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Audio Conferencing

real codec live compression

Definition: Audio conferencing allows participants in a live session to hear each other.

The audio is transmitted over the network between users, live and in real-time. Audio conferencing is one component of teleconferencing; the others are video conferencing, and data conferencing. Since the audio must be encoded, transmitted, and decoded in real-time, special compression and transmission techniques are typically used. In a teleconferencing system that is ITU-T H.323 compliant, the G.711 audio codec, which is basically uncompressed 8-bit PCM signal at 8KHz in either A-Law or µ-Law format, must be supported. This leads to bitrates of 56 or 64kbps, which are relatively high for audio but supported by today’s networks.

Support for other ITU-T audio recommendations and compression is optional, and its implementation specifics depend on the required speech quality, bit rate, computational power, and delay. Provisions for asymmetric operation of audio codecs have also been made; i.e., it is possible to send audio using one codec but receive audio using another codec. If the G.723.1 audio compression standard is provided, the terminal must be able to encode and decode at both the 5.3 kbps and the 6.3 kbps modes. If a terminal is audio only, it should also support the ITU-T G.729 recommendation. Note that if a terminals is known to be on a low-bandwidth network (<64kbps), it does not need to disclose capability to receive G.711 audio since it won’t practically be able to do so.

To transfer the live audio over the network, a protocol such as the RTP (Real-time Transport Protocol) or simple UDP (User Datagram Protocol) is used.

Audio Streaming - Introduction, Audio Compression, Dissemination over the Network, Real-time Transport Protocol (RTP) [next] [back] Audio Compression and Coding Techniques - Introduction – Audio Properties, Audio Digitization Codec, The Human Hearing/Vocal Systems and Audio Coding

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