RTCOutboundRtpStreamStats

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.

The RTCOutboundRtpStreamStats dictionary of the WebRTC API is used to report metrics and statistics related to an outbound RTP stream being sent by an RTCRtpSender.

The statistics can be obtained by iterating the RTCStatsReport returned by RTCPeerConnection.getStats() or RTCRtpSender.getStats() until you find a report with the type of outbound-rtp.

Instance properties

averageRtcpInterval

A floating-point value indicating the average RTCP interval between two consecutive compound RTCP packets.

framesEncoded

The number of frames that have been successfully encoded so far for sending on this RTP stream. Only valid for video streams.

perDscpPacketsSent

A record of key-value pairs with strings as the keys mapped to 32-bit integer values, each indicating the total number of packets this RTCRtpSender has transmitted for this source for each Differentiated Services Code Point (DSCP).

qualityLimitationDurations Experimental

A record mapping each of the quality limitation reasons in the RTCRemoteInboundRtpStreamStats enumeration to a floating-point value indicating the number of seconds the stream has spent with its quality limited for that reason.

qualityLimitationReason Experimental

One of the string none, cpu, bandwidth, or other, explaining why the resolution and/or frame rate is being limited for this RTP stream. Valid only for video streams.

remoteId

A string which identifies the RTCRemoteInboundRtpStreamStats object that provides statistics for the remote peer for this same SSRC. This ID is stable across multiple calls to getStats().

retransmittedBytesSent

The total number of bytes that have been retransmitted for this source as of the time the statistics were sampled. These retransmitted bytes comprise the packets included in the value returned by retransmittedPacketsSent.

retransmittedPacketsSent

The total number of packets that have needed to be retransmitted for this source as of the time the statistics were sampled. These retransmitted packets are included in the value returned by packetsSent.

senderId

The id of the RTCAudioSenderStats or RTCVideoSenderStats object containing statistics about this stream's RTCRtpSender.

sliCount

An integer indicating the number of times this sender received a Slice Loss Indication (SLI) frame from the remote peer, indicating that one or more consecutive video macroblocks have been lost or corrupted. Available only for video streams.

targetBitrate

A value indicating the bit rate the RTCRtpSender's codec is configured to attempt to achieve in its output media.

totalEncodedBytesTarget Experimental

A cumulative sum of the target frame sizes (the targeted maximum size of the frame in bytes when the codec is asked to compress it) for all of the frames encoded so far. This will likely differ from the total of the actual frame sizes.

totalEncodeTime

A floating-point value indicating the total number of seconds that have been spent encoding the frames encoded so far by this RTCRtpSender.

trackId

The id of the RTCSenderAudioTrackAttachmentStats or RTCSenderVideoTrackAttachmentStats object containing the current track attachment to the RTCRtpSender responsible for this stream.

Local-only measurements

These properties are computed locally, and are only available to the device receiving the media stream. Their primary purpose is to examine the error resiliency of the connection, as they provide information about lost packets, lost frames, and how heavily compressed the data is.

nackCount

An integer value indicating the total number of Negative ACKnowledgement (NACK) packets this RTCRtpSender has received from the remote RTCRtpReceiver.

qpSum

A 64-bit value containing the sum of the QP values for every frame encoded by this RTCRtpSender. Valid only for video streams.

Common RTP stream statistics

codecId Optional

A string that uniquely identifies the object that was inspected to produce the RTCCodecStats object associated with this RTP stream.

kind

A string indicating whether the MediaStreamTrack associated with the stream is an audio or a video track.

ssrc

A positive integer that identifies the SSRC of the RTP packets in this stream.

transportId Optional

A string that uniquely identifies the object which was inspected to produce the RTCTransportStats object associated with this RTP stream.

Common instance properties

The following properties are common to all WebRTC statistics objects.

id

A string that uniquely identifies the object that is being monitored to produce this set of statistics.

timestamp

A DOMHighResTimeStamp object indicating the time at which the sample was taken for this statistics object.

type

A string with the value "outbound-rtp", indicating the type of statistics that the object contains.

Examples

Specifications

Specification
Identifiers for WebRTC's Statistics API
# dom-rtcstatstype-outbound-rtp

Browser compatibility

BCD tables only load in the browser

See also