W3cubDocs

/Web APIs

RTCOutboundRtpStreamStats

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.

firCount

An integer value which indicates the total number of Full Intra Request (FIR) packets which this RTCRtpSender has sent to the remote RTCRtpReceiver. This is an indicator of how often the stream has lagged, requiring frames to be skipped in order to catch up. Valid only for video streams.

framesEncoded

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

nackCount

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

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).

pliCount

An integer specifying the number of times the remote receiver has notified this RTCRtpSender that some amount of encoded video data for one or more frames has been lost, using Picture Loss Indication (PLI) packets. Only available for video streams.

qpSum

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

qualityLimitationDurations

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

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

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.

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

Desktop Mobile
Chrome Edge Firefox Internet Explorer Opera Safari WebView Android Chrome Android Firefox for Android Opera Android Safari on IOS Samsung Internet
RTCOutboundRtpStreamStats 80 80 27 No 67 11 80 80 27 57 11 13.0
active 106 106 No No 92 No 106 106 No 72 No 20.0
bytesSent 80 80 72 No 67 13.1 80 80 79 57 13.4 13.0
codecId 80 80 98 No 67 11 80 80 98 57 11 13.0
headerBytesSent 80 80 100 No 67 14.1 80 80 100 57 14.5 13.0
id 80 80 72 No 67 13.1 80 80 79 57 13.4 13.0
kind 80 80 63 No 67 14 80 80 63 57 14 13.0
mediaSourceId 80 80 No No 67 14.1 80 80 No 57 14.5 13.0
mid 105 105 No No 91 No 105 105 No 72 No 20.0
nackCount 93 93 55 No 79 15.4 93 93 55 66 15.4 17.0
packetsSent 80 80 27 No 67 13.1 80 80 27 57 13.4 13.0
remoteId 81 81 27 No 68 15.3–15.4 81 81 No 58 No 13.0
retransmittedBytesSent 80 80 100 No 67 14.1 80 80 100 57 14.5 13.0
retransmittedPacketsSent 80 80 100 No 67 14.1 80 80 100 57 14.5 13.0
ssrc 80 80 27 No 67 11 80 80 27 57 11 13.0
targetBitrate 98 98 No No 84 16.4 98 98 No 68 16.4 18.0
timestamp 80 80 72 No 67 13.1 80 80 79 57 13.4 13.0
totalPacketSendDelay 80 80 No No 67 No 80 80 No 57 No 13.0
transportId 80 80 No No 67 11 80 80 No 57 11 13.0
type 80 80 72 No 67 13.1 80 80 79 57 13.4 13.0

See also

© 2005–2023 MDN contributors.
Licensed under the Creative Commons Attribution-ShareAlike License v2.5 or later.
https://developer.mozilla.org/en-US/docs/Web/API/RTCOutboundRtpStreamStats