VoIP Voice and Fax Signal Processing

A complete and systematic treatment of signal processing for VoIP voice and fax This book presents a consolidated view and basic approach to signal processing for VoIP voice and fax solutions. It provides readers with complete coverage of the topic, from how things work in voice and fax modules, to signal processing aspects, implementation, and testing. Beginning with an overview of VoIP infrastructure, interfaces, and signals, the book systematically covers: Voice compression Packet loss concealment techniques DTMF detection, generation, and rejection Wideband voice modules operation VoIP Voice-Network bit rate calculations VoIP voice testing Fax over IP and modem over IP Country deviations of PSTN mapped to VoIP VoIP on different processors and architectures Generic VAD-CNG for waveform codecs Echo cancellation Caller ID features in VoIP Packetization—RTP, RTCP, and jitter buffer Clock sources for VoIP applications Fax operation on PSTN, modulations, and fax messages Fax over IP payload formats and bit rate calculations Voice packets jitter with large data packets VoIP voice quality Over 100 questions and answers on voice and more than seventy questions and answers on fax are provided at the back of the book to reinforce the topics covered throughout the text. Additionally, several clarification, interpretation, and discussion sections are included in selected chapters to aide in readers' comprehension. VoIP Voice and Fax Signal Processing is an indispensable resource for professional electrical engineers, voice and fax solution developers, product and deployment support teams, quality assurance and test engineers, and computer engineers. It also serves as a valuable textbook for graduate-level students in electrical engineering and computer engineering courses.

Table of Contents:
Acknowledgments xix About the Author xxi Preface xxiii Glossary xxvii 1 PSTN Basic Infrastructure, Interfaces, and Signals 1 1.1 PSTN CO and DLC 2 1.2 PSTN User Interfaces 3 1.3 Data Services on Telephone Lines 7 1.4 Power Levels and Digital Quantization for G.711 μA-Law 9 1.5 Significance of Power Levels on Listening 11 1.6 TR-57, IEEE-743, and TIA Standards Overview 13 2 VoIP Overview and Infrastructure 19 2.1 PSTN and VoIP 20 2.2 Typical VoIP Deployment Example 25 2.3 Network and Acoustic Interfaces for VoIP 26 2.4 VoIP Systems Working Principles 27 2.5 VoIP Signaling 41 3 Voice Compression 49 3.1 Compression Codecs 50 3.2 G.711 Compression 50 3.3 Speech Redundancies and Compression 60 3.4 G.726 or ADPCM Compression 60 3.5 Wideband Voice 62 3.6 G.729 Family of Low-Bit-Rate Codecs 63 3.7 Miscellaneous Narrow and Wideband Codecs 67 3.8 Codecs and Overload Levels 70 3.9 Voice Quality of Codecs 70 3.10 C-Source Code for Codecs 74 3.11 Codecs in VoIP Deployment 74 4 Generic VADCNG for Waveform codecs 76 4.1 VADCNG and Codecs 77 4.2 Generic VADCNG Functionality 78 4.3 Comfort Noise Payload Format 78 4.4 G.711 Appendix II VADCNG Algorithm 80 4.5 Power-Based VADCNG 83 4.6 VADCNG in Low-Bit-Rate Codecs 85 4.7 Miscellaneous Aspects of VADCNG 86 4.8 Summary on VADCNG 89 5 Packet Loss Concealment Techniques 91 5.1 Packet Loss Concealment Overview 91 5.2 Packet Loss Concealment Techniques 92 5.3 Transmitter- and Receiver-Based Techniques 94 5.4 Decoder-Only Based PLC Techniques 99 5.5 PLC Techniques Description 101 5.6 PLC for Low-Bit-Rate Codecs 108 5.7 PLC Testing 110 5.8 PLC Summary and Discussion 111 6 ECHO Cancellation 113 6.1 Talker and Listener Echo in PSTN Voice Call 114 6.2 Naming Conventions in Echo Canceller 119 6.3 Line and Acoustic Echo Canceller 120 6.4 Talker Echo Levels and Delay 123 6.5 Echo Cancellation in VoIP Adapters 127 6.6 Echo Path 131 6.7 Adaptation Filtering Algorithms 132 6.8 Echo Canceller Control Functions 137 6.9 Echo Cancellation in Multiple VoIP Terminals 144 6.10 Echo Canceller Testing 145 7 DTMF Detection, Generation, and Rejection 151 7.1 Specifications of DTMF Tones 152 7.2 DTMF Tones Generation 152 7.3 DTMF Detection 156 7.4 Goertzel Filtering with Linear Filtering 158 7.5 Tone Detection Using Teager and Kaiser Energy Operator 167 7.6 DFT or FFT Processing 171 7.7 DTMF Rejection 171 7.8 DTMF RFC2833 Processing 174 7.9 DTMF Testing 177 7.10 Summary and Discussions 178 8 Caller ID Features in VoIP 179 8.1 FSK Caller ID on PSTN 180 8.2 FSK Caller ID Data Transport Protocol 183 8.3 DTMF-Based Caller ID 188 8.4 Country-Specific Caller ID Overview 190 8.5 Caller ID in VoIP 191 8.6 Call Wait Caller ID 193 8.7 Caller ID on FXO Interfaces 198 8.8 Summary and Discussions 202 9 Wideband Voice Modules Operation 203 9.1 Wideband Voice Examples 204 9.2 Wideband VoIP Adapter 207 9.3 Wideband Voice Summary 214 10 Packetization—RTP, RTCP, and Jitter Buffer 215 10.1 Real-Time Protocol (RTP) 215 10.2 RTP Control Protocol (RTCP) 218 10.3 VoIP Packet Impediments 219 10.4 Jitter Buffer 222 10.5 Adaptive Jitter Buffer 224 10.6 Adapting to Delay Variations 228 10.7 AJB Algorithms Overview 230 10.8 Adaptive Jitter Buffer Implementation Guidelines 239 10.9 Fixed Jitter Buffer Implementation Guidelines 241 11 VoIP Voice—Network Bit Rate Calculations 242 11.1 Voice Compression and Bit Rate Overview 243 11.2 Voice Payload and Headers 244 11.3 Ethernet, DSL, and Cable Interfaces for VoIP 245 11.4 VoIP Voice Packets on a DSL Interface 249 11.5 VoIP Voice Packets on a Cable Interface 249 11.6 Bit Rate Calculation for Different codecs 253 11.7 Bit Rate with VADCNG 253 11.8 Bit Rate with RTCP, RTCP-XR, and Signaling 254 11.9 Summary on VoIP Bit Rate 254 12 Clock Sources for VoIP Applications 257 12.1 PSTN Systems and Clocks 259 12.2 VoIP System Clock Options 259 12.3 Clock Timing Deviations Relating to VoIP Packets 263 12.4 Measuring Clock PPM 266 12.5 Clock Drift Influence on Voice and Fax Calls 268 13 VoIP Voice Testing 269 13.1 Basic Test Setup 269 13.2 First-Level VoIP Manual Tests 272 13.3 Analog Front-End Voice Transmission Tests 274 13.4 Telephone Line Monitor for Tones and Timing Characteristics 274 13.5 MOS—PSQM, PAMS, and PESQ Measurements 275 13.6 Bulk Calls for Stress Testing 276 13.7 Network Impediments Creation 277 13.8 VoIP Packets Analysis 278 13.9 Compliance Tests 278 13.10 VoIP Interoperability 278 13.11 Deployment Tests 279 13.12 Voice Quality Certifications 280 13.13 VoIP Speech Quality Tests by the ETSI 280 13.14 User Operational Considerations 281 14 Fax Operation on PSTN, Modulations, and Fax Messages 282 14.1 Fax Machine Overview 284 14.2 Fax Image Coding Schemes 286 14.3 Fax Modulation Rates 290 14.4 PSTN Fax Call Phases 291 14.5 Fax and Modem Tones Basics 300 14.6 Tones Detection 303 14.7 Fax Modulations and Demodulations 309 14.8 V.21 Fax Modem 311 14.9 V.27ter Fax Modem 313 14.10 V.29 Modem 318 14.11 V.17 Modem 321 14.12 V.34 Fax Modem 325 14.13 V.21 HDLC Framing and Deframing 326 14.14 HDLC Messages in ECM 331 14.15 Summary and Discussions on Fax 332 15 Fax Over IP and Modem Over IP 333 15.1 Fax over IP Overview 333 15.2 Fax over IP Benefits 336 15.3 Fax Basic Functionality and Detecting Fax Call 337 15.4 T.38 Fax Relay 339 15.5 Fax Pass-Through 346 15.6 Fax over IP Interoperability Challenges 348 15.7 Modem Basic Functions on PSTN 356 15.8 Migrating Modem Functions to IP 358 15.9 Guidelines for Fax and Modem Pass-Through in VoIP 362 15.10 VoIP Fax Tests 365 16 Fax Over IP Payload Formats and Bit Rate Calculations 371 16.1 Overview on T.38 and G.711 Pass-Through Bit Rate 372 16.2 G.711 Fax Pass-Through Bit Rate 374 16.3 T.38 Basic Payload Bytes for V.27ter, V.29, V.17, and V.34 374 16.4 Overview on Redundant and Duplicate Fax Packets 376 16.5 T.38 IFP Packets 378 16.6 IFP over TCP (TCPIPIFP) 381 16.7 IFP over UDP 382 16.8 T.38 UDPTL-Based Bit Rate Calculation with Redundancy 387 16.9 Fax UDPTL-Based Bit Rate on Ethernet and DSL Interfaces 388 16.10 T.38 Bit Rate Recommendations 392 17 Country Deviations of the PSTN Mapped to VoIP 393 17.1 Country-Specific Deviations 394 17.2 Country-Specific Deviations on VoIP Interfaces 396 17.3 Call Progress Tones for Multiple Countries 399 17.4 Call Progress Tone Detectors 404 18 Voice Packets Jitter with Large Data Packets 406 18.1 ATM Cells and Transmission 408 18.2 IPQoS and Queuing Jitter on an Interface 410 19 VoIP on Different Processors and Architectures 414 19.1 VoIP on Personal Computers 415 19.2 VoIP on PC Add-On Cards 416 19.3 VoIP on Dedicated Processors 417 19.4 Operating System Aspects on Different Platforms 419 19.5 Voice Processing Complexity 422 20 VoIP Voice Quality 425 20.1 Voice Quality Measurements 426 20.2 E-model-Based Voice Quality Estimation 435 20.3 VoIP Voice Quality Considerations 446 20.4 VoIP Voice Quality Summary 459 20.5 Voice Quality Monitoring and RTCP-XR 459 20.6 Summary and Discussions 463 21 VoIP Voice FAQs 464 22 Basic Fax and Fax Over IP FAQs 484 Index 517

About the Author :
Sivannarayana Nagireddi, PhD, is the architect of VoIP solutions at Ikanos Communications, Inc., and leads the DSP and VoIP team. Dr. Nagireddi and his team developed complete VoIP solutions including signal processing algorithms for voice-enabled residential gateway processors, which have been deployed by telecommunications providers. He is a member of the IEEE, a Fellow of IETE-India, and reviewer for the journal Medical Engineering & Physics.