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rfc:rfc8451

Internet Engineering Task Force (IETF) V. Singh Request for Comments: 8451 callstats.io Category: Informational R. Huang ISSN: 2070-1721 R. Even

                                                                Huawei
                                                          D. Romascanu
                                                            Individual
                                                               L. Deng
                                                          China Mobile
                                                        September 2018
      Considerations for Selecting RTP Control Protocol (RTCP)
     Extended Report (XR) Metrics for the WebRTC Statistics API

Abstract

 This document describes monitoring features related to media streams
 in Web real-time communication (WebRTC).  It provides a list of RTP
 Control Protocol (RTCP) Sender Report (SR), Receiver Report (RR), and
 Extended Report (XR) metrics, which may need to be supported by RTP
 implementations in some diverse environments.  It lists a set of
 identifiers for the WebRTC's statistics API.  These identifiers are a
 set of RTCP SR, RR, and XR metrics related to the transport of
 multimedia flows.

Status of This Memo

 This document is not an Internet Standards Track specification; it is
 published for informational purposes.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It represents the consensus of the IETF community.  It has
 received public review and has been approved for publication by the
 Internet Engineering Steering Group (IESG).  Not all documents
 approved by the IESG are a candidate for any level of Internet
 Standard; see Section 2 of RFC 7841.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 https://www.rfc-editor.org/info/rfc8451.

Singh, et al. Informational [Page 1] RFC 8451 RTCP XR Metrics for WebRTC September 2018

Copyright Notice

 Copyright (c) 2018 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (https://trustee.ietf.org/license-info) in effect on the date of
 publication of this document.  Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document.  Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.

Singh, et al. Informational [Page 2] RFC 8451 RTCP XR Metrics for WebRTC September 2018

Table of Contents

 1. Introduction ....................................................4
 2. Terminology .....................................................4
 3. RTP Statistics in WebRTC Implementations ........................5
 4. Considerations for Impact of Measurement Interval ...............5
 5. Candidate Metrics ...............................................6
    5.1. Network Impact Metrics .....................................6
         5.1.1. Loss and Discard Packet Count Metric ................6
         5.1.2. Burst/Gap Pattern Metrics for Loss and Discard ......7
         5.1.3. Run-Length Encoded Metrics for Loss and Discard .....8
    5.2. Application Impact Metrics .................................8
         5.2.1. Discarded Octets Metric .............................8
         5.2.2. Frame Impairment Summary Metrics ....................9
         5.2.3. Jitter Buffer Metrics ...............................9
    5.3. Recovery Metrics ..........................................10
         5.3.1. Post-Repair Packet Count Metrics ...................10
         5.3.2. Run-Length Encoded Metric for Post-Repair ..........10
 6. Identifiers from Sender, Receiver, and Extended Report Blocks ..11
    6.1. Cumulative Number of Packets and Octets Sent ..............11
    6.2. Cumulative Number of Packets and Octets Received ..........11
    6.3. Cumulative Number of Packets Lost .........................11
    6.4. Interval Packet Loss and Jitter ...........................12
    6.5. Cumulative Number of Packets and Octets Discarded .........12
    6.6. Cumulative Number of Packets Repaired .....................12
    6.7. Burst Packet Loss and Burst Discards ......................12
    6.8. Burst/Gap Rates ...........................................13
    6.9. Frame Impairment Metrics ..................................13
 7. Adding New Metrics to WebRTC Statistics API ....................13
 8. Security Considerations ........................................14
 9. IANA Considerations ............................................14
 10. References ....................................................14
    10.1. Normative References .....................................14
    10.2. Informative References ...................................16
 Acknowledgements ..................................................17
 Authors' Addresses ................................................18

Singh, et al. Informational [Page 3] RFC 8451 RTCP XR Metrics for WebRTC September 2018

1. Introduction

 Web real-time communication (WebRTC) [WebRTC-Overview] deployments
 are emerging, and applications need to be able to estimate the
 service quality.  If sufficient information (metrics or statistics)
 is provided to the application, it can attempt to improve the media
 quality.  [RFC7478] specifies a requirement for statistics:
 F38   The browser must be able to collect statistics, related to the
       transport of audio and video between peers, needed to estimate
       quality of experience.
 The WebRTC Stats API [W3C.webrtc-stats] currently lists metrics
 reported in the RTCP Sender Report and Receiver Report (SR/RR)
 [RFC3550] to fulfill this requirement.  However, the basic metrics
 from RTCP SR/RR are not sufficient for precise quality monitoring or
 diagnosing potential issues.
 Standards such as "RTP Control Protocol Extended Reports (RTCP XR)"
 [RFC3611] as well as other extensions standardized in the XRBLOCK
 Working Group, e.g., burst/gap loss metric reporting [RFC6958] and
 burst/gap discard metric reporting [RFC7003], have been produced for
 the purpose of collecting and reporting performance metrics from RTP
 endpoint devices that can be used to have end-to-end service
 visibility and to measure the delivery quality in various RTP
 services.  These metrics are able to complement those in [RFC3550].
 In this document, we provide rationale for choosing additional RTP
 metrics for the WebRTC getStats() API [W3C.webrtc].  All identifiers
 proposed in this document are recommended to be implemented by an
 WebRTC endpoint.  An endpoint may choose not to expose an identifier
 if it does not implement the corresponding RTCP Report.  This
 document only considers RTP-layer metrics.  Other metrics, e.g.,
 IP-layer metrics, are out of scope.

2. Terminology

 In addition to the terminology from [RFC3550], [RFC3611], and
 [RFC7478], this document uses the following term.
 ReportGroup: It is a set of metrics identified by a common
    synchronization source (SSRC).

Singh, et al. Informational [Page 4] RFC 8451 RTCP XR Metrics for WebRTC September 2018

3. RTP Statistics in WebRTC Implementations

 The RTCP Sender Reports (SRs) and Receiver Reports (RRs) [RFC3550]
 expose the basic metrics for the local and remote media streams.
 However, these metrics provide only partial or limited information,
 which may not be sufficient for diagnosing problems or monitoring
 quality.  For example, it may be useful to distinguish between
 packets lost and packets discarded due to late arrival.  Even though
 they have the same impact on the multimedia quality, it helps in
 identifying and diagnosing problems.  RTP Control Protocol Extended
 Reports (XRs) [RFC3611] and other extensions discussed in the XRBLOCK
 Working Group provide more detailed statistics, which complement the
 basic metrics reported in the RTCP SR and RRs.
 The WebRTC application extracts statistics from the browser by
 querying the getStats() API [W3C.webrtc].  The browser can easily
 report the local variables, i.e., the statistics related to the
 outgoing and incoming RTP media streams.  However, without the
 support of RTCP XRs or some other signaling mechanism, the WebRTC
 application cannot expose the remote endpoints' statistics.
 [WebRTC-RTP-USAGE] does not mandate the use of any RTCP XRs, and
 their usage is optional.  If the use of RTCP XRs is successfully
 negotiated between endpoints (via SDP), thereafter the application
 has access to both local and remote statistics.  Alternatively, once
 the WebRTC application gets the local information, it can report the
 information to an application server or a third-party monitoring
 system, which provides quality estimates or diagnostic services for
 application developers.  The exchange of statistics between endpoints
 or between a monitoring server and an endpoint is outside the scope
 of this document.

4. Considerations for Impact of Measurement Interval

 RTCP extensions like RTCP XR usually share the same timing interval
 with the RTCP SR/RR, i.e., they are sent as compound packets,
 together with the RTCP SR/RR.  Alternatively, if the RTCP XR uses a
 different measurement interval, all XRs using the same measurement
 interval are compounded together, and the measurement interval is
 indicated in a specific measurement information block defined in
 [RFC6776].
 When using WebRTC getStats() APIs (see "Statistics Model" in
 [W3C.webrtc]), the applications can query this information at
 arbitrary intervals.  For the statistics reported by the remote
 endpoint, e.g., those conveyed in an RTCP SR/RR/XR, these will not
 change until the next RTCP report is received.  However, statistics
 generated by the local endpoint have no such restrictions as long as
 the endpoint is sending and receiving media.  For example, an

Singh, et al. Informational [Page 5] RFC 8451 RTCP XR Metrics for WebRTC September 2018

 application may choose to poll the stack for statistics every 1
 second.  In that case, the underlying stack local will return the
 current snapshot of the local statistics (for incoming and outgoing
 media streams).  However, it may return the same remote statistics as
 previously, because no new RTCP reports may have been received in the
 past 1 second.  This can occur when the polling interval is shorter
 than the average RTCP reporting interval.

5. Candidate Metrics

 Since the following metrics are all defined in RTCP XR, which is not
 mandated in WebRTC, all of them are local.  However, if RTCP XR is
 supported by negotiation between two browsers, the following metrics
 can also be generated remotely and be sent to the local endpoint
 (that generated the media) via RTCP XR packets.
 The metrics are classified into 3 categories as follows: network
 impact metrics, application impact metrics, and recovery metrics.
 Network impact metrics are the statistics recording the information
 only for network transmission.  They are useful for network problem
 diagnosis.  Application impact metrics mainly collect the information
 from the viewpoint of the application, e.g., bit rate, frame rate, or
 jitter buffers.  Recovery metrics reflect how well the repair
 mechanisms perform, e.g., loss concealment, retransmission, or
 Forward Error Correction (FEC).  All 3 types of metrics are useful
 for quality estimations of services in WebRTC implementations.
 WebRTC applications can use these metrics to calculate the estimated
 Mean Opinion Score (MOS) [ITU-T_P.800.1] values or Media Delivery
 Index (MDI) [RFC4445] for their services.

5.1. Network Impact Metrics

5.1.1. Loss and Discard Packet Count Metric

 In multimedia transport, packets that are received abnormally are
 classified into 3 types: lost, discarded, and duplicate packets.
 Packet loss may be caused by network device breakdown, bit-error
 corruption, or network congestion (packets dropped by an intermediate
 router queue).  Duplicate packets may be a result of network delays
 that cause the sender to retransmit the original packets.  Discarded
 packets are packets that have been delayed long enough (perhaps they
 missed the playout time) and are considered useless by the receiver.
 Lost and discarded packets cause problems for multimedia services, as
 missing data and long delays can cause degradation in service
 quality, e.g., missing large blocks of contiguous packets (lost or
 discarded) may cause choppy audio, and long network transmission
 delay time may cause audio or video buffering.  The RTCP SR/RR
 defines a metric for counting the total number of RTP data packets

Singh, et al. Informational [Page 6] RFC 8451 RTCP XR Metrics for WebRTC September 2018

 that have been lost since the beginning of reception.  However, this
 statistic does not distinguish lost packets from discarded and
 duplicate packets.  Packets that arrive late will be discarded and
 are not reported as lost, and duplicate packets will be regarded as a
 normally received packet.  Hence, the loss metric can be misleading
 if many duplicate packets are received or packets are discarded,
 which causes the quality of the media transport to appear okay from a
 statistical point of view, while the users are actually experiencing
 bad service quality.  So, in such cases, it is better to use more
 accurate metrics in addition to those defined in RTCP SR/RR.
 The metrics for lost packets and duplicated packets defined in the
 Statistics Summary Report Block of [RFC3611] extend the information
 of loss carried in standard RTCP SR/RR.  They explicitly give an
 account of lost and duplicated packets.  Lost packet counts are
 useful for network problem diagnosis.  It is better to use the packet
 loss metrics of [RFC3611] to indicate the lost packet count instead
 of the cumulative number of packets lost metric of [RFC3550].
 Duplicated packets are usually rare and have little effect on QoS
 evaluation.  So it may not be suitable for use in WebRTC.
 Using loss metrics without considering discard metrics may result in
 inaccurate quality evaluation, as packet discard due to jitter is
 often more prevalent than packet loss in modern IP networks.  The
 discarded metric specified in [RFC7002] counts the number of packets
 discarded due to jitter.  It augments the loss statistics metrics
 specified in standard RTCP SR/RR.  For those WebRTC services with
 jitter buffers requiring precise quality evaluation and accurate
 troubleshooting, this metric is useful as a complement to the metrics
 of RTCP SR/RR.

5.1.2. Burst/Gap Pattern Metrics for Loss and Discard

 RTCP SR/RR defines coarse metrics regarding loss statistics: the
 metrics are all about per-call statistics and are not detailed enough
 to capture the transitory nature of some impairments like bursty
 packet loss.  Even if the average packet loss rate is low, the lost
 packets may occur during short dense periods, resulting in short
 periods of degraded quality.  Bursts cause lower quality experience
 than the non-bursts for low packet loss rates, whereas for high
 packet loss rates, the converse is true.  So capturing burst gap
 information is very helpful for quality evaluation and locating
 impairments.  If the WebRTC application needs to evaluate the service
 quality, burst gap metrics provide more accurate information than
 RTCP SR/RR.

Singh, et al. Informational [Page 7] RFC 8451 RTCP XR Metrics for WebRTC September 2018

 [RFC3611] introduces burst gap metrics in the VoIP Report Block.
 These metrics record the density and duration of burst and gap
 periods, which are helpful in isolating network problems since bursts
 correspond to periods of time during which the packet loss/discard
 rate is high enough to produce noticeable degradation in audio or
 video quality.  Metrics related to the burst gap are also introduced
 in [RFC7003] and [RFC6958], which define two new report blocks for
 use in a range of RTP applications beyond those described in
 [RFC3611].  These metrics distinguish discarded packets from loss
 packets that occur in the burst period and provide more information
 for diagnosing network problems.  Additionally, the block reports the
 frequency of burst events, which is useful information for evaluating
 the quality of experience.  Hence, if WebRTC applications need to do
 quality evaluation and observe when and why quality degrades, these
 metrics should be considered.

5.1.3. Run-Length Encoded Metrics for Loss and Discard

 Run-length encoding uses a bit vector to encode information about the
 packet.  Each bit in the vector represents a packet; depending on the
 signaled metric, it defines if the packet was lost, duplicated,
 discarded, or repaired.  An endpoint typically uses the run-length
 encoding to accurately communicate the status of each packet in the
 interval to the other endpoint.  [RFC3611] and [RFC7097] define run-
 length encoding for lost and duplicate packets, and discarded
 packets, respectively.
 The WebRTC application could benefit from the additional information.
 If losses occur after discards, an endpoint may be able to correlate
 the two run length vectors to identify congestion-related losses,
 e.g., a router queue became overloaded causing delays and then
 overflowed.  If the losses are independent, it may indicate bit-error
 corruption.  For the WebRTC Stats API [W3C.webrtc-stats], these types
 of metrics are not recommended for use due to the large amount of
 data and the computation involved.

5.2. Application Impact Metrics

5.2.1. Discarded Octets Metric

 The metric reports the cumulative size of the packets discarded in
 the interval.  It is complementary to the number of discarded
 packets.  An application measures sent octets and received octets to
 calculate the sending rate and receiving rate, respectively.  The
 application can calculate the actual bit rate in a particular
 interval by subtracting the discarded octets from the received
 octets.

Singh, et al. Informational [Page 8] RFC 8451 RTCP XR Metrics for WebRTC September 2018

 For WebRTC, the discarded octets metric supplements the metrics on
 sent and received octets and provides an accurate method for
 calculating the actual bit rate, which is an important parameter to
 reflect the quality of the media.  The Bytes Discarded metric is
 defined in [RFC7243].

5.2.2. Frame Impairment Summary Metrics

 RTP has different framing mechanisms for different payload types.
 For audio streams, a single RTP packet may contain one or multiple
 audio frames.  On the other hand, in video streams, a single video
 frame may be transmitted in multiple RTP packets.  The size of each
 packet is limited by the Maximum Transmission Unit (MTU) of the
 underlying network.  However, the statistics from standard SR/RR only
 collect information from the transport layer, so they may not fully
 reflect the quality observed by the application.  Video is typically
 encoded using two frame types, i.e., key frames and derived frames.
 Key frames are normally just spatially compressed, i.e., without
 prediction from other pictures.  The derived frames are temporally
 compressed, i.e., depend on the key frame for decoding.  Hence, key
 frames are much larger in size than derived frames.  The loss of
 these key frames results in a substantial reduction in video quality.
 Thus, it is reasonable to consider this application-layer information
 in WebRTC implementations, which influence sender strategies to
 mitigate the problem or require the accurate assessment of users'
 quality of experience.
 The metrics in this category include: number of discarded key frames,
 number of lost key frames, number of discarded derived frames, and
 number of lost derived frames.  These metrics can be used to
 calculate the Media Loss Rate (MLR) of the MDI [RFC4445].  Details of
 the definition of these metrics are described in [RFC7003].
 Additionally, the metric provides the rendered frame rate, an
 important parameter for quality estimation.

5.2.3. Jitter Buffer Metrics

 The size of the jitter buffer affects the end-to-end delay on the
 network and also the packet discard rate.  When the buffer size is
 too small, late-arriving packets are not played out and are dropped,
 while when the buffer size is too large, packets are held longer than
 necessary and consequently reduce conversational quality.
 Measurement of jitter buffer should not be ignored in the evaluation
 of end-user perception of conversational quality.  Metrics related to
 the jitter buffer, such as maximum and nominal jitter buffer, could
 be used to show how the jitter buffer behaves at the receiving
 endpoint.  They are useful for providing better end-user quality of
 experience (QoE) when jitter buffer factors are used as inputs to

Singh, et al. Informational [Page 9] RFC 8451 RTCP XR Metrics for WebRTC September 2018

 calculate estimated MOS values.  Thus, for those cases, jitter buffer
 metrics should be considered.  The definition of these metrics is
 provided in [RFC7005].

5.3. Recovery Metrics

 This document does not consider concealment metrics [RFC7294] as part
 of recovery metrics.

5.3.1. Post-Repair Packet Count Metrics

 Web applications can support certain RTP error-resilience mechanisms
 following the recommendations specified in [WebRTC-RTP-USAGE].  For
 these web applications using repair mechanisms, providing some
 statistics about the performance of their repair mechanisms could
 help provide a more accurate quality evaluation.
 The unrepaired packet count and repaired loss count defined in
 [RFC7509] provide the recovery information of the error-resilience
 mechanisms to the monitoring application or the sending endpoint.
 The endpoint can use these metrics to ascertain the ratio of repaired
 packets to lost packets.  Including post-repair packet count metrics
 helps the application evaluate the effectiveness of the applied
 repair mechanisms.

5.3.2. Run-Length Encoded Metric for Post-Repair

 [RFC5725] defines run-length encoding for post-repair packets.  When
 using error-resilience mechanisms, the endpoint can correlate the
 loss run length with this metric to ascertain where the losses and
 repairs occurred in the interval.  This provides more accurate
 information for recovery mechanisms evaluation than those in Section
 5.3.1.  However, when RTCP XR metrics are supported, using run-length
 encoded metrics is not suggested because the per-packet information
 yields an enormous amount of data that is not required in this case.
 For WebRTC, the application may benefit from the additional
 information.  If losses occur after discards, an endpoint may be able
 to correlate the two run-length vectors to identify congestion-
 related losses, e.g., a router queue became overloaded causing delays
 and then overflowed.  If the losses are independent, it may indicate
 bit-error corruption.  Lastly, when using error-resilience
 mechanisms, the endpoint can correlate the loss and post-repair run
 lengths to ascertain where the losses and repairs occurred in the
 interval.  For example, consecutive losses are likely not to be
 repaired by a simple FEC scheme.

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6. Identifiers from Sender, Receiver, and Extended Report Blocks

 This document describes a list of metrics and corresponding
 identifiers relevant to RTP media in WebRTC.  This group of
 identifiers are defined on a ReportGroup corresponding to a
 synchronization source (SSRC).  In practice, the application needs to
 be able to query the statistic identifiers on both an incoming
 (remote) and outgoing (local) media stream.  Since sending and
 receiving SRs and RRs are mandatory, the metrics defined in the SRs
 and RRs are always available.  For XR metrics, it depends on two
 factors: 1) if it is measured at the endpoint and 2) if it is
 reported by the endpoint in an XR block.  If a metric is only
 measured by the endpoint and not reported, the metrics will only be
 available for the incoming (remote) media stream.  Alternatively, if
 the corresponding metric is also reported in an XR block, it will be
 available for both the incoming (remote) and outgoing (local) media
 stream.
 For a remote statistic, the timestamp represents the timestamp from
 an incoming SR, RR, or XR packet.  Conversely, for a local statistic,
 it refers to the current timestamp generated by the local clock
 (typically the POSIX timestamp, i.e., milliseconds since January 1,
 1970).
 As per [RFC3550], the octets metrics represent the payload size
 (i.e., not including the header or padding).

6.1. Cumulative Number of Packets and Octets Sent

 Name: packetsSent
 Definition: Section 6.4.1 of [RFC3550].
 Name: bytesSent
 Definition: Section 6.4.1 of [RFC3550].

6.2. Cumulative Number of Packets and Octets Received

 Name: packetsReceived
 Definition: Section 6.4.1 of [RFC3550].
 Name: bytesReceived
 Definition: Section 6.4.1 of [RFC3550].

6.3. Cumulative Number of Packets Lost

 Name: packetsLost
 Definition: Section 6.4.1 of [RFC3550].

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6.4. Interval Packet Loss and Jitter

 Name: jitter
 Definition: Section 6.4.1 of [RFC3550].
 Name: fractionLost
 Definition: Section 6.4.1 of [RFC3550].

6.5. Cumulative Number of Packets and Octets Discarded

 Name: packetsDiscarded
 Definition: The cumulative number of RTP packets discarded due to
    late or early arrival; see item a of Appendix A of [RFC7002].
 Name: bytesDiscarded
 Definition: The cumulative number of octets discarded due to late or
    early arrival; see Appendix A of [RFC7243].

6.6. Cumulative Number of Packets Repaired

 Name: packetsRepaired
 Definition: The cumulative number of lost RTP packets repaired after
    applying a error-resilience mechanism; see item b of Appendix A of
    [RFC7509].  To clarify, the value is the upper bound on the
    cumulative number of lost packets.

6.7. Burst Packet Loss and Burst Discards

 Name: burstPacketsLost
 Definition: The cumulative number of RTP packets lost during loss
    bursts; see item c of Appendix A of [RFC6958].
 Name: burstLossCount
 Definition: The cumulative number of bursts of lost RTP packets; see
    item d of Appendix A of [RFC6958].
 Name: burstPacketsDiscarded
 Definition: The cumulative number of RTP packets discarded during
    discard bursts; see item b of Appendix A of [RFC7003].
 Name: burstDiscardCount
 Definition: The cumulative number of bursts of discarded RTP packets;
    see item e of Appendix A of [RFC8015].
 [RFC3611] recommends a Gmin (threshold) value of 16 for classifying
 packet loss or discard burst.

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6.8. Burst/Gap Rates

 Name: burstLossRate
 Definition: The fraction of RTP packets lost during bursts; see
    item a of Appendix A of [RFC7004].
 Name: gapLossRate
 Definition: The fraction of RTP packets lost during gaps; see item b
    of Appendix A of [RFC7004].
 Name: burstDiscardRate
 Definition: The fraction of RTP packets discarded during bursts; see
    item e of Appendix A of [RFC7004].
 Name: gapDiscardRate
 Definition: The fraction of RTP packets discarded during gaps; see
    item f of Appendix A of [RFC7004].

6.9. Frame Impairment Metrics

 Name: framesLost
 Definition: The cumulative number of full frames lost; see item i of
    Appendix A of [RFC7004].
 Name: framesCorrupted
 Definition: The cumulative number of frames partially lost; see
    item j of Appendix A of [RFC7004].
 Name: framesDropped
 Definition: The cumulative number of full frames discarded; see
    item g of Appendix A of [RFC7004].
 Name: framesSent
 Definition: The cumulative number of frames sent.
 Name: framesReceived
 Definition: The cumulative number of partial or full frames received.

7. Adding New Metrics to WebRTC Statistics API

 While this document was being drafted, the metrics defined herein
 were added to the W3C WebRTC specification.  The process to add new
 metrics in the future is to create an issue or pull request on the
 repository of the W3C WebRTC specification
 (https://github.com/w3c/webrtc-stats).

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8. Security Considerations

 This document focuses on listing the RTCP XR metrics defined in the
 corresponding RTCP reporting extensions and does not give rise to any
 security vulnerabilities beyond those described in [RFC3611] and
 [RFC6792].
 The overall security considerations for RTP used in WebRTC
 applications is described in [WebRTC-RTP-USAGE] and [WebRTC-Sec],
 which also apply to this memo.

9. IANA Considerations

 This document has no IANA actions.

10. References

10.1. Normative References

 [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
            July 2003, <https://www.rfc-editor.org/info/rfc3550>.
 [RFC3611]  Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
            "RTP Control Protocol Extended Reports (RTCP XR)",
            RFC 3611, DOI 10.17487/RFC3611, November 2003,
            <https://www.rfc-editor.org/info/rfc3611>.
 [RFC5725]  Begen, A., Hsu, D., and M. Lague, "Post-Repair Loss RLE
            Report Block Type for RTP Control Protocol (RTCP) Extended
            Reports (XRs)", RFC 5725, DOI 10.17487/RFC5725, February
            2010, <https://www.rfc-editor.org/info/rfc5725>.
 [RFC6776]  Clark, A. and Q. Wu, "Measurement Identity and Information
            Reporting Using a Source Description (SDES) Item and an
            RTCP Extended Report (XR) Block", RFC 6776,
            DOI 10.17487/RFC6776, October 2012,
            <https://www.rfc-editor.org/info/rfc6776>.
 [RFC6792]  Wu, Q., Ed., Hunt, G., and P. Arden, "Guidelines for Use
            of the RTP Monitoring Framework", RFC 6792,
            DOI 10.17487/RFC6792, November 2012,
            <https://www.rfc-editor.org/info/rfc6792>.

Singh, et al. Informational [Page 14] RFC 8451 RTCP XR Metrics for WebRTC September 2018

 [RFC6958]  Clark, A., Zhang, S., Zhao, J., and Q. Wu, Ed., "RTP
            Control Protocol (RTCP) Extended Report (XR) Block for
            Burst/Gap Loss Metric Reporting", RFC 6958,
            DOI 10.17487/RFC6958, May 2013,
            <https://www.rfc-editor.org/info/rfc6958>.
 [RFC7002]  Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol
            (RTCP) Extended Report (XR) Block for Discard Count Metric
            Reporting", RFC 7002, DOI 10.17487/RFC7002, September
            2013, <https://www.rfc-editor.org/info/rfc7002>.
 [RFC7003]  Clark, A., Huang, R., and Q. Wu, Ed., "RTP Control
            Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
            Discard Metric Reporting", RFC 7003, DOI 10.17487/RFC7003,
            September 2013, <https://www.rfc-editor.org/info/rfc7003>.
 [RFC7004]  Zorn, G., Schott, R., Wu, Q., Ed., and R. Huang, "RTP
            Control Protocol (RTCP) Extended Report (XR) Blocks for
            Summary Statistics Metrics Reporting", RFC 7004,
            DOI 10.17487/RFC7004, September 2013,
            <https://www.rfc-editor.org/info/rfc7004>.
 [RFC7005]  Clark, A., Singh, V., and Q. Wu, "RTP Control Protocol
            (RTCP) Extended Report (XR) Block for De-Jitter Buffer
            Metric Reporting", RFC 7005, DOI 10.17487/RFC7005,
            September 2013, <http://www.rfc-editor.org/info/rfc7005>.
 [RFC7097]  Ott, J., Singh, V., Ed., and I. Curcio, "RTP Control
            Protocol (RTCP) Extended Report (XR) for RLE of Discarded
            Packets", RFC 7097, DOI 10.17487/RFC7097, January 2014,
            <http://www.rfc-editor.org/info/rfc7097>.
 [RFC7243]  Singh, V., Ed., Ott, J., and I. Curcio, "RTP Control
            Protocol (RTCP) Extended Report (XR) Block for the Bytes
            Discarded Metric", RFC 7243, DOI 10.17487/RFC7243, May
            2014, <http://www.rfc-editor.org/info/rfc7243>.
 [RFC7509]  Huang, R. and V. Singh, "RTP Control Protocol (RTCP)
            Extended Report (XR) for Post-Repair Loss Count Metrics",
            RFC 7509, DOI 10.17487/RFC7509, May 2015,
            <http://www.rfc-editor.org/info/rfc7509>.
 [RFC8015]  Singh, V., Perkins, C., Clark, A., and R. Huang, "RTP
            Control Protocol (RTCP) Extended Report (XR) Block for
            Independent Reporting of Burst/Gap Discard Metrics",
            RFC 8015, DOI 10.17487/RFC8015, November 2016,
            <http://www.rfc-editor.org/info/rfc8015>.

Singh, et al. Informational [Page 15] RFC 8451 RTCP XR Metrics for WebRTC September 2018

10.2. Informative References

 [ITU-T_P.800.1]
            ITU-T, "Mean Opinion Score (MOS) terminology", ITU-T
            P.800.1, July 2016,
            <https://www.itu.int/rec/T-REC-P.800.1-201607-I>.
 [RFC4445]  Welch, J. and J. Clark, "A Proposed Media Delivery Index
            (MDI)", RFC 4445, DOI 10.17487/RFC4445, April 2006,
            <https://www.rfc-editor.org/info/rfc4445>.
 [WebRTC-Overview]
            Alverstrand, H., "Overview: Real Time Protocols for
            Browser-based Applications", Work in Progress,
            draft-ietf-rtcweb-overview-19, November 2017.
 [WebRTC-RTP-USAGE]
            Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
            Communication (WebRTC): Media Transport and Use of RTP",
            Work in Progress, draft-ietf-rtcweb-rtp-usage-26, March
            2016.
 [WebRTC-Sec]
            Rescorla, E., "Security Considerations for WebRTC", Work
            in Progress, draft-ietf-rtcweb-security-10, January 2018.
 [RFC7294]  Clark, A., Zorn, G., Bi, C., and Q. Wu, "RTP Control
            Protocol (RTCP) Extended Report (XR) Blocks for
            Concealment Metrics Reporting on Audio Applications",
            RFC 7294, DOI 10.17487/RFC7294, July 2014,
            <https://www.rfc-editor.org/info/rfc7294>.
 [RFC7478]  Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
            Time Communication Use Cases and Requirements", RFC 7478,
            DOI 10.17487/RFC7478, March 2015,
            <https://www.rfc-editor.org/info/rfc7478>.
 [W3C.webrtc]
            Bergkvist, A., Burnett, C., Jennings, C., Narayanan, A.,
            Aboba, B., Brandstetter, T., and J. Bruaroey, "WebRTC 1.0:
            Real-time Communication Between Browsers", W3C Candidate
            Recommendation, June 2018,
            <https://www.w3.org/TR/2018/CR-webrtc-20180621/>.
            Latest version available at
            <https://www.w3.org/TR/webrtc/>.

Singh, et al. Informational [Page 16] RFC 8451 RTCP XR Metrics for WebRTC September 2018

 [W3C.webrtc-stats]
            Alvestrand, H. and V. Singh, "Identifiers for WebRTC's
            Statistics API", W3C Candidate Recommendation, July 2018,
            <https://www.w3.org/TR/2018/CR-webrtc-stats-20180703/>.
            Latest version available at
            <https://www.w3.org/TR/webrtc-stats/>.

Acknowledgements

 The authors would like to thank Bernard Aboba, Harald Alvestrand, Al
 Morton, Colin Perkins, and Shida Schubert for their valuable comments
 and suggestions on earlier draft versions of this document.

Singh, et al. Informational [Page 17] RFC 8451 RTCP XR Metrics for WebRTC September 2018

Authors' Addresses

 Varun Singh
 CALLSTATS I/O Oy
 Annankatu 31-33 C 42
 Helsinki  00100
 Finland
 Email: varun@callstats.io
 URI:   https://www.callstats.io/about
 Rachel Huang
 Huawei
 101 Software Avenue, Yuhua District
 Nanjing  210012
 China
 Email: rachel.huang@huawei.com
 Roni Even
 Huawei
 14 David Hamelech
 Tel Aviv  64953
 Israel
 Email: roni.even@huawei.com
 Dan Romascanu
 Email: dromasca@gmail.com
 Lingli Deng
 China Mobile
 Email: denglingli@chinamobile.com

Singh, et al. Informational [Page 18]

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