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 being incubated in a Community Group
Status in Chromium
Enabled by default
Consensus & Standardization
- No signals
Search tagsWebRTC RTCRtpReceiver RTCRtpContributingSource getContributingSources RTP media audio video,
Last updated on 2020-11-09