RTCRtpReceiver.getSynchronizationSources()

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 components: 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.

Firefox: Shipped
Edge: Public support
Safari: Public support
Web Developers: No signals

Owner

hbos@chromium.org

Last updated on 2019-01-30