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Interface WebRTCSessionParams


  • WebRTCSessionParams



Optional audioMaxJitterBufferDuration

audioMaxJitterBufferDuration: number

The maximum jitter buffer duration. Units are seconds. The default is 1 second.

Set the jitter buffer duration high to reduce the possibility of audio dropouts at the cost of potentially higher round-trip audio latency on poor connections.

Optional audioMinJitterBufferDuration

audioMinJitterBufferDuration: number

The minimum jitter buffer duration. Units are seconds. The default is 0 seconds.

In practice, this should always be set to 0 seconds, which is the default. Setting the minimum jitter buffer duration to X seconds means that all audio sent to the server will always be buffered at least by X seconds. This is rarely desirable; lower latency is almost always preferred. You may, however, want to set the maximum jitter buffer duration if your users are experiencing frequent audio drop-outs; refer to audioMaxJitterBufferDuration below for more details.