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
Status in Chromium
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