9).

Let Pr,A = received power at BS A of a signal transmitted by a MS call in soft hand-off


with cell A and B, and Pr,B = received power at BS B of a signal transmitted by a MS

call in soft hand-off with cell A and B. If Pr,A > Pr,B, that is, if BS A is the controlling

BS of the call, the call is in the area of hand-off region for cell A, called as Group I. If

Pr,A < Pr,B, that is, if BS B is the controlling BS of the call, the call is in the area of

hand-off region for cell B, called as Group II.

A channel is borrowed from a call in soft hand-off when an incoming hand-off request

to cell occurs with no free channels. In this case, a channel is borrowed from a call in

the Group I of the cell to release a connection with the noncontrolling BS of the call. If

there are no calls in the Group I of the cell, the incoming hand-off request is placed into

a queue. If there are calls in the Group I of the cell, a channel is borrowed from a call in

the Group I and allocated to the hand-off request. In this case, channel borrowing from a

call in the Group I of one cell corresponds to channel borrowing from a call in the Group

II of a neighbor cell.

Each cell will reserve Ch channels out of a total of C available channels, exclusively for

hand-off calls, because a suddenly forced termination during a call session will be more

upsetting than a failure to connect. Every hand-off requirement is assumed to be perfectly

detected and the assignment of the channel is instantaneous, if the channel is available.

From the intrinsic property of soft hand-off process, a hand-off requirement must not be

denied immediately when there is no available channel. Instead, the hand-off requirement

shall be put into a queueing list. The available maximum queue length is equal to Q.

The ”call duration time” Tc is assumed to be exponentially distributed with mean Tc =

1/µc. Both new call arrivals and hand-off arrivals are assumed to be Poisson distributed

with rates λn and λh respectively. The new call arrival rates in two different regions

Group I and Group II are:

λn1 = the new call arrival rate in hand − off region = aλn (4.15)

and


λn2 = the new call arrival rate in hand− off region = (1− a)λn (4.16)

where

a =area of hand− off region

area of a cell(4.17)

Both the dwell times of a call in two distinct regions are assumed to be exponentially

distributed. The mean dwell time in hand-off region is Td1 = 1/µd and that in normal

region is Td2 = 1/µD.

We assume that when a new call is generated in the hand-off region, it can ask for

channel assignment from both the cell of interest and a neighboring cell with which it can

communicate. If the cell of interest has a channel available for such a generated call, the

new call is considered as a successful new call. If not, the new generated call is said to be

blocked from the point of view of the cell of interest. However, for the later case, the new

call may successfully enter the cellular system by getting a channel assignment from the

neighboring cell and becoming a hand-off arrival for the cell of interest. Of course, if the

new call cannot get channel assignment from either of the two cells, the call is said to be

blocked by this cellular system, and the mobile unit shall try its attempt later.

The basic parameters and the pseudo-code of the hand-off algorithm are given as

follows:

Basic Parameters:

Let C = number of channels, i = number of busy channels, and pi = probability that

i channels are busy.

Pseudo-Code:

1. begin

2. if i = C then

3. if new call then reject


4. if handoff then queue

5. if i << C then

6. if new call then compute pi

7. if rand() < pi then accept

8. else reject

9. if handoff then accept

10. end

Chapter 5

Simulation

We have used C++ programming for developing the protocol and it is purely a computer

simulation. We have simulated this protocol for the system model described earlier. For

the simulation, we have assumed the arrival of packets for all users as Poisson distributed

and the message length is exponentially distributed. The number of data slots in a frame

can be from 2 to 8 slots and the number of CDMA codes per data slot can be from 2

to 5. An efficient random binary back-off algorithm is simulated to take care of packet

collisions. The maximum number of reattempts for request to send data after collisions

permitted for each user is 6.

Parameters Used for Simulation of Random Access:

• number of nodes, N = 32 or 64

• arrival rate, λ = 0.1 to 0.5

• message length, Lm = 50

• code length, L = 15

• frame duration, T = 1.0 sec

• duration of RTS = 0.05 sec

• duration of CTS = 0.05 sec

82

Chapter 5. Simulation 83

• duration of DATA Slot = 0.18 sec

• number of data slots, S = 5

• number of CDMA Codes, U = 2 or 5

Parameters Used for Simulation of Random Access Polling:

• arrival rate, λ = greater than 0.5 and upto 1.0

• a, number of access slots=3

• k, number of slots in one polling cycle=5

• p, number of polling slots=3

• number of reattempts (backoff)=6

The results are obtained as follows :

1. The figure. 5.1 shows the throughput, delay and the rejection rate of the system

for 64 users and for 32 users. The simulation was carried out for 10000 frames. We have

also considered the CDMA codes as 5 per slot in one case and 2 per slot in another case.

2. The figure. 5.2 shows the same results as figure. 5.1 but this one was obtained for

20000 frames of simulation time.

3. The figure. 5.3 shows the throughput and delay performance for 32/64 users with

5/2 CDMA codes. These results are based on Markov Model, a theoretical approach.

So, from the performance curves, we conclude that the delay is more for more number

of users when the traffic is very high and the rejection rate is very low at this point. The

throughput is almost unity at 0.5 arrival rate for both cases of 5 CDMA codes per slot

and for the 2 CDMA codes per slot. But the rejection rate is very high for 2 CDMA codes

per slot compared to that of 5 CDMA codes per slot. The throughput is almost constant

(� 0.8) when the arrival rate is 0.5 and more. This is because of the RAP protocol where

we have used a = 3, p = 3 and k = 5. The rejection rate is very small but the delay


0 0.2 0.4 0.6 0.8 10

0.2

0.4

0.6

0.8

1

arrival rate

Thro

ughput

0 0.2 0.4 0.6 0.8 10

50

100

150

200

250

300

arrival rate

Dela

y(F

ram

es)

0 0.2 0.4 0.6 0.8 10

0.2

0.4

0.6

0.8

1

Arrival rate

Reje

ction r

ate

Simulation Time = 10000 frames

64 users; 5 codes64 users; 2 codes32 users; 5 codes32 users; 2 codes

Figure 5.1: Performance Curves for the Combined Protocol


0 0.2 0.4 0.6 0.8 10

0.2

0.4

0.6

0.8

1

arrival rate

Thro

ughput

0 0.2 0.4 0.6 0.8 10

50

100

150

200

250

300

arrival rate

Dela

y(F

ram

es)

0 0.2 0.4 0.6 0.8 10

0.2

0.4

0.6

0.8

1

Arrival rate

Reje

ction r

ate

Simulation Time = 20000 frames

64 users; 5 codes64 users; 2 codes32 users; 5 codes32 users; 2 codes

Figure 5.2: Performance Curves for the Combined Protocol


0 0.2 0.4 0.6 0.8 10

0.2

0.4

0.6

0.8

1

arrival rate

Th

rou

gh

pu

t

Computer Simulation

0 0.2 0.4 0.6 0.8 10

5

10

15

20

25

30

35

arrival rate

De

lay(F

ram

es)

Computer Simulation

0 0.2 0.4 0.6 0.8 10.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

arrival rate

Th

rou

gh

pu

t

Mathematical Analysis

0 0.2 0.4 0.6 0.8 1−5

0

5

10

15

arrival rate

De

lay(F

ram

es)

Mathematical Analysis

5 users; 2 codes 32 users; 2 codes64 users; 2 codes

Figure 5.3: Performance Curves for the Combined Protocol obtained from the mathemat-ical analysis.


increases with increasing arrival. We also found that the simulation and the theoretical

results are very closer in range.

5.1 Simulation and Results of the Power Control Al-

gorithm

The simulation works are carried out in C++ computer language and the results are

analysed and plotted using MATLAB. Results were generated using N=128, 1 ≤ K ≤ 20.

It was assumed that the chip period Tc = 1µsec. The PN sequences were generated

randomly. We applied it to wireless local area networks (WLANs) in the 2.4 GHz band

with 26 MHz bandwidth.

The following table 5.1 gives the details about the various simulation parameters:

Table 5.1: Simulation parameters used in MAC Power Control

i d v p fb fc Eb/N0

(movement) (distance) (velocity) (Tx. Power) (bit-rate) (chip-rate)1 100 30 0.1 192000 24384000 102 1000 30 0.05 96000 24480000 81 200 30 0.025 48000 24528000 52 1500 30 0.1 192000 24384000 101 300 30 0.05 96000 24480000 82 2000 30 0.025 48000 24528000 51 400 30 0.1 192000 24384000 102 2500 30 0.05 96000 24480000 81 100 30 0.1 192000 24384000 102 1000 30 0.05 96000 24480000 81 200 30 0.025 48000 24528000 52 1500 30 0.1 192000 24384000 101 300 30 0.05 96000 24480000 82 2000 30 0.025 48000 24528000 51 400 30 0.1 192000 24384000 102 2500 30 0.05 96000 24480000 8

In the table 5.1, i = 1 indicates that the mobile user is moving towards the base station


and i = 2 indicates that the mobile user is moving away from the base station.

0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 110

−20

10−15

10−10

10−5

100

data6

Tx Power in Watts

BE

R

12345678

0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 110

−20

10−15

10−10

10−5

100

Tx Power in Watts

BE

R

9 10111213141516

Figure 5.4: Analytical results for 16 Users in 0 to 1.0 Watt Tx. Power Range

These results show that when using a correlation receiver, even slight variation in the

received power levels seriously reduce the capacity of the system. So, permitting slight

variations in the received power levels for the transmitted power levels and plotted the

BER (or Eb

N0) graphs (lower and upper bounds) for different users versus time. We also

considered the fading (both short-term and long-term fadings) combined with the path-

loss and analysed the results (We cosidered an average terrain and hot humid areas with

the link reliability for all users as 99.99%. We also assumed that the short-term fading has

been compensated already). The plots for all these are presented here. The discussions

about each plot are as follows:

i) Figures 5.4 and 5.5 show the analytical results for the BER of 16 mobile users in a

single cell environment with 0 to 1.0 Watts and 0 to 1.0 µW transmitted power range

respectively.

ii) Figures 5.6 and 5.7 show the simulation results for the 16 mobile users of their desired

transmitted power levels to maintain their BER constant under no fading and fading


0 0.2 0.4 0.6 0.8 1 1.2

x 10−5

10−20

10−15

10−10

10−5

100

data6

Tx Power in Watts

BE

R

12345678

0 0.2 0.4 0.6 0.8 1 1.2

x 10−5

10−20

10−15

10−10

10−5

100

Tx Power in Watts

BE

R

9 10111213141516

Figure 5.5: Analytical results for 16 Users in 0 to 1.0 µW Tx. Power Range

0 1 2 3 4 5 6 7 8 9 1010

−4

10−2

100

102

TRANSMIT POWER LEVELS FOR 16 Users(without fading)

Time (sec)

Tx

pow

er w

ith c

ontr

ol (

Wat

ts)

12345678

0 1 2 3 4 5 6 7 8 9 1010

−4

10−2

100

102

Time (sec)

Tx

pow

er w

ith c

ontr

ol (

Wat

ts)

9 10111213141516

Figure 5.6: Power control analysis under no fading conditions


0 1 2 3 4 5 6 7 8 9 1010

−5

100

105

TRANSMIT POWER LEVELS FOR 16 Users(with fading)

Time (sec)

Tx

pow

er w

ith c

ontr

ol (

Wat

ts)

12345678

0 1 2 3 4 5 6 7 8 9 1010

−5

100

105

Time (sec)

Tx

pow

er w

ith c

ontr

ol (

Wat

ts)

9 10111213141516

Figure 5.7: Power control analysis under fading conditions

0 1 2 3 4 5 6 7 8 9 1010

−20

10−15

10−10

10−5

100

Time (sec)

BE

R w

ith p

ower

con

trol

BER PERFORMANCE FOR 16 Users(without fading)

12345678

0 1 2 3 4 5 6 7 8 9 1010

−20

10−15

10−10

10−5

100

Time (sec)

BE

R w

ith p

ower

con

trol

9 10111213141516

Figure 5.8: BER analysis under no fading conditions


0 1 2 3 4 5 6 7 8 9 1010

−10

10−5

100

Time (sec)

BE

R w

ith p

ower

con

trol

BER PERFORMANCE FOR 16 Users(with fading)

12345678

0 1 2 3 4 5 6 7 8 9 1010

−10

10−5

100

Time (sec)

BE

R w

ith p

ower

con

trol

9 10111213141516

Figure 5.9: BER analysis under fading conditions

conditions respectively.

iii) Figures 5.8 and 5.9 show the simulation results for the 16 mobile users of their BER

under no fading and fading conditions respectively.

iv) And, finally it can be seen that the analytical results of figures 5.4 and 5.5 and the

simulation results of figures 5.8 and 5.9 are coincident within the limits of approximation.

5.2 Results of the Admission Control of the Protocol

The simulation results for the call-admission control technique are given in figures 5.10

and 5.11. The new call blocking probability is increasing with increasing arrival of both

new calls and the hand-off calls. The CAC performance is better when there is no fading

in the wireless channel.


0 5 10 15 20 25 30 350.3

0.4

0.5

0.6

0.7

0.8

0.9

1

number of new calls arrival

new

cal

ls b

lock

ing

prob

abili

ty

with fading (32 users)

Figure 5.10: New Call Blocking Probability Performance with fading of channel

0 5 10 15 20 25 30 350.05

0.1

0.15

0.2

0.25

0.3

0.35

0.4

0.45

0.5

number of new calls arrival

new

cal

ls b

lock

ing

prob

abili

ty

without fading (32 users)

Figure 5.11: New Call Blocking Probability Performance without fading of channel


5.3 Results of Channel Allocation Scheme of the Pro-

tocol

The simulation results of DCA are given in figures 5.12 and 5.13. The DCA algorithm

performance is better with power control algorithms. The channel blocking probability is

very less with power control compared to the case of without power control. The results

can be tuned for various factors like different classes of traffic (CBR, VBR, etc.), voice

activity factor, and the CDMA processing gain.

0 5 10 15 20 25 30 350

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

0.9

1

Number of Users

Chan

nel B

lock

ing

Prob

abilit

y

Without Power Control

Figure 5.12: Channel Blocking Probability without Power Control

0 5 10 15 20 25 30 350

0.1

0.2

0.3

0.4

0.5

Number of Users

Chan

nel B

lock

ing

Prob

abilit

y

With Power Control

Figure 5.13: Channel Blocking Probability with Power Control


5.4 Results of Buffer Management Scheme

The simulation results for the buffer management system are given in figures 5.14 and

5.15. The packet loss probability performance is varied for various suitable optimization

factor P with different arrival rate. The packets loss curves are given for different buffer

threshold levels and so the packets loss can be set at minimum by choosing proper buffer

threshold in the flexible buffers.

0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 20

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

0.9

1

Parameter, P

Pack

et L

oss

Prob

abilit

y

mean message length = 50

arrival rate=0.1arrival rate=0.5arrival rate=1.0

Figure 5.14: Packet Loss Probability Vs. Buffer Threshold Optimization Parameter

0

2

4

6

8

10

12

14

16

18 20 22 24 26 28 30

Pac

ket L

oss

(num

ber o

f pac

kets

)

Buffer Threshold

"data.dat"

Figure 5.15: Packets Loss Vs. Buffer Threshold


5.5 Results of Maintenance of Data Transfer during

the Hand-offs

The simulation results of a soft hand-off algorithm are given in figures 5.16 and 5.17.

The hand-off calls blocking probability performance is produced here for both faded and

non-faded wireless channel cases. The power control algorithm reduces the hand-off calls

blocking probability in the faded channel.

0 5 10 15 20 25 30 350.2

0.25

0.3

0.35

0.4

0.45

0.5

number of hand−off calls arrival

hand

−off

calls

blo

ckin

g pr

obab

ility

with fading (32 users)

Figure 5.16: Hand-off Call Blocking Probability Performance with fading of channel

0 5 10 15 20 25 30 350.05

0.1

0.15

0.2

0.25

0.3

0.35

0.4

number of hand−off calls arrival

hand

−off

calls

blo

ckin

g pr

obab

ility

without fading (32 users)

Figure 5.17: Hand-off Call Blocking Probability Performance without fading of channel

Chapter 6

Testing of Protocol in Software

Radio Environment

Software radios have become well known as communication devices for performing most

of the signal processing in software. The use of DS/CDMA (Direct Sequence/Code Divi-

sion Multiple Access) technology in software radio is emerging as a topic of widespread

interest. The key issues in the implementation of software radios in a CDMA system

are the development of wide-band source coders, flexible channel codecs with powerful

error correction and small processing overhead and efficient digital modems with optimal

performance in the presence of Multiple Access Interference (MAI) and Additive White

Gaussian Noise (AWGN). This section gives a brief description of the simulation of certain

specific blocks of a software radio architecture like source coding (Huffman, Shannon-Fano

and PCM), channel coding ((17,9) double error correcting block code, (15,7) double error

correcting BCH code, (2,1) burst error correcting (upto 6 burst errors) convolution code),

digital modulation scheme (BPSK). We have considered a pico cell with a maximum of 7

users and studied the power efficiency of combined channel coding and modulation with

perfect power controlled CDMA system. Thus our simulation of the ”software radio”

has flexibility in choosing the proper channel coders dynamically depending upon the

variations of AWGN channel.

96

Chapter 6. Testing of Protocol in Software Radio Environment 97

6.1 Software Radio and its Layered Architecture

A software radio is a communication device that does all its digital signal processing in

user-level software. Wireless networks are statically specified by their built-in link and

physical layer functions. Conventional software radios effectively use improved analog

to digital converters and digital signal processing on a dedicated hardware, i.e., network

interface card (NIC). Implementing all the link and most of the physical layer functions

in user level software increases the flexibility and makes it possible to dynamically modify

functions such as modulation techniques and multiple access control techniques which are

otherwise fixed in traditional NICs. The signal processing involved in these layers has

been lumped into one layer because of its implementation in dedicated hardware. But to

interoperate with different networks, it may only be necessary to change small parts of

the existing layers. For instance two different computers may employ the same modula-

tion and coding but use different multiple access protocols or a given computer may need

to change the type of coding to dynamically adapt to changing channel conditions. To

facilitate this flexibility and modularity as much of the processing as possible is brought

under software control. New network interfaces can now be created by changing only a

small amount of code [53-60].

We consider a pico cell with fixed users and study the BER performance in an AWGN

channel. The generic block diagram of the software radio is shown in figure 6.1. Certain

specific blocks of these virtual radios (software radios) have been simulated. The layering

presented in figure 6.2, is a refinement of the OSI layering model which subdivides the

existing Data Link and Physical layers. The time to time variations of the AWGN channel

charecteristics are provided to the computers by a control DSP chip and the computers can

choose the type of channel coders dynamically so that optimum performance is achieved

for the applications that are in communication. We consider a DS-SS multiple access

system with BPSK signalling and a correlation receiver. There are K users in the system.


C H A N N E L

InformationSource

InformationSink

Digital Input

Source Bits Channel Bits

Channel Bits

From other

sources

Digital WaveformsBit Stream

Source

EncodingFormatting

Channel

Encoding

Multiplex−

ingEncryption Modulation

Multiple

Access

T

RX

R

XR

Multiple

AccessReceiver

Demodula−

tion

Demulti−

plexing

Channel

Decoding

DecryptionSource

DecodingFormatting Spectrum

Spread

SpreadSpectrum

Transmitter

To other sinks

Digital Output

Sink Bits

Figure 6.1: Software Radio Block Diagram

D A T A L I N K

P H Y S I C A L

D A T A L I N K

P H Y S I C A L

LINK FRAMING LINK FRAMING

MODULATION

MULTIPLE ACCESS MULTIPLE ACCESS

R F TRANSMITTER R F RECEIVER

DEMODULATION

CHANNEL DECODING

LINE DECODING

D/A CONVERSION A/D CONVERSION

LINE ENCODING

CHANNEL ENCODING

COMPUTER A COMPUTER B

Bytes Bytes

Bits Bits

Bits Bits

Symbols Symbols

Discrete Signal Discrete Signal

Discrete SignalDiscrete Signal

Continuous Signal

Continuous Signal

Continuous Signal

Continuous Signal

HARDWARE

SOFTWARE

MEDIUMWIRELESS

Figure 6.2: Software Radio Layering Model


6.2 Source Encoding and Decoding Methods

Analog-to-Digital Convertors (ADCs) and Digital-to-Analog Convertors (DACs) are crit-

ical components of software radios. Advances in processor technology combined with

availability of high density memory and improvement in I/O bandwidth have made soft-

ware approach an increasingly attractive alternative for implementing radio-based sys-

tems. The greatest advantage is high flexibility by allowing the support of a wide range

of modulation and coding schemes. ADCs can now be used to only sample an entire IF

band and the resultant samples streamed into computer’s memory for software processing.

That is the NIC can now be designed to contain the hardware part of A/D, i.e., the sam-

pler, producing discrete analog voltages. Quantisation and encoding can be implemented

in user-level software as shown in the figure 6.3.

S A M P L E R Q U A N T I Z E R E N C O D E R

Bit StreamContinuous-time

signal

Discrete-time

signal

SOFTWAREHARDWARE

Discrete-timesignal

H/S Interface

Figure 6.3: A/D Converter in Software Radio

It is assumed that the input signal has voltage levels lying within +10.0 to -10.0.

In uniform quantization the voltage range is divided into 64 discrete levels of step size

20.0/64. All sampled analog voltages lying within this range are encoded using one of the

source encoding techniques described below.


Pulse code modulation (PCM) encoding scheme at a resolution (number of bits per

sample) of 6 involves assigning a 6 bit code word for the quantized levels in sequence, i.e.,

from 000000 to 111111. The sampled voltage is shifted by adding +10.0. The resultant

voltage is closely approximated to one of the quantized levels and the code corresponding

to that level is assigned to the sample. The serious drawback of this encoding scheme is

that large number of bits have to be handled. For example, if a sample occurs a large

number of times, say 100, then a total of 600 bits must be transmitted.

Huffman and Shanon-Fano encoding algorithms have been used for data compression.

The pdf (probability density function) of the signal must be known and to achieve

data compression, the sample values of the signal must not be uniformly distributed.

Variable length code words are obtained if the pdf is not uniformly distributed. In the

simulation we have assumed that all voltage levels between -10.0 and +10.0 are equally

likely to occur and a fixed probability of 1.0/64 is assigned to each of the quantized levels.

Fixed length code words of 6 bits were obtained as a result.

The Huffman and Shanon-Fano codes have the property that no code word is a prefix

of another code word. Hence decoding algorithm is simple. A lookup table of all possible

codes generated at the transmitter must be maintained at the receiver for decoding. In

the case of PCM fixed length codes and fixed probability case in Huffman and Shanon-

Fano encoding, decoding is done by taking 6 bits at a time and comparing with the set

of 64 possible codes. The position of the code in the table is mapped on to the set of

quantized levels to get the analog voltage. The only error mechanism present in the A/D

convertor is quantizing error.

6.3 Channel Encoding and Decoding Techniques

The data link layer is responsible for error free transmission. It packages the raw bits

from the physical layer into blocks of data (frames) and sends these frames with the

necessary error control to the network layer. A channel code is designed specifically for


error detection, error correction or error prevention. Errors occur primarily in the channel

(wireless medium) due to multipath fading, shadowing, interference, reflections and path

loss and thermal noise interference in the digital hardware. To detect and correct errors

three powerful channel encoding and decoding techniques have been simulated.

The (17,9) Block Code (corrects upto 2 errors):

In this type of coding, the binary message sequence is divided into sequential blocks

each 9 bit long and each 9 bit block is coded into 17 bit block using systematic coding,

i.e., the first 9 bits of the 17 bit block are the message bits and the last 8 bits are the

parity check bits.

Let D = [d1 d2.......dk] be the data vector and C = [c1 c2.......cn] the code vector. G is

a k×n generator matrix defined by G =[Ik P T

], where Ik is a unit matrix of order k.

P is a (n−k)×k co-efficient matrix and P T is the transpose of P. The codes are generated

using the generator matrix as follows C = DG. Let H be a (n−k)×n parity-check matrix

defined by H =[P In−k

]. It is easily verified GHT = 0. Let R denote the received

vector that results from sending the code vector C over the a noisy channel. Let E be the

error vector such that R = C ⊕ E. Atmost t elements of E can be 1 and the remaining

must be 0 for correcting upto t errors. Post multiplying both sides of R = C ⊕ E by HT

and using that CHT = DGHT = 0 we get S = RHT = EHT where S is the syndrome

vector. If E = 0 i.e. no error then S = 0. If a single error has occured then S is identical

to a row of the HT matrix and that particular row position gives the position of the

element in R which is in error. If two errors have occured then S is identically equal to

the sum of some two unique rows of HT and these two row positions give the positions of

the elements in R which are in error. Error correction is simply negating the elements at

these positions. In all operations, addition means modulo 2 addition [61-68].

Decoding by syndrome calculation places restrictions in choosing the coefficient matrix

P. It is necessary that all rows of HT be distinct and the sum of any 2 rows of HT is

unique and not a row of HT . The HT matrix and the P T matrix for (17,9) code is built


by a simple algorithm given below.

a = n-k;

b = k-1;

while(true)

( P T [b] = get new (n-k) bit word;

HT =

P T [b]

In−k

ctr = 0;

for(i=0;i < a+ 1;i++)

( for(j=i+1;j < a+ 1;j++)

( sum[ctr] = HT [a+ 1 + i] +HT [a + 1 + j];

ctr++ ;)

)

for(i=0;i < (a + 1)C2;i++)

( if (sum[i] distinct && �= rows of HT )

( a = a + 1;

b = b + 1;

break;)

else goto P T [b];

)

if (b = 0) break;

)

For (17,9) double error correcting block code the P T matrix is shown below. Any

combination of 7 rows of this matrix can be used in (15,7) double error correcting block

code. (17,9) and (15,7) code have the same error correcting capability but (17,9) is more

bandwidth efficient.


PT =

1 1 1 1 1 1 1 1

0 0 1 0 0 1 1 1

1 0 0 1 0 0 1 1

1 1 0 0 1 1 1 0

1 1 1 1 0 0 0 0

0 1 1 1 1 0 0 1

1 0 0 1 1 1 0 1

1 1 0 0 0 1 0 1

1 0 1 0 1 0 1 1

The (15,7) BCH Code (corrects upto 2 errors):

(15,7) BCH code can correct upto 2 errors and non-systematic coding is used in the

simulation i.e. the message bits cannot be obtained directly from the coded vector. If

D(x) be the message polynomial, C(x) the code polynomial then C(x) = D(x)G(x) where

G(x) is the generator polynomial. The degree of the polynomials C(x) and G(x) are (n-1)

and (n-k) respectively. The generator polynomial of (15,7) BCH code is given by G(x) =

1⊕X4 ⊕X6 ⊕X7 ⊕X8.

If R(x) be the received vector then

R(x) = C(x)⊕ E(x)

E(x) is the error vector and α,α2,α3,.....α2t are the roots of G(x), α being the primitive

element of GF(2m).

R(αi) = C(αi) + E(αi), i = 1,2,3,....,2t.

R(αi) = D(αi)G(αi) + E(αi)

R(αi) = E(αi)


since G(αi) = 0, αi ’s being the roots of G(x). For the (15,7) BCH code m = 4 and t

= 2 in our case.

R(αi) =∑n−1j=0 ejα

i+j,ej = {0,1}.

This set of 2t equations involves only the components of the error pattern E(x). Solving

these 2t non-linear equations for ej the error pattern can be determined. The syndrome

vectors can now be defined as:

Si =∑n−1j=0 ejα

i+j.

In this case i = 1,2,3,4 and j = 0,1,2,......,14 and only 4 syndrome equations need be

solved to locate the error patterns. After computing the syndromes specific steps were

followed for decoding. Firstly, assume that 2 errors have occured in the received data

vectors and then computing the determinant as

∆ =

∣∣∣∣∣∣∣S1 S2

S2 S3

∣∣∣∣∣∣∣.

If ∆ �= 0 then 2 errors have occured in R(x) and

M =

S1 S2

S2 S3

.

Then compute the following matrix

λ1

λ2

= M−1

S3

S4

and the λ values give the eigen values. The roots of the quadratic equation


λ(z) = λ1z2 + λ2z + 1

give the positions at which error has occured. The quadratic equation cannot be solved

using the conventional algebraic method. The quadratic equation first has to be factorised

into 2 factors and the coefficient of z in each factor gives αr where r = 0,1,2,......,14. The

value of r in the two factors gives the positions at which error has occured. But if ∆ = 0

and S1 �= 0 then a single error has occured and M = [S1] and the entire procedure is

repeated. But if S1 = 0 there is no error in the received polynomial. After error detec-

tion and correction the message bits can be recovered by dividing the received, corrected

polynomial by the generator polynomial.

The (2,1) Convolution Code (corrects upto 6 burst errors):

The entire hardware for encoding and decoding of the information bits is simulated.

Initially, before the beginning of encoding all shift-registers are set to zero. The hardware

for the encoder is shown in figure 6.4. There is one sequence of input bits and two se-

quences of output bits, one a duplicate of the input bits and the other is the parity-check

sequence from the output of the EX-OR gate. Bits are transmitted alternately from the

two output sequences.

At the receiver, the received sequence is divided into two sequences, and from the

information bits the parity-check bits are recalculated. These are added (modulo 2 ad-

dition) to the received check bits to form the syndromes. In the absence of errors the

syndrome is a sequence of all zeros. If errors occur, a pattern occurs in the syndrome that

enables with this code the correction of any burst-error of length atmost 6. The hardware

for the decoder circuit is shown in figure 6.5.

The message bits are transmitted serially with continuous processing of information

and it is necessary to maintain an error-free guard space of 19 preceding and following

a block of 6 message bits which is difficult to maintain in a noisy dispersive channel.


Data Bits

Parity Bits

D D D D D D1 2 3 4 5 6

EX-OR

Coded Data Bits

Figure 6.4: Encoder for a (2,1) convolution code

Corrected Data Bits

D D D D D D DD D1 2 3 4 5 6 7 8 9

D D D D D D10 11 12 13 14 15

AND

NOT

EX-OR

Syndromes

EX-OR

Parity Bits

Data Bits

Figure 6.5: Syndrome calculation and error-correction for the convolution code


This coding produces optimum error correction for those channels which produce burst

errors. The tap positions in the encoder and decoder circuitry are determined by the

sub-generator polynomial and the sub-generator polynomial in this simulation is given by

G(x) = 1⊕X3.

6.4 TDMA/CDMA System and Results

The TDMA/CDMA MAC protocol is tested based on the software radio environment

(Digital Modulation Through AWGN, Rayleigh Fading, and Path-loss Channel).

In our TDMA/CDMA system, each user is given SS code for transmitting over the

common channel bandwidth. We make the following assumptions

• Perfect code acquisition and tracking is possible.

• All users in the DS/CDMA system communicate via a common channel bandwidth

using coherent BPSK data modulation at a common carrier frequency fc and data

rate Rb =1Tb, where Tb is the duration of a data bit.

• All users have the same chip rate Rc =1Tc

for their spread spectrum codes, where

Tc is the chip duration.

• The effect of the multiple access interference on the matched filter output of a desired

user will be assumed to be Additive White Gaussian Noise [69, 70].

The results are shown from figure 6.7 to figure 6.23 describing about the perfect power

control of all users’ received powers in the CDMA receiver considering a maximum of 7

users (the user transmits a speech signal as shown in figure 6.6) in the pico cell in an

AWGN channel.

This burst error correcting code should give better results than the block and BCH

code in a channel producing burst errors and is ideal for fading channels. It is necessary to

maintain an error-free guard space of 19 bits preceding and following a block of 6 message

bits in the convolution coding. This is difficult to maintain in a noisy, dispersive channel


0 20 40 60 80 100 120−0.4

−0.3

−0.2

−0.1

0

0.1

0.2

0.3

0.4

Time (msec)

Am

plitu

de

Figure 6.6: Speech Signal of user 1

1 2 3 4 5 6 710

−5

10−4

10−3

10−2

10−1

100

Number of Users

BE

R

Ideal (σN

= 0.0)σ

N = 1.0

σN

= 2.0 σ

N = 3.0

σN

= 4.0 σ

N = 5.0

Figure 6.7: Power Control for uncoded BPSK modulation with different noise varianceσN


1 2 3 4 5 6 710

−5

10−4

10−3

10−2

10−1

100

Number of Users

BE

R

Ideal (σN

= 0.0)σ

N = 1.0

σN

= 2.0 σ

N = 3.0

σN

= 4.0 σ

N = 5.0

Figure 6.8: Power Control for block coded BPSK modulation with different noise varianceσN

1 2 3 4 5 6 710

−5

10−4

10−3

10−2

10−1

100

Number of Users

BE

R

Ideal (σN

= 0.0)σ

N = 1.0

σN

= 2.0 σ

N = 3.0

σN

= 4.0 σ

N = 5.0

Figure 6.9: Power Control for cyclic coded BPSK modulation with different noise varianceσN


1 2 3 4 5 6 710

−5

10−4

10−3

10−2

10−1

100

Number of Users

BE

R

Ideal (σN

= 0.0)σ

N = 1.0

σN

= 2.0 σ

N = 3.0

σN

= 4.0 σ

N = 5.0

Figure 6.10: Power Control for convolution coded BPSK modulation with different noisevariance σN

1 2 3 4 5 6 710

−5

10−4

10−3

10−2

10−1

100

Number of Users

BE

R

Ideal (σN

= 0.0)

Uncoded Block coded Cyclic coded Convolution coded

Figure 6.11: Power Control for different channel coding shemes with BPSK modulationwith noise variance σN = 1.0


1 2 3 4 5 6 710

−5

10−4

10−3

10−2

10−1

100

Number of Users

BE

R

Ideal (σN

= 0.0)



1 2 3 4 5 6 710

−5

10−4

10−3

10−2

10−1

100

Number of Users

BE

R

Ideal (σN

= 0.0)




−15 −10 −5 0 5 10

10−3

10−2

10−1

100

Eb/N

0 (dB)

BE

R

Uncoded BPSK (analytical)Uncoded BPSK (simulation)without Power Control with Power Control

Figure 6.16: BER for user 1 with uncoded BPSK modulation when only one user is inthe cell

−15 −10 −5 0 5 10

10−3

10−2

10−1

100

Eb/N

0 (dB)

BE

R

Uncoded BPSK (analytical)Uncoded BPSK (simulation)without Power Control with Power Control

Figure 6.17: BER for user 1 with uncoded BPSK modulation when seven users are in thecell


−15 −10 −5 0 5 10

10−3

10−2

10−1

100

Eb/N

0 (dB)

BE

R

Uncoded BPSK (analytical) Block coded BPSK (simulation)without Power Control with Power Control

Figure 6.18: BER for user 1 with block coded BPSK modulation when only one user isin the cell

−15 −10 −5 0 5 10

10−3

10−2

10−1

100

Eb/N

0 (dB)

BE

R

Uncoded BPSK (analytical) Block coded BPSK (simulation)without Power Control with Power Control

Figure 6.19: BER for user 1 with block coded BPSK modulation when seven users are inthe cell


−15 −10 −5 0 5 10

10−3

10−2

10−1

100

Eb/N

0 (dB)

BE

R

Uncoded BPSK (analytical) Cyclic coded BPSK (simulation)without Power Control with Power Control

Figure 6.20: BER for user 1 with cyclic coded BPSK modulation when only one user isin the cell

−15 −10 −5 0 5 10

10−3

10−2

10−1

100

Eb/N

0 (dB)

BE

R

Uncoded BPSK (analytical) Cyclic coded BPSK (simulation)without Power Control with Power Control

Figure 6.21: BER for user 1 with cyclic coded BPSK modulation when seven users are inthe cell


−15 −10 −5 0 5 10

10−3

10−2

10−1

100

Eb/N

0 (dB)

BE

R

Uncoded BPSK (analytical) Convolution coded BPSK (simulation)without Power Control with Power Control

Figure 6.22: BER for user 1 with convolution coded BPSK modulation when only oneuser is in the cell

−15 −10 −5 0 5 10

10−3

10−2

10−1

100

Eb/N

0 (dB)

BE

R

Uncoded BPSK (analytical) Convolution coded BPSK (simulation)without Power Control with Power Control

Figure 6.23: BER for user 1 with convolution coded BPSK modulation when seven usersare in the cell


and can be avoided by splitting the continuous bit stream at the input of the encoder into

blocks of 6 bits and appending 6 zeros following the least significant information bit to

convert into 12 bit blocks. This is to ensure that all shift-registers are reset to zero and

the message bits are shifted out. The 6 information bits are now dispersed in 24 bits at

the output of the encoder. The continuous processing is now replaced by block processing

where the bits enter the encoder in blocks of 6 bits and leave in blocks of 24 bits. The

data rate is increased by 2 times but at the same time the guard space requirement is not

required and error performance in a bursty channel will be much better.

Chapter 7

Conclusions

In this research work, the hybrid TDMA/CDMA protocol combined with RAP has been

proposed to integrate light and heavy traffic that have to be accommodated for mobile

multimedia communications. The proposed protocol is verified by both simulation and

analytical modelling. This gives great advantages over the CSMA, polling, TDMA and

CDMA protocols in comparison to their packet delays and throughput. This proposed

protocol gives more throughput and less packet delay compared other protocols. Since,

the CDMA codes are more expensive and we need more number of CDMA codes in

CDMA system, we use only a minimum of 2 CDMA codes and a maximum of 5 CDMA

codes to get very high throughput and so our system is more economical than other MAC

protocol systems for WLANs. The rejection rate can be brought to very low level by RAP

method when the the arrival rate of users is very high. Thus we have combined the merits

of TDMA, CDMA and polling into a more attractive protocol which is very useful for

the current multimedia WLANs. The protocol performance is also tested by considering

the various digital modulation schemes, channel coding schemes and the various channel

fading conditions and that makes this new protocol system to be more realistic for UMTS

and IMT-2000. [71, 72]

The wideband channel measurement results for an indoor office environment at fre-

quency 2.4 GHz are presented and analytical BER model for the fading is simulated.

i) Path-loss: The parameters of a simple path-loss model have been studied. It can be

118

Chapter 7. Conclusions 119

concluded that this model gives accurate results for LOS situations.

ii). Fading: The short-term and long-term fadings are analysed using well known proba-

bilistic models.

iii) Power control: We found that the system capacity is improved with this Efficient Power

Control Mechanism (EPCM) considering MAI, path-loss and fading. We also found that

the transmitted power levels are within the reasonable limits and that are possible in

practice.

iv) BER results: The BER for a particular user is within the reasonable limit with rea-

sonable transmitted power levels which can change with time as the user moves randomly

with variable data rate (for a given application like voice,video or data). It is concluded

that this power control mechanism has more flexibility in the sytem for a given quality of

service.

v) The results of the analytical model show that the assumptions made in the derivation of

the end equations are correct because the results are more coincident with the simulation

results well within the limits of approximations.

The various applications of this proposed protocols such as Call Admission Con-

trol, Dynamic Channel Allocation, Buffer Management and the Hand-off methods were

simulated and their performance results were obtained in terms of arrival rate of new

calls/hand-off calls and the call blocking probabilities. We have tested the designed MAC

protocol in Software Radio environment and found that the QoS parameters of the pro-

tocol can be tuned to a desired level with this flexbile testing method.

7.1 Future Work

In future it should be possible to install a software radio just by down-loading the software

and with the necessary hardware of the work-station. The final conclusions of this work

for a better future generation multimedia wireless networks are given as follows:

The users should be able to have the features of the 4G MWNs so that they could

communicate at anytime, from anywhere and in anyform. The future wireless networks

may operate in THz frequencies with Gbits of data transmission.

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