Abstract

This document defines a set of Javascript APIs that allow access to the statistical information about a PeerConnection.

Status of This Document

This section describes the status of this document at the time of its publication. Other documents may supersede this document. A list of current W3C publications and the latest revision of this technical report can be found in the W3C technical reports index at https://www.w3.org/TR/.

This document is incomplete, and as such is not yet suitable for implementation. However, early experimentation is encouraged.

This document was published by the Web Real-Time Communications Working Group as an Editor's Draft. If you wish to make comments regarding this document, please send them to public-webrtc@w3.org (subscribe, archives). All comments are welcome.

Publication as an Editor's Draft does not imply endorsement by the W3C Membership. This is a draft document and may be updated, replaced or obsoleted by other documents at any time. It is inappropriate to cite this document as other than work in progress.

This document was produced by a group operating under the 5 February 2004 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 September 2015 W3C Process Document.

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 APIs and 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 word MUST is to be interpreted as described in [RFC2119].

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

Implementations that use ECMAScript to implement the APIs 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.

3. Terminology

The EventHandler represents a callback used for event handlers as defined in [HTML5].

The concepts queue a task, and fires a simple event are defined in [HTML5].

The terms event, event handlers, and event handler event types are defined in [HTML5].

The terms MediaStream, MediaStreamTrack, and Consumer are defined in [GETUSERMEDIA].

The terms RTCPeerConnection, RTCDataChannel are defined in [ WEBRTC].

4. RTCStatsType

RTCStatsType object is initialized to the name of the dictionary that the RTCStats represents.
Note
OPEN ISSUE: Need to define an IANA registry for the RTCStatsType and populate it with the following set as baseline.

4.1 RTCStatsType DOMString

RTCStatsType is a equal to one of the following strings defined in [IANA-TOBE]:

"inbound-rtp"

Statistics for the inbound RTP stream that is currently received with this RTCPeerConnection object. It is accessed by the RTCInboundRTPStreamStats .

"outbound-rtp"

Statistics for the outbound RTP stream that is currently sent with this RTCPeerConnection object. It is accessed by the RTCOutboundRTPStreamStats .

"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 .

"track"

Contains the sequence of tracks related to a specific media stream and the corresponding media-level metrics. It is accessed by the RTCMediaStreamStats .

"transport"

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

"candidate-pair"

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

"local-candidate"

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

"remote-candidate"

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

5. Stats dictionaries

5.1 RTCRTPStreamStats dictionary 

dictionary RTCRTPStreamStats : RTCStats {
    DOMString     ssrc;
    DOMString     associateStatsId;
    boolean       isRemote = false;
    DOMString     mediaType;
    DOMString     mediaTrackId;
    DOMString     transportId;
    DOMString     codecId;
    unsigned long firCount;
    unsigned long pliCount;
    unsigned long nackCount;
    unsigned long sliCount;
};

Dictionary RTCRTPStreamStats Members

ssrc of type DOMString
associateStatsId of type DOMString

The associateStatsId is used for looking up the corresponding (local/remote) RTCStats object for a given SSRC.

isRemote of type boolean, defaulting to false

false indicates that the statistics are measured locally, while true indicates that the measurements were done at the remote endpoint and reported in an RTCP RR/XR.

mediaType of type DOMString

Either "audio" or "video". This MUST match the media type part of the information in the corresponding codec member of RTCCodecStats.

mediaTrackId of type DOMString
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
firCount of type unsigned long

Count the total number of Full Intra Request (FIR) packets received by the sender. This metric is only valid for video and is sent by receiver. Calculated as defined in [RFC5104] section 4.3.1. and does not use the metric indicated in [RFC2032], because it was deprecated by [RFC4587].

pliCount of type unsigned long

Count the total number of Packet Loss Indication (PLI) packets received by the sender and is sent by receiver. Calculated as defined in [RFC4585] section 6.3.1.

nackCount of type unsigned long

Count the total number of Negative ACKnowledgement (NACK) packets received by the sender and is sent by receiver. Calculated as defined in [RFC4585] section 6.2.1.

sliCount of type unsigned long

Count the total number of Slice Loss Indication (SLI) packets received by the sender. This metric is only valid for video and is sent by receiver. Calculated as defined in [RFC4585] section 6.3.2.

5.2 RTCCodecStats dictionary 

dictionary RTCCodecStats : RTCStats {
    unsigned long payloadType;
    DOMString     codec;
    unsigned long clockRate;
    unsigned long channels;
    DOMString     parameters;
    DOMString     implementation;
};

Dictionary RTCCodecStats Members

payloadType of type unsigned long

Payload type as used in RTP encoding.

codec of type DOMString

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.

parameters of type DOMString

From the SDP description line.

implementation of type DOMString

Identifies the implementation used. This is useful for diagnosing interoperability issues.

If too much information is given here, it increases the fingerprint surface. Since it is only given for active tracks, the incremental exposure is small. (This is a fingerprinting vector.)

5.3 RTCInboundRTPStreamStats dictionary

RTCInboundRTPStreamStats dictionary represents the measurement metrics for the incoming media stream. 

dictionary RTCInboundRTPStreamStats : RTCRTPStreamStats {
    unsigned long      packetsReceived;
    unsigned long long bytesReceived;
    unsigned long      packetsLost;
    double             jitter;
    double             fractionLost;
    unsigned long      packetsDiscarded;
    unsigned long      packetsRepaired;
    unsigned long      burstPacketsLost;
    unsigned long      burstPacketsDiscarded;
    unsigned long      burstLossCount;
    unsigned long      burstDiscardCount;
    double             burstLossRate;
    double             burstDiscardRate;
    double             gapLossRate;
    double             gapDiscardRate;
};

Dictionary RTCInboundRTPStreamStats Members

packetsReceived of type unsigned long

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

bytesReceived of type unsigned long long

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

packetsLost of type unsigned long

Total number of RTP packets lost for this SSRC. Calculated as defined in [ RFC3550] section 6.4.1.

jitter of type double

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

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.

packetsDiscarded of type unsigned long

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

packetsRepaired of type unsigned long

The cumulative number of lost RTP packets repaired after applying an error-resilience mechanism [XRBLOCK-STATS]. It is measured for the primary source RTP packets and only counted for RTP packets that have no further chance of repair. To clarify, the value is upper-bound to the cumulative number of lost packets. Calculated as defined in [ RFC7509] section 3.1 and Appendix A.b.

burstPacketsLost of type unsigned long

The cumulative number of RTP packets lost during loss bursts, Appendix A (c) of [RFC6958].

burstPacketsDiscarded of type unsigned long

The cumulative number of RTP packets discarded during discard bursts, Appendix A (b) of [RFC7003].

burstLossCount of type unsigned long

The cumulative number of bursts of lost RTP packets, Appendix A (e) of [RFC6958].

[RFC3611] recommends a Gmin (threshold) value of 16 for classifying a sequence of packet losses or discards as a burst.

burstDiscardCount of type unsigned long

The cumulative number of bursts of discarded RTP packets, Appendix A (e) of [burst-gap-discard].

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.

5.4 RTCOutboundRTPStreamStats dictionary

RTCOutboundRTPStreamStats dictionary represents the measurement metrics for the outgoing media stream. 

dictionary RTCOutboundRTPStreamStats : RTCRTPStreamStats {
    unsigned long      packetsSent;
    unsigned long long bytesSent;
    double             targetBitrate;
    double             roundTripTime;
};

Dictionary RTCOutboundRTPStreamStats Members

packetsSent of type unsigned 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.

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.

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].

5.5 RTCPeerConnectionStats dictionary 

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

Dictionary RTCPeerConnectionStats Members

dataChannelsOpened of type unsigned long

Represents the number of unique datachannels opened.

dataChannelsClosed of type unsigned long

Represents the number of unique datachannels closed.

5.6 RTCMediaStreamStats dictionary 

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

Dictionary RTCMediaStreamStats Members

streamIdentifier of type DOMString

stream.id property

trackIds of type sequence<DOMString>

This is the id of the stats object, not the track.id.

5.7 RTCMediaStreamTrackStats dictionary

An RTCMediaStreamTrackStats object represents the stats about a MediaStreamTrack's connection to the PeerConnection 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 RTPSender object, or via being created on an RTPReceiver object). If it is detached (via removeTrack or via replaceTrack), it continues to appear, but with the "detached" member set to True. 

dictionary RTCMediaStreamTrackStats : RTCStats {
    DOMString           trackIdentifier;
    boolean             remoteSource;
    boolean             ended;
    boolean             detached;
    sequence<DOMString> ssrcIds;
    unsigned long       frameWidth;
    unsigned long       frameHeight;
    double              framesPerSecond;
    unsigned long       framesSent;
    unsigned long       framesReceived;
    unsigned long       framesDecoded;
    unsigned long       framesDropped;
    unsigned long       framesCorrupted;
    unsigned long       partialFramesLost;
    unsigned long       fullFramesLost;
    double              audioLevel;
    double              echoReturnLoss;
    double              echoReturnLossEnhancement;
};

Dictionary RTCMediaStreamTrackStats Members

trackIdentifier of type DOMString

Represents the track.id property.

remoteSource of type boolean
ended of type boolean

Reflects the "ended" state of the track.

detached of type boolean

True if the track has been detached from the PeerConnection object. If true, all stats reflect their values at the time when the track was detached.

ssrcIds of type sequence<DOMString>
frameWidth of type unsigned long

Only makes sense for video media streams and represents the width of the video frame for this SSRC.

frameHeight of type unsigned long

Only makes sense for video media streams and represents the height of the video frame for this SSRC.

framesPerSecond of type double

Only valid for video. It represents the nominal FPS value.

framesSent of type unsigned long

Only valid for video. It represents the total number of frames sent for this SSRC.

framesReceived of type unsigned long

Only valid for video and when remoteSource is set to true. It represents the total number of frames received for this SSRC.

framesDecoded of type unsigned long

Only valid for video. It represents the total number of frames correctly decoded for this SSRC. Same definition as totalVideoFrames in Section 5 of [MEDIA-SOURCE].

framesDropped of type unsigned long

Only valid for video. It is the total number of frames dropped predecode or dropped because the frame missed its display deadline for this SSRC. It is the same definition as droppedVideoFrames in Section 5 of [MEDIA-SOURCE].

framesCorrupted of type unsigned long

Only valid for video. It is the total number of corrupted frames that have been detected for this SSRC. It is the same definition as corruptedVideoFrames in Section 5 of [MEDIA-SOURCE].

partialFramesLost of type unsigned long

Only valid for video. partialFramesLost is the cumulative number of partial frames lost, as defined in Appendix A (j) of [RFC7004].

fullFramesLost of type unsigned long

Only valid for video. fullFramesLost is the cumulative number of full frames lost, as defined in Appendix A (i) of [RFC7004].

audioLevel of type double

Only valid for audio, and the value is between 0..1 (linear), where 1.0 represents 0 dBov. Calculated as defined in [RFC6464].

echoReturnLoss of type double

Only present on audio tracks sourced from a microphone where echo cancellation is applied. Calculated in decibels, as defined in [ECHO] (2012) section 3.14.

echoReturnLossEnhancement of type double

Only present on audio tracks sourced from a microphone where echo cancellation is applied. Calculated in decibels, as defined in [ECHO] (2012) section 3.15.

5.8 RTCDataChannelStats dictionary 

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

Dictionary RTCDataChannelStats Members

label of type DOMString
protocol of type DOMString
datachannelid of type long

the "id" attribute of the RTCDataChannel object

state of type RTCDataChannelState
messagesSent of type unsigned long

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

bytesSent of type unsigned long long

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

messagesReceived of type unsigned long

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

bytesReceived of type unsigned long long

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

5.9 RTCTransportStats dictionary

A Transport carries a part of an SDP session, consisting of RTP and RTCP. When Bundle is in use, an SDP session will have only one Transport per Bundle group. When Bundle is not in use, there is one Transport per m-line. 

dictionary RTCTransportStats : RTCStats {
    unsigned long long bytesSent;
    unsigned long long bytesReceived;
    DOMString          rtcpTransportStatsId;
    boolean            activeConnection;
    DOMString          selectedCandidatePairId;
    DOMString          localCertificateId;
    DOMString          remoteCertificateId;
};

Dictionary RTCTransportStats Members

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.

activeConnection of type boolean

Set to true when transport is active.

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.

5.10 RTCIceCandidateStats dictionary

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

dictionary RTCIceCandidateStats : RTCStats {
    DOMString           ip;
    long                port;
    DOMString           protocol;
    RTCIceCandidateType candidateType;
    long                priority;
    DOMString           url;
};

Dictionary RTCIceCandidateStats Members

ip of type DOMString

It is the IP address of the candidate, allowing for IPv4 addresses and IPv6 addresses, but fully qualified domain names (FQDNs) are not allowed. See [RFC5245] section 15.1 for details.

port of type long

It is the port number of the candidate.

protocol of type DOMString

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

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

The URL of the TURN or STUN server indicated in the that translated this IP address. It is the URL address surfaced in an RTCPeerConnectionIceEvent.

5.11 RTCIceCandidatePairStats dictionary 

dictionary RTCIceCandidatePairStats : RTCStats {
    DOMString                     transportId;
    DOMString                     localCandidateId;
    DOMString                     remoteCandidateId;
    RTCStatsIceCandidatePairState state;
    unsigned long long            priority;
    boolean                       nominated;
    boolean                       writable;
    boolean                       readable;
    unsigned long long            bytesSent;
    unsigned long long            bytesReceived;
    double                        totalRtt;
    double                        currentRtt;
    double                        availableOutgoingBitrate;
    double                        availableIncomingBitrate;
    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            consentRequestsReceived;
    unsigned long long            consentRequestsSent;
    unsigned long long            consentResponsesReceived;
    unsigned long long            consentResponsesSent;
};

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 RTCIceCandidateAttributes for the local candidate associated with this candidate pair.

remoteCandidateId of type DOMString

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

state of type RTCStatsIceCandidatePairState

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

priority of type unsigned long long

Calculated from candidate priorities as defined in [RFC5245] section 5.7.2.

nominated of type boolean

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

writable of type boolean

Has gotten ACK to an ICE request.

readable of type boolean

Has gotten a valid incoming ICE request.

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.

totalRtt of type double

Represents the sum of all RTT measurements in seconds since the beginning of the session, based on both STUN connectivity check [ STUN-PATH-CHAR] responses (responsesReceived) and consent [ RFC7675] responses (consentResponsesReceived). The average RTT can be computed from totalRtt by dividing it by (responsesReceived + consentResponsesReceived).

currentRtt of type double

Represents the latest RTT measured in seconds, computed from both STUN connectivity checks [STUN-PATH-CHAR] and consent responses [RFC7675].

availableOutgoingBitrate of type double

It is calculated by the underlying congestion control by combining the available bitrate for all the outgoing media streams. 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.

availableIncomingBitrate of type double

It is calculated by the underlying congestion control by combining the available bitrate for all the incoming media streams. 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.

requestsReceived of type unsigned long long

Represents the total number of connectivity check requests received (including retransmissions).

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.

retransmissionsReceived of type unsigned long long

Represents the total number of connectivity check retransmissions received.

retransmissionsSent of type unsigned long long

Represents the total number of connectivity check retransmissions sent.

consentRequestsReceived of type unsigned long long

Represents the total number of consent requests received.

consentRequestsSent of type unsigned long long

Represents the total number of consent requests sent.

consentResponsesReceived of type unsigned long long

Represents the total number of consent responses received.

consentResponsesSent of type unsigned long long

Represents the total number of consent responses sent.

5.11.1 RTCStatsIceCandidatePairState enum 

enum RTCStatsIceCandidatePairState {
    "frozen",
    "waiting",
    "inprogress",
    "failed",
    "succeeded",
    "cancelled"
};
Enumeration description
frozen

Defined in Section 5.7.4 of [RFC5245].

waiting

Defined in Section 5.7.4 of [RFC5245].

inprogress

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].

cancelled

Defined in Section 5.7.4 of [RFC5245].

5.12 RTCCertificateStats dictionary 

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

Dictionary RTCCertificateStats Members

fingerprint of type DOMString

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

fingerprintAlgorithm of type DOMString

For instance, "sha-256".

base64Certificate of type DOMString

For example, DER-encoded, base-64 representation of a certifiate.

issuerCertificateId of type DOMString

6. Examples

6.1 Example of association of local and remote statistics

The following example code shows the association of remote statistics with local statistics in a RTCStats dictionary.

Example 1
stats [
   dictionary (of type RTPoutboundStatsDictionary) {
      ssrc=1234
      id=foobar
      type=outboundrtp
      isRemote=false
      associateStatsId=barfoo
  },
  dictionary (of type RTPinboundStatsDictionary) {
      ssrc=1234
      id=barfoo
      type=inboundrtp
      isRemote=true
      associateStatsId=foobar
  }
]

6.2 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 2
var baselineReport, currentReport;
var selector = pc.getRemoteStreams()[0].getAudioTracks()[0];

pc.getStats(selector, function (report) {
    baselineReport = report;
}, logError);

// ... wait a bit
setTimeout(function () {
    pc.getStats(selector, function (report) {
        currentReport = report;
        processStats();
    }, logError);
}, aBit);

function processStats() {
    // compare the elements from the current report with the baseline
    for (var i in currentReport) {
        var now = currentReport[i];
        if (now.type != "outbund-rtp")
            continue;

        // get the corresponding stats from the baseline report
        base = baselineReport[now.id];

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

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

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

function logError(error) {
    log(error.name + ": " + error.message);
}

7. Security Considerations

Some stats identifiers may expose personally identifiable information, for example the IP addresses of the participating endpoints when a TURN relay is not used.

8. Change Log

This section will be removed before publication. The entries are in reverse chronological order.

8.1 Changes since 26 May 2016

8.2 Changes since 23 October 2015

8.3 Changes since 03 February 2015

  1. [#10] Added RTCRTPStreamStats.mediaType.

8.4 Changes since 30 September 2014

  1. kept getStats() in webrtc-pc. Changed RTCStatsType from enum to DOMString.
  2. Added "datachannel" to RTCStatsType.
  3. Added fractionLost to RTCInboundRTPStreamStats.
  4. Clarified that bytesSent and bytesReceived do no include headers or paddings.

8.5 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, Jan-Ivar Bruaroey, and Cullen Jennings for their contributions to this specification.

A. References

A.1 Normative references

[ECHO]
ITU-T G.168. ITU-T. Digital network echo cancellers. Standard. URL: https://www.itu.int/rec/T-REC-G.168/en
[GETUSERMEDIA]
Daniel Burnett; Adam Bergkvist; Cullen Jennings; Anant Narayanan; Bernard Aboba. W3C. Media Capture and Streams. 19 May 2016. W3C Candidate Recommendation. URL: https://www.w3.org/TR/mediacapture-streams/
[HTML5]
Ian Hickson; Robin Berjon; Steve Faulkner; Travis Leithead; Erika Doyle Navara; Theresa O'Connor; Silvia Pfeiffer. W3C. HTML5. 28 October 2014. W3C Recommendation. URL: https://www.w3.org/TR/html5/
[MEDIA-SOURCE]
Matthew Wolenetz; Jerry Smith; Mark Watson; Aaron Colwell; Adrian Bateman. W3C. Media Source Extensions. 5 July 2016. W3C Candidate Recommendation. URL: https://www.w3.org/TR/media-source/
[RFC2119]
S. Bradner. IETF. Key words for use in RFCs to Indicate Requirement Levels. March 1997. Best Current Practice. URL: https://tools.ietf.org/html/rfc2119
[RFC3550]
H. Schulzrinne; S. Casner; R. Frederick; V. Jacobson. IETF. RTP: A Transport Protocol for Real-Time Applications. July 2003. Internet Standard. URL: https://tools.ietf.org/html/rfc3550
[RFC3611]
T. Friedman, Ed.; R. Caceres, Ed.; A. Clark, Ed.. IETF. RTP Control Protocol Extended Reports (RTCP XR). November 2003. Proposed Standard. URL: https://tools.ietf.org/html/rfc3611
[RFC3890]
M. Westerlund. IETF. A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP). September 2004. Proposed Standard. URL: https://tools.ietf.org/html/rfc3890
[RFC4572]
J. Lennox. IETF. Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP). July 2006. Proposed Standard. URL: https://tools.ietf.org/html/rfc4572
[RFC4585]
J. Ott; S. Wenger; N. Sato; C. Burmeister; J. Rey. IETF. Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF). July 2006. Proposed Standard. URL: https://tools.ietf.org/html/rfc4585
[RFC5104]
S. Wenger; U. Chandra; M. Westerlund; B. Burman. IETF. Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF). February 2008. Proposed Standard. URL: https://tools.ietf.org/html/rfc5104
[RFC5245]
J. Rosenberg. IETF. Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols. April 2010. Proposed Standard. URL: https://tools.ietf.org/html/rfc5245
[RFC6464]
J. Lennox, Ed.; E. Ivov; E. Marocco. IETF. A Real-time Transport Protocol (RTP) Header Extension for Client-to-Mixer Audio Level Indication. December 2011. Proposed Standard. URL: https://tools.ietf.org/html/rfc6464
[RFC6958]
A. Clark; S. Zhang; J. Zhao; Q. Wu, Ed.. IETF. RTP Control Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Loss Metric Reporting. May 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc6958
[RFC7002]
A. Clark; G. Zorn; Q. Wu. IETF. RTP Control Protocol (RTCP) Extended Report (XR) Block for Discard Count Metric Reporting. September 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc7002
[RFC7003]
A. Clark; R. Huang; Q. Wu, Ed.. IETF. RTP Control Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Discard Metric Reporting. September 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc7003
[RFC7004]
G. Zorn; R. Schott; Q. Wu, Ed.; R. Huang. IETF. RTP Control Protocol (RTCP) Extended Report (XR) Blocks for Summary Statistics Metrics Reporting. September 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc7004
[RFC7509]
R. Huang; V. Singh. IETF. RTP Control Protocol (RTCP) Extended Report (XR) for Post-Repair Loss Count Metrics. May 2015. Proposed Standard. URL: https://tools.ietf.org/html/rfc7509
[RFC7675]
M. Perumal; D. Wing; R. Ravindranath; T. Reddy; M. Thomson. IETF. Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness. October 2015. Proposed Standard. URL: https://tools.ietf.org/html/rfc7675
[STUN-PATH-CHAR]
T. Reddy; D. Wing; P. Martinsen; V. Singh. IETF. Discovery of path characteristics using STUN. Internet Draft. URL: https://tools.ietf.org/html/draft-reddy-tram-stun-path-data
[WEBRTC]
Adam Bergkvist; Daniel Burnett; Cullen Jennings; Anant Narayanan; Bernard Aboba. W3C. WebRTC 1.0: Real-time Communication Between Browsers. 13 September 2016. W3C Working Draft. URL: https://www.w3.org/TR/webrtc/
[burst-gap-discard]
Varun Singh; Colin Perkins; Alan Clark; Rachel Huang. IETF. RTCP XR Block for Independent Reporting of Burst/Gap Discard Metric. Internet Draft. URL: https://tools.ietf.org/html/draft-ietf-xrblock-independent-burst-gap-discard

A.2 Informative references

[RFC2032]
T. Turletti; C. Huitema. IETF. RTP Payload Format for H.261 Video Streams. October 1996. Proposed Standard. URL: https://tools.ietf.org/html/rfc2032
[RFC4587]
R. Even. IETF. RTP Payload Format for H.261 Video Streams. August 2006. Proposed Standard. URL: https://tools.ietf.org/html/rfc4587
[WEBIDL]
Cameron McCormack; Boris Zbarsky. W3C. WebIDL Level 1. 15 September 2016. W3C Proposed Recommendation. URL: https://www.w3.org/TR/WebIDL-1/
[XRBLOCK-STATS]
Varun Singh; Rachel Huang; Roni Even; Dan Romascanu; Lingli Deng. IETF. RTCP XR Metrics for WebRTC. Internet Draft. URL: https://tools.ietf.org/html/draft-ietf-xrblock-rtcweb-rtcp-xr-metrics