The getSynchronizationSources() method returns the latest playout timestamps of RTP packets for audio and video receivers. This is useful for determining in real time which streams are active, such as for the use case of audio meters or prioritizing displaying active participant streams in the UI.

Specification

Editor's draft

Status in Chromium

Blink>WebRTC>PeerConnection


Enabled by default (tracking bug) in:

  • Chrome for desktop release 73
  • Chrome for Android release 73
  • Android WebView release 73

Consensus & Standardization

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

  • Shipped/Shipping
  • Positive
  • Positive
  • No signals

Owner

Search tags

WebRTC RTCRtpReceiver RTCRtpContributingSource getContributingSources RTP media audio video,

Last updated on 2020-11-09