[webrtc-stats] New metric to report minimum jitter buffer delay (#634)

ivocreusen has just created a new issue for https://github.com/w3c/webrtc-stats:

== New metric to report minimum jitter buffer delay ==
There are various reasons why the jitter buffer delay might be increased to a higher value, such as to achieve AV sync or because a  playoutDelay was set on a RTCRtpReceiver (see https://w3c.github.io/webrtc-extensions/#dom-rtcrtpreceiver-playoutdelay). When using one of these mechanisms, it can be useful to keep track of the minimal jitter buffer delay that could have been achieved, so WebRTC clients can track the amount of extra delay that is being added.

I would like to propose a new stat to track this minimum achievable jitter buffer delay. This stat should work similar to RTCInboundRtpStreamStats.jitterBufferTargetDelay, except that it should not be affected by playoutDelay (see link above) or AV sync or any other mechanisms. The metric should be purely based on the network characteristics such as jitter and packet loss. A proposal for a name is RTCInboundRtpStreamStats.estimatedMinimumJitterBufferTargetDelay

Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/634 using your GitHub account

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Received on Thursday, 2 June 2022 14:12:06 UTC