RTCRtpReceiver

Baseline Widely available *

This feature is well established and works across many devices and browser versions. It’s been available across browsers since January 2020.

* 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.

Instance properties

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.

Obsolete properties

rtcpTransport Deprecated

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

Static methods

RTCRtpReceiver.getCapabilities()

Returns the most optimistic view of the capabilities of the system for receiving media of the given kind.

Instance methods

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.

Specifications

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

Browser compatibility

Report problems with this compatibility data on GitHub
desktopmobile
Chrome
Edge
Firefox
Opera
Safari
Chrome Android
Firefox for Android
Opera Android
Safari on iOS
Samsung Internet
WebView Android
WebView on iOS
RTCRtpReceiver
createEncodedStreams
getCapabilities() static method
getContributingSources
audioLevel property in returned object
Audio tracks supported
rtpTimestamp property in returned object
source property in returned object
timestamp property in returned object
Video tracks supported
getParameters
codecs property in returned object
headerExtensions property in returned object
rtcp property in returned object
getStats
getSynchronizationSources
Audio tracks supported
Video tracks supported
jitterBufferTarget
rtcpTransport
Deprecated
track
transform
transport

Legend

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Full support
Full support
No support
No support
Deprecated. Not for use in new websites.
See implementation notes.

See also