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.
Docs: https://github.com/henbos/webrtc-provisional-stats/issues/13 https://github.com/henbos/webrtc-provisional-stats/pull/14
Explainers: https://github.com/henbos/webrtc-provisional-stats/pull/14