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Summary
Summary
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.
Author Notes
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.
Table of Contents
| 1 PSTN Basic Infrastructure, Interfaces and Signals |
| 1.1 PSTN central office, and DLC |
| 2 VoIP Overview and Infrastructure |
| 2.1 PSTN and VoIP |
| 2.2 Typical VoIP deployment example |
| 2.3 Network and acoustic interfaces for VoIP |
| 2.4 VoIP systems working principles |
| 2.5 VoIP Signaling |
| 3 Voice Compression |
| 3.1 Compression codecs |
| 3.2 G.711 Compression |
| 3.3 Speech redundancies and compression |
| 3.4 G.726 or ADPCM compression |
| 3.5 Wideband voice |
| 3.6 G.729 Family of low bit rate codecs |
| 3.7 Miscellaneous narrow and wideband codecs |
| 3.8 Codecs and overload levels |
| 3.9 Voice quality of codecs |
| 3.10 C-source code for codecs |
| 3.11 Codecs in VoIP deployment |
| 4 Generic VAD/CNG for Waveform Codes |
| 4.1 VAD/CNG and Codecs |
| 4.2 Generic VAD/CNG functionality |
| 4.3 Comfort noise payload format |
| 4.4 G.711 Appendix-II VAD/CNG algorithm |
| 4.5 Power based VAD/CNG |
| 4.6 VAD/CNG in low bit rate codecs |
| 4.7 Miscellaneous aspects of VAD/CNG |
| 4.8 Summary on VAD/CNG |
| 5 Packet Loss Concealment Techniques |
| 5.1 Packet loss concealment overview |
| 5.2 Packet Loss concealment techniques |
| 5.3 Transmitter and receiver based techniques |
| 5.4 Decoder only based PLC techniques |
| 5.5 PLC techniques description |
| 5.6 PLC for Low bit rate codecs |
| 5.7 PLC testing |
| 5.8 PLC summary and discussion |
| 6 Echo Cancellation |
| 6.1 Talker and listener echo in PSTN voice call |
| 6.2 Naming conventions in echo canceller |
| 6.3 Line and Acoustic echo canceller |
| 6.4 Talker echo levels and delay |
| 6.5 Echo cancellation in VoIP adapters |
| 6.6 Echo path |
| 6.7 Adaptation filtering algorithms |
| 6.8 Echo canceller control functions |
| 6.9 Echo cancellation in multiple VoIP terminals |
| 6.10 Echo canceller testing |
| 7 DTMF Detection, Generation, and Rejection |
| 7.1 Specifications of DTMF tones |
| 7.2 DTMF tones generation |
| 7.3 DTMF detection |
| 7.4 Goetzel filtering with linear filtering |
| 7.5 Tone detection using Teager and Kaiser Energy operator |
| 7.6 DFT or FFT processing |
| 7.7 DTMF rejection |
| 7.8 DTMF RFC2833 processing |
| 7.9 DTMF testing |
| 7.10 Summary and discussions |
| 8 Caller ID Features in VOIP |
| 8.1 FSK caller ID on PSTN |
| 8.2 FSK caller ID data transport protocol |
| 8.3 DTMF based caller ID |
| 8.4 Country specific caller ID overview |
| 8.5 Caller ID in VoIP |
| 8.6 Call wait caller ID |
| 8.7 Caller ID on FXO interfaces |
| 8.8 Summary and discussions |
| 9 Wideband Voice Modules Operation |
| 9.1 Wideband voice examples |
| 9.2 Wideband VoIP adapter |
| 9.3 Wideband voice summary |
| 10 Packetization - RTP, RTCP, and Jitter Buffer |
| 10.1 Real time protocol (RTP) |
| 10.2 RTP control protocol (RTCP) |
| 10.3 VoIP Packet impediments |
| 10.4 Jitter Buffer |
| 10.5 Adaptive Jitter Buffer |
| 10.6 Adapting to delay variations |
| 10.7 AJB algorithms overview |
| 10.8 Adaptive Jitter Buffer implementation guidelines |
| 10.9 Fixed Jitter Buffer implementation guidelines |
| 11 VOIP Vocie - Network BIT Rate calculations |
| 11.1 Voice compression and bit rate overview |
| 11.2 Voice payload and headers |
| 11.3 Ethernet, DSL and Cable interfaces for VoIP |
| 11.4 VoIP voice packets on DSL interface |
| 11.5 VoIP voice packets on Cable interface |
| 11.6 Bit rate calculation for different codec |
| 11.7 Bit rate with VAD/CNG |
| 11.8 Bit rate with RTCP, RTCP-XR, and Signaling |
| 11.9 Summary on VoIP bit rate |
| 12 Clock Sources for VoIP Applications |
| 12.1 PSTN systems and clocks |
| 12.2 VoIP system clock options |
| 12.3 Clock timing deviations relating to V |
