Page History: SIP audio settings
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Page Revision: Monday, 13 September 2010 11:15
The jitter is set centrally on the server using the server.ini file;
[audio]
jitterMs=50
Fine tuning of the audio server can be made locally on each audio server;
audioWaveInPrefetch=20
- Controls the number of milliseconds of data that should be received from the audio input before allowing that data to be used. This prevents under-runs of the audio input.
WaveInMinBufferSize=50
- Controls the buffer size (milliseconds) used to retrieve audio from the sound card. Larger buffers will cause more latency whilst they are filled before being handed to the audio server. Most consumer sound cards cannot cope with a buffer smaller than 50ms.
WaveOutPrequeue=20
- Controls the number of milliseconds of audio that should be queued to the audio output before allowing it to play. This prevents under-runs of the audio output.
WaveOutMaxBufferSize=100
- Controls the maximum number of milliseconds of audio that should be queued for playback. This prevents the output latency growing excessively over time if the clock of the soundcard varies slightly to that of the PC.