Internet DRAFT - draft-singh-xrblock-webrtc-additional-stats

draft-singh-xrblock-webrtc-additional-stats







XR Block Working Group                                          V. Singh
Internet-Draft                                          Aalto University
Intended status: Standards Track                                R. Huang
Expires: August 18, 2014                                         R. Even
                                                                  Huawei
                                                       February 14, 2014


Additional RTP Control Protocol (RTCP) Extended Report (XR) Metrics for
                         WebRTC Statistics API
             draft-singh-xrblock-webrtc-additional-stats-02

Abstract

   This document describes a list of additional identifiers used in
   WebRTC's Javascript 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
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   This Internet-Draft will expire on August 18, 2014.

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
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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Candidate XR Block Metrics for WebRTC   Statistics API  . . .   3
     2.1.  Variables from XR Blocks  . . . . . . . . . . . . . . . .   3
       2.1.1.  Packets and Octets Discarded  . . . . . . . . . . . .   3
       2.1.2.  Cumulative Number of Packets Repaired . . . . . . . .   4
       2.1.3.  Burst Packet Loss or Discarded  . . . . . . . . . . .   4
       2.1.4.  Frame Impairment Metrics  . . . . . . . . . . . . . .   4
   3.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   5
   4.  Security Considerations . . . . . . . . . . . . . . . . . . .   5
   5.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   5
   6.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   5
     6.1.  Normative References  . . . . . . . . . . . . . . . . . .   5
     6.2.  Informative References  . . . . . . . . . . . . . . . . .   6
   Appendix A.  Change Log . . . . . . . . . . . . . . . . . . . . .   7
     A.1.  changes in draft-singh-xrblock-webrtc-additional-stats-01   7
     A.2.  changes in draft-singh-xrblock-webrtc-additional-stats-00   7
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   7

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 Javascript API.  It currently
   lists basic metrics reported in the RTCP Sender and Receiver Report
   (SR/RR) to fulfill this requirement.  However, the basic metrics from



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   RTCP SR/RR are not sufficient for precise quality monitoring or
   troubleshooting.  This document proposes to expose the RTCP XR
   metrics to complement the identifiers already in the statistics
   registry [I-D.alvestrand-rtcweb-stats-registry].  In depth discussion
   about the XR metrics candidates is carried out in
   [I-D.huang-xrblock-rtcweb-rtcp-xr-metrics].

   The WebRTC application has two options to extract the statistics: 1)
   the browser monitors the local (outgoing) and remote (incoming) RTP
   stream and exposes metrics to the application via the Stats API
   [W3C.WD-webrtc-20130910], or 2) the browser measures the remote
   (incoming) RTP stream and exposes the metrics to the other
   participant by sending the appropriate RTCP XR.  At the moment
   [I-D.ietf-rtcweb-rtp-usage] does not specify the use of any RTCP XRs
   and since their usage is optional, the exchange of statistics between
   participants or a monitoring server is outside the scope of this
   document.

2.  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
   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


2.1.  Variables from XR Blocks

2.1.1.  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



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   Definition: Cumulative Number of octets discarded due to late or
   early-arrival, Appendix A of
   [I-D.ietf-xrblock-rtcp-xr-bytes-discarded-metric]

2.1.2.  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.huang-xrblock-post-repair-loss-count].  To clarify, the value is
   upper bound to the cumulative number of lost packets.

2.1.3.  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

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

   [RFC3611] recommends a Gmin value of 16.

2.1.4.  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]




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3.  IANA Considerations

   This document requests IANA to update the registry described in
   [I-D.alvestrand-rtcweb-stats-registry] with the identifiers defined
   in Section 2.

4.  Security Considerations

   The security considerations of
   [I-D.alvestrand-rtcweb-stats-registry], apply.

5.  Acknowledgements

   This document is a product of discussion in XRBLOCK WG, initial
   motivation for this documented is discussed in
   [I-D.huang-xrblock-rtcweb-rtcp-xr-metrics]

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

6.  References

6.1.  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.

   [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.



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   [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.

   [I-D.ietf-xrblock-rtcp-xr-bytes-discarded-metric]
              Singh, V., Ott, J., and I. Curcio, "RTP Control Protocol
              (RTCP) Extended Report (XR) for Bytes Discarded Metric",
              draft-ietf-xrblock-rtcp-xr-bytes-discarded-metric-00 (work
              in progress), October 2013.

   [I-D.huang-xrblock-post-repair-loss-count]
              Huang, R. and V. Singh, "RTP Control Protocol (RTCP)
              Extended Report (XR) for Post-Repair Non-Run Length
              Encoding (RLE) Loss Count Metrics", draft-huang-xrblock-
              post-repair-loss-count-00 (work in progress), 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.

6.2.  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-10 (work in
              progress), December 2012.

   [I-D.huang-xrblock-rtcweb-rtcp-xr-metrics]
              Huang, R., Even, R., and V. Singh, "Consideration for
              Selecting RTCP Extended Report (XR) Metrics for RTCWEB
              Statistics API", draft-huang-xrblock-rtcweb-rtcp-xr-
              metrics-01 (work in progress), July 2013.

   [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]




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              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-11 (work in progress),
              December 2013.

Appendix A.  Change Log

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

A.1.  changes in draft-singh-xrblock-webrtc-additional-stats-01

   o  Addressed comments from the Vancouver IETF meeting.

   o  Added Burst discard/loss metric.

   o  Removed retransmission metric.

A.2.  changes in draft-singh-xrblock-webrtc-additional-stats-00

   o  Clarified measurement points for remote (incoming) media stream
      and local (outgoing) media stream.

   o  Added this section.

Authors' Addresses

   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/


   Rachel Huang
   Huawei
   101 Software Avenue, Yuhua District
   Nanjing, CN  210012
   China

   Email: rachel.huang@huawei.com






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   Roni Even
   Huawei
   14 David Hamelech
   Tel Aviv  64953
   Israel

   Email: roni.even@mail01.huawei.com











































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