This document defines a set of WebIDL objects that allow access to the statistical information about a {{RTCPeerConnection}}.

These objects are returned from the getStats API that is specified in [[WEBRTC]].

Since the previous publication as a Candidate Recommendation, the stats objects were significantly reorganized to better match the underlying data sources. In addition, the networkType property was deprecated for preserving privacy, and the statsended event was removed as no longer needed.

Introduction

Audio, video, or data packets transmitted over a peer-connection can be lost, and experience varying amounts of network delay. A web application implementing WebRTC expects to monitor the performance of the underlying network and media pipeline.

This document defines the statistic identifiers used by the web application to extract metrics from the user agent.

This specification defines the conformance criteria that applies to a single product: the user agent.

Implementations that use ECMAScript to implement the objects defined in this specification MUST implement them in a manner consistent with the ECMAScript Bindings defined in the Web IDL specification [[WEBIDL]], as this document uses that specification and terminology.

This specification does not define what objects a conforming implementation should generate. Specifications that refer to this specification have the need to specify conformance. They should put in their document text that could like this (EXAMPLE ONLY):

Terminology

The terms {{RTCPeerConnection}}, {{RTCDataChannel}}, {{RTCDtlsTransport}}, {{RTCDtlsTransportState}}, {{RTCIceTransport}}, {{RTCIceRole}}, {{RTCIceTransportState}}, {{RTCDataChannelState}}, {{RTCIceCandidateType}}, {{RTCStats}}, {{RTCCertificate}} are defined in [[!WEBRTC]].

RTCPriorityType is defined in [[WEBRTC-PRIORITY]].

The term RTP stream is defined in [[RFC7656]].

The terms Synchronization Source (SSRC), RTCP Sender Report, RTCP Receiver Report are defined in [[RFC3550]].

The term RTCP Extended Report is defined in [[RFC3611]].

An audio sample refers to having a sample in any channel of an audio track - if multiple audio channels are used, metrics based on samples do not increment at a higher rate, simultaneously having samples in multiple channels counts as a single sample.

Basic concepts

The basic object of the stats model is the [= stats object =]. The following terms are defined to describe it:

Monitored object

An internal object that keeps a set of data values. Most monitored objects are object defined in the WebRTC API; they may be thought of as being internal properties of those objects.

Stats object
This is a set of values, copied out from a monitored object at a specific moment in time. It is returned as a WebIDL dictionary through the getStats API call.
Stats object reference

A monitored object has a stable identifier id, which is reflected in all stats objects produced from the monitored object. Stats objects may contain references to other stats objects using this id value. In a [= stats object =], these references are represented by a {{DOMString}} containing id value of the referenced stats object.

All stats object references have type {{DOMString}} and member names ending in Id, or they have type sequence<{{DOMString}}> and member names ending in Ids.

Stats value
Refers to a single value within a stats object.

A monitored object changes the values it contains continuously over its lifetime, but is never visible through the getStats API call. A stats object, once returned, never changes.

The stats API is defined in [[!WEBRTC]]. It is defined to return a collection of [= stats object =]s, each of which is a dictionary inheriting directly or indirectly from the {{RTCStats}} dictionary. This API is normatively defined in [[!WEBRTC]], but is reproduced here for ease of reference.

dictionary RTCStats {
    required DOMHighResTimeStamp timestamp;
    required RTCStatsType        type;
    required DOMString           id;
};
      

Timestamps are expressed with {{DOMHighResTimeStamp}} [[HIGHRES-TIME]], and are defined as {{Performance.timeOrigin}} + {{Performance.now()}} at the time the information is collected.

Guidelines for design of stats objects

When introducing a new stats object, the following principles should be followed:

The new members of the stats dictionary need to be named according to standard practice (camelCase), as per [[API-DESIGN-PRINCIPLES]].

Names ending in Id (such as {{RTCRtpStreamStats/transportId}}) are always a [= stats object reference =]; names ending in Ids (such as {{RTCMediaStreamStats/trackIds}}) are always of type sequence<DOMString>, where each {{DOMString}} is a [= stats object reference =].

If the natural name for a stats value would end in id (such as when the stats value is an in-protocol identifier for the monitored object), the recommended practice is to let the name end in identifier, such as {{RTCDataChannelStats/dataChannelIdentifier}}.

Stats are sampled by Javascript. In general, an application will not have overall control over how often stats are sampled, and the implementation cannot know what the intended use of the stats is. There is, by design, no control surface for the application to influence how stats are generated.

Therefore, letting the implementation compute "average" rates is not a good idea, since that implies some averaging time interval that can't be set beforehand. Instead, the recommended approach is to count the number of measurements of a value and sum the measurements given even if the sum is meaningless in itself; the JS application can then compute averages over any desired time interval by calling getStats() twice, taking the difference of the two sums and dividing by the difference of the two counts.

For stats that are measured against time, such as byte counts, no separate counter is needed; one can instead divide by the difference in the timestamps.

Guidelines for implementing stats objects

When implementing stats objects, the following guidelines should be adhered to:

Lifetime considerations for monitored objects

The object descriptions will say what the lifetime of a [= monitored object =] from the perspective of stats is. When a monitored object is deleted, it no longer appears in stats; until this happens, it will appear. This may or may not correspond to the actual lifetime of an object in an implementation; what matters for this specification is what appears in stats.

If a monitored object can only exist in a few instances over the lifetime of a {{RTCPeerConnection}}, it may be simplest to consider it "eternal" and never delete it from the set of objects reported on in stats. This type of object will remain visible until the {{RTCPeerConnection}} is no longer available; it is also visible in {{RTCPeerConnection/getStats()}} after pc.close(). This is the default when no lifetime is mentioned in its specification.

Objects that might exist in many instances over time should have a defined time at which they are [= deleted =], at which time they stop appearing in subsequent calls to {{RTCPeerConnection/getStats()}}. When an object is [= deleted =], we can guarantee that no subsequent {{RTCPeerConnection/getStats()}} call will contain a [= stats object reference =] that references the deleted object. We also guarantee that the stats id of the deleted object will never be reused for another object. This ensures that an application that collects [= stats object =]s for deleted [= monitored object =]s will always be able to uniquely identify the object pointed to in the result of any {{RTCPeerConnection/getStats()}} call.

Guidelines for {{RTCPeerConnection/getStats()}} results caching/throttling

A call to {{RTCPeerConnection/getStats()}} touches many components of WebRTC and may take significant time to execute. The implementation may or may not utilize caching or throttling of {{RTCPeerConnection/getStats()}} calls for performance benefits, however any implementation must adhere to the following:

When the state of the {{RTCPeerConnection}} visibly changes as a result of an API call, a promise resolving or an event firing, subsequent new {{RTCPeerConnection/getStats()}} calls must return up-to-date dictionaries for the affected objects.

When a stats object is [= deleted =], subsequent {{RTCPeerConnection/getStats()}} calls MUST NOT return stats for that [= monitored object =].

Maintenance procedures for stats object types

Adding new stats objects

This document, in its editors' draft form, serves as the repository for the currently defined set of stats object types including proposals for new standard types.

This document specifies the interoperable stats object types. Proposals for new object types may be made in the editors draft maintained on GitHub. New standard types may appear in future revisions of the W3C Recommendation.

If a need for a new stats object type or stats value within a stats object is found, an issue should be raised on Github, and a review process will decide on whether the stat should be added to the editors draft or not.

A pull request for a change to the editors draft may serve as guidance for the discussion, but the eventual merge is dependent on the review process.

While the WebRTC WG exists, it will serve as the review body; once it has disbanded, the W3C will have to establish appropriate review.

The level of review sought is that of the IETF process' "expert review", as defined in [[RFC5226]] section 4.1. The documentation needed includes the names of the new stats, their data types, and the definitions they are based on, specified to a level that allows interoperable implementation. The specification may consist of references to other documents.

Another specification that wishes to refer to a specific version (for instance for conformance) should refer to a dated version; these will be produced regularly when updates happen.

Retiring stats objects

At times, it makes sense to retire the definition for a stats object or a stats value. When this happens, it is not advisable to simply delete it from the spec, since there may be implementations out there that use it, and it is important that the name is reserved from re-use for another, incompatible definition.

Therefore, retired stats objects are moved to a separate section in this document. Retired stats objects are moved there in their entirety; retired stats values are moved to a "partial dictionary".

If there is no evidence that the retired object definition has ever been used (such as an object that is added to the spec and renamed, redefined or removed prior to implementation), the editors can decide to just remove the object from the spec.

Procedures for mitigating privacy concerns

The WebRTC's Statistics API exposes information about the system, including hardware capabilities and network characteristics. To limit the finger printing surface imposed by this API, some metrics are only exposed if allowed by the algorithms in this section.

Limiting exposure of hardware capabilities

To avoid passive fingerprinting, hardware capabilities should only be exposed in capturing or fullscreen contexts. This is tested using the algorithm below.

To check if hardware exposure is allowed, run the following steps:

  1. If the context capturing state is true, return true.

  2. If the current Document's fullscreenElement getter returns a non-null value, return true.

  3. Otherwise return false.

{{RTCStatsType}}

The {{RTCStats/type}} member, of type {{RTCStatsType}}, indicates the type of the object that the {{RTCStats}} object represents. An object with a given {{RTCStats/type}} can have only one IDL dictionary type, but multiple {{RTCStats/type}} values may indicate the same IDL dictionary type; for example, {{RTCStatsType/"local-candidate"}} and {{RTCStatsType/"remote-candidate"}} both use the IDL dictionary type {{RTCIceCandidateStats}}.

This specification is normative for the allowed values of {{RTCStatsType}}.

RTCStatsType enum

enum RTCStatsType {
"codec",
"inbound-rtp",
"outbound-rtp",
"remote-inbound-rtp",
"remote-outbound-rtp",
"media-source",
"media-playout",
"peer-connection",
"data-channel",
"stream",
"track",
"transport",
"candidate-pair",
"local-candidate",
"remote-candidate",
"certificate"
};
        

The following strings are valid values for {{RTCStatsType}}:

codec

Statistics for a codec that is currently being used by RTP streams being sent or received by this {{RTCPeerConnection}} object. It is accessed by the {{RTCCodecStats}}.

inbound-rtp

Statistics for an inbound RTP stream that is currently received with this {{RTCPeerConnection}} object. It is accessed by the {{RTCInboundRtpStreamStats}}.

outbound-rtp

Statistics for an outbound RTP stream that is currently sent with this {{RTCPeerConnection}} object. It is accessed by the {{RTCOutboundRtpStreamStats}}.

When there are multiple RTP streams connected to the same sender due to using simulcast, there will be one {{RTCOutboundRtpStreamStats}} per RTP stream, with distinct values of the {{RTCRtpStreamStats/ssrc}} member, and all these streams will have a reference to the same {{RTCStatsType/"track"}} object (of type {{RTCMediaStreamTrackStats}}). RTX streams do not show up as separate {{RTCOutboundRtpStreamStats}} objects but affect the {{RTCSentRtpStreamStats/packetsSent}}, {{RTCSentRtpStreamStats/bytesSent}}, {{RTCOutboundRtpStreamStats/retransmittedPacketsSent}} and {{RTCOutboundRtpStreamStats/retransmittedBytesSent}} counters of the relevant {{RTCOutboundRtpStreamStats}} objects.

remote-inbound-rtp

Statistics for the remote endpoint's inbound RTP stream corresponding to an outbound stream that is currently sent with this {{RTCPeerConnection}} object. It is measured at the remote endpoint and reported in an RTCP Receiver Report (RR) or RTCP Extended Report (XR). It is accessed by the {{RTCRemoteInboundRtpStreamStats}}.

remote-outbound-rtp

Statistics for the remote endpoint's outbound RTP stream corresponding to an inbound stream that is currently received with this {{RTCPeerConnection}} object. It is measured at the remote endpoint and reported in an RTCP Sender Report (SR). It is accessed by the {{RTCRemoteOutboundRtpStreamStats}}.

media-source

Statistics for the media produced by a {{MediaStreamTrack}} that is currently attached to an {{RTCRtpSender}}. This reflects the media that is fed to the encoder; after getUserMedia() constraints have been applied (i.e. not the raw media produced by the camera). It is either an {{RTCAudioSourceStats}} or {{RTCVideoSourceStats}} depending on its kind.

media-playout

Statistics related to audio playout. It is accessed by the {{RTCAudioPlayoutStats}}.

peer-connection

Statistics related to the {{RTCPeerConnection}} object. It is accessed by the {{RTCPeerConnectionStats}}.

data-channel

Statistics related to each {{RTCDataChannel}} id. It is accessed by the {{RTCDataChannelStats}}.

stream

This is now obsolete. Contains statistics related to a specific {{MediaStream}}. It is accessed by the obsolete dictionary {{RTCMediaStreamStats}}.

track

This is now obsolete. Statistics related to a specific {{MediaStreamTrack}}'s attachment to an {{RTCRtpSender}} or {{RTCRtpReceiver}}. It is accessed by the obsolete dictionary {{RTCMediaStreamTrackStats}}.

The monitored {{RTCStatsType/"track"}} object is [= deleted =] when the sender it reports on has its {{RTCRtpSender/track}} value changed to no longer refer to the same track.

All "track" stats have been made obsolete. The relevant metrics have been moved to "media-source", "outbound-rtp" and "inbound-rtp" stats.

transport

Transport statistics related to the {{RTCPeerConnection}} object. It is accessed by the {{RTCTransportStats}}.

candidate-pair

ICE candidate pair statistics related to the {{RTCIceTransport}} objects. It is accessed by the {{RTCIceCandidatePairStats}}.

A candidate pair that is not the current pair for a transport is [= deleted =] when the {{RTCIceTransport}} does an ICE restart, at the time the state changes to {{RTCIceTransportState/"new"}}. The candidate pair that is the current pair for a transport is [= deleted =] after an ICE restart when the {{RTCIceTransport}} switches to using a candidate pair generated from the new candidates; this time doesn't correspond to any other externally observable event.

local-candidate

ICE local candidate statistics related to the {{RTCIceTransport}} objects. It is accessed by the {{RTCIceCandidateStats}} for the local candidate.

A local candidate is [= deleted =] when the {{RTCIceTransport}} does an ICE restart, and the candidate is no longer a member of any non-deleted candidate pair.

remote-candidate

ICE remote candidate statistics related to the {{RTCIceTransport}} objects. It is accessed by the {{RTCIceCandidateStats}} for the remote candidate.

A remote candidate is [= deleted =] when the {{RTCIceTransport}} does an ICE restart, and the candidate is no longer a member of any non-deleted candidate pair.

certificate

Information about a certificate used by an {{RTCIceTransport}}. It is accessed by the {{RTCCertificateStats}}.

Stats dictionaries

The RTP statistics hierarchy

The dictionaries for RTP statistics are structured as a hierarchy, so that those stats that make sense in many different contexts are represented just once in IDL.

The metrics exposed here correspond to local measurements and those reported by RTCP packets. Compound RTCP packets contain multiple RTCP report blocks, such as Sender Report (SR) and Receiver Report (RR) whereas a non-compound RTCP packets may contain just a single RTCP SR or RR block.

The lifetime of all RTP [= monitored object =]s starts when the RTP stream is first used: When the first RTP packet is sent or received on the SSRC it represents, or when the first RTCP packet is sent or received that refers to the SSRC of the RTP stream.

RTP monitored objects are deleted when the corresponding RTP sender or RTP receiver is reconfigured to remove the corresponding RTP stream. This happens for the old SSRC when the {{RTCRtpStreamStats/ssrc}} changes, a simulcast layer is dropped or the {{RTCRtpTransceiver}}'s {{RTCRtpTransceiver/currentDirection}} becomes "stopped". The monitored object is not deleted if the transceiver is made "inactive" or if the encoding's {{RTCRtpEncodingParameters/active}} parameter is set to false. If an SSRC is recycled after a deletion event has happened, this is considered a new RTP monitored object and the new RTP stream stats will have reset counters and a new ID.

For a given RTP stats object, its total counters must always increase, but due to changes in SSRC, simulast layers dropping or transceivers stopping, an RTP stats object can be deleted and/or replaced by a new RTP stats object. The caller will need to be aware of this when aggregating packet counters accross multiple RTP stats objects (the aggregates may decrease due to deletions).

          An RTCRtpSender is sending two layers of simulcast (on SSRC=111 and
          SSRC=222). Two "outbound-rtp" stats objects are observed, one with
          ssrc=111, and the other with ssrc=222. Both objects' packet counters are
          increasing.

          One layer is inactivated using RTCRtpSender.setParameters(). While
          this pauses one of the layers (its packet counter freezes), the RTP
          monitored objects are not deleted. The RTCRtpTransceiver is negotiated
          as "inactive" and the RTP monitored objects are still not deleted.
          When the RTCRtpTransceiver becomes "sendonly" again, the same
          "outbound-rtp" objects continue to be used.

          Later, RTCRtpTransceiver.stop() is called. The "outbound-rtp" objects
          still exist but their packet counters have frozen. Renegotiation
          happens and the transceiver.currentDirection becomes "stopped", now
          both "outbound-rtp" objects have been deleted.
        

The hierarchy is as follows:

{{RTCRtpStreamStats}}: Stats that apply to any end of any RTP stream

RTCRtpStreamStats dictionary

dictionary RTCRtpStreamStats : RTCStats {
             required unsigned long       ssrc;
             required DOMString           kind;
             DOMString           transportId;
             DOMString           codecId;
};

Dictionary {{RTCRtpStreamStats}} Members

ssrc of type unsigned long

The synchronization source (ssrc) identifier is an unsigned integer value per [[RFC3550]] used to identify the stream of RTP packets that this stats object is describing.

For outbound and inbound local, ssrc describes the stats for the track that were sent and received, respectively by those endpoints. For the remote inbound and remote outbound, ssrc describes the stats for the track that were received by and sent to the remote endpoint.

kind of type DOMString

Either "audio" or "video". This MUST match the kind attribute of the related {{MediaStreamTrack}}.

transportId of type DOMString

It is a unique identifier that is associated to the object that was inspected to produce the {{RTCTransportStats}} associated with this RTP stream.

codecId of type DOMString

It is a unique identifier that is associated to the object that was inspected to produce the {{RTCCodecStats}} associated with this RTP stream.

RTCCodecStats dictionary

Codecs are created when registered for an RTP transport, but only the subset of codecs that are in use (referenced by an RTP stream) are exposed in getStats().

The {{RTCCodecStats}} object is created when one or more {{RTCRtpStreamStats/codecId}} references the codec. When there no longer exists any reference to the {{RTCCodecStats}}, the stats object is deleted. If the same codec is used again in the future, the {{RTCCodecStats}} object is revived with the same {{RTCStats/id}} as before.

Codec objects may be referenced by multiple RTP streams in media sections using the same transport, but similar codecs in different transports have different {{RTCCodecStats}} objects.

User agents are expected to coalesce information into a single "codec" entry per payload type per transport, unless {{RTCCodecStats/sdpFmtpLine}} differs per direction, in which case two entries (one for encode and one for decode) are needed.

dictionary RTCCodecStats : RTCStats {
             required unsigned long payloadType;
             required DOMString     transportId;
             required DOMString     mimeType;
             unsigned long clockRate;
             unsigned long channels;
             DOMString     sdpFmtpLine;
};

Dictionary {{RTCCodecStats}} Members

payloadType of type unsigned long

Payload type as used in RTP encoding or decoding.

transportId of type DOMString

The unique identifier of the transport on which this codec is being used, which can be used to look up the corresponding {{RTCTransportStats}} object.

mimeType of type DOMString

The codec MIME media type/subtype. e.g., video/vp8 or equivalent.

clockRate of type unsigned long

Represents the media sampling rate.

channels of type unsigned long

When present, indicates the number of channels (mono=1, stereo=2).

sdpFmtpLine of type DOMString

The a=fmtp line in the SDP corresponding to the codec, i.e., after the colon following the PT. This defined by [[!JSEP]] in Section 5.7.

RTCReceivedRtpStreamStats dictionary

dictionary RTCReceivedRtpStreamStats : RTCRtpStreamStats {
             unsigned long long   packetsReceived;
             long long            packetsLost;
             double               jitter;
};

Dictionary {{RTCReceivedRtpStreamStats}} Members

packetsReceived of type unsigned long long

Total number of RTP packets received for this SSRC. At the receiving endpoint, this is calculated as defined in [[!RFC3550]] section 6.4.1. At the sending endpoint the {{packetsReceived}} is estimated by subtracting the Cumulative Number of Packets Lost from the Extended Highest Sequence Number Received, both reported in the RTCP Receiver Report, and then subtracting the initial Extended Sequence Number that was sent to this SSRC in a RTCP Sender Report and then adding one, to mirror what is discussed in Appendix A.3 in [[!RFC3550]], but for the sender side. If no RTCP Receiver Report has been received yet, then return 0.

packetsLost of type long long

Total number of RTP packets lost for this SSRC. Calculated as defined in [[!RFC3550]] section 6.4.1. Note that because of how this is estimated, it can be negative if more packets are received than sent.

jitter of type double

Packet Jitter measured in seconds for this SSRC. Calculated as defined in section 6.4.1. of [[!RFC3550]].

{{RTCInboundRtpStreamStats}} dictionary

The RTCInboundRtpStreamStats dictionary represents the measurement metrics for the incoming RTP media stream. The timestamp reported in the statistics object is the time at which the data was sampled.

dictionary RTCInboundRtpStreamStats : RTCReceivedRtpStreamStats {
             required DOMString   trackIdentifier;
             required DOMString   kind;
             DOMString            mid;
             DOMString            remoteId;
             unsigned long        framesDecoded;
             unsigned long        keyFramesDecoded;
             unsigned long        framesDropped;
             unsigned long        frameWidth;
             unsigned long        frameHeight;
             double               framesPerSecond;
             unsigned long long   qpSum;
             double               totalDecodeTime;
             double               totalInterFrameDelay;
             double               totalSquaredInterFrameDelay;
             unsigned long        pauseCount;
             double               totalPausesDuration;
             unsigned long        freezeCount;
             double               totalFreezesDuration;
             DOMHighResTimeStamp  lastPacketReceivedTimestamp;
             unsigned long long   headerBytesReceived;
             unsigned long long   packetsDiscarded;
             unsigned long long   fecPacketsReceived;
             unsigned long long   fecPacketsDiscarded;
             unsigned long long   bytesReceived;
             unsigned long        nackCount;
             unsigned long        firCount;
             unsigned long        pliCount;
             double               totalProcessingDelay;
             DOMHighResTimeStamp  estimatedPlayoutTimestamp;
             double               jitterBufferDelay;
             double               jitterBufferTargetDelay;
             unsigned long long   jitterBufferEmittedCount;
             double               jitterBufferMinimumDelay;
             unsigned long long   totalSamplesReceived;
             unsigned long long   concealedSamples;
             unsigned long long   silentConcealedSamples;
             unsigned long long   concealmentEvents;
             unsigned long long   insertedSamplesForDeceleration;
             unsigned long long   removedSamplesForAcceleration;
             double               audioLevel;
             double               totalAudioEnergy;
             double               totalSamplesDuration;
             unsigned long        framesReceived;
             DOMString            decoderImplementation;
             DOMString            playoutId;
             boolean              powerEfficientDecoder;
             unsigned long        framesAssembledFromMultiplePackets;
             double               totalAssemblyTime;
            };

Dictionary {{RTCInboundRtpStreamStats}} Members

trackIdentifier of type DOMString

The value of the {{MediaStreamTrack}}'s id attribute.

kind of type DOMString

The value of the {{MediaStreamTrack}}'s kind attribute. This is either "audio" or "video".

mid of type DOMString

If the {{RTCRtpTransceiver}} owning this stream has a {{RTCRtpTransceiver/mid}} value that is not null, this is that value, otherwise this member is not present.

remoteId of type DOMString

The {{remoteId}} is used for looking up the remote {{RTCRemoteOutboundRtpStreamStats}} object for the same SSRC.

framesDecoded

Only [= map/exist =]s for video. It represents the total number of frames correctly decoded for this RTP stream, i.e., frames that would be displayed if no frames are dropped.

keyFramesDecoded of type unsigned long

Only [= map/exist =]s for video. It represents the total number of key frames, such as key frames in VP8 [[RFC6386]] or IDR-frames in H.264 [[RFC6184]], successfully decoded for this RTP media stream. This is a subset of {{framesDecoded}}. framesDecoded - keyFramesDecoded gives you the number of delta frames decoded.

framesDropped of type unsigned long

Only [= map/exist =]s for video. The total number of frames dropped prior to decode or dropped because the frame missed its display deadline for this receiver's track. The measurement begins when the receiver is created and is a cumulative metric as defined in Appendix A (g) of [[!RFC7004]].

frameWidth of type unsigned long

Only [= map/exist =]s for video. Represents the width of the last decoded frame. Before the first frame is decoded this member does not [= map/exist =].

frameHeight of type unsigned long

Only [= map/exist =]s for video. Represents the height of the last decoded frame. Before the first frame is decoded this member does not [= map/exist =].

framesPerSecond of type double

Only [= map/exist =]s for video. The number of decoded frames in the last second.

qpSum of type unsigned long long

Only [= map/exist =]s for video. The sum of the QP values of frames decoded by this receiver. The count of frames is in {{framesDecoded}}.

The definition of QP value depends on the codec; for VP8, the QP value is the value carried in the frame header as the syntax element y_ac_qi, and defined in [[RFC6386]] section 19.2. Its range is 0..127.

Note that the QP value is only an indication of quantizer values used; many formats have ways to vary the quantizer value within the frame.

totalDecodeTime of type double

Total number of seconds that have been spent decoding the {{framesDecoded}} frames of this stream. The average decode time can be calculated by dividing this value with {{framesDecoded}}. The time it takes to decode one frame is the time passed between feeding the decoder a frame and the decoder returning decoded data for that frame.

totalInterFrameDelay of type double

Sum of the interframe delays in seconds between consecutively decoded frames, recorded just after a frame has been decoded. The interframe delay variance be calculated from {{totalInterFrameDelay}}, {{totalSquaredInterFrameDelay}}, and {{framesDecoded}} according to the formula: ({{totalSquaredInterFrameDelay}} - {{totalInterFrameDelay}}^2/ {{framesDecoded}})/{{framesDecoded}}.

totalSquaredInterFrameDelay of type double

Sum of the squared interframe delays in seconds between consecutively decoded frames, recorded just after a frame has been decoded. See {{totalInterFrameDelay}} for details on how to calculate the interframe delay variance.

pauseCount of type unsigned long

Count the total number of video pauses experienced by this receiver. Video is considered to be paused if time passed since last rendered frame exceeds 5 seconds. {{pauseCount}} is incremented when a frame is rendered after such a pause.

totalPausesDuration of type double

Total duration of pauses (for definition of pause see {{pauseCount}}), in seconds. This value is updated when a frame is rendered.

freezeCount of type unsigned long

Count the total number of video freezes experienced by this receiver. It is a freeze if frame duration, which is time interval between two consecutively rendered frames, is equal or exceeds Max(3 * avg_frame_duration_ms, avg_frame_duration_ms + 150), where avg_frame_duration_ms is linear average of durations of last 30 rendered frames.

totalFreezesDuration of type double

Total duration of rendered frames which are considered as frozen (for definition of freeze see {{freezeCount}}), in seconds. This value is updated when a frame is rendered.

lastPacketReceivedTimestamp of type DOMHighResTimeStamp

Represents the timestamp at which the last packet was received for this SSRC. This differs from {{RTCStats/timestamp}}, which represents the time at which the statistics were generated by the local endpoint.

headerBytesReceived of type unsigned long long

Total number of RTP header and padding bytes received for this SSRC. This does not include the size of transport layer headers such as IP or UDP. headerBytesReceived + bytesReceived equals the number of bytes received as payload over the transport.

packetsDiscarded of type unsigned long long

The cumulative number of RTP packets discarded by the jitter buffer due to late or early-arrival, i.e., these packets are not played out. RTP packets discarded due to packet duplication are not reported in this metric [[XRBLOCK-STATS]]. Calculated as defined in [[!RFC7002]] section 3.2 and Appendix A.a.

fecPacketsReceived of type unsigned long long

Total number of RTP FEC packets received for this SSRC. This counter can also be incremented when receiving FEC packets in-band with media packets (e.g., with Opus).

fecPacketsDiscarded of type unsigned long long

Total number of RTP FEC packets received for this SSRC where the error correction payload was discarded by the application. This may happen 1. if all the source packets protected by the FEC packet were received or already recovered by a separate FEC packet, or 2. if the FEC packet arrived late, i.e., outside the recovery window, and the lost RTP packets have already been skipped during playout. This is a subset of {{fecPacketsReceived}}.

bytesReceived of type unsigned long long

Total number of bytes received for this SSRC. Calculated as defined in [[!RFC3550]] section 6.4.1.

firCount of type unsigned long

Only [= map/exist =]s for video. Count the total number of Full Intra Request (FIR) packets, as defined in [[!RFC5104]] section 4.3.1, sent by this receiver. Does not count the RTCP FIR indicated in [[?RFC2032]] which was deprecated by [[?RFC4587]].

pliCount of type unsigned long

Only [= map/exist =]s for video. Count the total number of Picture Loss Indication (PLI) packets, as defined in [[!RFC4585]] section 6.3.1, sent by this receiver.

totalProcessingDelay of type double

It is the sum of the time, in seconds, each audio sample or video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the corresponding sample or frame is decoded (decoded timestamp). At this point the audio sample or video frame is ready for playout by the MediaStreamTrack. Typically ready for playout here means after the audio sample or video frame is fully decoded by the decoder.

Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible and the decoded timestamp is measured as soon as the complete sample or frame is decoded.

In the case of audio, several samples are received in the same RTP packet, all samples will share the same reception timestamp and different decoded timestamps. In the case of video, the frame is received over several RTP packets, in this case the earliest timestamp containing the frame is counted as the reception timestamp, and the decoded timestamp corresponds to when the complete frame is decoded.

This metric is not incremented for frames that are not decoded, i.e. {{RTCInboundRtpStreamStats/framesDropped}}. The average processing delay can be calculated by dividing the {{totalProcessingDelay}} with the {{framesDecoded}} for video (or povisional stats spec totalSamplesDecoded for audio).

nackCount of type unsigned long

Count the total number of Negative ACKnowledgement (NACK) packets, as defined in [[!RFC4585]] section 6.2.1, sent by this receiver.

estimatedPlayoutTimestamp of type DOMHighResTimeStamp

This is the estimated playout time of this receiver's track. The playout time is the NTP timestamp of the last playable audio sample or video frame that has a known timestamp (from an RTCP SR packet mapping RTP timestamps to NTP timestamps), extrapolated with the time elapsed since it was ready to be played out. This is the "current time" of the track in NTP clock time of the sender and can be present even if there is no audio currently playing.

This can be useful for estimating how much audio and video is out of sync for two tracks from the same source, audioTrackStats.{{estimatedPlayoutTimestamp}} - videoTrackStats.{{estimatedPlayoutTimestamp}}.

jitterBufferDelay of type double

The purpose of the jitter buffer is to recombine RTP packets into frames (in the case of video) and have smooth playout. The model described here assumes that the samples or frames are still compressed and have not yet been decoded. It is the sum of the time, in seconds, each audio sample or a video frame takes from the time the first packet is received by the jitter buffer (ingest timestamp) to the time it exits the jitter buffer (emit timestamp). In the case of audio, several samples belong to the same RTP packet, hence they will have the same ingest timestamp but different jitter buffer emit timestamps. In the case of video, the frame maybe is received over several RTP packets, hence the ingest timestamp is the earliest packet of the frame that entered the jitter buffer and the emit timestamp is when the whole frame exits the jitter buffer. This metric increases upon samples or frames exiting, having completed their time in the buffer (and incrementing {{jitterBufferEmittedCount}}). The average jitter buffer delay can be calculated by dividing the {{jitterBufferDelay}} with the {{jitterBufferEmittedCount}}.

jitterBufferTargetDelay of type double

This value is increased by the target jitter buffer delay every time a sample is emitted by the jitter buffer. The added target is the target delay, in seconds, at the time that the sample was emitted from the jitter buffer. To get the average target delay, divide by {{jitterBufferEmittedCount}}.

jitterBufferEmittedCount of type unsigned long long

The total number of audio samples or video frames that have come out of the jitter buffer (increasing {{jitterBufferDelay}}).

jitterBufferMinimumDelay of type double

There are various reasons why the jitter buffer delay might be increased to a higher value, such as to achieve AV synchronization or because a playoutDelay was set on a RTCRtpReceiver. When using one of these mechanisms, it can be useful to keep track of the minimal jitter buffer delay that could have been achieved, so WebRTC clients can track the amount of additional delay that is being added.

This metric works the same way as {{jitterBufferTargetDelay}}, except that it is not affected by external mechanisms that increase the jitter buffer target delay, such as playoutDelay (see link above), AV sync, or any other mechanisms. This metric is purely based on the network characteristics such as jitter and packet loss, and can be seen as the minimum obtainable jitter buffer delay if no external factors would affect it. The metric is updated every time {{jitterBufferEmittedCount}} is updated.

totalSamplesReceived of type unsigned long long

Only [= map/exist =]s for audio. The total number of samples that have been received on this RTP stream. This includes {{concealedSamples}}.

concealedSamples of type unsigned long long

Only [= map/exist =]s for audio. The total number of samples that are concealed samples. A concealed sample is a sample that was replaced with synthesized samples generated locally before being played out. Examples of samples that have to be concealed are samples from lost packets (reported in {{RTCReceivedRtpStreamStats/packetsLost}}) or samples from packets that arrive too late to be played out (reported in {{RTCInboundRtpStreamStats/packetsDiscarded}}).

silentConcealedSamples of type unsigned long long

Only [= map/exist =]s for audio. The total number of concealed samples inserted that are "silent". Playing out silent samples results in silence or comfort noise. This is a subset of {{concealedSamples}}.

concealmentEvents of type unsigned long long

Only [= map/exist =]s for audio. The number of concealment events. This counter increases every time a concealed sample is synthesized after a non-concealed sample. That is, multiple consecutive concealed samples will increase the {{concealedSamples}} count multiple times but is a single concealment event.

insertedSamplesForDeceleration of type unsigned long long

Only [= map/exist =]s for audio. When playout is slowed down, this counter is increased by the difference between the number of samples received and the number of samples played out. If playout is slowed down by inserting samples, this will be the number of inserted samples.

removedSamplesForAcceleration of type unsigned long long

Only [= map/exist =]s for audio. When playout is sped up, this counter is increased by the difference between the number of samples received and the number of samples played out. If speedup is achieved by removing samples, this will be the count of samples removed.

audioLevel of type double

Only [= map/exist =]s for audio. Represents the audio level of the receiving track. For audio levels of tracks attached locally, see {{RTCAudioSourceStats}} instead.

The value is between 0..1 (linear), where 1.0 represents 0 dBov, 0 represents silence, and 0.5 represents approximately 6 dBSPL change in the sound pressure level from 0 dBov.

The {{audioLevel}} is averaged over some small interval, using the algorithm described under {{totalAudioEnergy}}. The interval used is implementation dependent.

totalAudioEnergy of type double

Only [= map/exist =]s for audio. Represents the audio energy of the receiving track. For audio energy of tracks attached locally, see {{RTCAudioSourceStats}} instead.

This value MUST be computed as follows: for each audio sample that is received (and thus counted by {{totalSamplesReceived}}), add the sample's value divided by the highest-intensity encodable value, squared and then multiplied by the duration of the sample in seconds. In other words, duration * Math.pow(energy/maxEnergy, 2).

This can be used to obtain a root mean square (RMS) value that uses the same units as {{audioLevel}}, as defined in [[RFC6464]]. It can be converted to these units using the formula Math.sqrt(totalAudioEnergy/totalSamplesDuration). This calculation can also be performed using the differences between the values of two different {{RTCPeerConnection/getStats()}} calls, in order to compute the average audio level over any desired time interval. In other words, do Math.sqrt((energy2 - energy1)/(duration2 - duration1)).

For example, if a 10ms packet of audio is produced with an RMS of 0.5 (out of 1.0), this should add 0.5 * 0.5 * 0.01 = 0.0025 to {{totalAudioEnergy}}. If another 10ms packet with an RMS of 0.1 is received, this should similarly add 0.0001 to {{totalAudioEnergy}}. Then, Math.sqrt(totalAudioEnergy/totalSamplesDuration) becomes Math.sqrt(0.0026/0.02) = 0.36, which is the same value that would be obtained by doing an RMS calculation over the contiguous 20ms segment of audio.

If multiple audio channels are used, the audio energy of a sample refers to the highest energy of any channel.

totalSamplesDuration of type double

Only [= map/exist =]s for audio. Represents the audio duration of the receiving track. For audio durations of tracks attached locally, see {{RTCAudioSourceStats}} instead.

Represents the total duration in seconds of all samples that have been received (and thus counted by {{totalSamplesReceived}}). Can be used with {{totalAudioEnergy}} to compute an average audio level over different intervals.

framesReceived of type unsigned long

Only [= map/exist =]s for video. Represents the total number of complete frames received on this RTP stream. This metric is incremented when the complete frame is received.

decoderImplementation of type DOMString

Only defined when [= exposing hardware is allowed =].

Only [= map/exist =]s for video. Identifies the decoder implementation used. This is useful for diagnosing interoperability issues.

playoutId of type DOMString

If audio playout is happening, this is used to look up the corresponding {{RTCAudioPlayoutStats}}.

powerEfficientDecoder of type boolean

Only defined when [= exposing hardware is allowed =].

Whether the decoder currently used is considered power efficient by the user agent. This SHOULD reflect if the configuration results in hardware acceleration, but the user agent MAY take other information into account when deciding if the configuration is considered power efficient.

framesAssembledFromMultiplePackets of type unsigned long

Only [= map/exist =]s for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet. For such frames the {{totalAssemblyTime}} is incremented. The average frame assembly time can be calculated by dividing the {{totalAssemblyTime}} with {{framesAssembledFromMultiplePackets}}.

totalAssemblyTime of type double

Only [= map/exist =]s for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received. Only incremented for frames consisting of more than one RTP packet.

Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible. This metric is not incremented for frames that are not decoded, i.e., {{framesDropped}} or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet.

{{RTCRemoteInboundRtpStreamStats}} dictionary

The RTCRemoteInboundRtpStreamStats dictionary represents the remote endpoint's measurement metrics for a particular incoming RTP stream (corresponding to an outgoing RTP stream at the sending endpoint). The timestamp reported in the statistics object is the time at which the corresponding RTCP RR was received.

dictionary RTCRemoteInboundRtpStreamStats : RTCReceivedRtpStreamStats {
             DOMString            localId;
             double               roundTripTime;
             double               totalRoundTripTime;
             double               fractionLost;
             unsigned long long   roundTripTimeMeasurements;
};

Dictionary {{RTCRemoteInboundRtpStreamStats}} Members

localId of type DOMString

The {{localId}} is used for looking up the local {{RTCOutboundRtpStreamStats}} object for the same SSRC.

roundTripTime of type double

Estimated round trip time for this SSRC based on the RTCP timestamps in the RTCP Receiver Report (RR) and measured in seconds. Calculated as defined in section 6.4.1. of [[!RFC3550]]. If no RTCP Receiver Report is received with a DLSR value other than 0, the round trip time is left undefined.

totalRoundTripTime of type double

Represents the cumulative sum of all round trip time measurements in seconds since the beginning of the session. The individual round trip time is calculated based on the RTCP timestamps in the RTCP Receiver Report (RR) [[!RFC3550]], hence undefined roundtrip times are not added. The average round trip time can be computed from {{totalRoundTripTime}} by dividing it by {{roundTripTimeMeasurements}}.

fractionLost of type double

The fraction packet loss reported for this SSRC. Calculated as defined in [[!RFC3550]] section 6.4.1 and Appendix A.3.

roundTripTimeMeasurements of type unsigned long long

Represents the total number of RTCP RR blocks received for this SSRC that contain a valid round trip time. This counter will not increment if the {{roundTripTime}} is undefined.

RTCSentRtpStreamStats dictionary

dictionary RTCSentRtpStreamStats : RTCRtpStreamStats {
             unsigned long      packetsSent;
             unsigned long long bytesSent;
};

Dictionary {{RTCSentRtpStreamStats}} Members

packetsSent of type unsigned long long

Total number of RTP packets sent for this SSRC. This includes retransmissions. Calculated as defined in [[!RFC3550]] section 6.4.1.

bytesSent of type unsigned long long

Total number of bytes sent for this SSRC. This includes retransmissions. Calculated as defined in [[!RFC3550]] section 6.4.1.

{{RTCOutboundRtpStreamStats}} dictionary

The RTCOutboundRtpStreamStats dictionary represents the measurement metrics for the outgoing RTP stream. The timestamp reported in the statistics object is the time at which the data was sampled.

dictionary RTCOutboundRtpStreamStats : RTCSentRtpStreamStats {
             DOMString            mid;
             DOMString            mediaSourceId;
             DOMString            remoteId;
             DOMString            rid;
             unsigned long long   headerBytesSent;
             unsigned long long   retransmittedPacketsSent;
             unsigned long long   retransmittedBytesSent;
             double               targetBitrate;
             unsigned long long   totalEncodedBytesTarget;
             unsigned long        frameWidth;
             unsigned long        frameHeight;
             double               framesPerSecond;
             unsigned long        framesSent;
             unsigned long        hugeFramesSent;
             unsigned long        framesEncoded;
             unsigned long        keyFramesEncoded;
             unsigned long long   qpSum;
             double               totalEncodeTime;
             double               totalPacketSendDelay;
             RTCQualityLimitationReason                 qualityLimitationReason;
             record<DOMString, double> qualityLimitationDurations;
             unsigned long        qualityLimitationResolutionChanges;
             unsigned long        nackCount;
             unsigned long        firCount;
             unsigned long        pliCount;
             DOMString            encoderImplementation;
             boolean              powerEfficientEncoder;
             boolean              active;
             DOMString            scalabilityMode;
};

Dictionary {{RTCOutboundRtpStreamStats}} Members

mid of type DOMString

If the {{RTCRtpTransceiver}} owning this stream has a {{RTCRtpTransceiver/mid}} value that is not null, this is that value, otherwise this member is not present.

mediaSourceId of type DOMString

The identifier of the stats object representing the track currently attached to the sender of this stream, an {{RTCMediaSourceStats}}.

remoteId of type DOMString

The {{remoteId}} is used for looking up the remote {{RTCRemoteInboundRtpStreamStats}} object for the same SSRC.

rid of type DOMString

Exposes the {{RTCRtpCodingParameters/rid}} encoding parameter of this RTP stream if it has been set, otherwise it is undefined. If set this value will be present regardless if the RID RTP header extension has been negotiated.

headerBytesSent of type unsigned long long

Total number of RTP header and padding bytes sent for this SSRC. This does not include the size of transport layer headers such as IP or UDP. headerBytesSent + bytesSent equals the number of bytes sent as payload over the transport.

retransmittedPacketsSent of type unsigned long long

The total number of packets that were retransmitted for this SSRC. This is a subset of {{RTCSentRtpStreamStats/packetsSent}}. If RTX is not negotiated, retransmitted packets are sent over this {{RTCRtpStreamStats/ssrc}}. If RTX was negotiated, retransmitted packets are sent over a separate ssrc but is still accounted for here.

retransmittedBytesSent of type unsigned long long

The total number of bytes that were retransmitted for this SSRC, only including payload bytes. This is a subset of {{RTCSentRtpStreamStats/bytesSent}}. If RTX is not negotiated, retransmitted bytes are sent over this {{RTCRtpStreamStats/ssrc}}. If RTX was negotiated, retransmitted bytes are sent over a separate ssrc but is still accounted for here.

targetBitrate of type double

Reflects the current encoder target in bits per second. The target is an instantanous value reflecting the encoder's settings, but the resulting payload bytes sent per second, excluding retransmissions, SHOULD closely correlate to the target. See also {{RTCSentRtpStreamStats/bytesSent}} and {{RTCOutboundRtpStreamStats/retransmittedBytesSent}}. The {{targetBitrate}} is defined in the same way as the Transport Independent Application Specific (TIAS) bitrate [[!RFC3890]].

totalEncodedBytesTarget of type unsigned long long

This value is increased by the target frame size in bytes every time a frame has been encoded. The actual frame size may be bigger or smaller than this number. This value goes up every time {{framesEncoded}} goes up.

frameWidth of type unsigned long

Only [= map/exist =]s for video. Represents the width of the last encoded frame. The resolution of the encoded frame may be lower than the media source (see {{RTCVideoSourceStats.width}}). Before the first frame is encoded this member does not [= map/exist =].

frameHeight of type unsigned long

Only [= map/exist =]s for video. Represents the height of the last encoded frame. The resolution of the encoded frame may be lower than the media source (see {{RTCVideoSourceStats.height}}). Before the first frame is encoded this member does not [= map/exist =].

framesPerSecond of type double

Only [= map/exist =]s for video. The number of encoded frames during the last second. This may be lower than the media source frame rate (see {{RTCVideoSourceStats.framesPerSecond}}).

framesSent of type unsigned long

Only [= map/exist =]s for video. Represents the total number of frames sent on this RTP stream.

hugeFramesSent of type unsigned long

Only [= map/exist =]s for video. Represents the total number of huge frames sent by this RTP stream. Huge frames, by definition, are frames that have an encoded size at least 2.5 times the average size of the frames. The average size of the frames is defined as the target bitrate per second divided by the target FPS at the time the frame was encoded. These are usually complex to encode frames with a lot of changes in the picture. This can be used to estimate, e.g slide changes in the streamed presentation.

The multiplier of 2.5 is choosen from analyzing encoded frame sizes for a sample presentation using WebRTC standalone implementation. 2.5 is a reasonably large multiplier which still caused all slide change events to be identified as a huge frames. It, however, produced 1.4% of false positive slide change detections which is deemed reasonable.

framesEncoded of type unsigned long

Only [= map/exist =]s for video. It represents the total number of frames successfully encoded for this RTP media stream.

keyFramesEncoded of type unsigned long

Only [= map/exist =]s for video. It represents the total number of key frames, such as key frames in VP8 [[RFC6386]] or IDR-frames in H.264 [[RFC6184]], successfully encoded for this RTP media stream. This is a subset of {{framesEncoded}}. framesEncoded - keyFramesEncoded gives you the number of delta frames encoded.

qpSum of type unsigned long long

Only [= map/exist =]s for video. The sum of the QP values of frames encoded by this sender. The count of frames is in {{framesEncoded}}.

The definition of QP value depends on the codec; for VP8, the QP value is the value carried in the frame header as the syntax element y_ac_qi, and defined in [[RFC6386]] section 19.2. Its range is 0..127.

Note that the QP value is only an indication of quantizer values used; many formats have ways to vary the quantizer value within the frame.

totalEncodeTime of type double

Total number of seconds that has been spent encoding the {{framesEncoded}} frames of this stream. The average encode time can be calculated by dividing this value with {{framesEncoded}}. The time it takes to encode one frame is the time passed between feeding the encoder a frame and the encoder returning encoded data for that frame. This does not include any additional time it may take to packetize the resulting data.

totalPacketSendDelay of type double

The total number of seconds that packets have spent buffered locally before being transmitted onto the network. The time is measured from when a packet is emitted from the RTP packetizer until it is handed over to the OS network socket. This measurement is added to {{totalPacketSendDelay}} when {{RTCSentRtpStreamStats/packetsSent}} is incremented.

qualityLimitationReason of type {{RTCQualityLimitationReason}}

Only [= map/exist =]s for video. The current reason for limiting the resolution and/or framerate, or {{RTCQualityLimitationReason/"none"}} if not limited.

The implementation reports the most limiting factor. If the implementation is not able to determine the most limiting factor because multiple may exist, the reasons MUST be reported in the following order of priority: "bandwidth", "cpu", "other".

The consumption of CPU and bandwidth resources is interdependent and difficult to estimate, making it hard to determine what the "most limiting factor" is. The priority order promoted here is based on the heuristic that "bandwidth" is generally more varying and thus a more likely and more useful signal than "cpu".

qualityLimitationDurations of type record<DOMString, double>

Only [= map/exist =]s for video. A record of the total time, in seconds, that this stream has spent in each quality limitation state. The record includes a mapping for all {{RTCQualityLimitationReason}} types, including {{RTCQualityLimitationReason/"none"}}.

The sum of all entries minus {{qualityLimitationDurations}}[{{RTCQualityLimitationReason/"none"}}] gives the total time that the stream has been limited.

qualityLimitationResolutionChanges of type unsigned long

Only [= map/exist =]s for video. The number of times that the resolution has changed because we are quality limited ({{qualityLimitationReason}} has a value other than {{RTCQualityLimitationReason/"none"}}). The counter is initially zero and increases when the resolution goes up or down. For example, if a 720p track is sent as 480p for some time and then recovers to 720p, {{qualityLimitationResolutionChanges}} will have the value 2.

nackCount of type unsigned long

Count the total number of Negative ACKnowledgement (NACK) packets, as defined in [[!RFC4585]] section 6.2.1, received by this sender.

firCount of type unsigned long

Only [= map/exist =]s for video. Count the total number of Full Intra Request (FIR) packets, as defined in [[!RFC5104]] section 4.3.1, received by this sender. Does not count the RTCP FIR indicated in [[?RFC2032]] which was deprecated by [[?RFC4587]].

pliCount of type unsigned long

Only [= map/exist =]s for video. Count the total number of Picture Loss Indication (PLI) packets, as defined in [[!RFC4585]] section 6.3.1, received by this sender

encoderImplementation of type DOMString

Only defined when [= exposing hardware is allowed =].

Only [= map/exist =]s for video. Identifies the encoder implementation used. This is useful for diagnosing interoperability issues.

powerEfficientEncoder of type boolean

Only defined when [= exposing hardware is allowed =].

Whether the encoder currently used is considered power efficient by the user agent. This SHOULD reflect if the configuration results in hardware acceleration, but the user agent MAY take other information into account when deciding if the configuration is considered power efficient.

active of type boolean

Indicates whether this RTP stream is configured to be sent or disabled. Note that an active stream can still not be sending, e.g. when being limited by network conditions.

scalabilityMode of type DOMString

The currently configured scalability mode of this RTP stream. If one is not configured or not applicable, this metric is left undefined.

RTCQualityLimitationReason enum

enum RTCQualityLimitationReason {
            "none",
            "cpu",
            "bandwidth",
            "other",
          };
{{RTCQualityLimitationReason}} Enumeration description
Enum valueDescription
none

The resolution and/or framerate is not limited.

cpu

The resolution and/or framerate is primarily limited due to CPU load.

bandwidth

The resolution and/or framerate is primarily limited due to congestion cues during bandwidth estimation. Typical, congestion control algorithms use inter-arrival time, round-trip time, packet or other congestion cues to perform bandwidth estimation.

other

The resolution and/or framerate is primarily limited for a reason other than the above.

{{RTCRemoteOutboundRtpStreamStats}} dictionary

The RTCRemoteOutboundRtpStreamStats dictionary represents the remote endpoint's measurement metrics for its outgoing RTP stream (corresponding to an outgoing RTP stream at the sending endpoint). The timestamp reported in the statistics object is the time at which the corresponding RTCP SR was received.

dictionary RTCRemoteOutboundRtpStreamStats : RTCSentRtpStreamStats {
             DOMString           localId;
             DOMHighResTimeStamp remoteTimestamp;
             unsigned long long  reportsSent;
             double              roundTripTime;
             double              totalRoundTripTime;
             unsigned long long  roundTripTimeMeasurements;
};

Dictionary {{RTCRemoteOutboundRtpStreamStats}} Members

localId of type DOMString

The {{localId}} is used for looking up the local {{RTCInboundRtpStreamStats}} object for the same SSRC.

remoteTimestamp of type DOMHighResTimeStamp

{{remoteTimestamp}}, of type {{DOMHighResTimeStamp}} [[!HIGHRES-TIME]], represents the remote timestamp at which these statistics were sent by the remote endpoint. This differs from {{RTCStats/timestamp}}, which represents the time at which the statistics were generated or received by the local endpoint. The {{remoteTimestamp}}, if present, is derived from the NTP timestamp in an RTCP Sender Report (SR) block, which reflects the remote endpoint's clock. That clock may not be synchronized with the local clock.

reportsSent of type unsigned long long

Represents the total number of RTCP SR blocks sent for this SSRC.

roundTripTime of type double

Estimated round trip time for this SSRC based on the latest RTCP Sender Report (SR) that contains a DLRR report block as defined in [[!RFC3611]]. The Calculation of the round trip time is defined in section 4.5. of [[!RFC3611]]. If the latest SR does not contain the DLRR report block, or if the last RR timestamp in the DLRR report block is zero, or if the delay since last RR value in the DLRR report block is zero, the round trip time is left undefined.

totalRoundTripTime of type double

Represents the cumulative sum of all round trip time measurements in seconds since the beginning of the session. The individual round trip time is calculated based on the DLRR report block in the RTCP Sender Report (SR) [[!RFC3611]], hence undefined roundtrip times are not added. The average round trip time can be computed from {{totalRoundTripTime}} by dividing it by {{roundTripTimeMeasurements}}.

roundTripTimeMeasurements of type unsigned long long

Represents the total number of RTCP Sender Report (SR) blocks received for this SSRC that can derive a valid round trip time according to [[!RFC3611]]. This counter will not increment if the {{roundTripTime}} is undefined.

{{RTCMediaSourceStats}} dictionary

The RTCMediaSourceStats dictionary represents a track that is currently attached to one or more senders. It contains information about media sources such as frame rate and resolution prior to encoding. This is the media passed from the {{MediaStreamTrack}} to the {{RTCRtpSender}}s. This is in contrast to {{RTCOutboundRtpStreamStats}} whose members describe metrics as measured after the encoding step. For example, a track may be captured from a high-resolution camera, its frames downscaled due to track constraints and then further downscaled by the encoders due to CPU and network conditions. This dictionary reflects the video frames or audio samples passed out from the track - after track constraints have been applied but before any encoding or further donwsampling occurs.

Media source objects are of either subdictionary {{RTCAudioSourceStats}} or {{RTCVideoSourceStats}}. The {{RTCStats/type}} is the same ({{RTCStatsType/"media-source"}}) but {{RTCMediaSourceStats/kind}} is different ("audio" or "video") depending on the kind of track.

The media source stats objects are created when a track is attached to any {{RTCRtpSender}} and may subsequently be attached to multiple senders during its life. The life of this object ends when the track is no longer attached to any sender of the same {{RTCPeerConnection}}. If a track whose media source object ended is attached again this results in a new media source stats object whose counters (such as number of frames) are reset.

dictionary RTCMediaSourceStats : RTCStats {
             required DOMString       trackIdentifier;
             required DOMString       kind;
};

Dictionary {{RTCMediaSourceStats}} Members

trackIdentifier of type DOMString

The value of the {{MediaStreamTrack}}'s id attribute.

kind of type DOMString

The value of the {{MediaStreamTrack}}'s kind attribute. This is either "audio" or "video". If it is "audio" then this stats object is of type {{RTCAudioSourceStats}}. If it is "video" then this stats object is of type {{RTCVideoSourceStats}}.

{{RTCAudioSourceStats}} dictionary

The RTCAudioSourceStats dictionary represents an audio track that is attached to one or more senders. It is an {{RTCMediaSourceStats}} whose {{RTCMediaSourceStats/kind}} is "audio".

dictionary RTCAudioSourceStats : RTCMediaSourceStats {
              double              audioLevel;
              double              totalAudioEnergy;
              double              totalSamplesDuration;
              double              echoReturnLoss;
              double              echoReturnLossEnhancement;
              double              droppedSamplesDuration;
              unsigned long       droppedSamplesEvents;
              double              totalCaptureDelay;
              unsigned long long  totalSamplesCaptured;
};

Dictionary {{RTCAudioSourceStats}} Members

audioLevel of type double

Represents the audio level of the media source. For audio levels of remotely sourced tracks, see {{RTCInboundRtpStreamStats}} instead.

The value is between 0..1 (linear), where 1.0 represents 0 dBov, 0 represents silence, and 0.5 represents approximately 6 dBSPL change in the sound pressure level from 0 dBov.

The {{audioLevel}} is averaged over some small interval, using the algorithm described under {{totalAudioEnergy}}. The interval used is implementation dependent.

totalAudioEnergy of type double

Represents the audio energy of the media source. For audio energy of remotely sourced tracks, see {{RTCInboundRtpStreamStats}} instead.

This value MUST be computed as follows: for each audio sample produced by the media source during the lifetime of this stats object, add the sample's value divided by the highest-intensity encodable value, squared and then multiplied by the duration of the sample in seconds. In other words, duration * Math.pow(energy/maxEnergy, 2).

This can be used to obtain a root mean square (RMS) value that uses the same units as {{audioLevel}}, as defined in [[RFC6464]]. It can be converted to these units using the formula Math.sqrt(totalAudioEnergy/totalSamplesDuration). This calculation can also be performed using the differences between the values of two different {{RTCPeerConnection/getStats()}} calls, in order to compute the average audio level over any desired time interval. In other words, do Math.sqrt((energy2 - energy1)/(duration2 - duration1)).

For example, if a 10ms packet of audio is produced with an RMS of 0.5 (out of 1.0), this should add 0.5 * 0.5 * 0.01 = 0.0025 to {{totalAudioEnergy}}. If another 10ms packet with an RMS of 0.1 is received, this should similarly add 0.0001 to {{totalAudioEnergy}}. Then, Math.sqrt(totalAudioEnergy/totalSamplesDuration) becomes Math.sqrt(0.0026/0.02) = 0.36, which is the same value that would be obtained by doing an RMS calculation over the contiguous 20ms segment of audio.

If multiple audio channels are used, the audio energy of a sample refers to the highest energy of any channel.

totalSamplesDuration of type double

Represents the audio duration of the media source. For audio durations of remotely sourced tracks, see {{RTCInboundRtpStreamStats}} instead.

Represents the total duration in seconds of all samples that have been produced by this source for the lifetime of this stats object. Can be used with {{totalAudioEnergy}} to compute an average audio level over different intervals.

{{totalSamplesDuration}} does not include samples dropped before reaching this media source, see {{droppedSamplesDuration}}.

echoReturnLoss of type double

Only [= map/exist =]s when the {{MediaStreamTrack}} is sourced from a microphone where echo cancellation is applied. Calculated in decibels, as defined in [[!ECHO]] (2012) section 3.14.

If multiple audio channels are used, the channel of the least audio energy is considered for any sample.

echoReturnLossEnhancement of type double

Only [= map/exist =]s when the {{MediaStreamTrack}} is sourced from a microphone where echo cancellation is applied. Calculated in decibels, as defined in [[!ECHO]] (2012) section 3.15.

If multiple audio channels are used, the channel of the least audio energy is considered for any sample.

droppedSamplesDuration of type double

Only applicable if this media source is backed by an audio capture device. The total duration, in seconds, of samples produced by the device that got dropped before reaching the media source.

droppedSamplesEvents of type unsigned long

The number of dropped samples events. This counter increases every time a sample is dropped after a non-dropped sample. That is, multiple consecutive dropped samples will increase {{droppedSamplesDuration}} multiple times but is a single dropped samples event.

totalCaptureDelay of type double

Only applicable if the audio source represents an audio capture device. This is the total delay, in seconds, for each audio sample between the time the sample was emitted by the capture device and the sample reaching the source. This can be used together with {{totalSamplesCaptured}} to calculate the average capture delay per sample.

totalSamplesCaptured of type unsigned long long

Only applicable if the audio source represents an audio capture device. This is the total number of captured samples reaching the audio source, i.e. that were not dropped by the capture pipeline. The frequency of the media source is not necessarily the same as the frequency of encoders later in the pipeline.

{{RTCVideoSourceStats}} dictionary

The RTCVideoSourceStats dictionary represents a video track that is attached to one or more senders. It is an {{RTCMediaSourceStats}} whose {{RTCMediaSourceStats/kind}} is "video".

dictionary RTCVideoSourceStats : RTCMediaSourceStats {
             unsigned long   width;
             unsigned long   height;
             unsigned long   frames;
             double          framesPerSecond;
};

Dictionary {{RTCVideoSourceStats}} Members

width of type unsigned long

The width, in pixels, of the last frame originating from this source. Before a frame has been produced this member does not [= map/exist =].

height of type unsigned long

The height, in pixels, of the last frame originating from this source. Before a frame has been produced this member does not [= map/exist =].

frames of type unsigned long

The total number of frames originating from this source.

framesPerSecond of type double

The number of frames originating from this source, measured during the last second. For the first second of this object's lifetime this member does not [= map/exist =].

{{RTCAudioPlayoutStats}} dictionary

Only applicable if the playout path represents an audio device. Represents one playout path - if the same playout stats object is referenced by multiple {{RTCInboundRtpStreamStats}} this is an indication that audio mixing is happening in which case sample counters in this stats object refer to the samples after mixing.

dictionary RTCAudioPlayoutStats : RTCStats {
             double        synthesizedSamplesDuration;
             unsigned long synthesizedSamplesEvents;
             double        totalSamplesDuration;
             double        totalPlayoutDelay;
             double        totalSamplesCount;
};

Dictionary {{RTCAudioPlayoutStats}} Members

synthesizedSamplesDuration of type double

If the playout path is unable to produce audio samples on time for device playout, samples are synthesized to be playout out instead. {{synthesizedSamplesDuration}} is measured in seconds and is incremented each time an audio sample is synthesized by this playout path. This metric can be used together with {{totalSamplesDuration}} to calculate the percentage of played out media being synthesized.

Synthesization typically only happens if the pipeline is underperforming. Samples synthesized by the {{RTCInboundRtpStreamStats}} are not counted for here, but in {{RTCInboundRtpStreamStats}}.{{RTCInboundRtpStreamStats/concealedSamples}}.

synthesizedSamplesEvents of type unsigned long

The number of synthesized samples events. This counter increases every time a sample is synthesized after a non-synthesized sample. That is, multiple consecutive synthesized samples will increase {{synthesizedSamplesDuration}} multiple times but is a single synthesization samples event.

totalSamplesDuration of type double

The total duration, in seconds, of all audio samples that have been playout. Includes both synthesized and non-synthesized samples.

totalPlayoutDelay of type double

When audio samples are pulled by the playout device, this counter is incremented with the estimated delay of the playout path for that audio sample. The playout delay includes the delay from being emitted to the actual time of playout on the device. This metric can be used together with {{totalSamplesCount}} to calculate the average playout delay per sample.

totalSamplesCount of type double

When audio samples are pulled by the playout device, this counter is incremented with the number of samples emitted for playout.

RTCPeerConnectionStats dictionary

dictionary RTCPeerConnectionStats : RTCStats {
            unsigned long dataChannelsOpened;
            unsigned long dataChannelsClosed;
};

Dictionary {{RTCPeerConnectionStats}} Members

dataChannelsOpened of type unsigned long

Represents the number of unique {{RTCDataChannel}}s that have entered the {{RTCDataChannelState/"open"}} state during their lifetime.

dataChannelsClosed of type unsigned long

Represents the number of unique {{RTCDataChannel}}s that have left the {{RTCDataChannelState/"open"}} state during their lifetime (due to being closed by either end or the underlying transport being closed). {{RTCDataChannel}}s that transition from {{RTCDataChannelState/"connecting"}} to {{RTCDataChannelState/"closing"}} or {{RTCDataChannelState/"closed"}} without ever being {{RTCDataChannelState/"open"}} are not counted in this number.

The total number of open data channels at any time can be calculated as {{RTCPeerConnectionStats/dataChannelsOpened}} - {{RTCPeerConnectionStats/dataChannelsClosed}}. This number is always positive.

RTCDataChannelStats dictionary

dictionary RTCDataChannelStats : RTCStats {
             DOMString           label;
             DOMString           protocol;
             unsigned short      dataChannelIdentifier;
             required RTCDataChannelState state;
             unsigned long       messagesSent;
             unsigned long long  bytesSent;
             unsigned long       messagesReceived;
             unsigned long long  bytesReceived;
};

Dictionary {{RTCDataChannelStats}} Members

label of type DOMString
The {{RTCDataChannel/label}} value of the {{RTCDataChannel}} object.
protocol of type DOMString
The {{RTCDataChannel/protocol}} value of the {{RTCDataChannel}} object.
dataChannelIdentifier of type unsigned short

The {{RTCDataChannel/id}} attribute of the {{RTCDataChannel}} object.

state of type {{RTCDataChannelState}}
The {{RTCDataChannel/readyState}} value of the {{RTCDataChannel}} object.
messagesSent of type unsigned long

Represents the total number of API "message" events sent.

bytesSent of type unsigned long long

Represents the total number of payload bytes sent on this {{RTCDataChannel}}, i.e., not including headers or padding.

messagesReceived of type unsigned long

Represents the total number of API "message" events received.

bytesReceived of type unsigned long long

Represents the total number of bytes received on this {{RTCDataChannel}}, i.e., not including headers or padding.

RTCTransportStats dictionary

An {{RTCTransportStats}} object represents the stats corresponding to an {{RTCDtlsTransport}} and its underlying {{RTCIceTransport}}. When bundling is used, a single transport will be used for all {{MediaStreamTrack}}s in the bundle group. If bundling is not used, different {{MediaStreamTrack}} will use different transports. Bundling is described in [[!WEBRTC]].

dictionary RTCTransportStats : RTCStats {
             unsigned long long    packetsSent;
             unsigned long long    packetsReceived;
             unsigned long long    bytesSent;
             unsigned long long    bytesReceived;
             RTCIceRole            iceRole;
             DOMString             iceLocalUsernameFragment;
             required RTCDtlsTransportState dtlsState;
             RTCIceTransportState  iceState;
             DOMString             selectedCandidatePairId;
             DOMString             localCertificateId;
             DOMString             remoteCertificateId;
             DOMString             tlsVersion;
             DOMString             dtlsCipher;
             RTCDtlsRole           dtlsRole;
             DOMString             srtpCipher;
             unsigned long         selectedCandidatePairChanges;
};

Dictionary {{RTCTransportStats}} Members

packetsSent of type unsigned long long

Represents the total number of packets sent over this transport.

packetsReceived of type unsigned long long

Represents the total number of packets received on this transport.

bytesSent of type unsigned long long

Represents the total number of payload bytes sent on this {{RTCIceTransport}}, i.e., not including headers, padding or ICE connectivity checks.

bytesReceived of type unsigned long long

Represents the total number of payload bytes received on this {{RTCIceTransport}}, i.e., not including headers, padding or ICE connectivity checks.

iceRole of type {{RTCIceRole}}

Set to the current value of the {{RTCIceTransport/role}} attribute of the underlying {{RTCDtlsTransport}}.{{RTCDtlsTransport/iceTransport}}.

iceLocalUsernameFragment of type DOMString

Set to the current value of the local username fragment used in message validation procedures [[RFC5245]] for this {{RTCIceTransport}}. It may be updated on setLocalDescription() and on ICE restart.

dtlsState of type {{RTCDtlsTransportState}}

Set to the current value of the {{RTCDtlsTransport/state}} attribute of the underlying {{RTCDtlsTransport}}.

iceState of type {{RTCIceTransportState}}

Set to the current value of the {{RTCIceTransport/state}} attribute of the underlying {{RTCIceTransport}}.

selectedCandidatePairId of type DOMString

It is a unique identifier that is associated to the object that was inspected to produce the {{RTCIceCandidatePairStats}} associated with this transport.

localCertificateId of type DOMString

For components where DTLS is negotiated, give local certificate.

remoteCertificateId of type DOMString

For components where DTLS is negotiated, give remote certificate.

tlsVersion of type DOMString

For components where DTLS is negotiated, the TLS version agreed. Only [= map/exist =]s after DTLS negotiation is complete.

The value comes from ServerHello.supported_versions if present, otherwise from ServerHello.version. It is represented as four upper case hexadecimal digits, representing the two bytes of the version.

dtlsCipher of type DOMString

Descriptive name of the cipher suite used for the DTLS transport, as defined in the "Description" column of the IANA cipher suite registry [[!IANA-TLS-CIPHERS]].

dtlsRole of type RTCDtlsRole

{{RTCDtlsRole/"client"}} or {{RTCDtlsRole/"server"}} depending on the DTLS role. {{RTCDtlsRole/"unknown"}} before the DTLS negotiation starts.

srtpCipher of type DOMString

Descriptive name of the protection profile used for the SRTP transport, as defined in the "Profile" column of the IANA DTLS-SRTP protection profile registry [[!IANA-DTLS-SRTP]] and described further in [[RFC5764]].

selectedCandidatePairChanges of type unsigned long

The number of times that the selected candidate pair of this transport has changed. Going from not having a selected candidate pair to having a selected candidate pair, or the other way around, also increases this counter. It is initially zero and becomes one when an initial candidate pair is selected.

RTCDtlsRole enum

enum RTCDtlsRole {
      "client",
      "server",
      "unknown",
};
{{RTCDtlsRole}} Enumeration description
Enum valueDescription
client

The RTCPeerConnection is acting as a DTLS client as defined in [[RFC6347]].

server

The RTCPeerConnection is acting as a DTLS server as defined in [[RFC6347]].

unknown

The DTLS role of the RTCPeerConnection has not been determined yet.

RTCIceCandidateStats dictionary

{{RTCIceCandidateStats}} reflects the properties of a candidate in Section 15.1 of [[!RFC5245]]. It corresponds to a {{RTCIceCandidate}} object.

dictionary RTCIceCandidateStats : RTCStats {
             required DOMString       transportId;
             DOMString?               address;
             long                     port;
             DOMString                protocol;
             required RTCIceCandidateType candidateType;
             long                     priority;
             DOMString                url;
             RTCIceServerTransportProtocol relayProtocol;
             DOMString                foundation;
             DOMString                relatedAddress;
             long                     relatedPort;
             DOMString                usernameFragment;
             RTCIceTcpCandidateType   tcpType;
};

Dictionary {{RTCIceCandidateStats}} Members

transportId of type DOMString

It is a unique identifier that is associated to the object that was inspected to produce the {{RTCTransportStats}} associated with this candidate.

address of type DOMString

It is the address of the candidate, allowing for IPv4 addresses, IPv6 addresses, and fully qualified domain names (FQDNs). See [[!RFC5245]] section 15.1 for details.

The user agent should make sure that only remote candidate addresses that the web application has configured on the corresponding {{RTCPeerConnection}} are exposed; This is especially important for peer reflexive remote candidates. By default, the user agent MUST leave the {{RTCIceCandidateStats/address}} member as null in the {{RTCIceCandidateStats}} dictionary of any remote candidate. Once a {{RTCPeerConnection}} instance learns on an address by the web application using {{RTCPeerConnection/addIceCandidate()}}, the user agent can expose the 'address' member value in any remote {{RTCIceCandidateStats}} dictionary of the corresponding {{RTCPeerConnection}} that matches the newly learnt address.

port of type long

It is the port number of the candidate.

protocol of type DOMString

Valid values for transport is one of "udp" and "tcp". Based on the "transport" defined in [[!RFC5245]] section 15.1.

candidateType of type {{RTCIceCandidateType}}

This enumeration is defined in [[WEBRTC]].

priority of type long

Calculated as defined in [[!RFC5245]] section 15.1.

url of type DOMString

For local candidates this is the URL of the ICE server from which the candidate was obtained. It is the same as the {{RTCPeerConnectionIceEvent/url}} surfaced in the {{RTCPeerConnectionIceEvent}}.

For remote candidates, this property is not present.

relayProtocol of type {{RTCIceServerTransportProtocol}}

It is the protocol used by the endpoint to communicate with the TURN server. This is only present for local relay candidates and defined in [[WEBRTC]].

foundation of type DOMString

The ICE foundation as defined in [[!RFC5245]] section 15.1.

relatedAddress of type DOMString

The ICE rel-addr defined in [[!RFC5245]] section 15.1. Only set for serverreflexive and relay candidates.

relatedPort of type long

The ICE rel-addr defined in [[!RFC5245]] section 15.1. Only set for serverreflexive and relay candidates.

usernameFragment of type DOMString

The ICE username fragment as defined in [[!RFC5245]] section 7.1.2.3.

tcpType of type {{RTCIceTcpCandidateType}}

The ICE candidate TCP type, as defined іn {{RTCIceTcpCandidateType}} and used in {{RTCIceCandidate}}.

RTCIceCandidatePairStats dictionary

dictionary RTCIceCandidatePairStats : RTCStats {
             required DOMString            transportId;
             required DOMString            localCandidateId;
             required DOMString            remoteCandidateId;
             required RTCStatsIceCandidatePairState state;
             boolean                       nominated;
             unsigned long long            packetsSent;
             unsigned long long            packetsReceived;
             unsigned long long            bytesSent;
             unsigned long long            bytesReceived;
             DOMHighResTimeStamp           lastPacketSentTimestamp;
             DOMHighResTimeStamp           lastPacketReceivedTimestamp;
             double                        totalRoundTripTime;
             double                        currentRoundTripTime;
             double                        availableOutgoingBitrate;
             double                        availableIncomingBitrate;
             unsigned long long            requestsReceived;
             unsigned long long            requestsSent;
             unsigned long long            responsesReceived;
             unsigned long long            responsesSent;
             unsigned long long            consentRequestsSent;
             unsigned long                 packetsDiscardedOnSend;
             unsigned long long            bytesDiscardedOnSend;
};

Dictionary {{RTCIceCandidatePairStats}} Members

transportId of type DOMString

It is a unique identifier that is associated to the object that was inspected to produce the {{RTCTransportStats}} associated with this candidate pair.

localCandidateId of type DOMString

It is a unique identifier that is associated to the object that was inspected to produce the {{RTCIceCandidateStats}} for the local candidate associated with this candidate pair.

remoteCandidateId of type DOMString

It is a unique identifier that is associated to the object that was inspected to produce the {{RTCIceCandidateStats}} for the remote candidate associated with this candidate pair.

state of type {{RTCStatsIceCandidatePairState}}

Represents the state of the checklist for the local and remote candidates in a pair.

nominated of type boolean

Related to updating the nominated flag described in Section 7.1.3.2.4 of [[!RFC5245]].

packetsSent of type unsigned long long

Represents the total number of packets sent on this candidate pair.

packetsReceived of type unsigned long long

Represents the total number of packets received on this candidate pair.

bytesSent of type unsigned long long

Represents the total number of payload bytes sent on this candidate pair, i.e., not including headers, padding or ICE connectivity checks.

bytesReceived of type unsigned long long

Represents the total number of payload bytes received on this candidate pair, i.e., not including headers, padding or ICE connectivity checks.

lastPacketSentTimestamp of type DOMHighResTimeStamp

Represents the timestamp at which the last packet was sent on this particular candidate pair, excluding STUN packets.

lastPacketReceivedTimestamp of type DOMHighResTimeStamp

Represents the timestamp at which the last packet was received on this particular candidate pair, excluding STUN packets.

totalRoundTripTime of type double

Represents the sum of all round trip time measurements in seconds since the beginning of the session, based on STUN connectivity check [[!STUN-PATH-CHAR]] responses (responsesReceived), including those that reply to requests that are sent in order to verify consent [[!RFC7675]]. The average round trip time can be computed from {{totalRoundTripTime}} by dividing it by {{responsesReceived}}.

currentRoundTripTime of type double

Represents the latest round trip time measured in seconds, computed from both STUN connectivity checks [[!STUN-PATH-CHAR]], including those that are sent for consent verification [[!RFC7675]].

availableOutgoingBitrate of type double

It is calculated by the underlying congestion control by combining the available bitrate for all the outgoing RTP streams using this candidate pair. The bitrate measurement does not count the size of the IP or other transport layers like TCP or UDP. It is similar to the TIAS defined in [[!RFC3890]], i.e., it is measured in bits per second and the bitrate is calculated over a 1 second window.

Implementations that do not calculate a sender-side estimate MUST leave this undefined. Additionally, the value MUST be undefined for candidate pairs that were never used. For pairs in use, the estimate is normally no lower than the bitrate for the packets sent at {{lastPacketSentTimestamp}}, but might be higher. For candidate pairs that are not currently in use but were used before, implementations MUST return undefined.

availableIncomingBitrate of type double

It is calculated by the underlying congestion control by combining the available bitrate for all the incoming RTP streams using this candidate pair. The bitrate measurement does not count the size of the IP or other transport layers like TCP or UDP. It is similar to the TIAS defined in [[!RFC3890]], i.e., it is measured in bits per second and the bitrate is calculated over a 1 second window.

Implementations that do not calculate a receiver-side estimate MUST leave this undefined. Additionally, the value should be undefined for candidate pairs that were never used. For pairs in use, the estimate is normally no lower than the bitrate for the packets received at {{lastPacketReceivedTimestamp}}, but might be higher. For candidate pairs that are not currently in use but were used before, implementations MUST return undefined.

requestsReceived of type unsigned long long

Represents the total number of connectivity check requests received (including retransmissions). It is impossible for the receiver to tell whether the request was sent in order to check connectivity or check consent, so all connectivity checks requests are counted here.

requestsSent of type unsigned long long

Represents the total number of connectivity check requests sent (not including retransmissions).

responsesReceived of type unsigned long long

Represents the total number of connectivity check responses received.

responsesSent of type unsigned long long

Represents the total number of connectivity check responses sent. Since we cannot distinguish connectivity check requests and consent requests, all responses are counted.

consentRequestsSent of type unsigned long long

Represents the total number of consent requests sent.

packetsDiscardedOnSend of type unsigned long

Total number of packets for this candidate pair that have been discarded due to socket errors, i.e. a socket error occurred when handing the packets to the socket. This might happen due to various reasons, including full buffer or no available memory.

bytesDiscardedOnSend of type unsigned long long

Total number of bytes for this candidate pair that have been discarded due to socket errors, i.e. a socket error occurred when handing the packets containing the bytes to the socket. This might happen due to various reasons, including full buffer or no available memory. Calculated as defined in [[!RFC3550]] section 6.4.1.

RTCStatsIceCandidatePairState enum

enum RTCStatsIceCandidatePairState {
    "frozen",
    "waiting",
    "in-progress",
    "failed",
    "succeeded"
};
{{RTCStatsIceCandidatePairState}} Enumeration description
Enum valueDescription
frozen

Defined in Section 5.7.4 of [[!RFC5245]].

waiting

Defined in Section 5.7.4 of [[!RFC5245]].

in-progress

Defined in Section 5.7.4 of [[!RFC5245]].

failed

Defined in Section 5.7.4 of [[!RFC5245]].

succeeded

Defined in Section 5.7.4 of [[!RFC5245]].

RTCCertificateStats dictionary

dictionary RTCCertificateStats : RTCStats {
             required DOMString fingerprint;
             required DOMString fingerprintAlgorithm;
             required DOMString base64Certificate;
             DOMString issuerCertificateId;
};

Dictionary {{RTCCertificateStats}} Members

fingerprint of type DOMString

The fingerprint of the certificate. Only use the fingerprint value as defined in Section 5 of [[!RFC4572]].

fingerprintAlgorithm of type DOMString

The hash function used to compute the certificate fingerprint. For instance, "sha-256".

base64Certificate of type DOMString

The DER-encoded base-64 representation of the certificate.

issuerCertificateId of type DOMString

The {{issuerCertificateId}} refers to the stats object that contains the next certificate in the certificate chain. If the current certificate is at the end of the chain (i.e. a self-signed certificate), this will not be set.

Obsolete stats

Obsolete RTCMediaStreamStats members

dictionary RTCMediaStreamStats : RTCStats {
  DOMString streamIdentifier;
  sequence<DOMString> trackIds;
};

This entire dictionary was made obsolete in September, 2019 due to sender, receiver and transceiver stats objects being a better fit to describe the modern {{RTCPeerConnection}} model (Unified Plan).

streamIdentifier of type DOMString

stream.id property

trackIds of type sequence<DOMString>

This is the {{RTCStats/id}} of the stats object, not the track.id.

Obsolete {{RTCCodecStats}} members

partial dictionary RTCCodecStats {
    DOMString implementation;
};
          
implementation of type DOMString

This was moved to {{RTCInboundRtpStreamStats}}.{{RTCInboundRtpStreamStats}}/decoderImplementation}} and {{RTCOutboundRtpStreamStats}}.{{RTCOutboundRtpStreamStats/encoderImplementation}} in August 2019.

Obsolete {{RTCIceCandidateStats}} members

partial dictionary RTCIceCandidateStats {
    boolean deleted = false;
    boolean isRemote;
};
        
deleted of type boolean

This field was obsoleted because if the ICE candidate is deleted it no longer appears in {{RTCPeerConnection/getStats()}}.

isRemote of type boolean

false indicates that this represents a local candidate; true indicates that this represents a remote candidate.

Obsolete {{RTCIceCandidatePairStats}} members

partial dictionary RTCIceCandidatePairStats {
    double totalRtt;
    double currentRtt;
    unsigned long long priority;
};
        
totalRtt

This field got renamed to {{totalRoundTripTime}} in Dec 2016.

currentRtt

This field got renamed to {{currentRoundTripTime}} in Dec 2016.

priority

This field got removed in Feb 2018, as it cannot be represented in 53 bits. It can be recalculated if needed as defined in [[RFC5245]] section 5.7.2.

Obsolete {{RTCRtpStreamStats}} members

partial dictionary RTCRtpStreamStats {
    DOMString mediaType;
};
mediaType of type DOMString

This field got renamed to {{kind}} in Feb 2018.

Obsolete {{RTCInboundRtpStreamStats}} members

partial dictionary RTCInboundRtpStreamStats {
    DOMString trackId;
    double fractionLost;
};
trackId of type DOMString

The identifier of the stats object representing the receiving track, an {{RTCMediaStreamTrackStats}}.

This field was made obsolete in April 2020 as a follow-up to "track" stats having been made obsolete.

fractionLost of type double

This field was moved to {{RTCRemoteInboundRtpStreamStats}} in December 2017.

Obsolete {{RTCOutboundRtpStreamStats}} members

partial dictionary RTCOutboundRtpStreamStats {
    DOMString trackId;
};
trackId of type DOMString

The identifier of the stats object representing the current track attachment to the sender of this stream, an {{RTCMediaStreamTrackStats}}.

This field was made obsolete in April 2020 as a follow-up to "track" stats having been made obsolete.

Obsolete {{RTCMediaStreamTrackStats}} members

dictionary RTCMediaStreamTrackStats {
    DOMString           trackIdentifier;
    boolean             ended;
    required DOMString  kind;
    RTCPriorityType     priority;
    boolean             remoteSource;
    DOMHighResTimeStamp estimatedPlayoutTimestamp;
    double              jitterBufferDelay;
    unsigned long long  jitterBufferEmittedCount;

    // Audio-only, send and receive side.
    double              audioLevel;
    double              totalAudioEnergy;
    double              totalSamplesDuration;

    // Audio-only, send side.
    double              echoReturnLoss;
    double              echoReturnLossEnhancement;

    // Audio-only, receive side.
    unsigned long long  totalSamplesReceived;
    unsigned long long  concealedSamples;
    unsigned long long  silentConcealedSamples;
    unsigned long long  concealmentEvents;
    unsigned long long  insertedSamplesForDeceleration;
    unsigned long long  removedSamplesForAcceleration;

    // Video-only, send and receive side.
    unsigned long       frameWidth;
    unsigned long       frameHeight;
    double              framesPerSecond;

    // Video-only, send side.
    unsigned long       keyFramesSent;
    unsigned long       framesCaptured;
    unsigned long       framesSent;
    unsigned long       hugeFramesSent;

    // Video-only, receive side.
    unsigned long       keyFramesReceived;
    unsigned long       framesReceived;
    unsigned long       framesDecoded;
    unsigned long       framesDropped;
};
        
trackIdentifier of type DOMString

Represents the id property of the track.

ended of type boolean

Reflects the ended state of the track.

kind of type DOMString

Either "audio" or "video". This reflects the kind attribute of the {{MediaStreamTrack}}, see [[!GETUSERMEDIA]].

priority of type {{RTCPriorityType}}

Indicates the priority that has been set for the track. It is defined in [[RTCWEB-TRANSPORT]], Section 4, but was removed from [[WEBRTC]] due to lack of implementation and moved to [[WEBRTC-PRIORITY]], with some changes to its definition.

remoteSource of type boolean

Only applicable for {{RTCStatsType/"track"}} stats. false if the track attachment is on an {{RTCRtpSender}}, true if the track attachment is on an {{RTCRtpReceiver}}.

With "track" stats made obsolete, and this information being available elsewhere, this metric was made obsolete in April, 2020.

estimatedPlayoutTimestamp of type DOMHighResTimeStamp

This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

jitterBufferDelay of type double

This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

jitterBufferEmittedCount of type unsigned long long

This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

audioLevel of type double

Audio-only. This field was moved to {{RTCAudioSourceStats}} and {{RTCInboundRtpStreamStats}} in June 2019.

totalAudioEnergy of type double

Audio-only. This field was moved to {{RTCAudioSourceStats}} and {{RTCInboundRtpStreamStats}} in June 2019.

totalSamplesDuration of type double

Audio-only. This field was moved to {{RTCAudioSourceStats}} and {{RTCInboundRtpStreamStats}} in June 2019.

echoReturnLoss of type double

Audio-only. This was moved to {{RTCAudioSourceStats}} in August 2019.

echoReturnLossEnhancement of type double

Audio-only. This was moved to {{RTCAudioSourceStats}} in August 2019.

totalSamplesReceived of type unsigned long long

Audio-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

concealedSamples of type unsigned long long

Audio-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

silentConcealedSamples of type unsigned long long

Audio-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

concealmentEvents of type unsigned long long

Audio-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

insertedSamplesForDeceleration of type unsigned long long

Audio-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

removedSamplesForAcceleration of type unsigned long long

Audio-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

frameWidth of type unsigned long

Video-only. This was moved to {{RTCOutboundRtpStreamStats}} and {{RTCInboundRtpStreamStats}} in August 2019.

frameHeight of type unsigned long

Video-only. This was moved to {{RTCOutboundRtpStreamStats}} and {{RTCInboundRtpStreamStats}} in August 2019.

framesPerSecond of type double

Video-only. For the sending case, this was replaced by {{RTCVideoSourceStats}}.{{RTCVideoSourceStats/framesPerSecond}} in May 2019 representing the frame rate of the track. For the receiving case, this was moved to {{RTCInboundRtpStreamStats}} in August 2019 representing the decoding frame rate. In August 2019, {{RTCOutboundRtpStreamStats/framesPerSecond}} was also added to {{RTCOutboundRtpStreamStats}}, representing the encoding frame rate (which may be lower than the source frame rate).

keyFramesSent of type unsigned long

Video-only. This field was replaced by {{RTCOutboundRtpStreamStats/keyFramesEncoded}} in {{RTCOutboundRtpStreamStats}} in June 2019. There were no known implementations supporting the old field at the time of the change.

framesCaptured of type unsigned long

Video-only. This was replaced by {{RTCVideoSourceStats}}.frames in May 2019.

framesSent of type unsigned long

Video-only. This was moved to {{RTCOutboundRtpStreamStats}} in August 2019.

hugeFramesSent of type unsigned long

Video-only. This was moved to {{RTCOutboundRtpStreamStats}} in August 2019.

keyFramesReceived of type unsigned long

Video-only. This field was replaced by {{RTCInboundRtpStreamStats/keyFramesDecoded}} in {{RTCInboundRtpStreamStats}} in June 2019. There were no known implementations supporting the old field at the time of the change.

framesReceived of type unsigned long

Video-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

framesDecoded of type unsigned long

Video-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

framesDropped of type unsigned long

Video-only. This was moved to {{RTCInboundRtpStreamStats}} in August 2019.

Examples

Example of a stats application

Consider the case where the user is experiencing bad sound and the application wants to determine if the cause of it is packet loss. The following example code might be used:

var baselineReport, currentReport;
var sender = pc.getSenders()[0];

sender.getStats().then(function (report) {
    baselineReport = report;
})
.then(function() {
    return new Promise(function(resolve) {
        setTimeout(resolve, aBit); // ... wait a bit
    });
})
.then(function() {
    return sender.getStats();
})
.then(function (report) {
    currentReport = report;
    processStats();
})
.catch(function (error) {
  console.log(error.toString());
});

function processStats() {
    // compare the elements from the current report with the baseline
    for (let now of currentReport.values()) {
        if (now.type != "outbound-rtp")
            continue;

        // get the corresponding stats from the baseline report
        let base = baselineReport.get(now.id);

        if (base) {
            remoteNow = currentReport[now.remoteId];
            remoteBase = baselineReport[base.remoteId];

            var packetsSent = now.packetsSent - base.packetsSent;
            var packetsReceived = remoteNow.packetsReceived - remoteBase.packetsReceived;

            // if intervalFractionLoss is > 0.3, we've probably found the culprit
            var intervalFractionLoss = (packetsSent - packetsReceived) / packetsSent;
        }
    });
}

Security and Privacy Considerations

The data exposed by WebRTC Statistics include most of the media and network data also exposed by [[!GETUSERMEDIA]] and [[!WEBRTC]] - as such, all the privacy and security considerations of these specifications related to data exposure apply as well to this specifciation.

In addition, the properties exposed by {{RTCReceivedRtpStreamStats}}, {{RTCRemoteInboundRtpStreamStats}}, {{RTCSentRtpStreamStats}}, {{RTCOutboundRtpStreamStats}}, {{RTCRemoteOutboundRtpStreamStats}}, {{RTCIceCandidatePairStats}}, {{RTCTransportStats}} expose network-layer data not currently available to the JavaScript layer.

Beyond the risks associated with revealing IP addresses as discussed in the WebRTC 1.0 specification, some combination of the network properties uniquely exposed by this specification can be correlated with location.

For instance, the round-trip time exposed in {{RTCRemoteInboundRtpStreamStats}} can give some coarse indication on how far aparts the peers are located, and thus, if one of the peer's location is known, this may reveal information about the other peer.

When applied to isolated streams, media metrics may allow an application to infer some characteristics of the isolated stream, such as if anyone is speaking (by watching the {{RTCAudioSourceStats/audioLevel}} statistic).

The following stats are deemed to be sensitive, and MUST NOT be reported for an isolated media stream:

Summary of WebRTC stats fields per type

{{RTCStatsType}}DictionaryFields
{{RTCStatsType/"codec"}} {{RTCCodecStats}}
{{RTCStatsType/"inbound-rtp"}} {{RTCInboundRtpStreamStats}}
{{RTCStatsType/"outbound-rtp"}} {{RTCOutboundRtpStreamStats}}
{{RTCStatsType/"remote-inbound-rtp"}} {{RTCRemoteInboundRtpStreamStats}}
{{RTCStatsType/"remote-outbound-rtp"}} {{RTCRemoteOutboundRtpStreamStats}}
{{RTCStatsType/"media-source"}} {{RTCAudioSourceStats}} {{RTCVideoSourceStats}}
{{RTCStatsType/"media-playout"}} {{RTCAudioPlayoutStats}}
{{RTCStatsType/"peer-connection"}} {{RTCPeerConnectionStats}}
{{RTCStatsType/"data-channel"}} {{RTCDataChannelStats}}
{{RTCStatsType/"transport"}} {{RTCTransportStats}}
{{RTCStatsType/"candidate-pair"}} {{RTCIceCandidatePairStats}}
{{RTCStatsType/"local-candidate"}} {{RTCIceCandidateStats}}
{{RTCStatsType/"remote-candidate"}} {{RTCIceCandidateStats}}
{{RTCStatsType/"certificate"}} {{RTCCertificateStats}}

Acknowledgements

The editors wish to thank the Working Group chairs, Stefan Håkansson, and the Team Contact, Dominique Hazaël-Massieux, for their support. The editors would like to thank Bernard Aboba, Taylor Brandstetter, Henrik Boström, Jan-Ivar Bruaroey, Karthik Budigere, Cullen Jennings, and Lennart Schulte for their contributions to this specification.