WebRTC relative packet arrival delay statistic

Web RTC

Add new non-standard audio receiver metric to the WebRTC getStats() API called relativePacketArrivalDelay. The metric estimates the delay of incoming packets relative to the first packet received.

The purpose of this metric is to identify networks which may cause bad audio due to the jitter buffer not adapting correctly.

Documentation

Status in Chromium

Blink>WebRTC


Origin trial (tracking bug) in:

  • Chrome for desktop release 75
  • Chrome for Android release 75
  • Android WebView release 75

Consensus & Standardization

After a feature ships in Chrome, the values listed here are not guaranteed to be up to date.

  • No public signals
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Owner

Last updated on 2019-04-01