Considerations for Selecting RTP Control Protocol (RTCP) Extended Report (XR) Metrics for the WebRTC Statistics API
RFC 8451

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.

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Copyright Notice

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   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   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.

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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

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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).

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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

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   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

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   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.

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   [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.

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   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

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   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>.

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   [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>.

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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/>.

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   [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.

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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

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