RTCRtpSynchronizationSource dictionary of the the WebRTC API is used by
getSynchronizationSources() to describe a particular synchronization source (SSRC). A synchronization source is a single source that shares timing and sequence number space. Since
RTCRtpContributingSource, its properties are also available.
The information provided is based on the last ten seconds of media received.
While the published specification describes this as an interface, it has since been changed to a dictionary in follow-up drafts.
Also implements the properties of
- A Boolean value indicating whether or not voice activity is included in the last RTP packet played from the source. If the peer has indicated that it's not supporting voice activity detection, this field is not provided.
|WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'RTCRtpSynchronizationSource' in that specification.
|Candidate Recommendation||Initial definition.|
|Feature||Android webview||Chrome for Android||Edge mobile||Firefox for Android||Opera Android||iOS Safari||Samsung Internet|
1. From version 59: this feature is behind the
media.peerconnection.rtpsourcesapi.enable preference (needs to be set to
true). To change preferences in Firefox, visit about:config.