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Contents Preface xi 1 Introduction 1 1.1 The rebirth of VoIP 1 1.2 Why beyond VoIP protocols? 2 1.2.1 Selecting a voice coder 2 1.2.2 Providing 'toll quality' . . . and more 2 1.2.3 Controlling IP quality of service 3 1.2.4 Dimensioning the network 4 1.2.5 Unleashing the potential of multicast 5 1.3 Scope of this book 5 1.4 Intended audience 6 1.5 Conclusion 7 1.6 References 7 2 Introduction to Speech-coding Techniques 9 2.1 A primer on digital signal processing 9 2.1.1 Introduction 9 2.1.2 Sampling and quantization 10 2.1.3 The sampling theorem 12 2.1.4 Quantization 14 2.1.5 ITU G.711 A-law or '-law, a basic coder at 64 kbit/s 16 2.2 The basic tools of digital signal processing 20 2.2.1 Why digital technology simplifies signal processing 20 2.2.2 The Z transform and the transfer function 22 2.2.3 Linear prediction for speech-coding schemes 30 2.3 Overview of speech signals 32 2.3.1 Narrow-band and wide-band encoding of audio signals 32 2.3.2 Speech production: voiced, unvoiced, and plosive sounds 32 2.3.3 A basic LPC vocoder: DOD LPC 10 36 2.3.4 Auditory perception used for speech and audio bitrate reduction 37 2.4 Advanced voice coder algorithms 39 2.4.1 Adaptive quantizers. NICAM and ADPCM coders 39 2.4.2 Differential predictive quantization 42 2.4.3 Long-term prediction for speech signal 45 2.4.4 Vector quantization 46 2.4.5 Entropy coding 47 2.5 Waveform coders. ADPCM ITU-T G.726 47 2.5.1 Coder specification . . . from digital test sequences to C code 50 2.5.2 Embedded version of the G.726 ADPCM coder G.727 51 2.5.3 Wide-band speech coding using a waveform-type coder 52 2.6 Hybrids and analysis by synthesis (ABS) speech coders 56 2.6.1 Principle 56 2.6.2 The GSM full-rate RPE-LTP speech coder (GSM 06.10) 58 2.7 Codebook-excited linear predictive (CELP) coders 61 2.7.1 ITU-T 8-kbit/s CS-ACELP G.729 64 2.7.2 ITU-T G.723.1: dual-rate speech coder for multimedia communications transmitting at 5.3 kbit/s and 6.3 kbit/s 66 2.7.3 The low-delay CELP coding scheme: ITU-T G.728 69 2.7.4 The AMR and AMR-WB coders 71 2.8 Quality of speech coders 74 2.8.1 Speech quality assessment 75 2.8.2 ACR subjective test, mean opinion score (MOS) 77 2.8.3 Other methods of assessing speech quality 80 2.8.4 Usage of MOS 81 2.9 Conclusion on speech-coding techniques and their near future 81 2.9.1 The race for low-bitrate coders 81 2.9.2 Optimization of source encoding and channel encoding 82 2.9.3 The future 83 2.10 References 84 2.10.1 Articles 84 2.10.2 Books 85 2.11 Annexes 86 2.11.1 Main characteristics of ITU-T standardized speech coders 86 2.11.2 Main characteristics of cellular mobile standardized speech coders 88 3 Voice Quality 89 3.1 Introduction 89 3.2 Reference VoIP media path 90 3.3 Echo in a telephone network 91 3.3.1 Talker echo, listener echo 91 3.3.2 Electric echo 92 3.3.3 Acoustic echo 94 3.3.4 How to limit echo 96 3.4 Delay 101 3.4.1 Influence of the operating system 101 3.4.2 The influence of the jitter buffer policy on delay 102 3.4.3 The influence of the codec, frame grouping, and redundancy 103 3.4.4 Measuring end-to-end delay 106 3.5 Acceptability of a phone call with echo and delay 107 3.5.1 The G.131 curve 107 3.5.2 Evaluation of echo attenuation (TELR) 108 3.5.3 Interactivity 111 3.5.4 Other requirements 112 3.5.5 Example of a speech quality prediction tool: the E-model 113 3.6 Conclusion 114 3.7 Standards 115 4 Quality of Service 117 4.1 Introduction: What is QoS? 117 4.2 Describing a data stream 118 4.3 Queuing techniques for QoS 120 4.3.1 Class-based queuing 120 4.3.2 Simple fair queuing: bitwise round robin fair queuing algorithm 121 4.3.3 GPS policy in a node 122 4.4 Signaling QoS requirements 130 4.4.1 The IP TOS octet 130 4.4.2 RSVP 139 4.4.3 Scaling issues with RSVP 145 4.4.4 Scaling RSVP with a layered architecture 148 4.5 The CableLabs??PacketCable??quality-of-service specification: DQoS 154 4.5.1 What is DQoS? 154 4.5.2 Session-per-session QoS reservation 155 4.5.3 Two-phase reservation mechanism 157 4.5.4 CMS to CMTS communications 160 4.6 Improving QoS in the best effort class 170 4.6.1 Issues with UDP traffic 171 4.6.2 Issues with TCP traffic 171 4.6.3 Using 'intelligent' packet discard 173 4.7 Issues with slow links 174 4.7.1 Queuing 174 4.7.2 Overhead 174 4.7.3 Overhead compression 175 4.7.4 Packet fragmentation, prioritization over serial links 177 4.8 Conclusion 179 4.9 References 181 4.9.1 Intserv over Diffserv 182 4.9.2 Aggregation 182 4.9.3 Security 182 4.9.4 RSVP simulator 182 4.10 Packet size annex 183 5 Network Dimensioning 219 5.1 Simple compressed voice flow model 219 5.1.1 Model of popular voice coders 219 5.1.2 Model for N simultaneous conversations using the same coder 222 5.1.3 Loss rate and dimensioning 225 5.1.4 Packet or frame loss? 233 5.1.5 Multiple coders 234 5.2 Building a network dedicated to IP telephony 235 5.2.1 Is it necessary? 235 5.2.2 Network dimensioning 235 5.3 Merging data communications and voice communications on one common IP backbone 239 5.3.1 Prioritization of voice flows 239 5.3.2 Impact on end-to-end delay 241 5.4 Multipoint communications 242 5.4.1 Audio multipoint conferences 242 5.4.2 Multipoint videoconferencing 247 5.5 Modeling call seizures 248 5.5.1 Model of call arrivals: the Poisson process 248 5.5.2 Model of a call server 249 5.5.3 Dimensioning call servers in small networks 252 5.5.4 Dimensioning call servers in large networks 256 5.6 Conclusion 260 5.7 References 261 6 IP Multicast Routing 263 6.1 Introduction 263 6.2 When to use multicast routing 263 6.2.1 A real-time technology 263 6.2.2 Network efficiency 265 6.2.3 Resource discovery 266 6.3 The multicast framework 266 6.3.1 Multicast address, multicast group 266 6.3.2 Multicast on ethernet 268 6.3.3 Group membership protocol 269 6.4 Controling scope in multicast applications 272 6.4.1 Scope versus initial TTL 272 6.4.2 TTL threshold 273 6.5 Building the multicast delivery tree 274 6.5.1 Flooding and spanning tree 274 6.5.2 Shared trees 274 6.5.3 Source-based trees 275 6.6 Multicast-routing protocols 277 6.6.1 Dense- and sparse-mode protocols 277 6.6.2 Other protocols 280 6.7 The mBone 285 6.7.1 An experimental network that triggered the deployment of commercial multicast networks 285 6.7.2 Routing protocols and topology 285 6.7.3 mBone applications 285 6.8 MULTICAST issues on non-broadcast media 289 6.8.1 Bridged LANs 289 6.8.2 IGMP snooping 289 6.8.3 Cisco group management protocol (CGMP) 290 6.8.4 IEEE GMRP 290 6.9 Conclusion 290 6.10 References 291 6.10.1 URLs 291 6.10.2 RFCs and internet drafts 291 6.10.3 Mailing lists 292 Index
Library of Congress Subject Headings for this publication:
Internet telephony.
Speech processing systems.