RTCRemoteInboundRtpStreamStats
Baseline Widely available *
This feature is well established and works across many devices and browser versions. It’s been available across browsers since February 2020.
* Some parts of this feature may have varying levels of support.
TheRTCRemoteInboundRtpStreamStats dictionary of theWebRTC API is used to report statistics from the remote endpoint about a particular incoming RTP stream.These will correspond to an outgoing RTP stream at the local end of theRTCPeerConnection.
The statistics can be obtained by iterating theRTCStatsReport returned byRTCPeerConnection.getStats() orRTCRtpReceiver.getStats() until you find a report with thetype ofremote-inbound-rtp.
In this article
Instance properties
>Remote inbound specific statistics
fractionLostOptionalA number indicating the fraction of packets lost for this SSRC since the last sender or receiver report.
localIdOptionalA string that is used to find the local
RTCOutboundRtpStreamStatsobject that shares the samesynchronization source (SSRC).roundTripTimeOptionalA number that indicates the estimated round trip time (RTT) for this SSRC, in seconds.This property will not exist until valid RTT data has been received.
roundTripTimeMeasurementsOptionalA positive integer indicating the total number of valid round trip time measurements received for thissynchronization source (SSRC).
totalRoundTripTimeOptionalA number indicating the cumulative sum of all round trip time measurements since the beginning of the session, in seconds.The average round trip time can be computed by dividing
totalRoundTripTimebyroundTripTimeMeasurements.
Received RTP stream statistics
jitterOptionalA number indicating thepacket jitter for this synchronization source, measured in seconds.
packetsLostOptionalAn integer indicating the total number of RTP packets lost for this SSRC, as measured at the remote endpoint.This value can be negative if duplicate packets were received.
packetsReceivedOptionalExperimentalA positive integer indicating the total number of RTP packets received for this SSRC, including retransmissions.
Common RTP stream statistics
codecIdOptionalA string that uniquely identifies the object that was inspected to produce the
RTCCodecStatsobject associated with thisRTP stream.kindA string indicating whether the
MediaStreamTrackassociated with the stream is an audio or a video track.ssrcA positive integer that identifies the SSRC of the RTP packets in this stream.
transportIdOptionalA string that uniquely identifies the object which was inspected to produce the
RTCTransportStatsobject associated with this RTP stream.
Common instance properties
The following properties are common to all WebRTC statistics objects.
idA string that uniquely identifies the object that is being monitored to produce this set of statistics.
timestampA
DOMHighResTimeStampobject indicating the time at which the sample was taken for this statistics object.typeA string with the value
"inbound-rtp", indicating the type of statistics that the object contains.
Examples
Given a variablepeerConnection that is an instance of anRTCPeerConnection, the code below usesawait to wait for the statistics report, and then iterates it usingRTCStatsReport.forEach().It then filters the dictionaries for just those reports that have the type ofremote-inbound-rtp and logs the result.
const stats = await myPeerConnection.getStats();stats.forEach((report) => { if (report.type === "remote-inbound-rtp") { console.log("Remote Inbound RTP Stream Stats:"); console.log(`id: ${report.id}`); console.log(`timestamp: ${report.timestamp}`); console.log(`transportId: ${report.transportId}`); console.log(`ssrc: ${report.ssrc}`); console.log(`kind: ${report.kind}`); console.log(`codecId: ${report.codecId}`); console.log(`packetsReceived: ${report.packetsReceived}`); console.log(`packetsLost: ${report.packetsLost}`); console.log(`jitter: ${report.jitter}`); console.log(`totalRoundTripTime: ${report.totalRoundTripTime}`); console.log( `roundTripTimeMeasurements: ${report.roundTripTimeMeasurements}`, ); console.log(`roundTripTime: ${report.roundTripTime}`); console.log(`localId: ${report.localId}`); console.log(`fractionLost: ${report.fractionLost}`); }});Specifications
| Specification |
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| Identifiers for WebRTC's Statistics API> # dom-rtcstatstype-remote-inbound-rtp> |