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Versions: (draft-huang-xrblock-rtcweb-rtcp-xr-metrics) 00 01 02 03 04 05

XR Block Working Group                                          V. Singh
Internet-Draft                                              callstats.io
Intended status: Standards Track                                R. Huang
Expires: May 31, 2017                                            R. Even
                                                                  Huawei
                                                            D. Romascanu

                                                                 L. Deng
                                                            China Mobile
                                                       November 27, 2016


 Considerations for Selecting RTCP Extended Report (XR) Metrics for the
                         WebRTC Statistics API
              draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-05

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

   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 May 31, 2017.

Copyright Notice

   Copyright (c) 2016 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  . . . . . . . . . .   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
   Appendix A.  Change Log . . . . . . . . . . . . . . . . . . . . .  16
     A.1.  changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-04 .  16
     A.2.  changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02,
           -03 . . . . . . . . . . . . . . . . . . . . . . . . . . .  16



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     A.3.  changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-01 .  16
     A.4.  changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00 .  16
     A.5.  changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04   16
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  17

1.  Introduction

   Web 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.  [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-20160527] 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.

   In this document, we provide rationale for choosing additional RTP
   metrics for the WebRTC getStats() API [W3C.WD-webrtc-20161124].  The
   document also creates a registry containing identifiers from the
   metrics reported in the RTCP Sender, Receiver, and Extended Reports.
   All identifiers proposed in this document are RECOMMENDED to be
   implemented by an endpoint.  An endpoint MAY choose not to expose an
   identifier if it does not implement the corresponding RTCP Report.

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].

   ReportGroup: It is a set of metrics identified by a common
   Synchronization source (SSRC).

3.  RTP Statistics in WebRTC Implementations

   The RTCP Sender Reports (SRs) and Receiver Reports (RRs) [RFC3550]
   exposes the basic metrics for the local and remote media streams.
   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



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   they have the same impact on the multimedia quality, it helps in
   identifying and diagnosing issues.

   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.  Section 5 discusses the use of XR metrics that
   may be useful for monitoring the performance of WebRTC applications.
   Section 6 proposes a set of candidate metrics.

   The WebRTC application extracts the statistic from the browser by
   querying the getStats() API [W3C.WD-webrtc-20161124], 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
   mechanism, 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 getStats() APIs (see section 7 of
   [W3C.WD-webrtc-20161124]), 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 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



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   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 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., bit rate, frames rate or jitter
   buffers.  Recovery metrics reflect how well the repair mechanisms
   perform, e.g. loss concealment, retransmission or FEC.  All of the 3
   types of metrics are useful for quality estimations of services in
   WebRTC implementations.  WebRTC application can use these metrics to
   calculate the Mean Opinion Score (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



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   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 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



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   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 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 Stats API [W3C.WD-webrtc-stats-20160527],
   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 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.
   The discarded bytes metric is defined in [RFC7243].







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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
   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].






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5.3.  Recovery metrics

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

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
   [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 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.






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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.  These group of
   identifiers are defined on a ReportGroup corresponding to an
   Synchronization source (SSRC).  In practice the application MUST be
   able to query the statistic identifiers on both an incoming (remote)
   and outgoing (local) media stream.  Since sending and 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 measured at the endpoint, 2) if it 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

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

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].

9.  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.

10.  References

10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/
              RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

   [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>.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,
              <http://www.rfc-editor.org/info/rfc4588>.

   [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>.

   [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>.




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   [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>.

10.2.  Informative References

   [RFC7478]  Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use Cases and Requirements", RFC 7478,
              DOI 10.17487/RFC7478, March 2015,
              <http://www.rfc-editor.org/info/rfc7478>.

   [W3C.WD-webrtc-20161124]
              Sporny, M. and D. Longley, "WebRTC 1.0: Real-time
              Communication Between Browsers", World Wide Web Consortium
              WD WD-webrtc-20161124, November 2016,
              <https://www.w3.org/TR/2016/WD-webrtc-20161124>.

   [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-08 (work in progress), February 2015.








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   [W3C.WD-webrtc-stats-20160527]
              Alvestrand, H. and V. Singh, "Identifiers for
              WebRTC&#039;s Statistics API", World Wide Web Consortium
              WD WD-webrtc-stats-20160527, May 2016,
              <http://www.w3.org/TR/2016/WD-webrtc-stats-20160527>.

   [RFC6390]  Clark, A. and B. Claise, "Guidelines for Considering New
              Performance Metric Development", BCP 170, RFC 6390, DOI
              10.17487/RFC6390, October 2011,
              <http://www.rfc-editor.org/info/rfc6390>.

Appendix A.  Change Log

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

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

   o  Removed IANA registry.

A.2.  changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02, -03

   o  Keep-alive versions, updates to references.

A.3.  changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-01

   o  Create new registry for WebRTC media metrics.

   o  Using camelCase instead of TitleCase for identifier names.

   o  Imported RTCP SR and RR metrics from the registry in alvestrand-
      rtcweb-stats-registry.

   o  Added Burst/Gap rate metrics.

   o  Added Frames sent and received metrics.

A.4.  changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00

   o  Submitted as WG Draft.

A.5.  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



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Authors' Addresses

   Varun Singh
   CALLSTATS I/O Oy
   Runeberginkatu 4c A 4
   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@mail01.huawei.com


   Dan Romascanu

   Email: dromasca@gmail.com


   Lingli Deng
   China Mobile

   Email: denglingli@chinamobile.com












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