This is an experimental technology
Because this technology's specification has not stabilized, check the compatibility table for usage in various browsers. Also note that the syntax and behavior of an experimental technology is subject to change in future versions of browsers as the specification changes.
RTCRtpContributingSource interface of the the WebRTC API provides contains information about a given contributing source (CSRC) including the most recent time a packet that the source contributed was played out.
- Returns a
DOMHighResTimeStampindicating the most recent time of playout of an RTP packet from the source.
- Returns the CSRC identifier of the contributing source.
- Indicates the audio level contained in the last RTP packet played from this source. audioLevel will be the level value defined in [RFC6465] if the RFC 6465 header extension is present, and otherwise null.
|WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'RTCRtpContributingSource' in that specification.
|Working Draft||Initial definition.|
|Feature||Chrome||Firefox (Gecko)||Internet Explorer||Opera||Safari (WebKit)|
|Feature||Android Webview||Chrome for Android||Firefox Mobile (Gecko)||Firefox OS||IE Mobile||Opera Mobile||Safari Mobile|
||No support||No support||?||?||?||?||?|