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
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.
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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 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.
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.
Contact: Varun Singh mailto:varun.singh@iki.fi tel:+358-9-470-24785
The following contact information is used for all registrations in this document:
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 [I-D.ietf-xrblock-rtcp-xr-bytes-discarded-metric]
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.
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.
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]
This document requests IANA to update the registry described in [I-D.alvestrand-rtcweb-stats-registry] with the identifiers defined in Section 2.
The security considerations of [I-D.alvestrand-rtcweb-stats-registry], apply.
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.
[I-D.ietf-rtcweb-use-cases-and-requirements] | Holmberg, C., Hakansson, S. and G. Eriksson, "Web Real-Time Communication Use-cases and Requirements", Internet-Draft draft-ietf-rtcweb-use-cases-and-requirements-10, December 2012. |
[I-D.huang-xrblock-rtcweb-rtcp-xr-metrics] | Huang, R., Even, R., Singh, V., Romascanu, D. and D. Lingli, "Considerations for Selecting RTCP Extended Report (XR) Metrics for the RTCWEB Statistics API", Internet-Draft draft-huang-xrblock-rtcweb-rtcp-xr-metrics-03, 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. |
[I-D.ietf-rtcweb-rtp-usage] | Perkins, C., Westerlund, M. and J. Ott, "Web Real-Time Communication (WebRTC): Media Transport and Use of RTP", Internet-Draft draft-ietf-rtcweb-rtp-usage-06, February 2013. |
Note to the RFC-Editor: please remove this section prior to publication as an RFC.