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Versions: 00 01 02 03 04 draft-ietf-xrblock-rtcweb-rtcp-xr-metrics

XR Block Working Group                                          R. Huang
Internet-Draft                                                   R. Even
Intended status: Standards Track                                  Huawei
Expires: January 5, 2015                                        V. Singh
                                                        Aalto University
                                                            D. Romascanu
                                                                   Avaya
                                                                 L. Deng
                                                            China Mobile
                                                            July 4, 2014


 Considerations for Selecting RTCP Extended Report (XR) Metrics for the
                         RTCWEB Statistics API
             draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04

Abstract

   This document describes monitoring features related to RTCWEB.  It
   provides a list of RTCP XR metrics, which are useful and may need to
   be supported in some RTCWEB implementations.  It also describes a
   list of additional identifiers for WebRTC's statistics API.  These
   identifiers are a set of RTCP XR metrics related to the transport of
   multimedia flows.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on January 5, 2015.

Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.





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   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://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.

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  . . . . . . . . . .   7
       5.2.3.  Jitter Buffer Metrics . . . . . . . . . . . . . . . .   8
     5.3.  Recovery metrics  . . . . . . . . . . . . . . . . . . . .   8
       5.3.1.  Post-repair Packet Count Metrics  . . . . . . . . . .   9
       5.3.2.  Run Length Encoded Metric for Post-repair . . . . . .   9
   6.  Candidate XR Block Metrics for WebRTC Statistics API  . . . .   9
     6.1.  Variables from XR Blocks  . . . . . . . . . . . . . . . .  10
       6.1.1.  Packets and Octets Discarded  . . . . . . . . . . . .  10
       6.1.2.  Cumulative Number of Packets Repaired . . . . . . . .  10
       6.1.3.  Burst Packet Loss or Discarded  . . . . . . . . . . .  10
       6.1.4.  Frame Impairment Metrics  . . . . . . . . . . . . . .  11
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  11
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  11
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  11
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  12
     10.2.  Informative References . . . . . . . . . . . . . . . . .  13
   Appendix A.  Change Log . . . . . . . . . . . . . . . . . . . . .  13
     A.1.  changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04   14
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  14








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1.  Introduction

   Web-based real-time communication (WebRTC) deployments are emerging
   and applications need to be able to estimate the service quality.  If
   sufficient information (metrics or statistics) are provided to the
   applications, it can attempt to improve the media quality.
   [I-D.ietf-rtcweb-use-cases-and-requirements] 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 [I-D.alvestrand-rtcweb-stats-registry] describes a registration
   procedure for metrics reported by the WebRTC Stats API
   [W3C.WD-webrtc-20130910].  It currently lists basic 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.

   In this document, we provide some guidelines on choosing additional
   RTP metrics for the WebRTC Stats API [W3C.WD-webrtc-20130910].
   Furthermore, we expose additional RTCP XR metrics to complement the
   identifiers that already exist in the statistics registry
   [I-D.alvestrand-rtcweb-stats-registry].

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

3.  RTP Statistics in WebRTC Implementations

   Currently, the statistics registry
   [I-D.alvestrand-rtcweb-stats-registry] exposes the basic RTCP SR and
   RR metrics for the local and remote media streams.  The exposed
   identifiers are: SentPacketCount, SentOctetCount, packetsLost,
   Jitter, ReceivedPacketCount, ReceivedOctetCount.  However, these
   metrics provides 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 diagnosing and
   identifying issues.





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   RTP Control Protocol Extended Reports [RFC3611] and other extensions
   discussed in the XRBLOCK working group provide more detailed
   statistics, which complement the basic metrics reported in the RTCP
   Sender and Receiver Reports.  Section Section 5 discusses the use of
   XR metrics that may be useful for monitoring the performance of
   WebRTC applications.

   The WebRTC application extracts the statistic from the browser by
   querying the Stats API [W3C.WD-webrtc-20130910], but the browser
   currently only reports the local variables i.e., the statistics
   related to the outgoing RTP media streams and the incoming RTP media
   streams.  Without the support of RTCP XRs or some other signaling
   mechianism, the WebRTC application cannot expose the remote
   endpoints' statistics.  At the moment [I-D.ietf-rtcweb-rtp-usage]
   does not mandate the use of any RTCP XRs and since 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 Statistics APIs (see section 7 of
   [W3C.WD-webrtc-20130910]), 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.  Some
   applications may choose 1 second or a different polling interval, but
   the statistics from the remote endpoint may not change when using
   intervals shorter than the average RTCP reporting interval.  However,
   statistics generated by the local endpoint have no such restrictions
   as long as the endpoint is sending and receiving media.







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5.  Candidate Metrics

   Since 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, following metrics can
   also be generated remotely and be sent to local by RTCP XR packets.

   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 in the
   viewpoint of application, e.g., bitrate, frames rate or jitter
   buffers.  Recovery metrics reflect how well the repair mechanisms,
   e.g. loss concealment, retransmission or FEC, perform.  All of the 3
   types of metrics are useful for quality estimations of services in
   WebRTC implementations.  WebRTC application can use these metrics to
   better calculate MoS values or Media Delivery Index (MDI) 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,
   which 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 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




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   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 packets 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 buffer 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 some transitory nature of the 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.  Distributed burst provides a higher
   subjective quality than a non-burst distribution 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,
   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



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   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 application 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,
   i.e., a router queue became overloaded causing delays and then
   overflowed.  If the losses are independent, it may indicate bit-error
   corruption.  For the WebRTC StatsAPI, 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.  Discard 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 bitrate 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 bitrate
   which is an important parameter to reflect the quality of the media.
   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, each of which has a fixed length.  On the other hand,
   in video streams, a single video frame may be transmitted in multiple



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   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 following metrics can also be considered for WebRTC's Statistics
   API: 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.
   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 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

   [Editor's Note: Concealment Metrics are currently not considered.]








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5.3.1.  Post-repair Packet Count Metrics

   Error-resilience mechanisms, like RTP retransmission or FEC, are
   optional in RTCWEB because the overhead of the repair bits adding to
   the original streams.  But they do help to greatly reduce the impact
   of packet loss and enhance the quality of transmission.  Web
   applications could support certain repair mechanism after negotiation
   between both sides of browsers when needed.  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 un-repaired packets count and repaired loss count defined in
   [I-D.ietf-xrblock-rtcp-xr-post-repair-loss-count] 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 this kind of 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, i.e., 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.  Candidate XR Block Metrics for WebRTC Statistics API

   This document describes a list of additional identifiers to
   complement the identifiers in Section 4.1 of
   [I-D.alvestrand-rtcweb-stats-registry] and these group of identifiers
   are defined on a ReportGroup corresponding to an SSRC.  In practice
   the application MUST be able to query the statistic identifiers on



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   both an incoming (remote) and outgoing (local) media stream.
   Depending on the support of the corresponding XR report the endpoint
   MAY be able to query the reception statistics for its outgoing
   (local) media stream.

   The following contact information is used for all registrations in
   this document:

       Contact:    Varun Singh
                   mailto:varun.singh@iki.fi
                   tel:+358-9-470-24785

6.1.  Variables from XR Blocks

6.1.1.  Packets and Octets Discarded

   Name: PacketsDiscarded

   Definition: Cumulative Number of RTP packets discarded due to late or
   early-arrival, Appendix A (a) of [RFC7002].

   Name: OctetsDiscarded

   Definition: Cumulative Number of octets discarded due to late or
   early-arrival, Appendix A of [RFC7243]

6.1.2.  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
   [I-D.ietf-xrblock-rtcp-xr-post-repair-loss-count].  To clarify, the
   value is upper bound to the cumulative number of lost packets.

6.1.3.  Burst Packet Loss or Discarded

   Name: BurstPacketDiscarded

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

   Name: BurstPacketLost

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

   Name: BurstCount



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

   [RFC3611] recommends a Gmin value of 16.

6.1.4.  Frame Impairment Metrics

   Name: FullFramesLostCount

   Definition: Number of full frames lost, Appendix A (i) of [RFC7004]

   Name: PartialFramesLostCount

   Definition: Number of frames partially lost, Appendix A (j) of
   [RFC7004]

   Name: FramesDiscardedCount

   Definition: Number of full frames discarded, Appendix A (g) of
   [RFC7004]

7.  IANA Considerations

   This document requests IANA to update the registry described in
   [I-D.alvestrand-rtcweb-stats-registry] with the identifiers defined
   in Section 6.

8.  Security Considerations

   The monitoring activities are implemented between two browsers or
   between a browser and a server.  Therefore encryption procedures,
   such as the ones suggested for a Secure RTCP (SRTCP), need to be
   used.  Currently, the monitoring in RTCWEB introduces no new security
   considerations beyond those described in [I-D.ietf-rtcweb-rtp-usage],
   [I-D.ietf-rtcweb-security], and
   [I-D.alvestrand-rtcweb-stats-registry].

9.  Acknowledgements

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

10.  References







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10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [I-D.alvestrand-rtcweb-stats-registry]
              Alvestrand, H., "A Registry for WebRTC statistics
              identifiers", draft-alvestrand-rtcweb-stats-registry-00
              (work in progress), September 2012.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3611]  Friedman, T., Caceres, R., and A. Clark, "RTP Control
              Protocol Extended Reports (RTCP XR)", RFC 3611, November
              2003.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

   [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, February 2010.

   [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, October 2012.

   [RFC6958]  Clark, A., Zhang, S., Zhao, J., and Q. Wu, "RTP Control
              Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
              Loss Metric Reporting", RFC 6958, May 2013.

   [RFC7002]  Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for Discard Count Metric
              Reporting", RFC 7002, September 2013.

   [RFC7003]  Clark, A., Huang, R., and Q. Wu, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for Burst/Gap Discard
              Metric Reporting", RFC 7003, September 2013.

   [RFC7004]  Zorn, G., Schott, R., Wu, Q., and R. Huang, "RTP Control
              Protocol (RTCP) Extended Report (XR) Blocks for Summary
              Statistics Metrics Reporting", RFC 7004, September 2013.






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   [RFC7005]  Clark, A., Singh, V., and Q. Wu, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for De-Jitter Buffer
              Metric Reporting", RFC 7005, September 2013.

   [RFC7097]  Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol
              (RTCP) Extended Report (XR) for RLE of Discarded Packets",
              RFC 7097, January 2014.

   [RFC7243]  Singh, V., Ott, J., and I. Curcio, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for the Bytes Discarded
              Metric", RFC 7243, May 2014.

   [I-D.ietf-xrblock-rtcp-xr-post-repair-loss-count]
              Huang, R. and V. Singh, "RTP Control Protocol (RTCP)
              Extended Report (XR) for Post-Repair Loss Count Metrics",
              draft-ietf-xrblock-rtcp-xr-post-repair-loss-count-05 (work
              in progress), June 2014.

10.2.  Informative References

   [I-D.ietf-rtcweb-use-cases-and-requirements]
              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements", draft-
              ietf-rtcweb-use-cases-and-requirements-14 (work in
              progress), February 2014.

   [W3C.WD-webrtc-20130910]
              Bergkvist, A., Burnett, D., Jennings, C., and A.
              Narayanan, "WebRTC 1.0: Real-time Communication Between
              Browsers", World Wide Web Consortium WD WD-
              webrtc-20130910, September 2013,
              <http://www.w3.org/TR/2013/WD-webrtc-20130910>.

   [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-15 (work in progress), May
              2014.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-06 (work in progress), January 2014.

Appendix A.  Change Log

   Note to the RFC-Editor: please remove this section prior to
   publication as an RFC.




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A.1.  changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04

   o  Addressed comments from the London IETF meeting:

   o  Removed ECN metrics.

   o  Merged draft-singh-xrblock-webrtc-additional-stats-01

Authors' Addresses

   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@mail01.huawei.com


   Varun Singh
   Aalto University
   School of Electrical Engineering
   Otakaari 5 A
   Espoo, FIN  02150
   Finland

   Email: varun@comnet.tkk.fi
   URI:   http://www.netlab.tkk.fi/~varun/


   Dan Romascanu
   Avaya
   Park Atidim, Bldg. #3
   Tel Aviv  61581
   Israel

   Email: dromasca@avaya.com





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   Lingli Deng
   China Mobile

   Email: denglingli@chinamobile.com















































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