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.


Status in Chromium


Origin trial (tracking bug)

Consensus & Standardization

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

  • No signal
  • No signal
  • No signals


Last updated on 2021-12-13