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RTCRtpReceiver

Baseline Widely available *

This feature is well established and works across many devices and browser versions. It’s been available across browsers since ⁨September 2017⁩.

* Some parts of this feature may have varying levels of support.

The RTCRtpReceiver interface of the WebRTC API manages the reception and decoding of data for a MediaStreamTrack on an RTCPeerConnection.

RTCRtpReceiver.jitterBufferTarget: A DOMHighResTimeStamp that indicates an application's preferred hold time for media in the jitter buffer, allowing it influence the tradeoff between playout delay and the risk of running out of audio or video frames due to network jitter.
RTCRtpReceiver.track Read only: Returns the MediaStreamTrack associated with the current RTCRtpReceiver instance.
RTCRtpReceiver.transport Read only: Returns the RTCDtlsTransport instance over which the media for the receiver's track is received.
RTCRtpReceiver.transform: An RTCRtpScriptTransform is used to insert a transform stream (TransformStream) running in a worker thread into the receiver pipeline, allowing stream transforms to be applied to incoming encoded video and audio frames.

rtcpTransport Deprecated: This property has been removed; the RTP and RTCP transports have been combined into a single transport. Use the transport property instead.

RTCRtpReceiver.getCapabilities(): Returns the most optimistic view of the capabilities of the system for receiving media of the given kind.

RTCRtpReceiver.getContributingSources(): Returns an array that contains an object for each unique CSRC (contributing source) identifier received by the current RTCRtpReceiver in the last ten seconds.
RTCRtpReceiver.getParameters(): Returns an object that contains information about how the RTC data is to be decoded.
RTCRtpReceiver.getStats(): Returns a Promise whose fulfillment handler receives a RTCStatsReport which contains statistics about the incoming streams and their dependencies.
RTCRtpReceiver.getSynchronizationSources(): Returns an array that contains an object for each unique SSRC (synchronization source) identifier received by the current RTCRtpReceiver in the last ten seconds.

Specification
WebRTC: Real-Time Communication in Browsers> # rtcrtpreceiver-interface>

	Desktop					Mobile
	Chrome	Edge	Firefox	Opera	Safari	Chrome Android	Firefox for Android	Opera Android	Safari on IOS	Samsung Internet	WebView Android	WebView on iOS
`RTCRtpReceiver`	59	12	34	46	11	59	34	43	11	7.0	59	11
`createEncodedStreams`	86	86	No	72	No	86	No	61	No	14.0	86	No
`getCapabilities_static`	59	12	113	46	12.1	59	113	43	12.2	7.0	59	12.2
`getContributingSources`	59	12	59	46	12.1	59	59	43	12.2	7.0	59	12.2
`getParameters`	59	79	128	46	11	59	128	43	11	7.0	59	11
`getStats`	67	13	55	54	12.1	67	55	48	12.2	7.0	67	12.2
`getSynchronizationSources`	73	79	59	60	12.1	73	59	52	12.2	11.0	73	12.2
`jitterBufferTarget`	124	124	115	110	No	124	115	82	No	27.0	124	No
`rtcpTransport`	59	12	No	46	No	59	No	43	No	7.0	59	No
`track`	59	12	34	46	11	59	34	43	11	7.0	59	11
`transform`	141	141	117	No	15.4	141	117	No	15.4	No	141	15.4
`transport`	59	12	82	46	15.4	59	82	43	15.4	7.0	59	15.4