2020

Identifiers for WebRTC's Statistics API W3C Candidate Recommendation 14 January 2020 This version: https://www.w3.org/TR/2020/CR-webrtc-stats-20200114/ Latest published version: https://www.w3.org/TR/webrtc-stats/ Latest editor's draft: https://w3c.github.io/webrtc-stats/ Test suite: https://github.com/web-platform-tests/wpt/tree/master/webrtc-stats Implementation report: https://wpt.fyi/webrtc-stats Previous version: https://www.w3.org/TR/2018/CR-webrtc-stats-20180703/ Editors: Harald Alvestrand (Google) Varun Singh (callstats.io) Participate: Mailing list Browse open issues Copyright © 2014-2020 W3C ® ( MIT , ERCIM , Keio , Beihang ). W3C liability , trademark and permissive document license rules apply. Abstract 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/. Since the previous publication as a Candidate Recommendation, the stats objects were significantly reorganized to better match the underlying data sources. In addition, the networkType property was deprecated for preserving privacy, and the statsended event was removed as no longer needed. This document was published by the Web Real-Time Communications Working Group as a Candidate Recommendation. This document is intended to become a W3C Recommendation. Comments regarding this document are welcome. Please send them to [email protected] ( archives ). W3C publishes a Candidate Recommendation to indicate that the document is believed to be stable and to encourage implementation by the developer community. This Candidate Recommendation is expected to advance to Proposed Recommendation no earlier than 21 February 2020. Please see the Working Group's implementation report . Publication as a Candidate Recommendation 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. This document was produced by a group operating under the W3C Patent Policy . W3C maintains a public list of any patent disclosures made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes contains Essential Claim(s) must disclose the information in accordance with section 6 of the W3C Patent Policy . This document is governed by the 1 March 2019 W3C Process Document . Table of Contents 1. Introduction 2. Conformance 3. Terminology 4. Basic concepts 4.1 Guidelines for design of stats objects 4.2 Guidelines for implementing stats objects 4.3 Lifetime considerations for monitored objects 4.4 Guidelines for getStats() results caching/throttling 5. Maintenance procedures for stats object types 5.1 Adding new stats objects 5.2 Retiring stats objects 6. RTCStatsType 6.1 RTCStatsType enum 7. Stats dictionaries 7.1 The RTP statistics hierarchy 7.2 RTCRtpStreamStats dictionary 7.3 RTCCodecStats dictionary 7.3.1 RTCCodecType enum 7.4 RTCReceivedRtpStreamStats dictionary 7.5 RTCInboundRtpStreamStats dictionary 7.6 RTCRemoteInboundRtpStreamStats dictionary 7.7 RTCSentRtpStreamStats dictionary 7.8 RTCOutboundRtpStreamStats dictionary 7.9 RTCQualityLimitationReason enum 7.10 RTCRemoteOutboundRtpStreamStats dictionary 7.11 RTCMediaSourceStats dictionary 7.12 RTCAudioSourceStats dictionary 7.13 RTCVideoSourceStats dictionary 7.14 RTCRtpContributingSourceStats dictionary 7.15 RTCPeerConnectionStats dictionary 7.16 RTCRtpTransceiverStats dictionary 7.17 RTCMediaHandlerStats dictionary 7.18 RTCVideoHandlerStats dictionary 7.19 RTCVideoSenderStats dictionary 7.20 RTCSenderVideoTrackAttachmentStats dictionary 7.21 RTCVideoReceiverStats dictionary 7.22 RTCAudioHandlerStats dictionary 7.23 RTCAudioSenderStats dictionary 7.24 RTCSenderAudioTrackAttachmentStats dictionary 7.25 RTCAudioReceiverStats dictionary 7.26 RTCDataChannelStats dictionary 7.27 RTCTransportStats dictionary 7.28 RTCSctpTransportStats dictionary 7.29 RTCIceCandidateStats dictionary 7.30 RTCIceCandidatePairStats dictionary 7.30.1 RTCStatsIceCandidatePairState enum 7.31 RTCCertificateStats dictionary 7.32 RTCIceServerStats dictionary 8. Obsolete stats 8.1 Obsolete RTCReceiverVideoTrackAttachmentStats dictionary 8.2 Obsolete RTCReceiverAudioTrackAttachmentStats dictionary 8.3 Obsolete RTCCodecStats members 8.4 Obsolete RTCIceCandidateStats members 8.5 Obsolete RTCIceCandidatePairStats members 8.6 Obsolete RTCRtpStreamStats members 8.7 Obsolete RTCInboundRtpStreamStats members 8.8 Obsolete RTCAudioHandlerStats members 8.9 Obsolete RTCAudioSenderStats members 8.10 Obsolete RTCAudioReceiverStats members 8.11 Obsolete RTCVideoHandlerStats members 8.12 Obsolete RTCVideoSenderStats members 8.13 Obsolete RTCVideoReceiverStats members 9. Examples 9.1 Example of a stats application 10. Security and Privacy Considerations 11. Change Log 11.1 Changes between 31 Mar 2017 and 14 Jun 2017 11.2 Changes between 14 Dec 2016 and 30 Mar 2017 11.3 Changes since 21 sep 2016 11.4 Changes since 26 May 2016 11.5 Changes since 23 October 2015 11.6 Changes since 03 February 2015 11.7 Changes since 30 September 2014 11.8 Acknowledgements A. References A.1 Normative references A.2 Informative references 1. Introduction This section is non-normative. 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 in this document are to be interpreted as described in BCP 14 [ RFC2119 ] [ RFC8174 ] when, and only when, they appear in all capitals, as shown here. 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): An implementation MUST support generating statistics for the type RTCInboundRtpStreamStats, with attributes packetsReceived, bytesReceived, packetsLost, and jitter. 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. It MAY support generating other stats. 3. Terminology The terms MediaStream , MediaStreamTrack , and Consumer are defined in [ GETUSERMEDIA ]. The terms RTCPeerConnection , RTCDataChannel , RTCDtlsTransport , RTCDtlsTransportState , RTCIceTransport , RTCIceRole , RTCSctpTransport , 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 and RTCCertificate are defined in [ WEBRTC ]. The terms DOMHighResTimeStamp performance.timeOrigin and performance.now() are defined in [ HIGHRES-TIME ]. 4. Basic concepts This section is non-normative. 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 object s, each of which is a dictionary inheriting directly or indirectly from the RTCStats dictionary. This API is normatively defined in [ WEBRTC ], but is reproduced here for ease of reference. WebIDL 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. 4.1 Guidelines for design of stats objects 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 RTCStatsType 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 they stop appearing in subsequent calls to getStats(). 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 object s for deleted monitored object s 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 a stats object is deleted , subsequent getStats() calls MUST NOT return stats for that monitored object . 5. Maintenance procedures for stats object types 5.1 Adding new stats objects 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. 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 . 6.1 RTCStatsType enum WebIDL enum RTCStatsType { " codec " , " inbound-rtp " , " outbound-rtp " , " remote-inbound-rtp " , " remote-outbound-rtp " , " media-source " , " csrc " , " peer-connection " , " data-channel " , " stream " , " track " , " transceiver " , " sender " , " receiver " , " transport " , " sctp-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 This is now obsolete. Contains statistics related to a specific MediaStream. It is accessed by the obsolete dictionary 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. transceiver Statistics related to a specific RTCRtpTransceiver. It is accessed by the RTCRtpTransceiverStats dictionary. 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 . sctp-transport SCTP transport statistics related to an RTCSctpTransport object. It is accessed by the RTCSctpTransportStats dictionary. 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 . 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 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 object s starts when the RTP stream is first used: When the first RTP packet is sent or received on the SSRC it represents, or when the first RTCP packet is sent or received that refers to the SSRC of the RTP stream. RTP monitored objects are 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). RTCOutboundRtpStreamStats : Stats measured locally. RTCRemoteOutboundRtpStreamStats : Stats relevant to the remote sending end of an RTP stream, usually computed based on an RTCP SR. 7.2 RTCRtpStreamStats dictionary WebIDL dictionary RTCRtpStreamStats : RTCStats { unsigned long ssrc ; DOMString kind ; DOMString transportId ; DOMString codecId ; }; Dictionary RTCRtpStreamStats Members 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. 7.3 RTCCodecStats dictionary WebIDL dictionary RTCCodecStats : RTCStats { unsigned long payloadType ; RTCCodecType codecType ; DOMString transportId ; DOMString mimeType ; unsigned long clockRate ; unsigned long channels ; DOMString sdpFmtpLine ; }; Dictionary RTCCodecStats Members payloadType of type unsigned long Payload type as used in RTP encoding or decoding. 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. 7.3.1 RTCCodecType enum WebIDL enum RTCCodecType { " encode " , " decode " , }; Enumeration description encode The attached RTCCodecStats represents a media format that is being encoded, or that the implementation is prepared to encode. decode The attached RTCCodecStats represents a media format that the implementation is prepared to decode. 7.4 RTCReceivedRtpStreamStats dictionary WebIDL 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 ; }; Dictionary RTCReceivedRtpStreamStats Members packetsReceived of type unsigned long long Total number of RTP packets received for this SSRC. At the receiving endpoint, this is calculated as defined in [ RFC3550 ] section 6.4.1. At the sending endpoint the packetsReceived 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 ]. 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. WebIDL dictionary RTCInboundRtpStreamStats : RTCReceivedRtpStreamStats { DOMString trackId ; DOMString receiverId ; DOMString remoteId ; unsigned long framesDecoded ; unsigned long keyFramesDecoded ; unsigned long frameWidth ; unsigned long frameHeight ; unsigned long frameBitDepth ; 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 ; }; Dictionary RTCInboundRtpStreamStats Members trackId of type DOMString The identifier of the stats object representing the receiving track, an 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. frameBitDepth of type unsigned long Only valid for video. Represents the bit depth per pixel of the last decoded frame. Typical values are 24, 30, or 36 bits. 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 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 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 . Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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 ). Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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 . Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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 ). Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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 . Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. The "audio energy of a sample" refers to the highest energy of any channel. 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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. 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. WebIDL dictionary RTCRemoteInboundRtpStreamStats : RTCReceivedRtpStreamStats { DOMString localId ; double roundTripTime ; double totalRoundTripTime ; double fractionLost ; unsigned long long reportsReceived ; unsigned long long roundTripTimeMeasurements ; }; Dictionary RTCRemoteInboundRtpStreamStats Members localId of type DOMString The localId is used for looking up the local RTCOutboundRtpStreamStats object for the same SSRC. roundTripTime of type double Estimated round trip time for this SSRC based on the RTCP timestamps in the RTCP Receiver Report (RR) and measured in seconds. Calculated as defined in section 6.4.1. of [ RFC3550 ]. If no RTCP Receiver Report is received with a DLSR value other than 0, the round trip time is left undefined. totalRoundTripTime of type double Represents the cumulative sum of all round trip time measurements in seconds since the beginning of the session. The individual round trip time is calculated based on the RTCP timestamps in the RTCP Receiver Report (RR) [ RFC3550 ], hence undefined roundtrip 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. 7.7 RTCSentRtpStreamStats dictionary WebIDL dictionary RTCSentRtpStreamStats : RTCRtpStreamStats { unsigned long packetsSent ; unsigned long long bytesSent ; }; Dictionary RTCSentRtpStreamStats Members packetsSent of type unsigned long long Total number of RTP packets sent for this SSRC. 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. 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. WebIDL 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 ; unsigned long frameBitDepth ; 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 ; }; Dictionary RTCOutboundRtpStreamStats Members trackId of type DOMString The identifier of the stats object representing the current track attachment to the sender of this stream, an RTCSenderAudioTrackAttachmentStats or RTCSenderVideoTrackAttachmentStats . mediaSourceId of type DOMString The identifier of the stats object representing the track currently attached to the sender of this stream, an 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. frameBitDepth of type unsigned long Only valid for video. Represents the bit depth per pixel of the last encoded frame. Typical values are 24, 30, or 36 bits. 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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. 7.9 RTCQualityLimitationReason enum WebIDL 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. 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. WebIDL dictionary RTCRemoteOutboundRtpStreamStats : RTCSentRtpStreamStats { DOMString localId ; DOMHighResTimeStamp remoteTimestamp ; unsigned long long reportsSent ; }; Dictionary RTCRemoteOutboundRtpStreamStats Members localId of type DOMString The localId is used for looking up the local RTCInboundRtpStreamStats object for the same SSRC. remoteTimestamp of type DOMHighResTimeStamp remoteTimestamp , of type DOMHighResTimeStamp [ HIGHRES-TIME ], represents the remote timestamp at which these statistics were sent by the remote endpoint. This differs from 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. 7.11 RTCMediaSourceStats dictionary The RTCMediaSourceStats dictionary represents a track that is currently attached to one or more senders. It contains information about media sources such as frame rate and resolution prior to encoding. This is the media passed from the MediaStreamTrack to the RTCRtpSender s. This is in contrast to RTCOutboundRtpStreamStats whose members describe metrics as measured after the encoding step. For example, a track may be captured from a high-resolution camera, its frames downscaled due to track constraints and then further downscaled by the encoders due to CPU and network conditions. This dictionary reflects the video frames or audio samples passed out from the track - after track constraints have been applied but before any encoding or further donwsampling occurs. Media source objects are of either subdictionary RTCAudioSourceStats or RTCVideoSourceStats . The 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. WebIDL dictionary RTCMediaSourceStats : RTCStats { DOMString trackIdentifier ; DOMString kind ; }; Dictionary RTCMediaSourceStats Members trackIdentifier of type DOMString The value of the MediaStreamTrack 's id attribute. kind of type DOMString The value of the MediaStreamTrack 's kind attribute. This is either "audio" or "video" . If it is "audio" then this stats object is of type RTCAudioSourceStats . If it is "video" then this stats object is of type RTCVideoSourceStats . 7.12 RTCAudioSourceStats dictionary The RTCAudioSourceStats dictionary represents an audio track that is attached to one or more senders. It is an RTCMediaSourceStats whose kind is "audio" . WebIDL dictionary RTCAudioSourceStats : RTCMediaSourceStats { double audioLevel ; double totalAudioEnergy ; double totalSamplesDuration ; double echoReturnLoss ; double echoReturnLossEnhancement ; }; Dictionary RTCAudioSourceStats Members 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. The "audio energy of a sample" refers to the highest energy of any channel. totalSamplesDuration of type double Represents the audio duration of the media source. For audio durations of remotely sourced tracks, see 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. When calculating echo return loss, the channel of the least audio energy is considered for any sample. 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. Note: If multiple audio channels are used, metrics based on samples do not increment at a higher rate. "A sample" refers to having a sample in either channel - simultaneously having samples in multiple channels counts as a single sample. When calculating echo return loss enhancement, the channel of the least audio energy is considered for any sample. 7.13 RTCVideoSourceStats dictionary The RTCVideoSourceStats dictionary represents a video track that is attached to one or more senders. It is an RTCMediaSourceStats whose kind is "video" . WebIDL dictionary RTCVideoSourceStats : RTCMediaSourceStats { unsigned long width ; unsigned long height ; unsigned long bitDepth ; unsigned long frames ; unsigned long framesPerSecond ; }; Dictionary RTCVideoSourceStats Members width of type unsigned long The width, in pixels, of the last frame originating from this source. Before a frame has been produced this 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. bitDepth of type unsigned long The bit depth per pixel 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. 7.14 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 . WebIDL dictionary RTCRtpContributingSourceStats : RTCStats { unsigned long contributorSsrc ; DOMString inboundRtpStreamId ; unsigned long packetsContributedTo ; double audioLevel ; }; Dictionary RTCRtpContributingSourceStats Members 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) . 7.15 RTCPeerConnectionStats dictionary WebIDL dictionary RTCPeerConnectionStats : RTCStats { unsigned long dataChannelsOpened ; unsigned long dataChannelsClosed ; unsigned long dataChannelsRequested ; unsigned long dataChannelsAccepted ; }; Dictionary RTCPeerConnectionStats Members 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. 7.16 RTCRtpTransceiverStats dictionary An RTCRtpTransceiverStats stats object represents an RTCRtpTransceiver of an RTCPeerConnection. It appears as soon as the monitored RTCRtpTransceiver object is created, such as by invoking addTransceiver, addTrack or setRemoteDescription. RTCRtpTransceiverStats objects can only be deleted if the corresponding RTCRtpTransceiver is removed - this can only happen if a remote description is rolled back. WebIDL dictionary RTCRtpTransceiverStats { DOMString senderId ; DOMString receiverId ; DOMString mid ; }; Dictionary RTCRtpTransceiverStats Members senderId of type DOMString The identifier of the stats object representing the RTCRtpSender associated with the RTCRtpTransceiver represented by this stats object. receiverId of type DOMString The identifier of the stats object representing the RTCRtpReceiver associated with the RTCRtpTransceiver represented by this stats object. mid of type DOMString If the RTCRtpTransceiver that this stats object represents has a mid value that is not null, this is that value, otherwise this value is undefined. 7.17 RTCMediaHandlerStats dictionary WebIDL dictionary RTCMediaHandlerStats : RTCStats { DOMString trackIdentifier ; boolean remoteSource ; boolean ended ; DOMString kind ; RTCPriorityType priority ; }; Dictionary RTCMediaHandlerStats Members trackIdentifier of type DOMString Represents the id property of the track. remoteSource of type boolean 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 ]. priority of type RTCPriorityType Indicates the priority set for the track. It is specified in [ RTCWEB-TRANSPORT ], Section 4. 7.18 RTCVideoHandlerStats dictionary WebIDL dictionary RTCVideoHandlerStats : RTCMediaHandlerStats { }; Dictionary RTCVideoHandlerStats Members 7.19 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. WebIDL dictionary RTCVideoSenderStats : RTCVideoHandlerStats { DOMString mediaSourceId ; }; Dictionary RTCVideoSenderStats Members mediaSourceId of type DOMString The identifier of the stats object representing the track currently attached to this sender, an RTCMediaSourceStats . 7.20 RTCSenderVideoTrackAttachmentStats dictionary 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. WebIDL dictionary RTCSenderVideoTrackAttachmentStats : RTCVideoSenderStats { }; Dictionary RTCSenderVideoTrackAttachmentStats Members 7.21 RTCVideoReceiverStats dictionary 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. WebIDL dictionary RTCVideoReceiverStats : RTCVideoHandlerStats { }; Dictionary RTCVideoReceiverStats Members 7.22 RTCAudioHandlerStats dictionary WebIDL dictionary RTCAudioHandlerStats : RTCMediaHandlerStats { }; Dictionary RTCAudioHandlerStats Members 7.23 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. WebIDL dictionary RTCAudioSenderStats : RTCAudioHandlerStats { DOMString mediaSourceId ; }; Dictionary RTCAudioSenderStats Members mediaSourceId of type DOMString The identifier of the stats object representing the track currently attached to this sender, an RTCMediaSourceStats . 7.24 RTCSenderAudioTrackAttachmentStats dictionary 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. WebIDL dictionary RTCSenderAudioTrackAttachmentStats : RTCAudioSenderStats { }; Dictionary RTCSenderAudioTrackAttachmentStats Members 7.25 RTCAudioReceiverStats dictionary 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. WebIDL dictionary RTCAudioReceiverStats : RTCAudioHandlerStats { }; Dictionary RTCAudioReceiverStats Members 7.26 RTCDataChannelStats dictionary WebIDL 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 ; }; Dictionary RTCDataChannelStats Members label of type DOMString The "label" value of the RTCDataChannel object. protocol of type DOMString The "protocol" value of the RTCDataChannel object. dataChannelIdentifier of type long The "id" attribute of the RTCDataChannel object. transportId of type DOMString A stats object reference for the transport used to carry this datachannel. state of type RTCDataChannelState The "readyState" value of the RTCDataChannel object. messagesSent of type unsigned long Represents the total number of API "message" events sent. bytesSent of type unsigned long long Represents the total number of payload bytes sent on this RTCDatachannel , i.e., not including headers or padding. messagesReceived of type unsigned long Represents the total number of API "message" events received. bytesReceived of type unsigned long long Represents the total number of bytes received on this RTCDatachannel , i.e., not including headers or padding. 7.27 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 MediaStreamTrack s in the bundle group. If bundling is not used, different MediaStreamTrack will use different transports. RTCP multiplexing and bundling are described in [ WEBRTC ]. WebIDL 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 ; }; Dictionary RTCTransportStats Members packetsSent of type unsigned long long Represents the total number of packets sent over this transport. packetsReceived of type unsigned long long Represents the total number of packets received on this transport. bytesSent of type unsigned long long Represents the total number of payload bytes sent on this 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. 7.28 RTCSctpTransportStats dictionary An RTCSctpTransportStats object represents the stats corresponding to an RTCSctpTransport described in [ WEBRTC ]. WebIDL dictionary RTCSctpTransportStats : RTCStats { double smoothedRoundTripTime ; }; Dictionary RTCSctpTransportStats Members smoothedRoundTripTime of type double The latest smoothed round-trip time value, corresponding to spinfo_srtt defined in [ RFC6458 ] but converted to seconds. If there has been no round-trip time measurements yet, this value is undefined. 7.29 RTCIceCandidateStats dictionary RTCIceCandidateStats reflects the properties of a candidate in Section 15.1 of [ RFC5245 ]. It corresponds to a RTCIceCandidate object. WebIDL dictionary RTCIceCandidateStats : RTCStats { DOMString transportId ; DOMString ? address ; long port ; DOMString protocol ; RTCIceCandidateType candidateType ; long priority ; DOMString url ; DOMString relayProtocol ; }; Dictionary RTCIceCandidateStats Members transportId of type DOMString It is a unique identifier that is associated to the object that was inspected to produce the RTCTransportStats associated with this candidate. address of type DOMString It is the address of the candidate, allowing for IPv4 addresses, IPv6 addresses, and fully qualified domain names (FQDNs). See [ RFC5245 ] section 15.1 for details. The user agent should make sure that only remote candidate addresses that the web application has configured on the corresponding RTCPeerConnection are exposed; This is especially important for peer reflexive remote candidates. By default, the user agent MUST leave the '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. 7.30 RTCIceCandidatePairStats dictionary WebIDL 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 ; }; Dictionary RTCIceCandidatePairStats Members transportId of type DOMString It is a unique identifier that is associated to the object that was inspected to produce the RTCTransportStats associated with this candidate pair. localCandidateId of type DOMString It is a unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the local candidate associated with this candidate pair. remoteCandidateId of type DOMString It is a unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the remote candidate associated with this candidate pair. state of type RTCStatsIceCandidatePairState Represents the state of the checklist for the local and remote candidates in a pair. nominated of type boolean Related to updating the nominated flag described in Section 7.1.3.2.4 of [ RFC5245 ]. packetsSent of type unsigned long long Represents the total number of packets sent on this candidate pair. packetsReceived of type unsigned long long Represents the total number of packets received on this candidate pair. bytesSent of type unsigned long long Represents the total number of payload bytes sent on this candidate pair, i.e., not including headers 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. 7.30.1 RTCStatsIceCandidatePairState enum WebIDL 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 ]. 7.31 RTCCertificateStats dictionary WebIDL dictionary RTCCertificateStats : RTCStats { DOMString fingerprint ; DOMString fingerprintAlgorithm ; DOMString base64Certificate ; DOMString issuerCertificateId ; }; Dictionary RTCCertificateStats Members fingerprint of type DOMString The fingerprint of the certificate. Only use the fingerprint value as defined in Section 5 of [ RFC4572 ]. fingerprintAlgorithm of type DOMString The hash function used to compute the certificate fingerprint. For instance, "sha-256". base64Certificate of type DOMString The DER-encoded base-64 representation of the certificate. issuerCertificateId of type DOMString The issuerCertificateId refers to the stats object that contains the next certificate in the certificate chain. If the current certificate is at the end of the chain (i.e. a self-signed certificate), this will not be set. 7.32 RTCIceServerStats dictionary WebIDL dictionary RTCIceServerStats : RTCStats { DOMString url ; long port ; DOMString protocol ; unsigned long totalRequestsSent ; unsigned long totalResponsesReceived ; double totalRoundTripTime ; }; Dictionary RTCIceServerStats Members 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. 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. 8. Obsolete stats WebIDL dictionary RTCMediaStreamStats : RTCStats { DOMString streamIdentifier ; sequence < DOMString > trackIds ; }; Obsolete RTCMediaStreamStats members This entire dictionary was made obsolete in September, 2019 due to sender, receiver and transceiver stats objects being a better fit to describe the modern RTCPeerConnection model (Unified Plan). streamIdentifier of type DOMString stream.id property trackIds of type sequence<DOMString> This is the id of the stats object, not the track.id . 8.1 Obsolete RTCReceiverVideoTrackAttachmentStats dictionary The RTCReceiverVideoTrackAttachmentStats is a copy of RTCVideoReceiverStats . It adds no new information, it only exists for backwards-compatibility reasons as an obsolete dictionary. WebIDL dictionary RTCReceiverVideoTrackAttachmentStats : RTCVideoReceiverStats {}; 8.2 Obsolete RTCReceiverAudioTrackAttachmentStats dictionary The RTCReceiverAudioTrackAttachmentStats is a copy of RTCAudioReceiverStats . It adds no new information, it only exists for backwards-compatibility reasons as an obsolete dictionary. WebIDL dictionary RTCReceiverAudioTrackAttachmentStats : RTCAudioReceiverStats {}; 8.3 Obsolete RTCCodecStats members WebIDL partial dictionary RTCCodecStats { DOMString implementation ; }; implementation of type DOMString This was moved to RTCInboundRtpStreamStats.decoderImplementation and RTCOutboundRtpStreamStats.encoderImplementation in August 2019. 8.4 Obsolete RTCIceCandidateStats members WebIDL partial dictionary RTCIceCandidateStats { boolean deleted = false; boolean isRemote ; }; deleted of type boolean This field was obsoleted by the statsended event in [ WEBRTC ], which has now also been obsolete. deleted is no longer applicable beacause if the ice candidate is deleted it no longer appears in getStats(). isRemote of type boolean false indicates that this represents a local candidate; true indicates that this represents a remote candidate. 8.5 Obsolete RTCIceCandidatePairStats members WebIDL 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. 8.6 Obsolete RTCRtpStreamStats members WebIDL 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. 8.7 Obsolete RTCInboundRtpStreamStats members WebIDL partial dictionary RTCInboundRtpStreamStats { double fractionLost ; }; fractionLost of type double This field was moved to RTCRemoteInboundRtpStreamStats in December 2017. 8.8 Obsolete RTCAudioHandlerStats members WebIDL 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. 8.9 Obsolete RTCAudioSenderStats members WebIDL 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. 8.10 Obsolete RTCAudioReceiverStats members WebIDL 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. 8.11 Obsolete RTCVideoHandlerStats members WebIDL 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). 8.12 Obsolete RTCVideoSenderStats members WebIDL 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. 8.13 Obsolete RTCVideoReceiverStats members WebIDL 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. 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 ( ) { 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; } }); } 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. 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 : RTCAudioHandlerStats.audioLevel . RTCRtpContributingSourceStats (the whole dictionary). 11. Change Log This section will be removed before publication. The entries are in reverse chronological order. 11.1 Changes between 31 Mar 2017 and 14 Jun 2017 [#184] Stat for how much time it takes to encode video [#185] Added sample counters to RTCMediaStreamTrackStats [#187] Metrics to keep track of audio and video sync added to RTCMediaStreamTrackStats members [#191] Refactor out isRemote [#192] Update example to match webrtc spec's + senders.getStats [#194] Adding RTCRTPContributingSourceStats stats report object [#195] Adding "codec type" and transportId to RTCCodecStats [#196] Removed writable/readable, and added consentExpiredTimestamp to RTCIceCandidatePairStats [#197] packetsSent/Received added to RTCTransportStats [#198] packetsSent/Received added to RTCIceCandidatePairStats [#199] framesCaptured added to RTCMediaStreamTrackStats [#203] Packets discarded on send [#208] voiceActivityFlag added to RTCMediaStreamTrackStats [#209] definition for 'framesPerSecond' [#210] Added paragraph about timestamp and reference DOMHighResTimeStamp [#211] timestamps needed for circuit breakers and a/v sync [#212] added priority to RTCMediaStreamTrackStats [#213] added last[Request/Response]Timestamp and firstRequestTimestamp to RTCIceCandidatePairStats [#216] stats for number of circuit breaker triggered [#217] added jitter buffer delay added to RTCMediaStreamTrackStats [#218] added iceRole to RTCTransportStats [#221] Added audio level stat that can be used to compute averages 11.2 Changes between 14 Dec 2016 and 30 Mar 2017 [#102] certificate stats: describe issuerCertificateId [#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. A. References A.1 Normative references [ECHO] Digital network echo cancellers . ITU-T G.168. ITU-T. Standard. URL: https://www.itu.int/rec/T-REC-G.168/en [GETUSERMEDIA] Media Capture and Streams . Daniel Burnett; Adam Bergkvist; Cullen Jennings; Anant Narayanan; Bernard Aboba; Jan-Ivar Bruaroey; Henrik Boström. W3C. 2 July 2019. W3C Candidate Recommendation. URL: https://www.w3.org/TR/mediacapture-streams/ [HIGHRES-TIME] High Resolution Time Level 2 . Ilya Grigorik. W3C. 21 November 2019. W3C Recommendation. URL: https://www.w3.org/TR/hr-time-2/ [IANA-DTLS-SRTP] DTLS-SRTP Protection Profiles . IANA. URL: https://www.iana.org/assignments/srtp-protection/srtp-protection.xhtml [IANA-TLS-CIPHERS] TLS Cipher Suite Registry . IANA. URL: https://www.iana.org/assignments/tls-parameters/tls-parameters.xhtml#tls-parameters-4 [IANA-TLS-GROUPS] TLS Supported Groups Registry . IANA. URL: https://www.iana.org/assignments/tls-parameters/tls-parameters.xhtml#tls-parameters-8 [JSEP] Javascript Session Establishment Protocol . Justin Uberti; Cullen Jennings; Eric Rescorla. IETF. 16 January 2017. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/ [RFC2032] RTP Payload Format for H.261 Video Streams . T. Turletti; C. Huitema. IETF. October 1996. Proposed Standard. URL: https://tools.ietf.org/html/rfc2032 [RFC2119] Key words for use in RFCs to Indicate Requirement Levels . S. Bradner. IETF. March 1997. Best Current Practice. URL: https://tools.ietf.org/html/rfc2119 [RFC2474] Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers . K. Nichols; S. Blake; F. Baker; D. Black. IETF. December 1998. Proposed Standard. URL: https://tools.ietf.org/html/rfc2474 [RFC3550] RTP: A Transport Protocol for Real-Time Applications . H. Schulzrinne; S. Casner; R. Frederick; V. Jacobson. IETF. July 2003. Internet Standard. URL: https://tools.ietf.org/html/rfc3550 [RFC3611] RTP Control Protocol Extended Reports (RTCP XR) . T. Friedman, Ed.; R. Caceres, Ed.; A. Clark, Ed.. IETF. November 2003. Proposed Standard. URL: https://tools.ietf.org/html/rfc3611 [RFC3711] The Secure Real-time Transport Protocol (SRTP) . M. Baugher; D. McGrew; M. Naslund; E. Carrara; K. Norrman. IETF. March 2004. Proposed Standard. URL: https://tools.ietf.org/html/rfc3711 [RFC3890] A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP) . M. Westerlund. IETF. September 2004. Proposed Standard. URL: https://tools.ietf.org/html/rfc3890 [RFC4572] Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP) . J. Lennox. IETF. July 2006. Proposed Standard. URL: https://tools.ietf.org/html/rfc4572 [RFC4585] Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF) . J. Ott; S. Wenger; N. Sato; C. Burmeister; J. Rey. IETF. July 2006. Proposed Standard. URL: https://tools.ietf.org/html/rfc4585 [RFC4587] RTP Payload Format for H.261 Video Streams . R. Even. IETF. August 2006. Proposed Standard. URL: https://tools.ietf.org/html/rfc4587 [RFC5104] Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF) . S. Wenger; U. Chandra; M. Westerlund; B. Burman. IETF. February 2008. Proposed Standard. URL: https://tools.ietf.org/html/rfc5104 [RFC5226] Guidelines for Writing an IANA Considerations Section in RFCs . T. Narten; H. Alvestrand. IETF. May 2008. Best Current Practice. URL: https://tools.ietf.org/html/rfc5226 [RFC5245] Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols . J. Rosenberg. IETF. April 2010. Proposed Standard. URL: https://tools.ietf.org/html/rfc5245 [RFC5764] Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP) . D. McGrew; E. Rescorla. IETF. May 2010. Proposed Standard. URL: https://tools.ietf.org/html/rfc5764 [RFC6184] RTP Payload Format for H.264 Video . Y.-K. Wang; R. Even; T. Kristensen; R. Jesup. IETF. May 2011. Proposed Standard. URL: https://tools.ietf.org/html/rfc6184 [RFC6386] VP8 Data Format and Decoding Guide . J. Bankoski; J. Koleszar; L. Quillio; J. Salonen; P. Wilkins; Y. Xu. IETF. November 2011. Informational. URL: https://tools.ietf.org/html/rfc6386 [RFC6458] Sockets API Extensions for the Stream Control Transmission Protocol (SCTP) . R. Stewart; M. Tuexen; K. Poon; P. Lei; V. Yasevich. IETF. December 2011. Informational. URL: https://tools.ietf.org/html/rfc6458 [RFC6464] A Real-time Transport Protocol (RTP) Header Extension for Client-to-Mixer Audio Level Indication . J. Lennox, Ed.; E. Ivov; E. Marocco. IETF. December 2011. Proposed Standard. URL: https://tools.ietf.org/html/rfc6464 [RFC6465] A Real-time Transport Protocol (RTP) Header Extension for Mixer-to-Client Audio Level Indication . E. Ivov, Ed.; E. Marocco, Ed.; J. Lennox. IETF. December 2011. Proposed Standard. URL: https://tools.ietf.org/html/rfc6465 [RFC6958] RTP Control Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Loss Metric Reporting . A. Clark; S. Zhang; J. Zhao; Q. Wu, Ed.. IETF. May 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc6958 [RFC7002] RTP Control Protocol (RTCP) Extended Report (XR) Block for Discard Count Metric Reporting . A. Clark; G. Zorn; Q. Wu. IETF. September 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc7002 [RFC7003] RTP Control Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Discard Metric Reporting . A. Clark; R. Huang; Q. Wu, Ed.. IETF. September 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc7003 [RFC7004] RTP Control Protocol (RTCP) Extended Report (XR) Blocks for Summary Statistics Metrics Reporting . G. Zorn; R. Schott; Q. Wu, Ed.; R. Huang. IETF. September 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc7004 [RFC7509] RTP Control Protocol (RTCP) Extended Report (XR) for Post-Repair Loss Count Metrics . R. Huang; V. Singh. IETF. May 2015. Proposed Standard. URL: https://tools.ietf.org/html/rfc7509 [RFC7656] A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources . J. Lennox; K. Gross; S. Nandakumar; G. Salgueiro; B. Burman, Ed.. IETF. November 2015. Informational. URL: https://tools.ietf.org/html/rfc7656 [RFC7675] Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness . M. Perumal; D. Wing; R. Ravindranath; T. Reddy; M. Thomson. IETF. October 2015. Proposed Standard. URL: https://tools.ietf.org/html/rfc7675 [RFC7982] Measurement of Round-Trip Time and Fractional Loss Using Session Traversal Utilities for NAT (STUN) . P. Martinsen; T. Reddy; D. Wing; V. Singh. IETF. September 2016. Proposed Standard. URL: https://tools.ietf.org/html/rfc7982 [RFC8015] RTP Control Protocol (RTCP) Extended Report (XR) Block for Independent Reporting of Burst/Gap Discard Metrics . V. Singh; C. Perkins; A. Clark; R. Huang. IETF. November 2016. Proposed Standard. URL: https://tools.ietf.org/html/rfc8015 [RFC8083] Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions . C. Perkins; V. Singh. IETF. March 2017. Proposed Standard. URL: https://tools.ietf.org/html/rfc8083 [RFC8174] Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words . B. Leiba. IETF. May 2017. Best Current Practice. URL: https://tools.ietf.org/html/rfc8174 [RTCWEB-TRANSPORT] Transports for RTCWEB . H. Alvestrand. IETF. 31 October 2016. Active Internet-Draft. URL: https://tools.ietf.org/html/draft-ietf-rtcweb-transports [STUN-PATH-CHAR] Discovery of path characteristics using STUN . T. Reddy; D. Wing; P. Martinsen; V. Singh. IETF. Internet Draft. URL: https://tools.ietf.org/html/draft-reddy-tram-stun-path-data [WEBIDL] Web IDL . Boris Zbarsky. W3C. 15 December 2016. W3C Editor's Draft. URL: https://heycam.github.io/webidl/ [WEBRTC] WebRTC 1.0: Real-time Communication Between Browsers . Cullen Jennings; Henrik Boström; Jan-Ivar Bruaroey; Adam Bergkvist; Daniel Burnett; Anant Narayanan; Bernard Aboba; Taylor Brandstetter. W3C. 13 December 2019. W3C Candidate Recommendation. URL: https://www.w3.org/TR/webrtc/ [XRBLOCK-STATS] RTCP XR Metrics for WebRTC . Varun Singh; Rachel Huang; Roni Even; Dan Romascanu; Lingli Deng. IETF. Internet Draft. URL: https://tools.ietf.org/html/draft-ietf-xrblock-rtcweb-rtcp-xr-metrics A.2 Informative references [API-DESIGN-PRINCIPLES] API Design Principles . Domenic Denicola. 29 December 2015. URL: https://w3ctag.github.io/design-principles/ ↑

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