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The RTCStatsReport interface is used to provide statistics data about WebRTC connections as returned by the RTCPeerConnection.getStats(), RTCRtpReceiver.getStats(), and RTCRtpSender.getStats() methods. This object consists of a mapping of strings which identify a category of statistics to objects which contains the various measured values for each.

Calling getStats() on an RTCPeerConnection lets you specifiy whether you wish to obtain statistics for outbound, inbound, or all streams on the connection. The RTCRtpReceiver and RTCRtpSender versions of getStats() specifically only return the incoming and outgoing streams, respectively.

The RTCStats subdictionaries

The full RTCStatsReport interface is made up of a mapping of strings to subdictionaries which contain the statistics for each category, such as ICE candidate or media stream statistics. Each of these subdictionaries has in common the following values:

A DOMHighResTimeStamp object which indicates the time at which the data was handled by the local WebRTC layer; for example, for statistics related to received RCTP packets, this would indicate the time at which the data was received by the local endpoint (the remote timestamp is available as well in one of the fields defined within one of the other subdictionaries. The time is specified relative to the UNIX epoch (the first moment of January 1, 1970, UTC).
A string, taken from the enum type RTCStatsType,
A DOMString which uniquely identifies the object which was inspected to produce this object based on RTCStats. This allows any two RTCStats-based objects which were produced from the same data to be correlated to one another. The format of this ID is not defined by the specification, so it cannot be relied upon to have any particular format, structure, or meaning beyond that.

From a definition standpoint, these values are defined in the RTCStats dictionary in RTCStatsReport's WebIDL file.



The RTCRtpStreamStats dictionary is based on RTCStats, and inherits its properties. In addition, some or all of the following properties are available.

Standard fields included for all media types
A DOMString which uniquely identifies the object which was inspected to produce the RTCCodecStats object associated with this RTP stream.
A DOMString whose value is "audio" if the associated MediaStreamTrack is audio-only or "video" if the track contains video. This value will match that of the media type indicated by RTCCodecStats.codec, as well as the track's kind property. Previously called mediaType.
The 32-bit integer which identifies the source of the RTP packets this RTCRtpStreamStats object covers. This value is generated per the RFC 3550 specification.
A DOMString which uniquely identifies the RTCMediaStreamTrackStats object representing the associated MediaStreamTrack. This is not the same as the value of MediaStreamTrack.id.
A DOMString uniquely identifying the object which was inspected to produce the RTCTransportStats object associated with this RTP stream.
Obsolete fields
Renamed to kind in the specification in February 2018. See Browser compatibility in RTCRtpStreamStats.kind to determine when browsers made the transition.
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.

A count of the total number of Full Intra Request (FIR) packets received by the sender. This statistic is only available to the device which is receiving the stream and is only available for video tracks. A FIR packet is sent by the receiving end of the stream when it falls behind or has lost packets and is unable to continue decoding the stream; the sending end of the stream receives the FIR packet and responds by sending a full frame instead of a delta frame, thereby letting the receiver "catch up." The higher this number is, the more often a problem of this nature arose, which can be a sign of network congestion or an overburdened receiving device.
The number of times the receiver notified the sender that one or more RTP packets has been lost by sending a Negative ACKnowledgement (NACK, also called "Generic NACK") packet to the sender. This value is only available to the receiver.
The number of times the receiving end of the stream sent a Picture Loss Indiciation (PLI) packet to the sender, indicating that it has lost some amount of encoded video data for one or more frames. Only the receiver has this value, and it's only valid for video tracks.
The sum of the Quantization Parameter (QP) values associated with every frame received to date on the video track described by this RTCRtpStreamStats object. In general, the higher this number is, the more heavily compressed the video track was. Combined with RTCReceivedRtpStreamStats.framesDecoded or RTCSentRtpStreamStats.framesEncoded, you can approximate the average QP over those frames, keeping in mind that codecs often vary the quantizer values even within frames. Also keep in mind that the values of QP can vary from codec to codec, so this value is only potentially useful when compared against the same codec.
The number of times the receiver notified the sender that one or more consecutive (in scan order) encoded video macroblocks have been lost or corrupted; this notification is sent by the receiver to the sender using a Slice Loss Indication (SLI) packet. This is a fairly technical part of how codecs work and while the higher this value is, the more errors occurred in the stream, generally most of the time this value is only interesting to very intensively hardcore media developers.


Contains statistics about inbound RTP streaming. The RTCInboundRTPStreamStats dictionary is based upon both RTCStats and RTCRtpStreamStats and includes all properties from those dictionaries, as well as the following:



Contains statistics about outbound RTP streaming. The RTCOutboundRTPStreamStats dictionary is based upon both RTCStats and RTCRtpStreamStats and includes all properties from those dictionaries, as well as the following:

A long integer indicating the number of RTP packets sent...
A 64-bit integer indicating how many bytes of data have been sent on the stream...


The number of frames which have been
A long integer which indicates the total number of frames successfully encoded for this RTP stream. This field is only present on video connections.










Specification Status Comment
WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'RTCStatsReport' in that specification.
Candidate Recommendation Initial specification.

Browser compatibilityUpdate compatibility data on GitHub

Chrome Edge Firefox Internet Explorer Opera Safari
Basic support No ? No No No No
Android webview Chrome for Android Edge Mobile Firefox for Android Opera for Android iOS Safari Samsung Internet
Basic support No No ? No No No No

See also

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