RTCRtpContributingSource

The 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.

Properties

audioLevel Optional

A double-precision floating-point value between 0 and 1 specifying the audio level contained in the last RTP packet played from this source.

rtpTimestamp Optional

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.

source Optional

A 32-bit unsigned integer value specifying the CSRC identifier of the contributing source.

timestamp Optional

A DOMHighResTimeStamp indicating the most recent time at which a frame originating from this source was delivered to the receiver's MediaStreamTrack

Specifications

Specification
WebRTC 1.0: Real-Time Communication Between Browsers (WebRTC 1.0)
# dom-rtcrtpcontributingsource

Browser compatibility

Desktop Mobile
Chrome Edge Firefox Internet Explorer Opera Safari WebView Android Chrome Android Firefox for Android Opera Android Safari on IOS Samsung Internet
RTCRtpContributingSource
59
≤79
59
No
No
12.1
59
59
59
No
12.2
7.0
audioLevel
No
No
59
No
No
12.1
No
No
59
No
12.2
No
source
59
≤79
59
No
No
12.1
59
59
59
No
12.2
7.0
timestamp
59
≤79
59
Starting in version 60, the timestamp is correctly computed based on the window's Performance time, rather than Date.getTime().
No
No
12.1
59
59
59
Starting in version 60, the timestamp is correctly computed based on the window's Performance time, rather than Date.getTime().
No
12.2
7.0

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Licensed under the Creative Commons Attribution-ShareAlike License v2.5 or later.
https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpContributingSource