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]].
This document is incomplete, and as such is not yet suitable for implementation. However, early experimentation is encouraged.
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):
The concepts queue a task, and fires a simple event are defined in [[!HTML5]].
The terms event, event handlers, and event handler event types are defined in [[!HTML5]].
The terms MediaStream, MediaStreamTrack, and Consumer are defined in [[!GETUSERMEDIA]].
The terms RTCPeerConnection, RTCDataChannel, RTCDtlsTransport, RTCDtlsTransportState, RTCIceTransport, RTCIceRole, RTCDataChannelState, RTCIceCandidateType and RTCPriorityType are defined in [[!WEBRTC]].
The term RTP stream is defined in [[RFC7656]] section 2.1.10.
The terms
RTCStats,
RTCStats.timestamp,
RTCStats.type,
RTCStats.id,
RTCCertificate and
statsended are defined in
[[!WEBRTC]].
The terms DOMHighResTimeStamp
performance.timeOrigin and
performance.now() are
defined in [[!HIGHRES-TIME]].
The basic object of the stats model is the stats object. The following terms are defined to describe it:
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.
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 attribute names ending in 'Id', or they have type sequence<DOMString> and attribute names ending in 'Ids'.
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 objects, 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 {
DOMHighResTimeStamp timestamp;
RTCStatsType type;
DOMString id;
};
Timestamps are expressed with DOMHighResTimeStamp [[!HIGHRES-TIME]], and are defined as performance.timeOrigin + performance.now() at the time the information is collected.
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 "transportId") are always a stats object reference; names ending in "Ids" (such as "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 "ssrcIdentifier" or "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.
When implementing stats objects, the following guidelines should be adhered to:
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 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 a statsended event is fired on their behalf. Each event that causes deletions to happen MUST fire only one statsended event, but there are cases where one action causes multiple deletion events; for instance, an ICE restart will fire only one event containing the stats for all the discarded candidates and pairs, but will also cause a later event to be fired when the currently-in-use candidate pair and its candidates are discarded.
When a monitored object is deleted, a final stats object is produced, carrying the
values current at the time of deletion. This object will be made available using the
statsended event on the associated RTCPeerConnection. This is
important in order to report consistently on short-lived objects and to be able to
consistently report totals over the lifetime of a RTCPeerConnection.
When an object is deleted, we can guarantee that no subsequent 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 objects for deleted monitored objects will always be able to uniquely identify the object pointed to in the result of any getStats() call.
A call to getStats() touches many components of WebRTC and may take significant time to execute. The implementation may or may not utilize caching or throttling of 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 getStats() calls must return up-to-date dictionaries for the affected objects. For example, if a track is added with addTrack() subsequent getStats() calls must resolve with a corresponding RTCMediaHandlerStats object. If you call setRemoteDescription() removing a remote track, upon the promise resolving or an associated event (stream's onremovetrack or track's onmute) firing, calling getStats() must resolve with an up-to-date RTCMediaHandlerStats object.
When the statsended event is fired, subsequent getStats() calls MUST NOT return stats for the monitored object that was reported on in the statsended event.
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.
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.
The type element, of type RTCStatsType, indicates the type of the
object that the RTCStats object represents. An object with a given "type" can
have only one IDL dictionary type, but multiple "type" values may indicate the same IDL
dictionary type; for example, "local-candidate" and "remote-candidate" both use the IDL
dictionary type RTCIceCandidateStats.
This specification is normative for the allowed values of RTCStatsType.
enum RTCStatsType {
"codec",
"inbound-rtp",
"outbound-rtp",
"remote-inbound-rtp",
"remote-outbound-rtp",
"media-source",
"csrc",
"peer-connection",
"data-channel",
"stream",
"track",
"sender",
"receiver",
"transport",
"candidate-pair",
"local-candidate",
"remote-candidate",
"certificate",
"ice-server"
};
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, such as when using simulcast or RTX, there will be one RTCOutboundRtpStreamStats per RTP stream, with distinct values of the "ssrc" attribute, and all these senders will have a reference to the same "sender" object (of type RTCAudioSenderStats or RTCVideoSenderStats) and "track" object (of type RTCSenderAudioTrackAttachmentStats or RTCSenderVideoTrackAttachmentStats).
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.
csrc
Statistics for a contributing source (CSRC) that contributed to an inbound RTP
stream. It is accessed by the RTCRtpContributingSourceStats.
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
Contains statistics related to a specific MediaStream. It is accessed by the
RTCMediaStreamStats.
track
Statistics related to a specific MediaStreamTrack's attachment to an
RTCRtpSender and the corresponding media-level metrics. It is accessed by
RTCSenderVideoTrackAttachmentStats,
RTCSenderAudioTrackAttachmentStats,
RTCReceiverVideoTrackAttachmentStats or
RTCReceiverAudioTrackAttachmentStats, all
inherited from RTCMediaHandlerStats.
The monitored "track" object is deleted when the sender it reports on has its "track" value changed to no longer refer to the same track.
sender
Statistics related to a specific RTCRtpSender and the corresponding
media-level metrics. It is accessed by the RTCAudioSenderStats or
the RTCVideoSenderStats depending on kind.
receiver
Statistics related to a specific receiver and the corresponding media-level
metrics. It is accessed by the RTCAudioReceiverStats or the
RTCVideoReceiverStats depending on kind.
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 "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.
ice-server
Information about the connection to an ICE server (e.g. STUN or TURN). It is accessed by the
RTCIceServerStats.
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). Wheras, a non-compound RTCP packets may contain just a single RTCP SR or RR block.
The lifetime of all RTP monitored objects 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 not deleted.
The hierarchy is as follows:
RTCRtpStreamStats: Stats that apply to any end of any RTP stream
RTCReceivedRtpStreamStats: Stats measured at the receiving end of an RTP stream, known either because they're measured locally or transmitted via an RTCP Receiver Report (RR) or Extended Report (XR) block.
RTCInboundRtpStreamStats: Stats that can only be measured at the local receiving end of an RTP stream.
RTCRemoteInboundRtpStreamStats: Stats relevant to the remote receiving end of an RTP stream - usually computed by combining local data with data received via an RTCP RR or XR block.
RTCSentRtpStreamStats: Stats measured at the sending end of an RTP stream, known either because they're measured locally or because they're received via RTCP, usually in an RTCP Sender Report (SR).
RTCRemoteOutboundRtpStreamStats: Stats relevant to the remote sending end of an RTP stream, usually computed based on an RTCP SR.
dictionary RTCRtpStreamStats : RTCStats {
unsigned long ssrc;
DOMString kind;
DOMString transportId;
DOMString codecId;
};
ssrc of type unsigned
long
The 32-bit unsigned integer value per [[RFC3550]] used to identify the source of the stream of RTP packets that this stats object concerns.
kind of type DOMString
Either "audio" or "video". This MUST match the media
type part of the information in the corresponding codecType member of RTCCodecStats, and
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.
dictionary RTCCodecStats : RTCStats {
unsigned long payloadType;
RTCCodecType codecType;
DOMString transportId;
DOMString mimeType;
unsigned long clockRate;
unsigned long channels;
DOMString sdpFmtpLine;
};
payloadType of type unsigned
long
Payload type as used in RTP encoding or decoding.
codecType of type RTCCodecType
"encode" or "decode", depending on whether this object
represents a media format that the implementation is prepared to encode or
decode.
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
Use 2 for stereo, missing for most other cases.
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.
enum RTCCodecType {
"encode",
"decode",
};
| Enumeration description | |
|---|---|
encode
|
The attached |
decode
|
The attached |
dictionary RTCReceivedRtpStreamStats : RTCRtpStreamStats {
unsigned long long packetsReceived;
long long packetsLost;
double jitter;
unsigned long long packetsDiscarded;
unsigned long long packetsRepaired;
unsigned long long burstPacketsLost;
unsigned long long burstPacketsDiscarded;
unsigned long burstLossCount;
unsigned long burstDiscardCount;
double burstLossRate;
double burstDiscardRate;
double gapLossRate;
double gapDiscardRate;
unsigned long framesDropped;
unsigned long partialFramesLost;
unsigned long fullFramesLost;
};
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 can be calculated by subtracting the packets lost from the expected Highest Sequence Number reported in the RTCP Sender Report as discussed in Appendix A.3. in [[!RFC3550]].
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]].
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.
packetsRepaired of type unsigned long long
The cumulative number of lost RTP packets repaired after applying an error-resilience mechanism [[XRBLOCK-STATS]]. It is measured for the primary source RTP packets and only counted for RTP packets that have no further chance of repair. To clarify, the value is upper-bound to the cumulative number of lost packets. Calculated as defined in [[!RFC7509]] section 3.1 and Appendix A.b.
burstPacketsLost of type unsigned long long
The cumulative number of RTP packets lost during loss bursts, Appendix A (c) of [[!RFC6958]].
burstPacketsDiscarded of type unsigned long long
The cumulative number of RTP packets discarded during discard bursts, Appendix A (b) of [[!RFC7003]].
burstLossCount of type unsigned
long
The cumulative number of bursts of lost RTP packets, Appendix A (e) of [[!RFC6958]].
[[!RFC3611]] recommends a Gmin (threshold) value of 16 for classifying a sequence of packet losses or discards as a burst.
burstDiscardCount of type unsigned long
The cumulative number of bursts of discarded RTP packets, Appendix A (e) of [[!RFC8015]].
burstLossRate of type double
The fraction of RTP packets lost during bursts to the total number of RTP packets expected in the bursts. As defined in Appendix A (a) of [[!RFC7004]], however, the actual value is reported without multiplying by 32768.
burstDiscardRate of type double
The fraction of RTP packets discarded during bursts to the total number of RTP packets expected in bursts. As defined in Appendix A (e) of [[!RFC7004]], however, the actual value is reported without multiplying by 32768.
gapLossRate of type double
The fraction of RTP packets lost during the gap periods. Appendix A (b) of [[!RFC7004]], however, the actual value is reported without multiplying by 32768.
gapDiscardRate of type double
The fraction of RTP packets discarded during the gap periods. Appendix A (f) of [[!RFC7004]], however, the actual value is reported without multiplying by 32768.
framesDropped of type unsigned
long
Only valid 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]].
partialFramesLost of type unsigned long
Only valid for video. The cumulative number of partial frames lost. The measurement begins when the receiver is created and is a cumulative metric as defined in Appendix A (j) of [[!RFC7004]]. This metric is incremented when the frame is sent to the decoder. If the partial frame is received and recovered via retransmission or FEC before decoding, the framesReceived counter is incremented.
fullFramesLost of type unsigned
long
Only valid for video. The cumulative number of full frames lost. The measurement begins when the receiver is created and is a cumulative metric as defined in Appendix A (i) of [[!RFC7004]].
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 {
DOMString trackId;
DOMString receiverId;
DOMString remoteId;
unsigned long framesDecoded;
unsigned long keyFramesDecoded;
unsigned long frameWidth;
unsigned long frameHeight;
double framesPerSecond;
unsigned long long qpSum;
double totalDecodeTime;
double totalInterFrameDelay;
double totalSquaredInterFrameDelay;
boolean voiceActivityFlag;
DOMHighResTimeStamp lastPacketReceivedTimestamp;
double averageRtcpInterval;
unsigned long long headerBytesReceived;
unsigned long long fecPacketsReceived;
unsigned long long fecPacketsDiscarded;
unsigned long long bytesReceived;
unsigned long long packetsFailedDecryption;
unsigned long long packetsDuplicated;
record<USVString, unsigned long long> perDscpPacketsReceived;
unsigned long nackCount;
unsigned long firCount;
unsigned long pliCount;
unsigned long sliCount;
DOMHighResTimeStamp estimatedPlayoutTimestamp;
double jitterBufferDelay;
unsigned long long jitterBufferEmittedCount;
unsigned long long totalSamplesReceived;
unsigned long long samplesDecodedWithSilk;
unsigned long long samplesDecodedWithCelt;
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;
};
trackId of type DOMString
RTCReceiverAudioTrackAttachmentStats or
RTCReceiverVideoTrackAttachmentStats.
receiverId of type DOMString
The stats ID used to look up the RTCAudioReceiverStats or
RTCVideoReceiverStats object receiving this stream.
remoteId of type DOMString
The remoteId is used for looking up the remote
RTCRemoteOutboundRtpStreamStats object for the same SSRC.
framesDecoded
Only valid 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 valid 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.
frameWidth of type unsigned
long
Only valid for video. Represents the width of the last decoded frame. Before the first frame is decoded this attribute is missing.
frameHeight of type unsigned
long
Only valid for video. Represents the height of the last decoded frame. Before the first frame is decoded this attribute is missing.
framesPerSecond of type double
Only valid for video. The number of decoded frames in the last second.
qpSum of type unsigned long
long
Only valid 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.
voiceActivityFlag of type boolean
Only valid for audio. Whether the last RTP packet whose frame was delivered to the
RTCRtpReceiver's MediaStreamTrack for playout contained voice activity or not based
on the presence of the V bit in the extension header, as defined in [[RFC6464]]. This
is the stats-equivalent of RTCRtpSynchronizationSource.voiceActivityFlag
in [[WEBRTC].
lastPacketReceivedTimestamp of type DOMHighResTimeStamp
Represents the timestamp at which the last packet was received for this SSRC.
This differs from timestamp, which represents the time at which the
statistics were generated by the local endpoint.
averageRtcpInterval of type double
The average RTCP interval between two consecutive compound RTCP packets. This is calculated by the sending endpoint when sending compound RTCP reports. Compound packets must contain at least a RTCP RR or SR block and an SDES packet with the CNAME item.
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.
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.
packetsFailedDecryption of type unsigned long long
The cumulative number of RTP packets that failed to be decrypted according to the
procedures in [[!RFC3711]]. These packets are not counted by
packetsDiscarded.
packetsDuplicated of type unsigned long long
packetsDiscarded.perDscpPacketsReceived of type record<USVString, unsigned long long>
Total number of packets received for this SSRC, per Differentiated Services code point (DSCP) [[RFC2474]]. DSCPs are identified as decimal integers in string form. Note that due to network remapping and bleaching, these numbers are not expected to match the numbers seen on sending. Not all OSes make this information available.
firCount of type unsigned
long
Only valid for video. Count the total number of Full Intra Request (FIR) packets sent by this receiver. Calculated as defined in [[!RFC5104]] section 4.3.1. and does not use the metric indicated in [[RFC2032]], because it was deprecated by [[RFC4587]].
pliCount of type unsigned
long
Only valid for video. Count the total number of Picture Loss Indication (PLI) packets sent by this receiver. Calculated as defined in [[!RFC4585]] section 6.3.1.
nackCount of type unsigned
long
Count the total number of Negative ACKnowledgement (NACK) packets sent by this receiver. Calculated as defined in [[!RFC4585]] section 6.2.1.
sliCount of type unsigned
long
Only valid for video. Count the total number of Slice Loss Indication (SLI) packets sent by this receiver. Calculated as defined in [[!RFC4585]] section 6.3.2.
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
It is the sum of the time, in seconds, each audio sample or video frame takes from
the time it is received and to the time it exits the jitter buffer. This 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.
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).
totalSamplesReceived of type unsigned long long
Only valid for audio. The total number of samples that have been received on this RTP stream. This includes concealedSamples.
samplesDecodedWithSilk of type unsigned long long
Only valid for audio and when the audio codec is Opus. The total number of samples decoded by the SILK portion of the Opus codec.
samplesDecodedWithCelt of type unsigned long long
Only valid for audio and when the audio codec is Opus. The total number of samples decoded by the CELT portion of the Opus codec.
concealedSamples of type unsigned long long
Only valid 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 packetsLost) or samples from packets that arrive too late to be played out (reported in packetsDiscarded).
silentConcealedSamples of type unsigned long long
Only valid 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 valid 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 valid 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 valid 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 valid 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 algortihm described under totalAudioEnergy. The interval used is implementation dependent.
totalAudioEnergy of type double
Only valid 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
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.
totalSamplesDuration of type double
Only valid 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 valid 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
Identifies the decoder implementation used. This is useful for diagnosing interoperability issues.
If too much information is given here, it increases the fingerprint surface. Since it is only given for active tracks, the incremental exposure is small.
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 reportsReceived;
unsigned long long roundTripTimeMeasurements;
};
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 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.
reportsReceived of type unsigned long long
Represents the total number of RTCP RR blocks received for this SSRC.
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 increment if the roundTripTime is undefined.
dictionary RTCSentRtpStreamStats : RTCRtpStreamStats {
unsigned long packetsSent;
unsigned long long bytesSent;
};
packetsSent of type unsigned
long long
Total number of RTP packets sent for this SSRC. Calculated as defined in [[!RFC3550]] section 6.4.1.
bytesSent of type unsigned long
long
Total number of bytes sent for this SSRC. Calculated as defined in [[!RFC3550]] section 6.4.1.
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 trackId;
DOMString mediaSourceId;
DOMString senderId;
DOMString remoteId;
DOMString rid;
DOMHighResTimeStamp lastPacketSentTimestamp;
unsigned long long headerBytesSent;
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
unsigned long fecPacketsSent;
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 framesDiscardedOnSend;
unsigned long long qpSum;
unsigned long long totalSamplesSent;
unsigned long long samplesEncodedWithSilk;
unsigned long long samplesEncodedWithCelt;
boolean voiceActivityFlag;
double totalEncodeTime;
double totalPacketSendDelay;
double averageRtcpInterval;
RTCQualityLimitationReason qualityLimitationReason;
record<DOMString, double> qualityLimitationDurations;
unsigned long qualityLimitationResolutionChanges;
record<USVString, unsigned long long> perDscpPacketsSent;
unsigned long nackCount;
unsigned long firCount;
unsigned long pliCount;
unsigned long sliCount;
DOMString encoderImplementation;
};
trackId of type DOMString
RTCSenderAudioTrackAttachmentStats or
RTCSenderVideoTrackAttachmentStats.
mediaSourceId of type DOMString
RTCMediaSourceStats.
senderId of type DOMString
The stats ID used to look up the RTCAudioSenderStats or
RTCVideoSenderStats object sending this stream.
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 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.
lastPacketSentTimestamp of type DOMHighResTimeStamp
Represents the timestamp at which the last packet was sent for this SSRC. This
differs from timestamp, which represents the time at which the
statistics were generated by the local endpoint.
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.
packetsDiscardedOnSend of type unsigned long
Total number of RTP packets for this SSRC that have been discarded due to socket errors, i.e. a socket error occured 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 SSRC that have been discarded due to socket errors, i.e. a socket error occured 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.
fecPacketsSent of type unsigned
long
Total number of RTP FEC packets sent for this SSRC. This counter can also be incremented when sending FEC packets in-band with media packets (e.g., with Opus).
retransmittedPacketsSent of type unsigned long long
The total number of packets that were retransmitted for this SSRC. This is a
subset of packetsSent.
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 bytesSent.
targetBitrate of type double
It is the current target bitrate configured for this particular SSRC and is the Transport Independent Application Specific (TIAS) bitrate [[!RFC3890]]. Typically, the target bitrate is a configuration parameter provided to the codec's encoder and does not count the size of the IP or other transport layers like TCP or UDP. It is measured in bits per second and the bitrate is calculated over a 1 second window.
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 valid 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 attribute is missing.
frameHeight of type unsigned
long
Only valid 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 attribute is missing.
framesPerSecond of type double
Only valid 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 valid for video. Represents the total number of frames sent on this RTP stream.
hugeFramesSent of type unsigned
long
Only valid 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 long
Only valid for video. It represents the total number of frames successfully encoded for this RTP media stream.
keyFramesEncoded of type unsigned long
Only valid 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.
framesDiscardedOnSend of type unsigned long
Total number of video frames that have been discarded for this SSRC due to socket errors, i.e. a socket error occured when handing the packets to the socket. This might happen due to various reasons, including full buffer or no available memory.
qpSum of type unsigned long
long
Only valid 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.
totalSamplesSent of type unsigned long long
Only valid for audio. The total number of samples that have been sent over this RTP stream.
samplesEncodedWithSilk of type unsigned long long
Only valid for audio and when the audio codec is Opus. The total number of samples encoded by the SILK portion of the Opus codec.
samplesEncodedWithCelt of type unsigned long long
Only valid for audio and when the audio codec is Opus. The total number of samples encoded by the CELT portion of the Opus codec.
voiceActivityFlag of type boolean
Only valid for audio. Whether the last RTP packet sent contained voice activity or not based on the presence of the V bit in the extension header, as defined in [[RFC6464]].
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
packetsSent is incremented.
averageRtcpInterval of type double
The average RTCP interval between two consecutive compound RTCP packets. This is calculated by the sending endpoint when sending compound RTCP reports. Compound packets must contain at least a RTCP RR or SR block and an SDES packet with the CNAME item.
qualityLimitationReason of type RTCQualityLimitationReason
Only valid for video. The current reason for limiting the resolution and/or framerate, or "none" if not limited.
qualityLimitationDurations of type record<DOMString, double>
Only valid 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 "none".
The sum of all entries minus qualityLimidationDurations["none"]
gives the total time that the stream has been limited.
qualityLimitationResolutionChanges of type unsigned long
Only valid for video. The number of times that the resolution has changed because
we are quality limited (qualityLimitationReason has a value other than
"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.
perDscpPacketsSent of type record<USVString, unsigned long long>
Total number of packets sent for this SSRC, per DSCP. DSCPs are identified as decimal integers in string form.
nackCount of type unsigned
long
Count the total number of Negative ACKnowledgement (NACK) packets received by this sender. Calculated as defined in [[!RFC4585]] section 6.2.1.
firCount of type unsigned
long
Only valid for video. Count the total number of Full Intra Request (FIR) packets received by this sender. Calculated as defined in [[!RFC5104]] section 4.3.1. and does not use the metric indicated in [[RFC2032]], because it was deprecated by [[RFC4587]].
pliCount of type unsigned
long
Only valid for video. Count the total number of Picture Loss Indication (PLI) packets received by this sender. Calculated as defined in [[!RFC4585]] section 6.3.1.
sliCount of type unsigned
long
Only valid for video. Count the total number of Slice Loss Indication (SLI) packets received by this sender. Calculated as defined in [[!RFC4585]] section 6.3.2.
encoderImplementation of type DOMString
Identifies the encoder implementation used. This is useful for diagnosing interoperability issues.
If too much information is given here, it increases the fingerprint surface. Since it is only given for active tracks, the incremental exposure is small.
enum RTCQualityLimitationReason {
"none",
"cpu",
"bandwidth",
"other",
};
| Enumeration description | |
|---|---|
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. |
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;
};
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 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.
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 RTCRtpSenders. 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 type is the same
("media-source") but 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 {
DOMString trackIdentifier;
DOMString kind;
};
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.
The RTCAudioSourceStats dictionary represents an audio track that is attached
to one or more senders. It is an RTCMediaSourceStats whose
kind is "audio".
dictionary RTCAudioSourceStats : RTCMediaSourceStats {
double audioLevel;
double totalAudioEnergy;
double totalSamplesDuration;
double echoReturnLoss;
double echoReturnLossEnhancement;
};
audioLevel of type double
Represents the audio level of the media source. For audio levels of remotely
sourced tracks, see RTCAudioReceiverStats 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 algortihm 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 RTCAudioReceiverStats 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
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.
totalSamplesDuration of type double
Represents the audio duration of the media source. For audio durations of
remotely sourced tracks, see RTCAudioReceiverStats 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.
echoReturnLoss of type double
Only present when the MediaStreamTrack is sourced from a microphone where echo cancellation is applied. Calculated in decibels, as defined in [[!ECHO]] (2012) section 3.14.
echoReturnLossEnhancement of type double
Only present when the MediaStreamTrack is sourced from a microphone where echo cancellation is applied. Calculated in decibels, as defined in [[!ECHO]] (2012) section 3.15.
The RTCVideoSourceStats dictionary represents a video track that is attached
to one or more senders. It is an RTCMediaSourceStats whose
kind is "video".
dictionary RTCVideoSourceStats : RTCMediaSourceStats {
unsigned long width;
unsigned long height;
unsigned long frames;
unsigned long framesPerSecond;
};
width of type unsigned
long
The width, in pixels, of the last frame originating from this source. Before a frame has been produced this attribute is missing.
height of type unsigned
long
The height, in pixels, of the last frame originating from this source. Before a frame has been produced this attribute is missing.
frames of type frames
The total number of frames originating from this source.
framesPerSecond of type unsigned long
The number of frames originating from this source, measured during the last second. For the first second of this object's lifetime this attribute is missing.
The RTCRtpContributingSourceStats dictionary represents the measurement
metrics for a contributing source (CSRC) that is contributing to an incoming RTP stream.
Each contributing source produces a stream of RTP packets, which are combined by a mixer
into a single stream of RTP packets that is ultimately received by the WebRTC endpoint.
Information about the sources that contributed to this combined stream may be provided in
the CSRC list or [[RFC6465]] header extension of received RTP packets. The
timestamp of this stats object is the
most recent time an RTP packet the source contributed to was received and counted by
packetsContributedTo.
dictionary RTCRtpContributingSourceStats : RTCStats {
unsigned long contributorSsrc;
DOMString inboundRtpStreamId;
unsigned long packetsContributedTo;
double audioLevel;
};
contributorSsrc of type unsigned long
The SSRC identifier of the contributing source represented by this stats object, as defined by [[!RFC3550]]. It is a 32-bit unsigned integer that appears in the CSRC list of any packets the relevant source contributed to.
inboundRtpStreamId of type DOMString
The ID of the RTCInboundRtpStreamStats object representing
the inbound RTP stream that this contributing source is contributing to.
packetsContributedTo of type unsigned long
The total number of RTP packets that this contributing source contributed to.
This value is incremented each time a packet is counted by
RTCInboundRtpStreamStats.packetsReceived, and the packet's CSRC
list (as defined by [[!RFC3550]] section 5.1) contains the SSRC identifier of
this contributing source, contributorSsrc.
audioLevel of type double
Present if the last received RTP packet that this source contributed to contained
an [[!RFC6465]] mixer-to-client audio level header extension. The value of
audioLevel 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 [[!RFC6465]] header extension contains values in the range 0..127, in units
of -dBov, where 127 represents silence. To convert these values to the linear
0..1 range of audioLevel, a value of 127 is converted to 0, and all
other values are converted using the equation: f(rfc6465_level) =
10^(-rfc6465_level/20).
dictionary RTCPeerConnectionStats : RTCStats {
unsigned long dataChannelsOpened;
unsigned long dataChannelsClosed;
unsigned long dataChannelsRequested;
unsigned long dataChannelsAccepted;
};
dataChannelsOpened of type unsigned long
Represents the number of unique DataChannels that have entered the "open" state during their lifetime.
dataChannelsClosed of type unsigned long
Represents the number of unique DataChannels that have left the "open" state during their lifetime (due to being closed by either end or the underlying transport being closed). DataChannels that transition from "connecting" to "closing" or "closed" without ever being "open" are not counted in this number.
dataChannelsRequested of type unsigned long
Represents the number of unique DataChannels returned from a successful createDataChannel() call on the RTCPeerConnection. If the underlying data transport is not established, these may be in the "connecting" state.
dataChannelsAccepted of type unsigned long
Represents the number of unique DataChannels signaled in a "datachannel" event on the RTCPeerConnection.
The total number of open data channels at any time can be calculated as dataChannelsOpened - dataChannelsClosed. This number is always positive.
The sum of dataChannelsRequested and dataChannelsAccepted is always greater than or equal to dataChannelsOpened - the difference is equal to the number of channels that have been requested, but have not reached the "open" state.
dictionary RTCMediaStreamStats : RTCStats {
DOMString streamIdentifier;
sequence<DOMString> trackIds;
};
streamIdentifier of type DOMString
stream.id property
trackIds of type sequence<DOMString>
This is the id of the stats object, not the track.id.
dictionary RTCMediaHandlerStats : RTCStats {
DOMString trackIdentifier;
boolean remoteSource;
boolean ended;
DOMString kind;
RTCPriorityType priority;
};
trackIdentifier of type DOMString
Represents the id property of the track.
remoteSource of type boolean
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 set for the track. It is specified in [[!RTCWEB-TRANSPORT]], Section 4.
dictionary RTCVideoHandlerStats : RTCMediaHandlerStats {
};
An RTCVideoSenderStats object represents the stats about one video sender of a RTCPeerConnection object for which one calls getStats.
It appears in the stats as soon as the sender is added by either addTrack or addTransceiver, or by media negotiation.
dictionary RTCVideoSenderStats : RTCVideoHandlerStats {
DOMString mediaSourceId;
};
mediaSourceId of type DOMString
The identifier of the stats object representing the track currently attached to
this sender, an RTCMediaSourceStats.
An RTCSenderVideoTrackAttachmentStats object represents the stats about one attachment of a video MediaStreamTrack to the RTCPeerConnection object for which one calls getStats.
It appears in the stats as soon as it is attached (via addTrack, via addTransceiver, via replaceTrack on an RTCRtpSender object).
If a video track is attached twice (via addTransceiver or replaceTrack), there will be two RTCSenderVideoTrackAttachmentStats objects, one for each attachment. They will have the same "trackIdentifier" attribute, but different "id" attributes.
dictionary RTCSenderVideoTrackAttachmentStats : RTCVideoSenderStats {
};
An RTCVideoReceiverStats object represents the stats about one video receiver of a RTCPeerConnection object for which one calls getStats.
It appears in the stats as soon as the RTCRtpReceiver is added by either addTrack or addTransceiver, or by media negotiation.
dictionary RTCVideoReceiverStats : RTCVideoHandlerStats {
};
dictionary RTCAudioHandlerStats : RTCMediaHandlerStats {
};
An RTCAudioSenderStats object represents the stats about one audio sender of a RTCPeerConnection object for which one calls getStats.
It appears in the stats as soon as the RTCRtpSender is added by either addTrack or addTransceiver, or by media negotiation.
dictionary RTCAudioSenderStats : RTCAudioHandlerStats {
DOMString mediaSourceId;
};
mediaSourceId of type DOMString
The identifier of the stats object representing the track currently attached to
this sender, an RTCMediaSourceStats.
An RTCSenderAudioTrackAttachmentStats object represents the stats about one attachment of an audio MediaStreamTrack to the RTCPeerConnection object for which one calls getStats.
It appears in the stats as soon as it is attached (via addTrack, via addTransceiver, via replaceTrack on an RTCRtpSender object).
If an audio track is attached twice (via addTransceiver or replaceTrack), there will be two RTCSenderAudioTrackAttachmentStats objects, one for each attachment. They will have the same "trackIdentifier" attribute, but different "id" attributes.
dictionary RTCSenderAudioTrackAttachmentStats : RTCAudioSenderStats {
};
An RTCAudioReceiverStats object represents the stats about one audio receiver of a RTCPeerConnection object for which one calls getStats.
It appears in the stats as soon as the RTCRtpReceiver is added by either addTrack or addTransceiver, or by media negotiation.
dictionary RTCAudioReceiverStats : RTCAudioHandlerStats {
};
dictionary RTCDataChannelStats : RTCStats {
DOMString label;
DOMString protocol;
long dataChannelIdentifier;
DOMString transportId;
RTCDataChannelState state;
unsigned long messagesSent;
unsigned long long bytesSent;
unsigned long messagesReceived;
unsigned long long bytesReceived;
};
label of type DOMString
protocol of type DOMString
dataChannelIdentifier of type long
The "id" attribute of the RTCDataChannel object.
transportId of type DOMString
state of type RTCDataChannelState
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.
An RTCTransportStats object represents the stats corresponding to an
RTCDtlsTransport and its underlying
RTCIceTransport. When RTCP multiplexing is used, one transport is
used for both RTP and RTCP. Otherwise, RTP and RTCP will be sent on separate transports,
and rtcpTransportStatsId can be used to pair the resulting
RTCTransportStats objects. Additionally, when bundling is used, a single
transport will be used for all MediaStreamTracks in the bundle group.
If bundling is not used, different MediaStreamTrack will use
different transports. RTCP multiplexing and bundling are described in [[!WEBRTC]].
dictionary RTCTransportStats : RTCStats {
unsigned long long packetsSent;
unsigned long long packetsReceived;
unsigned long long bytesSent;
unsigned long long bytesReceived;
DOMString rtcpTransportStatsId;
RTCIceRole iceRole;
RTCDtlsTransportState dtlsState;
DOMString selectedCandidatePairId;
DOMString localCertificateId;
DOMString remoteCertificateId;
DOMString tlsVersion;
DOMString dtlsCipher;
DOMString srtpCipher;
DOMString tlsGroup;
unsigned long selectedCandidatePairChanges;
};
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
PeerConnection, i.e., not including headers or padding.
bytesReceived of type unsigned
long long
Represents the total number of bytes received on this
PeerConnection, i.e., not including headers or padding.
rtcpTransportStatsId of type DOMString
If RTP and RTCP are not multiplexed, this is the id of the transport
that gives stats for the RTCP component, and this record has only the RTP
component stats.
iceRole of type RTCIceRole
Set to the current value of the "role" attribute of the underlying RTCDtlsTransport's "transport".
dtlsState of type RTCDtlsTransportState
Set to the current value of the "state" attribute of the underlying RTCDtlsTransport.
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 present 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]].
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]].
tlsGroup of type DOMString
Descriptive name of the group used for the encryption, as defined in the "Description" column of the IANA TLS Supported Groups registry [[!IANA-TLS-GROUPS]].
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.
RTCIceCandidateStats reflects the properties of a candidate in
Section 15.1 of [[!RFC5245]]. It corresponds to a RTCIceCandidate object.
dictionary RTCIceCandidateStats : RTCStats {
DOMString transportId;
RTCNetworkType networkType;
DOMString? address;
long port;
DOMString protocol;
RTCIceCandidateType candidateType;
long priority;
DOMString url;
DOMString relayProtocol;
};
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.
networkType of type RTCNetworkType
Represents the type of network interface used by the base of a local candidate (the address the ICE agent sends from). Only present for local candidates; it's not possible to know what type of network interface a remote candidate is using.
networkType of the
relevant candidate would be "wifi", even when the next hop is over a
cellular connection.
This reveals information that would otherwise not be available to web pages, which increases the fingerprint surface.
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 '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 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.
relayProtocol of type DOMString
It is the protocol used by the endpoint to communicate with the TURN server. This
is only present for local candidates. Valid values are udp,
tcp, or tls.
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 url surfaced in the RTCPeerConnectionIceEvent.
For remote candidates, this property is not present.
enum RTCNetworkType {
"bluetooth",
"cellular",
"ethernet",
"wifi",
"wimax",
"vpn",
"unknown"
};
| Enumeration description | |
|---|---|
bluetooth
|
A Bluetooth connection. |
cellular
|
A cellular connection (e.g., EDGE, HSPA, LTE, etc.). |
ethernet
|
An Ethernet connection. |
wifi
|
A Wi-Fi connection. |
wimax
|
A WiMAX connection. |
vpn
|
The connection runs over a VPN. The underlying network type is not available. |
unknown
|
The user agent is unable or unwilling to identify the underlying connection technology. |
dictionary RTCIceCandidatePairStats : RTCStats {
DOMString transportId;
DOMString localCandidateId;
DOMString remoteCandidateId;
RTCStatsIceCandidatePairState state;
boolean nominated;
unsigned long long packetsSent;
unsigned long long packetsReceived;
unsigned long long bytesSent;
unsigned long long bytesReceived;
DOMHighResTimeStamp lastPacketSentTimestamp;
DOMHighResTimeStamp lastPacketReceivedTimestamp;
DOMHighResTimeStamp firstRequestTimestamp;
DOMHighResTimeStamp lastRequestTimestamp;
DOMHighResTimeStamp lastResponseTimestamp;
double totalRoundTripTime;
double currentRoundTripTime;
double availableOutgoingBitrate;
double availableIncomingBitrate;
unsigned long circuitBreakerTriggerCount;
unsigned long long requestsReceived;
unsigned long long requestsSent;
unsigned long long responsesReceived;
unsigned long long responsesSent;
unsigned long long retransmissionsReceived;
unsigned long long retransmissionsSent;
unsigned long long consentRequestsSent;
DOMHighResTimeStamp consentExpiredTimestamp;
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
};
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 or padding.
bytesReceived of type unsigned
long long
Represents the total number of payload bytes received on this candidate pair, i.e., not including headers or padding.
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.
firstRequestTimestamp of type DOMHighResTimeStamp
Represents the timestamp at which the first STUN request was sent on this particular candidate pair.
lastRequestTimestamp of type DOMHighResTimeStamp
Represents the timestamp at which the last STUN request was sent on this
particular candidate pair. The average interval between two consecutive
connectivity checks sent can be calculated with (lastRequestTimestamp -
firstRequestTimestamp) / requestsSent.
lastResponseTimestamp of type DOMHighResTimeStamp
Represents the timestamp at which the last STUN response was received on this particular candidate pair.
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.
circuitBreakerTriggerCount of type unsigned long
Represents the number of times the circuit breaker is triggered for this particular 5-tuple. Ceasing transmission when a circuit breaker is triggered is defined in Section 4.5 of [[!RFC8083]]. The field MUST return undefined for user-agents that do not implement the circuit-breaker algorithm.
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.
retransmissionsReceived of type unsigned long long
Represents the total number of connectivity check request retransmissions received. Retransmissions are defined as connectivity check requests with a TRANSACTION_TRANSMIT_COUNTER attribute where the "req" field is larger than 1, as defined in [[!RFC7982]].
retransmissionsSent of type unsigned long long
Represents the total number of connectivity check request retransmissions sent.
consentRequestsSent of type unsigned long long
Represents the total number of consent requests sent.
consentExpiredTimestamp of type DOMHighResTimeStamp
Represents the timestamp at which the latest valid STUN binding response expired,
as defined in [[!RFC7675]] section 5.1. If a valid STUN binding response has not
been made (responsesReceived is zero) or the latest one has not
expired this value must be undefined.
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 occured 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 occured 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.
enum RTCStatsIceCandidatePairState {
"frozen",
"waiting",
"in-progress",
"failed",
"succeeded"
};
| Enumeration description | |
|---|---|
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]]. |
dictionary RTCCertificateStats : RTCStats {
DOMString fingerprint;
DOMString fingerprintAlgorithm;
DOMString base64Certificate;
DOMString issuerCertificateId;
};
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.
dictionary RTCIceServerStats : RTCStats {
DOMString url;
long port;
DOMString protocol;
RTCNetworkType networkType;
unsigned long totalRequestsSent;
unsigned long totalResponsesReceived;
double totalRoundTripTime;
};
url of type DOMString
The URL of the ICE server (e.g. TURN or STUN server).
port of type long
It is the port number used by the client.
protocol of type DOMString
Valid values for transport is one of udp and tcp. Based
on the "transport" defined in [RFC5245] section 15.1.
networkType of type RTCNetworkType
Represents the type of network interface used.
totalRequestsSent of type unsigned long
The total amount of requests that have been sent to this server.
totalResponsesReceived of type unsigned long
The total amount of responses received from this server.
totalRoundTripTime of type double
The sum of RTTs for all requests that have been sent where a response has been received.
dictionary RTCReceiverVideoTrackAttachmentStats : RTCVideoReceiverStats {
};
The RTCReceiverVideoTrackAttachmentStats is a copy of
RTCVideoReceiverStats. It adds no new information, it
only exists for backwards-compatibility reasons as an obsolete
dictionary.
dictionary RTCReceiverAudioTrackAttachmentStats : RTCAudioReceiverStats {
};
The RTCReceiverAudioTrackAttachmentStats is a copy of
RTCAudioReceiverStats. It adds no new information, it
only exists for backwards-compatibility reasons as an obsolete
dictionary.
partial dictionary RTCCodecStats {
DOMString implementation;
};
implementation of type DOMString
This was moved to RTCInboundRtpStreamStats.decoderImplementation and RTCOutboundRtpStreamStats.encoderImplementation in August 2019.
partial dictionary RTCIceCandidateStats {
boolean deleted = false;
boolean isRemote;
};
deleted of type boolean
This field is obsoleted by the statsended event.
isRemote of type boolean
false indicates that this represents a local candidate;
true indicates that this represents a remote candidate.
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.
partial dictionary RTCRtpStreamStats {
DOMString mediaType;
double averageRTCPInterval;
};
mediaType of type DOMString
This field got renamed to "kind" in Feb 2018.
averageRTCPInterval
This field got renamed to "averageRtcpInterval" in Jan 2018.
partial dictionary RTCInboundRtpStreamStats {
double fractionLost;
};
fractionLost of type double
This field was moved to RTCRemoteInboundRtpStreamStats in December 2017.
partial dictionary RTCAudioHandlerStats {
double audioLevel;
double totalAudioEnergy;
double totalSamplesDuration;
boolean voiceActivityFlag;
};
audioLevel of type double
This field was moved to RTCAudioReceiverStats and RTCAudioSourceStats in June 2019.
totalAudioEnergy of type double
This field was moved to RTCAudioReceiverStats and RTCAudioSourceStats in June 2019.
totalSamplesDuration of type double
This field was moved to RTCAudioReceiverStats and RTCAudioSourceStats in June 2019.
voiceActivityFlag of type boolean
This field was moved to RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats in August 2019.
partial dictionary RTCAudioSenderStats {
unsigned long long totalSamplesSent;
double echoReturnLoss;
double echoReturnLossEnhancement;
};
totalSamplesSent of type unsigned long long
This was moved to RTCOutboundRtpStreamStats in August 2019.
echoReturnLoss of type double
This was moved to RTCAudioSourceStats in August 2019.
echoReturnLossEnhancement of type double
This was moved to RTCAudioSourceStats in August 2019.
partial dictionary RTCAudioReceiverStats {
DOMHighResTimeStamp estimatedPlayoutTimestamp;
double jitterBufferDelay;
unsigned long long jitterBufferEmittedCount;
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;
};
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.
totalSamplesReceived of type unsigned long long
This was moved to RTCInboundRtpStreamStats in August 2019.
concealedSamples of type unsigned long long
This was moved to RTCInboundRtpStreamStats in August 2019.
silentConcealedSamples of type unsigned long long
This was moved to RTCInboundRtpStreamStats in August 2019.
concealmentEvents of type unsigned long long
This was moved to RTCInboundRtpStreamStats in August 2019.
insertedSamplesForDeceleration of type unsigned long long
This was moved to RTCInboundRtpStreamStats in August 2019.
removedSamplesForAcceleration of type unsigned long long
This was moved to RTCInboundRtpStreamStats in August 2019.
audioLevel of type double
This was moved to RTCInboundRtpStreamStats in August 2019.
totalAudioEnergy of type double
This was moved to RTCInboundRtpStreamStats in August 2019.
totalSamplesDuration of type double
This was moved to RTCInboundRtpStreamStats in August 2019.
partial dictionary RTCVideoHandlerStats {
unsigned long frameWidth;
unsigned long frameHeight;
double framesPerSecond;
};
frameWidth of type unsigned
long
This was moved to RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats in August 2019.
frameHeight of type unsigned
long
This was moved to RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats in August 2019.
framesPerSecond of type double
For the sending case, this was replaced by 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, framesPerSecond was also added to RTCOutboundRtpStreamStats, representing the encoding frame rate (which may be lower than the source frame rate).
partial dictionary RTCVideoSenderStats {
unsigned long keyFramesSent;
unsigned long framesCaptured;
unsigned long framesSent;
unsigned long hugeFramesSent;
};
keyFramesSent of type unsigned
long
This field was replaced by 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
This was replaced by RTCVideoSourceStats.frames in May 2019.
framesSent of type unsigned
long
This was moved to RTCOutboundRtpStreamStats in August 2019.
hugeFramesSent of type unsigned
long
This was moved to RTCOutboundRtpStreamStats in August 2019.
partial dictionary RTCVideoReceiverStats {
unsigned long keyFramesReceived;
DOMHighResTimeStamp estimatedPlayoutTimestamp;
double jitterBufferDelay;
unsigned long long jitterBufferEmittedCount;
unsigned long framesReceived;
unsigned long framesDecoded;
unsigned long framesDropped;
unsigned long partialFramesLost;
unsigned long fullFramesLost;
};
keyFramesReceived of type unsigned long
This field was replaced by keyFramesDecoded in RTCInboundRtpStreamStats in June 2019. There were no known implementations supporting the old field at the time of the change.
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.
framesReceived of type unsigned
long
This was moved to RTCInboundRtpStreamStats in August 2019.
framesDecoded of type unsigned
long
This was moved to RTCInboundRtpStreamStats in August 2019.
framesDropped of type unsigned
long
This was moved to RTCInboundRtpStreamStats in August 2019.
partialFramesLost of type unsigned long
This was moved to RTCInboundRtpStreamStats in August 2019.
fullFramesLost of type unsigned
long
This was moved to RTCInboundRtpStreamStats in August 2019.
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;
}
});
}
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
audioLevel statistic).
The following stats are deemed to be sensitive, and MUST NOT be reported for an isolated media stream:
This section will be removed before publication. The entries are in reverse chronological order.
This list does not include infrastructure and minor editorials.
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.