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].
Status of This Document
This section describes the status of this document at the time of its publication. Other documents may supersede this document. A list of current W3C publications and the latest revision of this technical report can be found in the W3C technical reports index at https://www.w3.org/TR/.
This document is incomplete, and as such is not yet suitable for implementation. However,
early experimentation is encouraged.
Publication as a Working Draft does not imply endorsement by the W3C
Membership. This is a draft document and may be updated, replaced or obsoleted by other
documents at any time. It is inappropriate to cite this document as other than work in
progress.
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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.
2. Conformance
As well as sections marked as non-normative, all authoring guidelines, diagrams, examples,
and notes in this specification are non-normative. Everything else in this specification is
normative.
The key words MAY, MUST, and MUST NOT are
to be interpreted as described in [RFC2119].
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 like this:
An implementation MUST support generating statistics for the type
RTCInboundRtpStreamStats, with attributes packetsReceived, bytesReceived, packetsLost,
jitter, and fractionLost.
It MUST support generating statistics for the type RTCOutboundRtpStreamStats, with
attributes packetsSent, bytesSent.
For all subclasses of RTCRtpStreamStats, it MUST include ssrc and kind. When stats
exist for both sides of a connection, in the form of an inbound-rtp / remote-outbound-rtp
pair or an outbound-rtp / remote-inbound-rtp pair, the members remoteId and localId MUST
also be present.
The terms MediaStream, MediaStreamTrack, and Consumer are
defined in [GETUSERMEDIA].
The terms RTCPeerConnection, RTCDataChannel,
RTCDtlsTransport, RTCDtlsTransportState, RTCIceTransport,
RTCIceRole and RTCPriorityType are defined in [WEBRTC].
The term RTP stream is defined in [RFC7656] section 2.1.10.
The basic object of the stats model is the stats object. The following terms are
defined to describe it:
Monitored object
An internal object that keeps a set of data values. Most monitored objects are object
defined in the WebRTC API; they may be thought of as being internal properties of those
objects.
Stats object
This is a set of values, copied out from a monitored object at a specific moment in time.
It is returned as a WebIDL dictionary through the getStats API call.
Stats object reference
A monitored object has a stable identifier "id", which is reflected in all stats
objects produced from the monitored object. Stats objects may contain references to
other stats objects using this "id" value. In a stats object, these references
are represented by a DOMString containing "id" value of the referenced stats object.
All stats object references have type DOMString and attribute names ending in 'Id', or
they have type sequence<DOMString> and attribute names ending in 'Ids'.
Stats value
Refers to a single value within a stats object.
A monitored object changes the values it contains continuously over its lifetime, but is
never visible through the getStats API call. A stats object, once returned, never changes.
The stats API is defined in [WEBRTC]. It is defined to return a collection of stats
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.
When introducing a new stats object, the following principles should be followed:
An RTCStats object should correspond to an object defined in the specification it
supports.
The object MUST define a new value in the "RTCStastType" enum, and MUST define the
syntax of the stats object it returns either by reference to an existing
sub-dictionary of RTCStats or by defining a new sub-dictionary of RTCStats.
All members of the new object need to have definitions that make them consistently
implementable. References to other specifications are a good way of doing this.
All members need to have defined behavior for what happens before the thing it counts
happens, or when the information it's supposed to show is not available. Usually, this
will be "start at zero" or "do not populate the value".
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.
4.2
Guidelines for implementing stats objects
When implementing stats objects, the following guidelines should be adhered to:
When a feature is not implemented on the platform, omit the dictionary member that is
tracking usage of the feature.
When a feature is not applicable to an instance of an object (for example audioLevel
on a video stream), omit the dictionary member. Do NOT report a count of zero, -1 or
"empty string".
When a counted feature hasn't been used yet, but may happen in the future, report a
count of zero.
4.3
Lifetime considerations for monitored objects
The object descriptions will say what the lifetime of a monitored object from the
perspective of stats is. When a monitored object is "deleted", it no longer appears in
stats; until this happens, it will appear. This may or may not correspond to the actual
lifetime of an object in an implementation; what matters for this specification is what
appears in stats.
If a monitored object can only exist in a few instances over the lifetime of a
RTCPeerConnection, it may be simplest to consider it "eternal" and never delete it from the
set of objects reported on in stats. This type of object will remain visible until the
RTCPeerConnection is no longer available; it is also visible in 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.
4.4
Guidelines for getStats() results caching/throttling
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.
5.
Maintenance procedures for stats object types
5.1
Adding new stats objects
This document, in its editors' draft form, serves as the repository for the currently
defined set of stats object types.
If a need for a new stats object type or stats value within a stats object is found, an
issue should be raised against this spec on Github, and a review process will decide on
whether the stat should be added to the specification or not.
A pull request for a change to this document may serve as guidance for the discussion,
but the eventual merge is dependent on the review process.
While the WebRTC WG exist, 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.
5.2
Retiring stats objects
At times, it makes sense to retire the definition for a stats object or a stats value.
When this happens, it is not advisable to simply delete it from the spec, since there may
be implementations out there that use it, and it is important that the name is reserved
from re-use for another, incompatible definition.
Therefore, retired stats objects are moved to a separate section in this document.
Retired stats objects are moved there in their entirety; retired stats values are moved
to a "partial dictionary".
If there is no evidence that the retired object definition has ever been used (such as an
object that is added to the spec and renamed, redefined or removed prior to
implementation), the editors can decide to just remove the object from the spec.
6.
RTCStatsType
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.
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.
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.
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.
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
Contains 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
Contains statistics related to a specific receiver and the corresponding media-level
metrics. It is accessed by the RTCAudioReceiverStats or the
RTCVideoSenderStats 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.
7.
Stats dictionaries
7.1
The RTP statistics hierarchy
The dictionaries for RTP statistics are structured as a hierarchy, so that those stats
that make sense in many different contexts are represented just once in IDL.
The 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.
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).
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.
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).
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 codec
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.
firCount of type unsigned
long
Count the total number of Full Intra Request (FIR) packets received by the
sender. This metric is only valid for video and is sent by 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
Count the total number of Picture Loss Indication (PLI) packets received by the
sender. This metric is only valid for video and is sent by receiver. Calculated
as defined in [RFC4585] section 6.3.1.
nackCount of type unsigned
long
Count the total number of Negative ACKnowledgement (NACK) packets received by the
sender and is sent by receiver. Calculated as defined in [RFC4585] section
6.2.1.
sliCount of type unsigned
long
Count the total number of Slice Loss Indication (SLI) packets received by the
sender. This metric is only valid for video and is sent by receiver. Calculated
as defined in [RFC4585] section 6.3.2.
qpSum of type unsigned long
long
The sum of the QP values of frames passed. The count of frames is in
framesDecoded for inbound stream stats, and in framesEncoded for
outbound stream stats.
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.
"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.
implementation of type DOMString
Identifies the 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.
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
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
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
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
The cumulative number of RTP packets lost during loss bursts, Appendix A (c) of
[RFC6958].
burstPacketsDiscarded of type unsigned 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.
7.5
RTCInboundRtpStreamStats dictionary
The RTCInboundRtpStreamStats dictionary represents the measurement metrics for
the incoming RTP media stream. The timestamp reported in the statistics object is the
time at which the data was sampled.
Only valid for video. It represents the total number of frames correctly decoded
for this SSRC, i.e., frames that would be displayed if no frames are dropped.
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 packet and an SDES packet with the
CNAME item.
fecPacketsReceived of type unsigned 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).
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
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
The cumulative number of packets discarded because they are duplicated. Duplicate
packets are not counted in packetsDiscarded.
Duplicated packets have the same RTP sequence number and content as a previously
received packet. If multiple duplicates of a packet are received, all of them are
counted.
An improved estimate of lost packets can be calculated by adding
packetsDuplicated to packetsLost; this will always result in a
positive number, but not the same number as RFC 3550 would calculate.
perDscpPacketsReceived of type record<USVString, unsigned 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.
7.6
RTCRemoteInboundRtpStreamStats dictionary
The RTCRemoteInboundRtpStreamStats dictionary represents the remote endpoint's
measurement metrics for a particular incoming RTP stream (corresponding to an outgoing
RTP stream at the sending endpoint). The timestamp reported in the statistics object is
the time at which the corresponding RTCP RR was received.
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.
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.
Total number of RTP packets sent for this SSRC. Calculated as defined in
[RFC3550] section 6.4.1.
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.
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).
bytesSent of type unsigned
long long
Total number of bytes sent for this SSRC. Calculated as defined in [RFC3550]
section 6.4.1.
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.
7.8
RTCOutboundRtpStreamStats dictionary
The RTCOutboundRtpStreamStats dictionary represents the measurement metrics
for the outgoing RTP stream. The timestamp reported in the statistics object is the time
at which the data was sampled.
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.
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.
framesEncoded of type long
Only valid for video. It represents the total number of frames successfully
encoded for this RTP media stream.
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.
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 packet and an SDES packet with the
CNAME item.
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.
perDscpPacketsSent of type record<USVString, unsigned long>
Total number of packets sent for this SSRC, per DSCP. DSCPs are identified as
decimal integers in string form.
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.
7.10
RTCRemoteOutboundRtpStreamStats dictionary
The RTCRemoteOutboundRtpStreamStats dictionary represents the remote
endpoint's measurement metrics for its outgoing RTP stream (corresponding to an outgoing
RTP stream at the sending endpoint). The timestamp reported in the statistics object is
the time at which the corresponding RTCP SR was received.
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) packet, which reflects the remote
endpoint's clock. That clock may not be synchronized with the local clock.
7.11 RTCRtpContributingSourceStats dictionary
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.
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).
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.
Only applicable for 'track' stats. True if the source is remote, for instance if it
is sourced from another host via an RTCPeerConnection. False otherwise.
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].
Represents the width of the last processed frame for this track. Before the first
frame is processed this attribute is missing.
frameHeight of type unsigned
long
Represents the height of the last processed frame for this track. Before the
first frame is processed this attribute is missing.
framesPerSecond of type double
Represents the nominal FPS value before the degradation preference is applied. It
is the number of complete frames in the last second. For sending tracks it is the
current captured FPS and for the receiving tracks it is the current decoding
framerate.
7.16 RTCVideoSenderStats dictionary
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.
Represents the total number of frames captured, before encoding, for this
RTCRtpSender (or for this MediaStreamTrack, if type is
"track"). For example, if type is "sender"
and this sender's track represents a camera, then this is the number of frames
produced by the camera for this track while being sent by this sender, combined
with the number of frames produced by all tracks previously attached to this
sender while being sent by this sender. Framerates can vary due to hardware
limitations or environmental factors such as lighting conditions.
framesSent of type unsigned
long
Represents the total number of frames sent by this RTCRtpSender (or for this
MediaStreamTrack, if type is "track").
hugeFramesSent of type unsigned long
Represents the total number of huge frames sent by this RTCRtpSender (or for this
MediaStreamTrack, if type is "track"). 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. If
a huge frame is also a key frame, then both counters hugeFramesSent and
keyFramesSent are incremented.
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.
keyFramesSent of type unsigned long
Represents the total number of key frames sent by this RTCRtpSender (or for this
MediaStreamTrack, if type is "track"), such as
Infra-frames in VP8 [RFC6386] or I-frames in H.264 [RFC6184]. This is a
subset of framesSent. framesSent - keyFramesSent gives
you the number of delta frames sent.
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.
If the track is detached from the RTCPeerConnection (via removeTrack or via replaceTrack),
it continues to appear, but with the "objectDeleted" member set to true.
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 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 video currently playing.
This can be useful for estimating how much audio and video is out of sync for two
tracks from the same remote source,
audioTrackStats.estimatedPlayoutTimestamp -
videoTrackStats.estimatedPlayoutTimestamp.
jitterBufferDelay of type double
It is the sum of the time, in seconds, each frame takes from the time it is
received and to the time it exits the jitter buffer. This increases upon frames
exiting, having completed their time in the buffer (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 frames that have come out of the jitter buffer (increasing
jitterBufferDelay).
framesReceived of type unsigned long
Represents the total number of complete frames received for this receiver. This
metric is incremented when the complete frame is received.
keyFramesReceived of type unsigned long
Represents the total number of complete key frames received for this
MediaStreamTrack, such as Infra-frames in VP8 [RFC6386] or I-frames in H.264
[RFC6184]. This is a subset of framesReceived.
framesReceived - keyFramesReceived gives you the number of delta
frames received. This metric is incremented when the complete key frame is
received. It is not incremented if a partial key frames is received and sent for
decoding, i.e., the frame could not be recovered via retransmission or FEC.
framesDecoded of type unsigned long
Only valid for video. It represents the total number of frames correctly decoded
for this SSRC, i.e., frames that would be displayed if no frames are dropped.
framesDropped of type unsigned long
The total number of frames dropped predecode or dropped because the frame missed
its display deadline for this receiver's track. As defined in Appendix A (g) of
[RFC7004].
partialFramesLost of type unsigned long
The cumulative number of partial frames lost, 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
The cumulative number of full frames lost, as defined in Appendix A (i) of
[RFC7004].
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 "audio level" value defined in [RFC6464] (as 0..127, where 0 represents 0
dBov, 126 represents -126 dBov and 127 represents silence) is obtained by the
calculation given in appendix A of [RFC6465]: informally, level =
-round(log10(audioLevel) * 20), with audioLevel 0.0 and values above 127 mapped
to 127.
The audioLevel represents the output audio level of the track; thus, if
the track is sourced from an RTCReceiver, does no audio processing, has a
constant level, and has a volume setting of 1.0, the audio level is
expected to be the same as the audio level of the source SSRC, while if the
volume setting is 0.5, the audioLevel is expected to be half that value.
For outgoing audio tracks, the audioLevel is the level of the audio being sent.
The audioLevel is averaged over some small interval, using the algortihm
described under totalAudioEnergy. The interval used is implementation
dependent.
totalAudioEnergy of type double
This value MUST be computed as follows: for each audio sample sent/received for
this object (and counted by totalSamplesSent or
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 received 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.
voiceActivityFlag of type boolean
Whether the last RTP packet sent or played out by this track contained voice
activity or not based on the presence of the V bit in the extension header, as
defined in [RFC6464].
This value indicates the voice activity in the latest RTP packet played out from
a given SSRC, and is defined in the
RTCRtpSynchronizationSource.voiceActivityFlag of [[WEBRTC].
totalSamplesDuration of type double
Represents the total duration in seconds of all samples that have sent or
received (and thus counted by totalSamplesSent or
totalSamplesReceived). Can be used with
totalAudioEnergy to compute an average audio level over
different intervals.
7.20 RTCAudioSenderStats dictionary
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.
Only present while the sender is sending a track 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 while the sender is sending a track sourced from a microphone where
echo cancellation is applied. Calculated in decibels, as defined in [ECHO]
(2012) section 3.15.
totalSamplesSent of type unsigned long long
The total number of samples that have been sent by this sender.
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.
If the track is detached from the RTCPeerConnection (via removeTrack or via replaceTrack),
it continues to appear, but with the "objectDeleted" member set to true.
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 sample 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 sample takes from the time it is
received and to the time it exits the jitter buffer. This increases upon samples
exiting, having completed their time in the buffer (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 samples that have come out of the jitter buffer (increasing
jitterBufferDelay).
totalSamplesReceived of type unsigned long long
The total number of samples that have been received by this receiver. This
includes concealedSamples.
concealedSamples of type unsigned long long
The total number of samples that are concealed samples. A concealed sample is a
sample that is based on data that was synthesized to conceal packet loss and does
not represent incoming data.
concealmentEvents of type unsigned long long
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.
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.
7.24 RTCTransportStats dictionary
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].
Represents the total number of packets sent over this transport.
packetsReceived of type unsigned 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.
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.
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].
7.25 RTCIceCandidateStats dictionary
RTCIceCandidateStats reflects the properties of a candidate in
Section 15.1 of [RFC5245]. It corresponds to a RTCIceCandidate object.
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.
Note
This stat only tells you about the network interface used by the first "hop";
it's possible that a connection will be bottlenecked by another type of network.
For example, when using Wi-Fi tethering, the networkType of the
relevant candidate would be "wifi", even when the next hop is over a
cellular connection.
ip of type DOMString
It is the IP address of the candidate, allowing for IPv4 addresses and IPv6
addresses, but fully qualified domain names (FQDNs) are not allowed. See
[RFC5245] section 15.1 for details.
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 for the TURN URL protocol is
one of udp, tcp, or tls.
The URL of the TURN or STUN server indicated in the that translated this IP
address. It is the URL address surfaced in an
RTCPeerConnectionIceEvent.
deleted of type boolean, defaulting to false
For local candidates, true indicates that the candidate has been
deleted/freed as described by [RFC5245]. For host candidates, this means that
any network resources (typically a socket) associated with the candidate have
been released. For TURN candidates, this means the TURN allocation is no longer
active.
For remote candidates, this property is not applicable.
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 RTCIceCandidateAttributes 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 RTCIceCandidateAttributes for the remote candidate
associated with this candidate pair.
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
Represents the total number of packets sent on this candidate pair.
packetsReceived of type unsigned 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.
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.
This field got renamed to "averageRtcpInterval" in Jan 2018.
9.
Examples
9.1
Example of a stats application
Consider the case where the user is experiencing bad sound and the application wants to
determine if the cause of it is packet loss. The following example code might be used:
Example 1
var baselineReport, currentReport;
var sender = pc.getSenders()[0];
sender.getStats().then(function (report) {
baselineReport = report;
})
.then(function() {
returnnewPromise(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());
});
functionprocessStats() {
// compare the elements from the current report with the baselinefor (let now of currentReport.values()) {
if (now.type != "outbound-rtp")
continue;
// get the corresponding stats from the baseline reportlet 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 culpritvar intervalFractionLoss = (packetsSent - packetsReceived) / packetsSent;
}
});
}
10.
Security and Privacy Considerations
The data exposed by WebRTC Statistics include most of the media and network data also
exposed by [GETUSERMEDIA] and [WEBRTC] - as such, all the privacy and security
considerations of these specifications related to data exposure apply as well to this
specifciation.
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
volume statistic).
[#114] Minor clarification regarding stats object lifetime
[#157] Change type of RTCRTPStreamStats.ssrc from string to unsigled long
[#149] Remove references saying "defines an API"
[#148] Explanation for "remoteSource"
[#156] frameWidth/frameHeight: use last decoded value
[#123] Explain "sum and count" design paradigm
[#126] Fix RTCStatsType for "stream"
[#127] Added "kind" to RTCMediaStreamTrackStats
[#129] Remove ssrcids field
[#128] Define audio level rigidly
[#122] Replace RTCTransportStats.active with .dtlsState
[#125] Added more datachannel counters, with definitions
[#142] Rename RTCRtpMediaStreamStats.trackId
[#167] Moving roundTripTime from outbound to inbound
[#139] Define terminology for "stats object" et al
[#169] Fix issues with TURN URL protocol
[#168] Align codec types with webrtc-pc
[#166] Removed cancelled and renamed inprogress to in-progress
[#164] Added remoteTimestamp to RTCRtpStreamTrackStats
[#138] Make a RTCMediaStreamTackStats object per track attachment
[#165] Remove separation of received consent and connectivity requests
[#179] Adds definitions for RTCDataChannelStats members
[#176] Add link from datachannel to transport
[#181] Add section on obsoleted stats
[#182] Bandwidth estimations again
[#188] RTT undefined when no RTCP RR
11.3
Changes since 21 sep 2016
[#64] Added text about which specification is authoriative
[#59] Example conformance specification
[#71] Introduced the term "RTP stream"
[#76] Adding missing "codec" RTCStatsType
[#68] Design considerations section
[#73] Fix the summary of RTCTransportStats
[#75] Clarify that pliCount is only valid for video
[#70] Added QP statistcs
[#90] Adding "deleted" property to RTCIceCandidate
[#93] Adding isRemote to RTCIceCandidate
[#95] Rename "RTT" to RoundTripTime
[#94] Add TransportId to RTCIceCandidateStats
[#43] Added procedures for new stats
This list does not include infrastructure and minor editorials.
11.4
Changes since 26 May 2016
[#54] Debug problems with ICE.
[#52] adding XRBLOCK metrics
[#51] Dashed enums and crosslinking to stats objects.
[#37] Clarified RTT units.
11.5
Changes since 23 October 2015
[#18] Updated spec changes.
[#17] Changed "remoteId" to "associateStatsId".
[#8] Ended and detached stats for a track.
[#33] Added the codec "implementation" variable.
[#34]Converted to WebIDL contiguious mode.
[#36] Aligned RTCIceCandidateStats with RTCIceCandidate.
[#24] Added packetsDiscarded and packetsRepaired to stats.
[#13] Aligned bitrate to the TIAS definition.
[#47] Changed RTCCodec to RTCCodecStats
Various formatting, layout and link fixes.
11.6
Changes since 03 February 2015
[#10] Added RTCRTPStreamStats.mediaType.
11.7
Changes since 30 September 2014
kept getStats() in webrtc-pc. Changed RTCStatsType from enum to DOMString.
Added "datachannel" to RTCStatsType.
Added fractionLost to RTCInboundRTPStreamStats.
Clarified that bytesSent and bytesReceived do no include headers or paddings.
11.8
Acknowledgements
The editors wish to thank the Working Group chairs, Stefan Håkansson, and the Team
Contact, Dominique Hazaël-Massieux, for their support. The editors would like to thank
Bernard Aboba, Taylor Brandstetter, Henrik Boström, Jan-Ivar Bruaroey, Karthik Budigere,
Cullen Jennings, and Lennart Schulte for their contributions to this specification.
