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VoIP Voice and Fax Signal Processing
book

VoIP Voice and Fax Signal Processing

by Sivannarayana Nagireddi
September 2008
Intermediate to advanced
550 pages
18h 10m
English
Wiley-Interscience
Content preview from VoIP Voice and Fax Signal Processing

3.11 Codecs IN VoIP DEPLOYMENT

VoIP deployments use multiple codecs. It is difficult to arrive at one generic conclusion on the selection of codecs. Codec selection will consider several aspects of quality, compression, packet loss performance, frame size and end-to-end delays, processing, memory requirements, any backward compatibility, roadmap and life of codec, ability to cater to multiple deployments, and finally acceptance in the market. An overview on this selection is provided in this section.

As mentioned, retail market products keep several codecs in the VoIP box to cater to wider requirements. VoIP-to-PSTN high-channel gateways like CISCO gateways Donohue et al. (2006) keep several codecs to take care of wider market requirements. VoIP customer premises equipments (CPEs) mainly work with a particular deployment. The requirements vary with the region of interest and the service provider, but the requirements will narrow down to two to three codecs from actually working deployment. The service provider controls these parameters.

As a common practice, G.711 is used in all VoIP deployments. It is the best narrowband codec. It is helpful to use G.711 to get better branding on voice quality. Japan VoIP deployments use mainly G.711 because of higher Internet bandwidth support in most deployments and as it provides better quality. G.729A is supported on most VoIP CPEs, but G.729 is not used in actual VoIP calls. Europe is using G.711 and G.729AB, and some deployments are using ...

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