Considerations for Selecting RTP Control Protocol (RTCP) Extended Report (XR) Metrics for the WebRTC Statistics API
draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-10

XR Block Working Group                                          V. Singh
Internet-Draft                                              callstats.io
Intended status: Informational                                  R. Huang
Expires: November 26, 2018                                       R. Even
                                                                  Huawei
                                                            D. Romascanu
                                                              Individual
                                                                 L. Deng
                                                            China Mobile
                                                            May 25, 2018

                                    
                                    
                                    

 Considerations for Selecting RTCP Extended Report (XR) Metrics for the
                         WebRTC Statistics API
              draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-10

Abstract

   This document describes monitoring features related to media streams
   in Web real-time communication (WebRTC).  It provides a list of RTCP
   Sender Report, Receiver Report and Extended Report 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

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   This Internet-Draft will expire on January 21, 2018.

Copyright Notice
 

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   Copyright (c) 2018 IETF Trust and the persons identified as the
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Table of Contents

1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
3.  RTP Statistics in WebRTC Implementations  . . . . . . . . . .   3
4.  Considerations for Impact of Measurement Interval . . . . . .   4
5.  Candidate Metrics . . . . . . . . . . . . . . . . . . . . . .   5
  5.1.  Network Impact Metrics  . . . . . . . . . . . . . . . . .   5
    5.1.1.  Loss and Discard Packet Count Metric  . . . . . . . .   5
    5.1.2.  Burst/Gap Pattern Metrics for Loss and Discard  . . .   6
    5.1.3.  Run Length Encoded Metrics for Loss, Discard  . . . .   7
  5.2.  Application Impact Metrics  . . . . . . . . . . . . . . .   7
    5.2.1.  Discard Octets Metric . . . . . . . . . . . . . . . .   7
    5.2.2.  Frame Impairment Summary Metrics  . . . . . . . . . .   8
    5.2.3.  Jitter Buffer Metrics . . . . . . . . . . . . . . . .   8
  5.3.  Recovery metrics  . . . . . . . . . . . . . . . . . . . .   9
    5.3.1.  Post-repair Packet Count Metrics  . . . . . . . . . .   9
    5.3.2.  Run Length Encoded Metric for Post-repair . . . . . .   9
6.  Identifiers from Sender, Receiver, and Extended Report Blocks  10
  6.1.  Cumulative Number of Packets and Octets Sent  . . . . . .  10
  6.2.  Cumulative Number of Packets and Octets Received  . . . .  10
  6.3.  Cumulative Number of Packets Lost . . . . . . . . . . . .  11
  6.4.  Interval Packet Loss and Jitter . . . . . . . . . . . . .  11
  6.5.  Cumulative Number of Packets and Octets Discarded . . . .  11
  6.6.  Cumulative Number of Packets Repaired . . . . . . . . . .  11
  6.7.  Burst Packet Loss and Burst Discards  . . . . . . . . . .  11
  6.8.  Burst/Gap Rates . . . . . . . . . . . . . . . . . . . . .  12
  6.9.  Frame Impairment Metrics  . . . . . . . . . . . . . . . .  12
7.  Adding new metrics to WebRTC Statistics API . . . . . . . . .  13
8.  Security Considerations . . . . . . . . . . . . . . . . . . .  13
9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  13
10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  13
  10.1.  Normative References . . . . . . . . . . . . . . . . . .  13
  10.2.  Informative References . . . . . . . . . . . . . . . . .  15
 

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Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  16

1.  Introduction

   Web real-time communication (WebRTC) [I-D.ietf-rtcweb-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.WD-webrtc-stats] currently lists metrics
   reported in the RTCP Sender 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],
   burst/gap discard metric reporting [RFC7003], and etc., have been
   produced for the purpose of collecting and reporting performance
   metrics from RTP endpoint devices that can be used to have a end-to-
   end service visibility and measure the delivering 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.WD-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

   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 quality
   monitoring.  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 the statistic from the browser by
   querying the getStats() API [W3C.WD-webrtc]. The browser can easily
   report the local variables i.e., the statistics related to the
   outgoing RTP media streams and the 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. [I-D.ietf-rtcweb-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, they can report it to an application server or a third-
   party monitoring system, which provides quality estimations or
   diagnosis 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 section 7 of [W3C.WD-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 application may choose to poll
 

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   the stack for statistics every 1 second. In this 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 before for the remote statistics, as 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 local by RTCP XR
   packets.

   The following metrics are classified into 3 categories: 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 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 of the 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 which 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 causes 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.  But 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
   the statistic point of view, but meanwhile the users may actually be
   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 lost packets and duplicated packets metrics defined in 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 loss packets
   metrics of [RFC3611] to indicate the packet lost 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 the jitter.  It augments the loss statistics metrics
   specified in standard RTCP SR/RR.  For those RTCWEB 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
   services quality, burst gap metrics provides more accurate
   information than RTCP SR/RR.

   [RFC3611] introduces burst gap metrics in VoIP report block.  These
   metrics record the density and duration of burst and gap periods,
 

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   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.  Burst gap related metrics are also introduced in
   [RFC7003] and [RFC6958] which define two new report blocks for usage
   in a range of RTP applications beyond those described in [RFC3611].
   These metrics distinguish discarded packets from loss packets that
   occur in the bursts period and provides 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, Discard

   Run-length encoding uses a bit vector to encode information about the
   packet.  Each bit in the vector represents a packet and 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], [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.WD-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 number of discarded packets.  An
   application measures sent octets and received octets to calculate
   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.

   For WebRTC, discarded octets supplements the 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.
 

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   The discarded bytes 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, statistics from standard SR/RR only collect
   information from transport layer, which 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, number
   of lost derived frames. These metrics can be used to calculate Media
   Loss Rate (MLR) of 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, slower packets are not played out and 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.  Jitter buffer related metrics,
   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 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
 

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   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 [draft-ietf-rtcweb-rtp-
   usage]. For these web applications using repair mechanisms, providing
   some statistic information for the performance of their repair
   mechanisms could help to have 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, it is not suggested to use due to their enormous
   amount of data when RTCP XR are supported.

   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.

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
 

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   receiving SR and RR are mandatory, the metrics defined in the SR and
   RR report blocks are always available.  For XR metrics, it depends on
   two factors: 1) if it is measured at the endpoint, 2) if it is
   reported by the endpoint in an XR report.  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 report, 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/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 Jan 1,
   1970).

   As per [RFC3550], the octets metrics represent the payload size
   (i.e., not including header or padding).

6.1.  Cumulative Number of Packets and Octets Sent

   Name: packetsSent

   Definition: section 6.4.1 in [RFC3550].

   Name: bytesSent

   Definition: section 6.4.1 in [RFC3550].

6.2.  Cumulative Number of Packets and Octets Received

   Name: packetsReceived

   Definition: section 6.4.1 in [RFC3550].

   Name: bytesReceived

   Definition: section 6.4.1 in [RFC3550].

 

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6.3.  Cumulative Number of Packets Lost

   Name: packetsLost

   Definition: section 6.4.1 in [RFC3550].

6.4.  Interval Packet Loss and Jitter

   Name: jitter

   Definition: section 6.4.1 in [RFC3550].

   Name: fractionLost

   Definition: section 6.4.1 in [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, Appendix A (a) of [RFC7002].

   Name: bytesDiscarded

   Definition: The cumulative number of octets discarded due to late or
   early-arrival, 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, Appendix A (b) of [RFC7509].
   To clarify, the value is upper bound to 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, Appendix A (c) of [RFC6958].

   Name: burstLossCount

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

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   Name: burstPacketsDiscarded

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

   Name: burstDiscardCount

   Definition: The cumulative number of bursts of discarded RTP packets,
   Appendix A (e) of [RFC8015].

   [RFC3611] recommends a Gmin (threshold) value of 16 for classifying
   packet loss or discard burst.

6.8.  Burst/Gap Rates

   Name: burstLossRate

   Definition: The fraction of RTP packets lost during bursts,
   Appendix A (a) of [RFC7004].

   Name: gapLossRate

   Definition: The fraction of RTP packets lost during gaps, Appendix A
   (b) of [RFC7004].

   Name: burstDiscardRate

   Definition: The fraction of RTP packets discarded during bursts,
   Appendix A (e) of [RFC7004].

   Name: gapDiscardRate

   Definition: The fraction of RTP packets discarded during gaps,
   Appendix A (f) of [RFC7004].

6.9.  Frame Impairment Metrics

   Name: framesLost

   Definition: The cumulative number of full frames lost, Appendix A (i)
   of [RFC7004].

   Name: framesCorrupted

   Definition: The cumulative number of frames partially lost,
   Appendix A (j) of [RFC7004].

   Name: framesDropped
 

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   Definition: The cumulative number of full frames discarded,
   Appendix A (g) 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

   During the progress of this work, the metrics defined in this draft
   have already been added to the W3C WebRTC specification.  The working
   process to add new metrics for future is to create an issue or pull
   request on the repository of the W3C WebRTC specification
   (https://github.com/w3c/webrtc-stats).

8.  Security Considerations

   This document focuses on listing the RTCP XR metrics defined in the
   corresponding RTCP reporting extensions and do not give rise to any
   new security vulnerabilities beyond those described in [RFC3611] and
   [RFC6792].

   The overall security considerations for RTP used in WebRTC
   applications is described in [I-D.ietf-rtcweb-rtp-usage] and       
   [I-D.ietf-rtcweb-security], which are also apply to this memo.

9.  IANA Consideration

   This document requests no action by IANA.

10.  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 version of this document.

11.  References

11.1.  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, <http://www.rfc-editor.org/info/rfc3550>.
 

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   [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,
              <http://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, <http://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,
              <http://www.rfc-editor.org/info/rfc6776>.

   [RFC6792]  Qu, Q. and P. Arden, "Guidelines for Use of the RTP
              Monitoring Framework", RFC 6792, November 2012,
              <http://www.rfc-editor.org/info/rfc6792>

   [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,
              <http://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, <http://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, <http://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,
              <http://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
 

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              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>.

11.2.  Informative References

   [I-D.ietf-rtcweb-overview] H. Alverstrand, "Overview: Real Time
              Protocols for Browser-based Applications", draft-ietf-
              rtcweb-overview-19  (work in progress), November 2017.

   [ITU-T P.800.1] "Mean Opinion Score (MOS) terminology", ITU-T
              P.800.1, July 2016.

   [RFC4445]  Welch, J. and J. Clark, "A Proposed Media Delivery Index
              (MDI)", RFC4445, April 2006.

   [I-D.ietf-rtcweb-rtp-usage] Perkins, C., Westerlund, M., and J. Ott,
              "Web Real-Time Communication (WebRTC): Media Transport and
              Use of RTP", draft-ietf-rtcweb-rtp-usage-26 (work in
              progress), March 2016.

   [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for
              WebRTC", draft-ietf-rtcweb-security-10 (work in progress),
              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, July 2014, <http://www.rfc-editor.org/info/rfc7294>

   [RFC7478]  Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use Cases and Requirements", RFC 7478,
 

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              DOI 10.17487/RFC7478, March 2015, <http://www.rfc-
              editor.org/info/rfc7478>.

   [W3C.WD-webrtc] Bergkvist, A., Burnett, C., Jennings, C., Narayanan,
              A., Aboba B. and T. Brandstetter, "WebRTC 1.0: Real-time
              Communication Between Browsers", World Wide Web Consortium
              WD WD-webrtc-20171102, November 2017,
              <https://www.w3.org/TR/2017/CR-webrtc-20171102/>.

   [W3C.WD-webrtc-stats] Alvestrand, H. and V. Singh, "Identifiers for
              WebRTC's Statistics API", World Wide Web Consortium WD WD-
              webrtc-stats-20180519, May 2018,
              <https://www.w3.org/TR/2018/WD-webrtc-stats-20180519/>.

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, CN  210012
   China

   Email: rachel.huang@huawei.com

   Roni Even
   Huawei
   14 David Hamelech
   Tel Aviv  64953
   Israel

   Email: roni.even@huawei.com

   Dan Romascanu

 

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   Email: dromasca@gmail.com

   Lingli Deng
   China Mobile

   Email: denglingli@chinamobile.com

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