The RTCRtpContributingSource dictionary of the 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.

While the published specification describes this as an interface, it has since been changed to a dictionary in follow-up drafts.


audioLevel Read only 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.
source Read only
A 32-bit unsigned integer value specifying the CSRC identifier of the contributing source.
timestamp Read only
A DOMHighResTimeStamp indicating the most recent time of playout of an RTP packet from the source.


Specification Status Comment
WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'RTCRtpContributingSource' in that specification.
Candidate Recommendation Initial definition.

Browser compatibility

FeatureChromeEdgeFirefoxInternet ExplorerOperaSafari
Basic support59 ?591 No No ?
timestamp59 ?591 2 No No ?
source59 ?591 No No ?
audioLevel No ?591 No No ?
FeatureAndroid webviewChrome for AndroidEdge mobileFirefox for AndroidOpera AndroidiOS SafariSamsung Internet
Basic support5959 ?591 No ?7.0
timestamp5959 ?591 2 No ?7.0
source5959 ?591 No ?7.0
audioLevel No No ?591 No ? No

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.

2. Starting in version 60, the timestamp is correctly computed based on the window's Performance time, rather than Date.getTime().

Document Tags and Contributors

Contributors to this page: fscholz, Sheppy, Onkar316, jpmedley
Last updated by: fscholz,