VoIP Voice and Fax Signal Processing

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VOICE COMPRESSION

Voice compression is essential for interactive voice communication systems of VoIP, mobile, public switched telephone network (PSTN), and satellite. Voice compression reduces the network bit rate or bandwidth on the communication channel. VoIP makes use of several compression codecs for minimizing the Internet bandwidth requirements. PSTN also makes use of voice compression except in small local and regional systems with analog switching. At a digital interface for PSTN, every 13- or 14-bit sample at 8-kHz sampling is converted to logarithmic 8-bit compression as per G.711 [ITU-T-G.711 (1988)] making it 64 kbps on a digital circuit switched network. G.711 is also used in VoIP. Voice with VoIP evolved to consider compression as one of the main parameters, and IP networks were considered band limited, at least in the early stage of VoIP considerations. Compression codecs such as G.729AB and G.723.1 are considered in VoIP to reduce network bandwidth. Wideband G.722 codecs are used to improve voice quality and to create better perception than PSTN. PSTN also makes use of higher compression of 16-, 24-, 32-, and 40-kbps adaptive differential pulse code modulation (ADPCM), which are usually referenced with an ITU codec name as G.726 [ITU-T-G.726 (1990)] to send more voice channels on the same digital network. In codec selection, the main parameters [Nikhil (2000)] considered are bit rate, quality, delays, and complexity (processing and memory) requirements. In this ...