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.

Motivation

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)

Consensus & Standardization

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

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Owner

Last updated on 2021-08-26