Internet Engineering Task Force (IETF) V. Singh Request for Comments: 8451 callstats.io Category: Informational R. Huang ISSN: 2070-1721 R. Even Huawei D. Romascanu Individual L. Deng China Mobile September 2018 Considerations for Selecting RTP Control Protocol (RTCP) Extended Report (XR) Metrics for the WebRTC Statistics API
AbstractThis document describes monitoring features related to media streams in Web real-time communication (WebRTC). It provides a list of RTP Control Protocol (RTCP) Sender Report (SR), Receiver Report (RR), and Extended Report (XR) 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 document is not an Internet Standards Track specification; it is published for informational purposes. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at https://www.rfc-editor.org/info/rfc8451.
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1. Introduction ....................................................4 2. Terminology .....................................................4 3. RTP Statistics in WebRTC Implementations ........................5 4. Considerations for Impact of Measurement Interval ...............5 5. Candidate Metrics ...............................................6 5.1. Network Impact Metrics .....................................6 5.1.1. Loss and Discard Packet Count Metric ................6 5.1.2. Burst/Gap Pattern Metrics for Loss and Discard ......7 5.1.3. Run-Length Encoded Metrics for Loss and Discard .....8 5.2. Application Impact Metrics .................................8 5.2.1. Discarded Octets Metric .............................8 5.2.2. Frame Impairment Summary Metrics ....................9 5.2.3. Jitter Buffer Metrics ...............................9 5.3. Recovery Metrics ..........................................10 5.3.1. Post-Repair Packet Count Metrics ...................10 5.3.2. Run-Length Encoded Metric for Post-Repair ..........10 6. Identifiers from Sender, Receiver, and Extended Report Blocks ..11 6.1. Cumulative Number of Packets and Octets Sent ..............11 6.2. Cumulative Number of Packets and Octets Received ..........11 6.3. Cumulative Number of Packets Lost .........................11 6.4. Interval Packet Loss and Jitter ...........................12 6.5. Cumulative Number of Packets and Octets Discarded .........12 6.6. Cumulative Number of Packets Repaired .....................12 6.7. Burst Packet Loss and Burst Discards ......................12 6.8. Burst/Gap Rates ...........................................13 6.9. Frame Impairment Metrics ..................................13 7. Adding New Metrics to WebRTC Statistics API ....................13 8. Security Considerations ........................................14 9. IANA Considerations ............................................14 10. References ....................................................14 10.1. Normative References .....................................14 10.2. Informative References ...................................16 Acknowledgements ..................................................17 Authors' Addresses ................................................18
WebRTC-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.webrtc-stats] currently lists metrics reported in the RTCP Sender Report 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] and burst/gap discard metric reporting [RFC7003], have been produced for the purpose of collecting and reporting performance metrics from RTP endpoint devices that can be used to have end-to-end service visibility and to measure the delivery 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.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. RFC3550], [RFC3611], and [RFC7478], this document uses the following term. ReportGroup: It is a set of metrics identified by a common synchronization source (SSRC).
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 monitoring quality. 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 statistics from the browser by querying the getStats() API [W3C.webrtc]. The browser can easily report the local variables, i.e., the statistics related to the outgoing and 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. [WebRTC-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, it can report the information to an application server or a third-party monitoring system, which provides quality estimates or diagnostic 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. RFC6776]. When using WebRTC getStats() APIs (see "Statistics Model" in [W3C.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 the stack for statistics every 1 second. In that 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 previously, because 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. ITU-T_P.800.1] values or Media Delivery Index (MDI) [RFC4445] for their services.
that have been lost since the beginning of reception. However, 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 a statistical point of view, while the users are actually 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 metrics for lost packets and duplicated packets defined in the 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 packet loss metrics of [RFC3611] to indicate the lost packet 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 jitter. It augments the loss statistics metrics specified in standard RTCP SR/RR. For those WebRTC 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.
[RFC3611] introduces burst gap metrics in the 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. Metrics related to the burst gap are also introduced in [RFC7003] and [RFC6958], which define two new report blocks for use in a range of RTP applications beyond those described in [RFC3611]. These metrics distinguish discarded packets from loss packets that occur in the burst period and provide 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. RFC3611] and [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.webrtc-stats], these types of metrics are not recommended for use due to the large amount of data and the computation involved.
For WebRTC, the discarded octets metric supplements the metrics on 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 Bytes Discarded metric is defined in [RFC7243]. 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.
calculate estimated MOS values. Thus, for those cases, jitter buffer metrics should be considered. The definition of these metrics is provided in [RFC7005]. RFC7294] as part of recovery metrics. WebRTC-RTP-USAGE]. For these web applications using repair mechanisms, providing some statistics about the performance of their repair mechanisms could help provide 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. 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, when RTCP XR metrics are supported, using run-length encoded metrics is not suggested because the per-packet information yields an enormous amount of data that is not required in this case. 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.
RFC3550], the octets metrics represent the payload size (i.e., not including the header or padding). Section 6.4.1 of [RFC3550]. Name: bytesSent Definition: Section 6.4.1 of [RFC3550]. Section 6.4.1 of [RFC3550]. Name: bytesReceived Definition: Section 6.4.1 of [RFC3550]. Section 6.4.1 of [RFC3550].
Section 6.4.1 of [RFC3550]. Name: fractionLost Definition: Section 6.4.1 of [RFC3550]. Appendix A of [RFC7002]. Name: bytesDiscarded Definition: The cumulative number of octets discarded due to late or early arrival; see Appendix A of [RFC7243]. Appendix A of [RFC7509]. To clarify, the value is the upper bound on the cumulative number of lost packets. Appendix A of [RFC6958]. Name: burstLossCount Definition: The cumulative number of bursts of lost RTP packets; see item d of Appendix A of [RFC6958]. Name: burstPacketsDiscarded Definition: The cumulative number of RTP packets discarded during discard bursts; see item b of Appendix A of [RFC7003]. Name: burstDiscardCount Definition: The cumulative number of bursts of discarded RTP packets; see item e of Appendix A of [RFC8015]. [RFC3611] recommends a Gmin (threshold) value of 16 for classifying packet loss or discard burst.
Appendix A of [RFC7004]. Name: gapLossRate Definition: The fraction of RTP packets lost during gaps; see item b of Appendix A of [RFC7004]. Name: burstDiscardRate Definition: The fraction of RTP packets discarded during bursts; see item e of Appendix A of [RFC7004]. Name: gapDiscardRate Definition: The fraction of RTP packets discarded during gaps; see item f of Appendix A of [RFC7004]. Appendix A of [RFC7004]. Name: framesCorrupted Definition: The cumulative number of frames partially lost; see item j of Appendix A of [RFC7004]. Name: framesDropped Definition: The cumulative number of full frames discarded; see item g of Appendix A of [RFC7004]. Name: framesSent Definition: The cumulative number of frames sent. Name: framesReceived Definition: The cumulative number of partial or full frames received. https://github.com/w3c/webrtc-stats).
RFC3611] and [RFC6792]. The overall security considerations for RTP used in WebRTC applications is described in [WebRTC-RTP-USAGE] and [WebRTC-Sec], which also apply to this memo. [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, <https://www.rfc-editor.org/info/rfc3550>. [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, <https://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, <https://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, <https://www.rfc-editor.org/info/rfc6776>. [RFC6792] Wu, Q., Ed., Hunt, G., and P. Arden, "Guidelines for Use of the RTP Monitoring Framework", RFC 6792, DOI 10.17487/RFC6792, November 2012, <https://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, <https://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, <https://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, <https://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, <https://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 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>.
[ITU-T_P.800.1] ITU-T, "Mean Opinion Score (MOS) terminology", ITU-T P.800.1, July 2016, <https://www.itu.int/rec/T-REC-P.800.1-201607-I>. [RFC4445] Welch, J. and J. Clark, "A Proposed Media Delivery Index (MDI)", RFC 4445, DOI 10.17487/RFC4445, April 2006, <https://www.rfc-editor.org/info/rfc4445>. [WebRTC-Overview] Alverstrand, H., "Overview: Real Time Protocols for Browser-based Applications", Work in Progress, draft-ietf-rtcweb-overview-19, November 2017. [WebRTC-RTP-USAGE] Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Communication (WebRTC): Media Transport and Use of RTP", Work in Progress, draft-ietf-rtcweb-rtp-usage-26, March 2016. [WebRTC-Sec] Rescorla, E., "Security Considerations for WebRTC", Work in Progress, draft-ietf-rtcweb-security-10, 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, DOI 10.17487/RFC7294, July 2014, <https://www.rfc-editor.org/info/rfc7294>. [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Time Communication Use Cases and Requirements", RFC 7478, DOI 10.17487/RFC7478, March 2015, <https://www.rfc-editor.org/info/rfc7478>. [W3C.webrtc] Bergkvist, A., Burnett, C., Jennings, C., Narayanan, A., Aboba, B., Brandstetter, T., and J. Bruaroey, "WebRTC 1.0: Real-time Communication Between Browsers", W3C Candidate Recommendation, June 2018, <https://www.w3.org/TR/2018/CR-webrtc-20180621/>. Latest version available at <https://www.w3.org/TR/webrtc/>.
[W3C.webrtc-stats] Alvestrand, H. and V. Singh, "Identifiers for WebRTC's Statistics API", W3C Candidate Recommendation, July 2018, <https://www.w3.org/TR/2018/CR-webrtc-stats-20180703/>. Latest version available at <https://www.w3.org/TR/webrtc-stats/>.
https://www.callstats.io/about Rachel Huang Huawei 101 Software Avenue, Yuhua District Nanjing 210012 China Email: firstname.lastname@example.org Roni Even Huawei 14 David Hamelech Tel Aviv 64953 Israel Email: email@example.com Dan Romascanu Email: firstname.lastname@example.org Lingli Deng China Mobile Email: email@example.com