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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 XR Block Working Group V. Singh 3 Internet-Draft callstats.io 4 Intended status: Standards Track R. Huang 5 Expires: May 31, 2017 R. Even 6 Huawei 7 D. Romascanu 9 L. Deng 10 China Mobile 11 November 27, 2016 13 Considerations for Selecting RTCP Extended Report (XR) Metrics for the 14 WebRTC Statistics API 15 draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-05 17 Abstract 19 This document describes monitoring features related to media streams 20 in Web real-time communication (WebRTC). It provides a list of RTCP 21 Sender Report, Receiver Report and Extended Report metrics, which may 22 need to be supported by RTP implementations in some diverse 23 environments. It lists a set of identifiers for the WebRTC's 24 statistics API. These identifiers are a set of RTCP SR, RR, and XR 25 metrics related to the transport of multimedia flows. 27 Status of This Memo 29 This Internet-Draft is submitted in full conformance with the 30 provisions of BCP 78 and BCP 79. 32 Internet-Drafts are working documents of the Internet Engineering 33 Task Force (IETF). Note that other groups may also distribute 34 working documents as Internet-Drafts. The list of current Internet- 35 Drafts is at http://datatracker.ietf.org/drafts/current/. 37 Internet-Drafts are draft documents valid for a maximum of six months 38 and may be updated, replaced, or obsoleted by other documents at any 39 time. It is inappropriate to use Internet-Drafts as reference 40 material or to cite them other than as "work in progress." 42 This Internet-Draft will expire on May 31, 2017. 44 Copyright Notice 46 Copyright (c) 2016 IETF Trust and the persons identified as the 47 document authors. All rights reserved. 49 This document is subject to BCP 78 and the IETF Trust's Legal 50 Provisions Relating to IETF Documents 51 (http://trustee.ietf.org/license-info) in effect on the date of 52 publication of this document. Please review these documents 53 carefully, as they describe your rights and restrictions with respect 54 to this document. Code Components extracted from this document must 55 include Simplified BSD License text as described in Section 4.e of 56 the Trust Legal Provisions and are provided without warranty as 57 described in the Simplified BSD License. 59 Table of Contents 61 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 62 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 63 3. RTP Statistics in WebRTC Implementations . . . . . . . . . . 3 64 4. Considerations for Impact of Measurement Interval . . . . . . 4 65 5. Candidate Metrics . . . . . . . . . . . . . . . . . . . . . . 5 66 5.1. Network Impact Metrics . . . . . . . . . . . . . . . . . 5 67 5.1.1. Loss and Discard Packet Count Metric . . . . . . . . 5 68 5.1.2. Burst/Gap Pattern Metrics for Loss and Discard . . . 6 69 5.1.3. Run Length Encoded Metrics for Loss, Discard . . . . 7 70 5.2. Application Impact Metrics . . . . . . . . . . . . . . . 7 71 5.2.1. Discard Octets Metric . . . . . . . . . . . . . . . . 7 72 5.2.2. Frame Impairment Summary Metrics . . . . . . . . . . 8 73 5.2.3. Jitter Buffer Metrics . . . . . . . . . . . . . . . . 8 74 5.3. Recovery metrics . . . . . . . . . . . . . . . . . . . . 9 75 5.3.1. Post-repair Packet Count Metrics . . . . . . . . . . 9 76 5.3.2. Run Length Encoded Metric for Post-repair . . . . . . 9 77 6. Identifiers from Sender, Receiver, and Extended Report Blocks 10 78 6.1. Cumulative Number of Packets and Octets Sent . . . . . . 10 79 6.2. Cumulative Number of Packets and Octets Received . . . . 10 80 6.3. Cumulative Number of Packets Lost . . . . . . . . . . . . 11 81 6.4. Interval Packet Loss and Jitter . . . . . . . . . . . . . 11 82 6.5. Cumulative Number of Packets and Octets Discarded . . . . 11 83 6.6. Cumulative Number of Packets Repaired . . . . . . . . . . 11 84 6.7. Burst Packet Loss and Burst Discards . . . . . . . . . . 11 85 6.8. Burst/Gap Rates . . . . . . . . . . . . . . . . . . . . . 12 86 6.9. Frame Impairment Metrics . . . . . . . . . . . . . . . . 12 87 7. Adding new metrics to WebRTC Statistics API . . . . . . . . . 13 88 8. Security Considerations . . . . . . . . . . . . . . . . . . . 13 89 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13 90 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 13 91 10.1. Normative References . . . . . . . . . . . . . . . . . . 13 92 10.2. Informative References . . . . . . . . . . . . . . . . . 15 93 Appendix A. Change Log . . . . . . . . . . . . . . . . . . . . . 16 94 A.1. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-04 . 16 95 A.2. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02, 96 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . . 16 98 A.3. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-01 . 16 99 A.4. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00 . 16 100 A.5. changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04 16 101 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17 103 1. Introduction 105 Web real-time communication (WebRTC) deployments are emerging and 106 applications need to be able to estimate the service quality. If 107 sufficient information (metrics or statistics) are provided to the 108 applications, it can attempt to improve the media quality. [RFC7478] 109 specifies a requirement for statistics: 111 F38 The browser must be able to collect statistics, related to the 112 transport of audio and video between peers, needed to estimate 113 quality of experience. 115 The WebRTC Stats API [W3C.WD-webrtc-stats-20160527] currently lists 116 metrics reported in the RTCP Sender and Receiver Report (SR/RR) 117 [RFC3550] to fulfill this requirement. However, the basic metrics 118 from RTCP SR/RR are not sufficient for precise quality monitoring, or 119 diagnosing potential issues. 121 In this document, we provide rationale for choosing additional RTP 122 metrics for the WebRTC getStats() API [W3C.WD-webrtc-20161124]. The 123 document also creates a registry containing identifiers from the 124 metrics reported in the RTCP Sender, Receiver, and Extended Reports. 125 All identifiers proposed in this document are RECOMMENDED to be 126 implemented by an endpoint. An endpoint MAY choose not to expose an 127 identifier if it does not implement the corresponding RTCP Report. 129 2. Terminology 131 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 132 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 133 document are to be interpreted as described in [RFC2119]. 135 ReportGroup: It is a set of metrics identified by a common 136 Synchronization source (SSRC). 138 3. RTP Statistics in WebRTC Implementations 140 The RTCP Sender Reports (SRs) and Receiver Reports (RRs) [RFC3550] 141 exposes the basic metrics for the local and remote media streams. 142 However, these metrics provides only partial or limited information, 143 which may not be sufficient for diagnosing problems or quality 144 monitoring. For example, it may be useful to distinguish between 145 packets lost and packets discarded due to late arrival, even though 146 they have the same impact on the multimedia quality, it helps in 147 identifying and diagnosing issues. 149 RTP Control Protocol Extended Reports (XRs) [RFC3611] and other 150 extensions discussed in the XRBLOCK working group provide more 151 detailed statistics, which complement the basic metrics reported in 152 the RTCP SR and RRs. Section 5 discusses the use of XR metrics that 153 may be useful for monitoring the performance of WebRTC applications. 154 Section 6 proposes a set of candidate metrics. 156 The WebRTC application extracts the statistic from the browser by 157 querying the getStats() API [W3C.WD-webrtc-20161124], but the browser 158 currently only reports the local variables i.e., the statistics 159 related to the outgoing RTP media streams and the incoming RTP media 160 streams. Without the support of RTCP XRs or some other signaling 161 mechanism, the WebRTC application cannot expose the remote endpoints' 162 statistics. At the moment [I-D.ietf-rtcweb-rtp-usage] does not 163 mandate the use of any RTCP XRs and since their usage is optional. 164 If the use of RTCP XRs is successfully negotiated between endpoints 165 (via SDP), thereafter the application has access to both local and 166 remote statistics. Alternatively, once the WebRTC application gets 167 the local information, they can report it to an application server or 168 a third-party monitoring system, which provides quality estimations 169 or diagnosis services for application developers. The exchange of 170 statistics between endpoints or between a monitoring server and an 171 endpoint is outside the scope of this document. 173 4. Considerations for Impact of Measurement Interval 175 RTCP extensions like RTCP XR usually share the same timing interval 176 with the RTCP SR/RR, i.e., they are sent as compound packets, 177 together with the RTCP SR/RR. Alternatively, if the RTCP XR uses a 178 different measurement interval, all XRs using the same measurement 179 interval are compounded together and the measurement interval is 180 indicated in a specific measurement information block defined in 181 [RFC6776]. 183 When using WebRTC getStats() APIs (see section 7 of 184 [W3C.WD-webrtc-20161124]), the applications can query this 185 information at arbitrary intervals. For the statistics reported by 186 the remote endpoint, e.g., those conveyed in an RTCP SR/RR/XR, these 187 will not change until the next RTCP report is received. However, 188 statistics generated by the local endpoint have no such restrictions 189 as long as the endpoint is sending and receiving media. For example, 190 an application may choose to poll the stack for statistics every 1 191 second, in this case the underlying stack local will return the 192 current snapshot of the local statistics (for incoming and outgoing 193 media streams). However it may return the same remote statistics as 194 before for the remote statistics, as no new RTCP reports may have 195 been received in the past 1 second. This can occur when the polling 196 interval is shorter than the average RTCP reporting interval. 198 5. Candidate Metrics 200 Since following metrics are all defined in RTCP XR which is not 201 mandated in WebRTC, all of them are local. However, if RTCP XR is 202 supported by negotiation between two browsers, following metrics can 203 also be generated remotely and be sent to local by RTCP XR packets. 205 Following metrics are classified into 3 categories: network impact 206 metrics, application impact metrics and recovery metrics. Network 207 impact metrics are the statistics recording the information only for 208 network transmission. They are useful for network problem diagnosis. 209 Application impact metrics mainly collect the information in the 210 viewpoint of application, e.g., bit rate, frames rate or jitter 211 buffers. Recovery metrics reflect how well the repair mechanisms 212 perform, e.g. loss concealment, retransmission or FEC. All of the 3 213 types of metrics are useful for quality estimations of services in 214 WebRTC implementations. WebRTC application can use these metrics to 215 calculate the Mean Opinion Score (MoS) values or Media Delivery Index 216 (MDI) for their services. 218 5.1. Network Impact Metrics 220 5.1.1. Loss and Discard Packet Count Metric 222 In multimedia transport, packets which are received abnormally are 223 classified into 3 types: lost, discarded and duplicate packets. 224 Packet loss may be caused by network device breakdown, bit-error 225 corruption or network congestion (packets dropped by an intermediate 226 router queue). Duplicate packets may be a result of network delays, 227 which causes the sender to retransmit the original packets. 228 Discarded packets are packets that have been delayed long enough 229 (perhaps they missed the playout time) and are considered useless by 230 the receiver. Lost and discarded packets cause problems for 231 multimedia services, as missing data and long delays can cause 232 degradation in service quality, e.g., missing large blocks of 233 contiguous packets (lost or discarded) may cause choppy audio, and 234 long network transmission delay time may cause audio or video 235 buffering. The RTCP SR/RR defines a metric for counting the total 236 number of RTP data packets that have been lost since the beginning of 237 reception. But this statistic does not distinguish lost packets from 238 discarded and duplicate packets. Packets that arrive late will be 239 discarded and are not reported as lost, and duplicate packets will be 240 regarded as a normally received packet. Hence, the loss metric can 241 be misleading if many duplicate packets are received or packets are 242 discarded, which causes the quality of the media transport to appear 243 okay from the statistic point of view, but meanwhile the users may 244 actually be experiencing bad service quality. So in such cases, it 245 is better to use more accurate metrics in addition to those defined 246 in RTCP SR/RR. 248 The lost packets and duplicated packets metrics defined in Statistics 249 Summary Report Block of [RFC3611] extend the information of loss 250 carried in standard RTCP SR/RR. They explicitly give an account of 251 lost and duplicated packets. Lost packets counts are useful for 252 network problem diagnosis. It is better to use the loss packets 253 metrics of [RFC3611] to indicate the packet lost count instead of the 254 cumulative number of packets lost metric of [RFC3550]. Duplicated 255 packets are usually rare and have little effect on QoS evaluation. 256 So it may not be suitable for use in WebRTC. 258 Using loss metrics without considering discard metrics may result in 259 inaccurate quality evaluation, as packet discard due to jitter is 260 often more prevalent than packet loss in modern IP networks. The 261 discarded metric specified in [RFC7002] counts the number of packets 262 discarded due to the jitter. It augments the loss statistics metrics 263 specified in standard RTCP SR/RR. For those RTCWEB services with 264 jitter buffer requiring precise quality evaluation and accurate 265 troubleshooting, this metric is useful as a complement to the metrics 266 of RTCP SR/RR. 268 5.1.2. Burst/Gap Pattern Metrics for Loss and Discard 270 RTCP SR/RR defines coarse metrics regarding loss statistics, the 271 metrics are all about per call statistics and are not detailed enough 272 to capture some transitory nature of the impairments like bursty 273 packet loss. Even if the average packet loss rate is low, the lost 274 packets may occur during short dense periods, resulting in short 275 periods of degraded quality. Distributed burst provides a higher 276 subjective quality than a non-burst distribution for low packet loss 277 rates whereas for high packet loss rates the converse is true. So 278 capturing burst gap information is very helpful for quality 279 evaluation and locating impairments. If the WebRTC application needs 280 to evaluate the services quality, burst gap metrics provides more 281 accurate information than RTCP SR/RR. 283 [RFC3611] introduces burst gap metrics in VoIP report block. These 284 metrics record the density and duration of burst and gap periods, 285 which are helpful in isolating network problems since bursts 286 correspond to periods of time during which the packet loss/discard 287 rate is high enough to produce noticeable degradation in audio or 288 video quality. Burst gap related metrics are also introduced in 289 [RFC7003] and [RFC6958] which define two new report blocks for usage 290 in a range of RTP applications beyond those described in [RFC3611]. 291 These metrics distinguish discarded packets from loss packets that 292 occur in the bursts period and provides more information for 293 diagnosing network problems. Additionally, the block reports the 294 frequency of burst events which is useful information for evaluating 295 the quality of experience. Hence, if WebRTC application need to do 296 quality evaluation and observe when and why quality degrades, these 297 metrics should be considered. 299 5.1.3. Run Length Encoded Metrics for Loss, Discard 301 Run-length encoding uses a bit vector to encode information about the 302 packet. Each bit in the vector represents a packet and depending on 303 the signaled metric it defines if the packet was lost, duplicated, 304 discarded, or repaired. An endpoint typically uses the run length 305 encoding to accurately communicate the status of each packet in the 306 interval to the other endpoint. [RFC3611], [RFC7097] define run- 307 length encoding for lost and duplicate packets, and discarded 308 packets, respectively. 310 The WebRTC application could benefit from the additional information. 311 If losses occur after discards, an endpoint may be able to correlate 312 the two run length vectors to identify congestion-related losses, 313 i.e., a router queue became overloaded causing delays and then 314 overflowed. If the losses are independent, it may indicate bit-error 315 corruption. For the WebRTC Stats API [W3C.WD-webrtc-stats-20160527], 316 these types of metrics are not recommended for use due to the large 317 amount of data and the computation involved. 319 5.2. Application Impact Metrics 321 5.2.1. Discard Octets Metric 323 The metric reports the cumulative size of the packets discarded in 324 the interval, it is complementary to number of discarded packets. An 325 application measures sent octets and received octets to calculate 326 sending rate and receiving rate, respectively. The application can 327 calculate the actual bit rate in a particular interval by subtracting 328 the discarded octets from the received octets. 330 For WebRTC, discarded octets supplements the sent and received octets 331 and provides an accurate method for calculating the actual bit rate 332 which is an important parameter to reflect the quality of the media. 333 The discarded bytes metric is defined in [RFC7243]. 335 5.2.2. Frame Impairment Summary Metrics 337 RTP has different framing mechanisms for different payload types. 338 For audio streams, a single RTP packet may contain one or multiple 339 audio frames, each of which has a fixed length. On the other hand, 340 in video streams, a single video frame may be transmitted in multiple 341 RTP packets. The size of each packet is limited by the Maximum 342 Transmission Unit (MTU) of the underlying network. However, 343 statistics from standard SR/RR only collect information from 344 transport layer, which may not fully reflect the quality observed by 345 the application. Video is typically encoded using two frame types 346 i.e., key frames and derived frames. Key frames are normally just 347 spatially compressed, i.e., without prediction from other pictures. 348 The derived frames are temporally compressed, i.e., depend on the key 349 frame for decoding. Hence, key frames are much larger in size than 350 derived frames. The loss of these key frames results in a 351 substantial reduction in video quality. Thus it is reasonable to 352 consider this application layer information in WebRTC 353 implementations, which influence sender strategies to mitigate the 354 problem or require the accurate assessment of users' quality of 355 experience. 357 The following metrics can also be considered for WebRTC's Statistics 358 API: number of discarded key frames, number of lost key frames, 359 number of discarded derived frames, number of lost derived frames. 360 These metrics can be used to calculate Media Loss Rate (MLR) of MDI. 361 Details of the definition of these metrics are described in 362 [RFC7003]. Additionally, the metric provides the rendered frame 363 rate, an important parameter for quality estimation. 365 5.2.3. Jitter Buffer Metrics 367 The size of the jitter buffer affects the end-to-end delay on the 368 network and also the packet discard rate. When the buffer size is 369 too small, slower packets are not played out and dropped, while when 370 the buffer size is too large, packets are held longer than necessary 371 and consequently reduce conversational quality. Measurement of 372 jitter buffer should not be ignored in the evaluation of end user 373 perception of conversational quality. Jitter buffer related metrics, 374 such as maximum and nominal jitter buffer, could be used to show how 375 the jitter buffer behaves at the receiving endpoint. They are useful 376 for providing better end-user quality of experience (QoE) when jitter 377 buffer factors are used as inputs to calculate MoS values. Thus for 378 those cases, jitter buffer metrics should be considered. The 379 definition of these metrics is provided in [RFC7005]. 381 5.3. Recovery metrics 383 This document does not consider concealment metrics as part of 384 recovery metrics. 386 5.3.1. Post-repair Packet Count Metrics 388 Error-resilience mechanisms, like RTP retransmission or FEC, are 389 optional in RTCWEB because the overhead of the repair bits adding to 390 the original streams. But they do help to greatly reduce the impact 391 of packet loss and enhance the quality of transmission. Web 392 applications could support certain repair mechanism after negotiation 393 between both sides of browsers when needed. For these web 394 applications using repair mechanisms, providing some statistic 395 information for the performance of their repair mechanisms could help 396 to have a more accurate quality evaluation. 398 The un-repaired packets count and repaired loss count defined in 399 [RFC7509] provide the recovery information of the error-resilience 400 mechanisms to the monitoring application or the sending endpoint. 401 The endpoint can use these metrics to ascertain the ratio of repaired 402 packets to lost packets. Including this kind of metrics helps the 403 application evaluate the effectiveness of the applied repair 404 mechanisms. 406 5.3.2. Run Length Encoded Metric for Post-repair 408 [RFC5725] defines run-length encoding for post-repair packets. When 409 using error-resilience mechanisms, the endpoint can correlate the 410 loss run length with this metric to ascertain where the losses and 411 repairs occurred in the interval. This provides more accurate 412 information for recovery mechanisms evaluation than those in 413 Section 5.3.1. However, it is not suggested to use due to their 414 enormous amount of data when RTCP XR are supported. 416 For WebRTC, the application may benefit from the additional 417 information. If losses occur after discards, an endpoint may be able 418 to correlate the two run length vectors to identify congestion- 419 related losses, i.e., a router queue became overloaded causing delays 420 and then overflowed. If the losses are independent, it may indicate 421 bit-error corruption. Lastly, when using error-resilience 422 mechanisms, the endpoint can correlate the loss and post-repair run 423 lengths to ascertain where the losses and repairs occurred in the 424 interval. For example, consecutive losses are likely not to be 425 repaired by a simple FEC scheme. 427 6. Identifiers from Sender, Receiver, and Extended Report Blocks 429 This document describes a list of metrics and corresponding 430 identifiers relevant to RTP media in WebRTC. These group of 431 identifiers are defined on a ReportGroup corresponding to an 432 Synchronization source (SSRC). In practice the application MUST be 433 able to query the statistic identifiers on both an incoming (remote) 434 and outgoing (local) media stream. Since sending and receiving SR 435 and RR are mandatory, the metrics defined in the SR and RR report 436 blocks are always available. For XR metrics, it depends on two 437 factors: 1) if it measured at the endpoint, 2) if it reported by the 438 endpoint in an XR report. If a metric is only measured by the 439 endpoint and not reported, the metrics will only be available for the 440 incoming (remote) media stream. Alternatively, if the corresponding 441 metric is also reported in an XR report, it will be available for 442 both the incoming (remote) and outgoing (local) media stream. 444 For a remote statistic, the timestamp represents the timestamp from 445 an incoming SR/RR/XR packet. Conversely, for a local statistic, it 446 refers to the current timestamp generated by the local clock 447 (typically the POSIX timestamp, i.e., milliseconds since Jan 1, 448 1970). 450 As per [RFC3550], the octets metrics represent the payload size 451 (i.e., not including header or padding). 453 6.1. Cumulative Number of Packets and Octets Sent 455 Name: packetsSent 457 Definition: section 6.4.1 in [RFC3550]. 459 Name: bytesSent 461 Definition: section 6.4.1 in [RFC3550]. 463 6.2. Cumulative Number of Packets and Octets Received 465 Name: packetsReceived 467 Definition: section 6.4.1 in [RFC3550]. 469 Name: bytesReceived 471 Definition: section 6.4.1 in [RFC3550]. 473 6.3. Cumulative Number of Packets Lost 475 Name: packetsLost 477 Definition: section 6.4.1 in [RFC3550]. 479 6.4. Interval Packet Loss and Jitter 481 Name: jitter 483 Definition: section 6.4.1 in [RFC3550]. 485 Name: fractionLost 487 Definition: section 6.4.1 in [RFC3550]. 489 6.5. Cumulative Number of Packets and Octets Discarded 491 Name: packetsDiscarded 493 Definition: The cumulative number of RTP packets discarded due to 494 late or early-arrival, Appendix A (a) of [RFC7002]. 496 Name: bytesDiscarded 498 Definition: The cumulative number of octets discarded due to late or 499 early-arrival, Appendix A of [RFC7243]. 501 6.6. Cumulative Number of Packets Repaired 503 Name: packetsRepaired 505 Definition: The cumulative number of lost RTP packets repaired after 506 applying a error-resilience mechanism, Appendix A (b) of [RFC7509]. 507 To clarify, the value is upper bound to the cumulative number of lost 508 packets. 510 6.7. Burst Packet Loss and Burst Discards 512 Name: burstPacketsLost 514 Definition: The cumulative number of RTP packets lost during loss 515 bursts, Appendix A (c) of [RFC6958]. 517 Name: burstLossCount 519 Definition: The cumulative number of bursts of lost RTP packets, 520 Appendix A (e) of [RFC6958]. 522 Name: burstPacketsDiscarded 524 Definition: The cumulative number of RTP packets discarded during 525 discard bursts, Appendix A (b) of [RFC7003]. 527 Name: burstDiscardCount 529 Definition: The cumulative number of bursts of discarded RTP packets, 530 Appendix A (e) of [RFC8015]. 532 [RFC3611] recommends a Gmin (threshold) value of 16 for classifying 533 packet loss or discard burst. 535 6.8. Burst/Gap Rates 537 Name: burstLossRate 539 Definition: The fraction of RTP packets lost during bursts, 540 Appendix A (a) of [RFC7004]. 542 Name: gapLossRate 544 Definition: The fraction of RTP packets lost during gaps, Appendix A 545 (b) of [RFC7004]. 547 Name: burstDiscardRate 549 Definition: The fraction of RTP packets discarded during bursts, 550 Appendix A (e) of [RFC7004]. 552 Name: gapDiscardRate 554 Definition: The fraction of RTP packets discarded during gaps, 555 Appendix A (f) of [RFC7004]. 557 6.9. Frame Impairment Metrics 559 Name: framesLost 561 Definition: The cumulative number of full frames lost, Appendix A (i) 562 of [RFC7004]. 564 Name: framesCorrupted 566 Definition: The cumulative number of frames partially lost, 567 Appendix A (j) of [RFC7004]. 569 Name: framesDropped 570 Definition: The cumulative number of full frames discarded, 571 Appendix A (g) of [RFC7004]. 573 Name: framesSent 575 Definition: The cumulative number of frames sent. 577 Name: framesReceived 579 Definition: The cumulative number of partial or full frames received. 581 7. Adding new metrics to WebRTC Statistics API 583 The metrics defined in this draft have already been added to the W3C 584 WebRTC specification. The current working process to add new metrics 585 is, create an issue or pull request on the repository of the W3C 586 WebRTC specification (https://github.com/w3c/webrtc-stats). 588 8. Security Considerations 590 The monitoring activities are implemented between two browsers or 591 between a browser and a server. Therefore encryption procedures, 592 such as the ones suggested for a Secure RTCP (SRTCP), need to be 593 used. Currently, the monitoring in RTCWEB introduces no new security 594 considerations beyond those described in [I-D.ietf-rtcweb-rtp-usage], 595 [I-D.ietf-rtcweb-security]. 597 9. Acknowledgements 599 The authors would like to thank Bernard Aboba, Harald Alvestrand, Al 600 Morton, Colin Perkins, and Shida Schubert for their valuable comments 601 and suggestions on earlier version of this document. 603 10. References 605 10.1. Normative References 607 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 608 Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/ 609 RFC2119, March 1997, 610 . 612 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 613 Jacobson, "RTP: A Transport Protocol for Real-Time 614 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 615 July 2003, . 617 [RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed., 618 "RTP Control Protocol Extended Reports (RTCP XR)", RFC 619 3611, DOI 10.17487/RFC3611, November 2003, 620 . 622 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 623 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 624 DOI 10.17487/RFC4588, July 2006, 625 . 627 [RFC5725] Begen, A., Hsu, D., and M. Lague, "Post-Repair Loss RLE 628 Report Block Type for RTP Control Protocol (RTCP) Extended 629 Reports (XRs)", RFC 5725, DOI 10.17487/RFC5725, February 630 2010, . 632 [RFC6776] Clark, A. and Q. Wu, "Measurement Identity and Information 633 Reporting Using a Source Description (SDES) Item and an 634 RTCP Extended Report (XR) Block", RFC 6776, DOI 10.17487/ 635 RFC6776, October 2012, 636 . 638 [RFC6958] Clark, A., Zhang, S., Zhao, J., and Q. Wu, Ed., "RTP 639 Control Protocol (RTCP) Extended Report (XR) Block for 640 Burst/Gap Loss Metric Reporting", RFC 6958, DOI 10.17487/ 641 RFC6958, May 2013, 642 . 644 [RFC7002] Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol 645 (RTCP) Extended Report (XR) Block for Discard Count Metric 646 Reporting", RFC 7002, DOI 10.17487/RFC7002, September 647 2013, . 649 [RFC7003] Clark, A., Huang, R., and Q. Wu, Ed., "RTP Control 650 Protocol (RTCP) Extended Report (XR) Block for Burst/Gap 651 Discard Metric Reporting", RFC 7003, DOI 10.17487/RFC7003, 652 September 2013, . 654 [RFC7004] Zorn, G., Schott, R., Wu, Q., Ed., and R. Huang, "RTP 655 Control Protocol (RTCP) Extended Report (XR) Blocks for 656 Summary Statistics Metrics Reporting", RFC 7004, DOI 657 10.17487/RFC7004, September 2013, 658 . 660 [RFC7005] Clark, A., Singh, V., and Q. Wu, "RTP Control Protocol 661 (RTCP) Extended Report (XR) Block for De-Jitter Buffer 662 Metric Reporting", RFC 7005, DOI 10.17487/RFC7005, 663 September 2013, . 665 [RFC7097] Ott, J., Singh, V., Ed., and I. Curcio, "RTP Control 666 Protocol (RTCP) Extended Report (XR) for RLE of Discarded 667 Packets", RFC 7097, DOI 10.17487/RFC7097, January 2014, 668 . 670 [RFC7243] Singh, V., Ed., Ott, J., and I. Curcio, "RTP Control 671 Protocol (RTCP) Extended Report (XR) Block for the Bytes 672 Discarded Metric", RFC 7243, DOI 10.17487/RFC7243, May 673 2014, . 675 [RFC7509] Huang, R. and V. Singh, "RTP Control Protocol (RTCP) 676 Extended Report (XR) for Post-Repair Loss Count Metrics", 677 RFC 7509, DOI 10.17487/RFC7509, May 2015, 678 . 680 [RFC8015] Singh, V., Perkins, C., Clark, A., and R. Huang, "RTP 681 Control Protocol (RTCP) Extended Report (XR) Block for 682 Independent Reporting of Burst/Gap Discard Metrics", RFC 683 8015, DOI 10.17487/RFC8015, November 2016, 684 . 686 10.2. Informative References 688 [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 689 Time Communication Use Cases and Requirements", RFC 7478, 690 DOI 10.17487/RFC7478, March 2015, 691 . 693 [W3C.WD-webrtc-20161124] 694 Sporny, M. and D. Longley, "WebRTC 1.0: Real-time 695 Communication Between Browsers", World Wide Web Consortium 696 WD WD-webrtc-20161124, November 2016, 697 . 699 [I-D.ietf-rtcweb-rtp-usage] 700 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 701 Communication (WebRTC): Media Transport and Use of RTP", 702 draft-ietf-rtcweb-rtp-usage-26 (work in progress), March 703 2016. 705 [I-D.ietf-rtcweb-security] 706 Rescorla, E., "Security Considerations for WebRTC", draft- 707 ietf-rtcweb-security-08 (work in progress), February 2015. 709 [W3C.WD-webrtc-stats-20160527] 710 Alvestrand, H. and V. Singh, "Identifiers for 711 WebRTC's Statistics API", World Wide Web Consortium 712 WD WD-webrtc-stats-20160527, May 2016, 713 . 715 [RFC6390] Clark, A. and B. Claise, "Guidelines for Considering New 716 Performance Metric Development", BCP 170, RFC 6390, DOI 717 10.17487/RFC6390, October 2011, 718 . 720 Appendix A. Change Log 722 Note to the RFC-Editor: please remove this section prior to 723 publication as an RFC. 725 A.1. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-04 727 o Removed IANA registry. 729 A.2. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02, -03 731 o Keep-alive versions, updates to references. 733 A.3. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-01 735 o Create new registry for WebRTC media metrics. 737 o Using camelCase instead of TitleCase for identifier names. 739 o Imported RTCP SR and RR metrics from the registry in alvestrand- 740 rtcweb-stats-registry. 742 o Added Burst/Gap rate metrics. 744 o Added Frames sent and received metrics. 746 A.4. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00 748 o Submitted as WG Draft. 750 A.5. changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04 752 o Addressed comments from the London IETF meeting: 754 o Removed ECN metrics. 756 o Merged draft-singh-xrblock-webrtc-additional-stats-01 758 Authors' Addresses 760 Varun Singh 761 CALLSTATS I/O Oy 762 Runeberginkatu 4c A 4 763 Helsinki 00100 764 Finland 766 Email: varun@callstats.io 767 URI: https://www.callstats.io/about 769 Rachel Huang 770 Huawei 771 101 Software Avenue, Yuhua District 772 Nanjing, CN 210012 773 China 775 Email: rachel.huang@huawei.com 777 Roni Even 778 Huawei 779 14 David Hamelech 780 Tel Aviv 64953 781 Israel 783 Email: roni.even@mail01.huawei.com 785 Dan Romascanu 787 Email: dromasca@gmail.com 789 Lingli Deng 790 China Mobile 792 Email: denglingli@chinamobile.com