RTCRtpContributingSource dictionary of the WebRTC API is used by
getContributingSources() to provide information about a given contributing source (CSRC), including the most recent time a packet that the source contributed was played out.
The information provided is based on the last ten seconds of media received.
A double-precision floating-point value between 0 and 1 specifying the audio level contained in the last RTP packet played from this source.
The RTP timestamp of the media played out at the time indicated by
timestamp. This value is a source-generated time value which can be used to help with sequencing and synchronization.
A 32-bit unsigned integer value specifying the CSRC identifier of the contributing source.
|WebRTC 1.0: Real-Time Communication Between Browsers (WebRTC 1.0)|
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