Optimizing Voice in ATM/ IP Mobile I Networks
(TELECOM ENGINEERING I)
Juliet Bates- McGrawHill
Contents
Foreword
Preface
Acknowledgments
Introduction
Chapter 1 Mobile Voice Access to ATM/ IP Networks 1
1.1 Second- and Third- Generation Mobile Networks Standards 2
1.2 ATM and
IP Transport in UMTS 3
1.3 ATM in UMTS 4
1.4 The UTRAN
1.5 Speech Coding in the UTRAN 5
1.6 QoS in IP Networks 6
1.7 Media Gateways (MGWs) 7
1.8 Media Gateway Controller 9
1.9 Interworking Between IP and ATM 11
Chapter 2 PBX Voice Access to ATM 13
2.1 Disadvantages of Inherited PBX Solutions 13
2.2 Advantages of Migrating Voice to ATM 14
2.3 Virtual Private Networks (VPNs) 5
2.3.1 Enterprise VPNs 1
2.3.2 PVC- Based Enterprise VPNs 15
2.3.3 SVC- Based Enterprise VPNs 6
2.4 G.711 Coding 17
2.5 Signaling Information 18
2.5.1 Channel Associated Signaling (CAS) 1
2.5.2 QSIG and DPNSS 9
2.5.3 Common Channel Signaling (CCS) 1
2.5.4 Private Signaling System No.1 (PSS1) 19
2.5.5 Narrowband ISDN and Q.931/ DSS1
2.6 Transporting PBX Signaling 20
2.6.1 Transporting Signaling Transparently 2
2.6.2 Interworking Signaling 21
Chapter 3 Mathematical Modeling of Voice Traffic 23
3.1 Statistical Gain in the ATM Core 23
3.2 Measurement of Conversational Voice 24
3.2.1 Gruber's Results 25
3.2.2 Brady's Results 6
3.2.3 Sriram's Results
3.2.4 Heffes and Lucantoni's Results 27
3.2.5 Sriram and Whitt's Results 8
3.2.6 Cheng's Results 31 Conclusions 33
Chapter 4 Transporting Voice over ATM 35
4.1 Adaptation to ATM 35
4.2 Alternative Adaptation Layers for Voice over ATM 35
4.3 Circuit Emulation Using AAL1 36
4.3.1 AAL1 Packet Data Unit (PDU) Format 3
4.3.2 Unstructured Circuit Emulation 37
4.3.3 Structured Circuit Emulation 8
4.4 Bandwidth Requirements for Circuit Emulation 3
4.4.1 Circuit Emulation in Unstructured Mode 8
4.4.2 Circuit Emulation in Structured Mode 39
4.5 Voiceband Services Using AAL2 41
4.5.1 AAL2 Common Part Sublayer (CPS) 43
4.6 Voiceband Services Using AAL5 45
4.6.1 Voice and Data Within an FRF.11 Subframe 4 Conclusions 47
Chapter 5 Voiceband Processing 49
5.1 Compression 50
5.2 Clear Channel 51
5.3 Compression Coder Units 5
5.4 Combining Coded Units in an AAL Frame 51
5.4.1 Combining Coder Units into an AAL2 Frame 53
5.4.2 Combining Coded Units into an AAL5 Frame 56
5.5 Comparison of AAL1, AAL2, and AAL5 Bandwidth 58
5.6 Comparison of Delay in AAL1, AAL2, and AAL5
5.7 Diverse Bandwidth Requirements of a Single Voice Source 63
5.8 Silence Detection and Suppression 64
5.9 Benefits of Silence Suppression
5.10 Silence Detection Overheads 65 Conclusions 66
Chapter 6 Quality of Service in an ATM Network 69
6.1 Traffic Management in an ATM Network 69
6.2 Service Categories and QoS Classes 70
6.2.1 Constant Bit Rate (CBR) 1
6.2.2 Variable Bit Rate (VBR) 72
6.2.3 Available Bit Rate (ABR)
6.2.4 Unspecified Bit Rate (UBR) 72
6.3 ATM QoS Classes 7
6.4 ATM QoS Class Parameters 3
6.4.1 Cell Loss Ratios (CLRs) Within a QoS Class 73
6.4.2 Cell Transfer Delay (CTD) Within a QoS Class 74
6.4.3 Cell Delay Variation (CDV) Within a QoS Class 75
6.4.4 End- to- End CTD and CDV 76
6.5 Relationship Between Load, CLR, and CDV in a Single Shared Buffer 77
6.6 QoS Targets 78
6.7 Traffic- Descriptor Parameters 79
6.8 Call Admission Control (CAC) Algorithm 80
6.8.1 Steady State Queue Behavior
6.8.2 Defining the Effective Bandwidth 81
6.9 Overbooking 8 Conclusions 2
Chapter 7 The Voice Model 85
7.1 Characterization of Voice 86
7.2 Uniform Arrival and Service Model 87
7.3 Notation of Queuing Systems 8
7.4 Queue Emptying 89
7.5 Stochastic Modeling of a Single Queue in an ATM Switch 89
7.5.1 Modeling at the Talkspurt Level 90
7.5.2 The Effects of Shaping 91
7.6 Sizing the Buffer 92
7.6.1 Meeting a Delay Target 9 Conclusions 3
Chapter 8 Traffic Modeling at the Talkspurt Level 95
8.1 Lifetime of a Talkspurt 95
8.2 Markovian Methods of Modeling the Buffer Queue 96
8.2.1 Continuous Time Markov Chain 9
8.2.2 Probability of the Last Cell
8.2.3 Birth and Death Process 98
8.2.4 A Pure Birth or Death Process 99
8.3 Estimating the Number of Calls Concurrently in a Talkspurt 99
8.4 Effective Bandwidth 100
8.5 A Method for Dimensioning the MBS 101
8.6 Setting the PCR, SCR, and MBS Traffic Descriptors 101 Conclusions 105
Chapter 9 Application of a Call Admission Control (CAC) Algorithm 107
9.1 Call Admission Control (CAC) 107
9.2 Scaling the PCR and MBS Traffic Descriptors 108
9.3 Application of CAC 109
9.4 Comparison of Effective Bandwidth Versus CLR Target and Queue Length 10
9.5 Calculating Effective Bandwidth for an Nrt- VBR Service 113
9.6 Analysis of Results for Nrt- VBR 115
9.7 Maximal- Rate Envelope 116
9.8 Tolerable Queue Length Due to Delay 117
9.8.1 Effective Bandwidth for an Rt- VBR Class of Service 118
9.8.2 Effective Bandwidth for an Nrt- VBR Class of Service 122
9.9 The Importance of Scaling Down the Traffic Descriptors 124
9.10 Analysis of Results of Table 9- 10 127
9.11 The Effect of Cell Loss on Voice Quality 128 Conclusions 130
Chapter 10 Queuing and Shaping 133
10.1 Sources of Delay in Voice Networks 133
10.2 Relationship Between CLR and Queue Delay 134
10.3 Delay Due to Queuing and Shaping 135
10.4 Bursting the Buffer Queue 136
10.5 Buffer Modeling 13
10.5.1 Steady State 7
10.6 Opnet Simulation 138
10.6.1 The M/ M/1 Queuing Model 13
10.7 The Opnet Queuing Model 13
10.8 Opnet Results 139
10.9 Calculation of Expected Average Delay per Talkspurt 141
10.10 Opnet Results on Bandwidth 145
10.10.1 Key to Table 10- 2 6
10.11 Dividing Channels into More Than One Connection 149
Conclusions
Chapter 11 Main Conclusions 153
Appendix and Other
Appendix A FRF.11, FRF.8, and FRF.5 157
Appendix B Talkspurt Probability Tables 159
Appendix C M/ M/1 Queuing System 177
Appendix D Markov's Inequality Algorithml 179
Appendix E Market Research into Voice over Alternate Networks 181
Appendix F Digital Private Network Signaling Subsystem (DPNSS) 187
Appendix G QSIG 189
Appendix H Formulas for Tables 191
Glossary 209
References 217
Bibliography 225
Index 229
About the Author
Foreword
For
years, the transmission of voice over the PSTN has been im
plemented over TDM networks. Traffic engineering rules for these
networks have been understood for well over half a century, but the
shift to voice over packet networking introduces a different
operat ing paradigm. The challenge to network operators is to
deliver a Quality of Service (QoS) That at least matches the voice
quality that TDM networks have achieved, while reducing costs
through im proved bandwidth efficiency and statistical gains.
The
TDM codec G.711 is inefficient with respect to the use of network
resources. Complex voice codecs, such as G.729A/ B, or mobile codecs
such as AMR, ACELP, or FR- GSM can encode voice signals into data
units that produce throughput rates at far less than the G.711 rate of
64 Kbps. Furthermore, conversational voice involves two states,
that of talking and that of listening. When a person is listening,
there is no need to send digitized voice samples into a packet or
cell relay network towards the active talker. This concept is
known as si lence suppression, voice activity detection, or
discontinuous trans mission and lends well to achieving
statistical gains in a packet network.
The
purpose of this book is to provide an insight into some of the
challenges and decisions facing network engineers and planners
as they deploy voice services over their packet networks. The ob
jective is to explain the dynamics of conversational speech and
to empower the network planner with the requisite knowledge to ac
curately size the traffic descriptors of multiplexed voice
channels.
To
that end, the author, Juliet Bates, seeks to define a represen
tative model of voice traffic in order to make recommendations
for the effective use of the sophisticated traffic management
capabili ties of ATM. This book describes how to determine the
extent to which overbooking can be applied in order to achieve
statistical gain of silence suppressed and compressed voice
channels, while providing a Cell Loss Ratio (CLR) Guarantee.
Taking
this a step further, it is proposed in this book that the
determination of the Peak Cell Rate (PCR), Sustainable Cell Rate
(SCR), and Maximum Burst Size (MBS) ATM traffic descriptors should
be based on the mathematical modeling of conversational speech.
Networks should be, and can be, engineered such that overpro
visioning is not necessary. ATM mechanisms for QoS are suitable
for voice services, despite the fact that conversational
behavior is memoryless (statistically speaking). This book
describes how to maintain QoS for voice traffic. ATM already has the
requisite QoS capabilities.
Some
of the principals are applicable to IP networks, but to truly
engineer QoS, service providers must look to ATM. Readers of this
book will learn about the opportunities for statistical gain,
what is practical, and what compromises or risks can be taken to
achieve a high QoS at opti mal network utilization. Brian A. Day
Product Line Manager Voice Gateways Broadband Networking
Division Alcatel Canada
Preface
Economic
gains in voice transmission over a network are made possible by
two factors: low bit rate through compression, and ex ploitation of
speech silence durations. Overestimation of band width
requirements will cause underutilization of the network
resources and increased call blocking probability. Underestima
tion may cause cell or packet loss.
The
potential for statistical gain in multiplexed voice sources is
inhibited by the intermittent use of single voice channels for
fax and uncompressed voice, which may complicate the normal
methods of setting traffic descriptors. Overbooking can be
applied to compensate. The extent to which overbooking can be
applied is a subject of this book together with calculations to
show the risk of cell loss and the associated delay (and delay
variation) Incurred in the buffer. The Quality of Service (QoS)
Parameters, Cell Loss Ratio (CLR) And Cell Delay Variation (CDV),
which apply to Asynchronous Transfer Mode (ATM), Constant Bit Rate
(CBR), and Variable Bit Rate (VBR) Service classes are controlled by
the size of the buffer at the egress port and the service rate of the
queue in the buffer.
The
target CLR will reflect the probability that the buffer could be
exceeded. In provisioning effective bandwidth, the Call
Admission Control (CAC) Algorithm is aware of the QoS targets
required by the selected service class. This book describes a
method for the re duction of bandwidth, due to silence suppression,
and the deter mination of a scaling factor which can be applied
with a risk of cell loss selected in order to maintain the CLR and
CDV of the chosen ATM service category.
Juliet Bates Principal Consultant Professional Services Group Broadband Network Division Alcatel Telecom UK
Acknowledgments
I
wish to acknowledge Brunei University, Uxbridge, UK for allow ing
me to reproduce material from my Ph. D. thesis (July 2001).
Also, I particularly thank Dr. Michael Berwick in the Depart ment
of Electronic and Computer Engineering at Brunei, and Pro fessor
Malcolm Irving, who both provided excellent support and advice
during my time at Brunei University. I gratefully acknowledge
the technical advice I have received from Mustapha Aissaoui. I
also thank Bryan Edwards for his encouragement, and Tim Forshaw,
Dave Hills, and Mike Wilkinson for kindly reviewing each draft and
providing helpful comments and advice.
Within
the Appendixes of this book, there are extracts from com
prehensive reports into the market development prospects for
voice over alternative technologies, and I thank the Yankee Group
for permitting this material to be included. LT.4.303
Introduction Familiarity with the fundamentals of Asynchronous
Transfer Mode (ATM) Networking is assumed. While a very large
number of books have been published on this subject, Bibliography
refer ences [1] and [2] will provide a grounding in the areas
covered in this work.
Chapter
1 provides an introduction and describes voice access models for
mobile voice networks, such as the Universal Mobile
Telecommunications System (UMTS), combining the ATM and In
ternet Protocol (IP) In the core.
Chapter
2 describes the voice access methods and signaling pro tocols
employed in fixed voice networks—for example, where a Private
Branch Exchange (PBX) Accesses an ATM core network.
Chapter
3 discusses recently completed research in voice char
acterization and defends the rationale for choosing the distribu
tion and parameters best for a voice model. Also included are
alternative ways of building a model that will approximate the ar
rival process using key characteristics obtained from real
traffic.
Chapter
4 describes the framing format when voice is trans mitted over an
ATM network. Three different types of ATM Adap tation Layer (AAL)
Can be used to encapsulate coded voice. This section helps you
assess the framing overheads incurred by each type of AAL.
Chapter
5 explores various methods of servicing the contribut ing
channels used by each AAL type and calculates the corre sponding
delay implied by each solution. The difficulties of sizing
bandwidth requirements for compressed and silence- suppressed
voice channels in the presence of other applications are
explained.
Chapter
6 sets out the Cell Loss Ratio (CLR) And Cell Delay Variation (CDV)
Performance targets, which must be achieved to ensure adherence to
a particular ATM Quality of Service (QoS)
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