Audio Encoding

Audio Encoding is a method by which calls are signalled, setup and then broken down again using Audio Codecs to relay Audio streams over an IP network. The International Telecommunication Union has defined multiple audio codecs for use with H.323, which incidentally are all also compatible with SIP as this technology is codec agnostic. The various codecs in use today include:

    G.711 is 3 kHz audio encoded at 64-kbps. G.711 is PCM audio, the format used for voice delivery over traditional telephone networks and exchanges.
    G.722 is high-quality 7kHz audio in 48-, 56-, or 64-kbps streams. Two lower-quality, narrow-band revisions exist: G.722.1 encodes the audio at 24- or 32-kbps, and G.722.2 encodes at around 16kbps.
    G.723.1 is used for compressing speech at very low bit rates: 5.3- and 6.3-kbps.
    G.728 is 3.4kHz audio encoded at 16-kbps, but uses much smaller packet sizes (.625 millisecond compared to 37.5ms with G.723.1) to guarantee lower delays.
    G.729 is a newer voice codec that utilises 8-kbps streams and 15ms packet sizes. Two variations, G.729 and G.729A, are available and which differ only in terms of their mathematical implementation.

Open Source Codecs:

    Speex is an open source speech codec. In contrast to the G-series codecs listed above, it is not protected by patents. It encodes at variable bitrates, from 2.15- to 44.2-kbps.
    GSM6.10 is another open source codec, encoding at 13.3-kbps. At this time there is an unresolved patent dispute surrounding the codec, but is still supported by multiple software programs.

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