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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 RMCAT WG V. Singh 3 Internet-Draft callstats.io 4 Intended status: Informational J. Ott 5 Expires: May 9, 2019 Technical University of Munich 6 S. Holmer 7 Google 8 November 5, 2018 10 Evaluating Congestion Control for Interactive Real-time Media 11 draft-ietf-rmcat-eval-criteria-08 13 Abstract 15 The Real-time Transport Protocol (RTP) is used to transmit media in 16 telephony and video conferencing applications. This document 17 describes the guidelines to evaluate new congestion control 18 algorithms for interactive point-to-point real-time media. 20 Status of This Memo 22 This Internet-Draft is submitted in full conformance with the 23 provisions of BCP 78 and BCP 79. 25 Internet-Drafts are working documents of the Internet Engineering 26 Task Force (IETF). Note that other groups may also distribute 27 working documents as Internet-Drafts. The list of current Internet- 28 Drafts is at http://datatracker.ietf.org/drafts/current/. 30 Internet-Drafts are draft documents valid for a maximum of six months 31 and may be updated, replaced, or obsoleted by other documents at any 32 time. It is inappropriate to use Internet-Drafts as reference 33 material or to cite them other than as "work in progress." 35 This Internet-Draft will expire on May 9, 2019. 37 Copyright Notice 39 Copyright (c) 2018 IETF Trust and the persons identified as the 40 document authors. All rights reserved. 42 This document is subject to BCP 78 and the IETF Trust's Legal 43 Provisions Relating to IETF Documents 44 (http://trustee.ietf.org/license-info) in effect on the date of 45 publication of this document. Please review these documents 46 carefully, as they describe your rights and restrictions with respect 47 to this document. Code Components extracted from this document must 48 include Simplified BSD License text as described in Section 4.e of 49 the Trust Legal Provisions and are provided without warranty as 50 described in the Simplified BSD License. 52 Table of Contents 54 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 55 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 56 3. Metrics . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 57 3.1. RTP Log Format . . . . . . . . . . . . . . . . . . . . . 5 58 4. List of Network Parameters . . . . . . . . . . . . . . . . . 5 59 4.1. One-way Propagation Delay . . . . . . . . . . . . . . . . 5 60 4.2. End-to-end Loss . . . . . . . . . . . . . . . . . . . . . 6 61 4.3. Drop Tail Router Queue Length . . . . . . . . . . . . . . 6 62 4.4. Loss generation model . . . . . . . . . . . . . . . . . . 7 63 4.5. Jitter models . . . . . . . . . . . . . . . . . . . . . . 7 64 4.5.1. Random Bounded PDV (RBPDV) . . . . . . . . . . . . . 8 65 4.5.2. Approximately Random Subject to No-Reordering Bounded 66 PDV (NR-RPVD) . . . . . . . . . . . . . . . . 9 67 4.5.3. Recommended distribution . . . . . . . . . . . . . . 9 68 5. WiFi or Cellular Links . . . . . . . . . . . . . . . . . . . 10 69 6. Traffic Models . . . . . . . . . . . . . . . . . . . . . . . 10 70 6.1. TCP traffic model . . . . . . . . . . . . . . . . . . . . 10 71 6.2. RTP Video model . . . . . . . . . . . . . . . . . . . . . 10 72 6.3. Background UDP . . . . . . . . . . . . . . . . . . . . . 11 73 7. Security Considerations . . . . . . . . . . . . . . . . . . . 11 74 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11 75 9. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 11 76 10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 11 77 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 12 78 11.1. Normative References . . . . . . . . . . . . . . . . . . 12 79 11.2. Informative References . . . . . . . . . . . . . . . . . 13 80 Appendix A. Application Trade-off . . . . . . . . . . . . . . . 14 81 A.1. Measuring Quality . . . . . . . . . . . . . . . . . . . . 14 82 Appendix B. Change Log . . . . . . . . . . . . . . . . . . . . . 14 83 B.1. Changes in draft-ietf-rmcat-eval-criteria-07 . . . . . . 14 84 B.2. Changes in draft-ietf-rmcat-eval-criteria-06 . . . . . . 14 85 B.3. Changes in draft-ietf-rmcat-eval-criteria-05 . . . . . . 14 86 B.4. Changes in draft-ietf-rmcat-eval-criteria-04 . . . . . . 14 87 B.5. Changes in draft-ietf-rmcat-eval-criteria-03 . . . . . . 15 88 B.6. Changes in draft-ietf-rmcat-eval-criteria-02 . . . . . . 15 89 B.7. Changes in draft-ietf-rmcat-eval-criteria-01 . . . . . . 15 90 B.8. Changes in draft-ietf-rmcat-eval-criteria-00 . . . . . . 15 91 B.9. Changes in draft-singh-rmcat-cc-eval-04 . . . . . . . . . 15 92 B.10. Changes in draft-singh-rmcat-cc-eval-03 . . . . . . . . . 15 93 B.11. Changes in draft-singh-rmcat-cc-eval-02 . . . . . . . . . 16 94 B.12. Changes in draft-singh-rmcat-cc-eval-01 . . . . . . . . . 16 95 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16 97 1. Introduction 99 This memo describes the guidelines to help with evaluating new 100 congestion control algorithms for interactive point-to-point real 101 time media. The requirements for the congestion control algorithm 102 are outlined in [I-D.ietf-rmcat-cc-requirements]). This document 103 builds upon previous work at the IETF: Specifying New Congestion 104 Control Algorithms [RFC5033] and Metrics for the Evaluation of 105 Congestion Control Algorithms [RFC5166]. 107 The guidelines proposed in the document are intended to help prevent 108 a congestion collapse, promote fair capacity usage and optimize the 109 media flow's throughput. Furthermore, the proposed algorithms are 110 expected to operate within the envelope of the circuit breakers 111 defined in RFC8083 [RFC8083]. 113 This document only provides broad-level criteria for evaluating a new 114 congestion control algorithm. The minimal requirement for congestion 115 control proposals is to produce or present results for the test 116 scenarios described in [I-D.ietf-rmcat-eval-test] (Basic Test Cases). 117 Additionally, proponents may produce evaluation results for the 118 wireless test scenarios [I-D.ietf-rmcat-wireless-tests]. 120 2. Terminology 122 The terminology defined in RTP [RFC3550], RTP Profile for Audio and 123 Video Conferences with Minimal Control [RFC3551], RTCP Extended 124 Report (XR) [RFC3611], Extended RTP Profile for RTCP-based Feedback 125 (RTP/AVPF) [RFC4585] and Support for Reduced-Size RTCP [RFC5506] 126 apply. 128 3. Metrics 130 This document specifies testing criteria for evaluating congestion 131 control algorithms for RTP media flows. Proposed algorithms are to 132 prove their performance by means of simulation and/or emulation 133 experiments for all the cases described. 135 Each experiment is expected to log every incoming and outgoing packet 136 (the RTP logging format is described in Section 3.1). The logging 137 can be done inside the application or at the endpoints using PCAP 138 (packet capture, e.g., tcpdump, wireshark). The following are 139 calculated based on the information in the packet logs: 141 1. Sending rate, Receiver rate, Goodput (measured at 200ms 142 intervals) 144 2. Packets sent, Packets received 145 3. Bytes sent, bytes received 147 4. Packet delay 149 5. Packets lost, Packets discarded (from the playout or de-jitter 150 buffer) 152 6. If using, retransmission or FEC: post-repair loss 154 7. Self-Fairness and Fairness with respect to cross traffic: 155 Experiments testing a given congestion control proposal must 156 report on relative ratios of the average throughput (measured at 157 coarser time intervals) obtained by each RTP media stream. In 158 the presence of background cross-traffic such as TCP, the report 159 must also include the relative ratio between average throughput 160 of RTP media streams and cross-traffic streams. 161 During static periods of a test (i.e., when bottleneck bandwidth 162 is constant and no arrival/departure of streams), these report 163 on relative ratios serve as an indicator of how fair the RTP 164 streams share bandwidth amongst themselves and against cross- 165 traffic streams. The throughput measurement interval should be 166 set at a few values (for example, at 1s, 5s, and 20s) in order 167 to measure fairness across different time scales. 168 As a general guideline, the relative ratio between congestion 169 controlled RTP flows with the same priority level and similar 170 path RTT should be bounded between (0.333 and 3.) For example, 171 see the test scenarios described in [I-D.ietf-rmcat-eval-test]. 173 8. Convergence time: The time taken to reach a stable rate at 174 startup, after the available link capacity changes, or when new 175 flows get added to the bottleneck link. 177 9. Instability or oscillation in the sending rate: The frequency or 178 number of instances when the sending rate oscillates between an 179 high watermark level and a low watermark level, or vice-versa in 180 a defined time window. For example, the watermarks can be set 181 at 4x interval: 500 Kbps, 2 Mbps, and a time window of 500ms. 183 10. Bandwidth Utilization, defined as ratio of the instantaneous 184 sending rate to the instantaneous bottleneck capacity. This 185 metric is useful only when a congestion controlled RTP flow is 186 by itself or competing with similar cross-traffic. 188 Note that the above metrics are all objective application-independent 189 metrics. Refer to Section 3, in [I-D.ietf-netvc-testing] for 190 objective metrics for evaluating codecs. 192 From the logs the statistical measures (min, max, mean, standard 193 deviation and variance) for the whole duration or any specific part 194 of the session can be calculated. Also the metrics (sending rate, 195 receiver rate, goodput, latency) can be visualized in graphs as 196 variation over time, the measurements in the plot are at 1 second 197 intervals. Additionally, from the logs it is possible to plot the 198 histogram or CDF of packet delay. 200 3.1. RTP Log Format 202 Having a common log format simplifies running analyses across and 203 comparing different measurements. The log file SHOULD be tab or 204 comma separated containing the following details: 206 Send or receive timestamp (unix) 207 RTP payload type 208 SSRC 209 RTP sequence no 210 RTP timestamp 211 marker bit 212 payload size 214 If the congestion control implements, retransmissions or FEC, the 215 evaluation should report both packet loss (before applying error- 216 resilience) and residual packet loss (after applying error- 217 resilience). 219 4. List of Network Parameters 221 The implementors initially are encouraged to choose evaluation 222 settings from the following values: 224 4.1. One-way Propagation Delay 226 Experiments are expected to verify that the congestion control is 227 able to work across a broad range of path characteristics, also 228 including challenging situations, for example over trans-continental 229 and/or satellite links. Tests thus account for the following 230 different latencies: 232 1. Very low latency: 0-1ms 234 2. Low latency: 50ms 236 3. High latency: 150ms 238 4. Extreme latency: 300ms 240 4.2. End-to-end Loss 242 Many paths in the Internet today are largely lossless but, with 243 wireless networks and interference, towards remote regions, or in 244 scenarios featuring high/fast mobility, media flows may exhibit 245 substantial packet loss. This variety needs to be reflected 246 appropriately by the tests. 248 To model a wide range of lossy links, the experiments can choose one 249 of the following loss rates, the fractional loss is the ratio of 250 packets lost and packets sent. 252 1. no loss: 0% 254 2. 1% 256 3. 5% 258 4. 10% 260 5. 20% 262 4.3. Drop Tail Router Queue Length 264 Routers SHOULD be configured to use Drop Trail queues in the 265 experiments due to their (still) prevalent nature. Experimentation 266 with AQM schemes is encouraged but not mandatory. 268 The router queue length is measured as the time taken to drain the 269 FIFO queue. It has been noted in various discussions that the queue 270 length in the current deployed Internet varies significantly. While 271 the core backbone network has very short queue length, the home 272 gateways usually have larger queue length. Those various queue 273 lengths can be categorized in the following way: 275 1. QoS-aware (or short): 70ms 277 2. Nominal: 300-500ms 279 3. Buffer-bloated: 1000-2000ms 281 Here the size of the queue is measured in bytes or packets and to 282 convert the queue length measured in seconds to queue length in 283 bytes: 285 QueueSize (in bytes) = QueueSize (in sec) x Throughput (in bps)/8 287 4.4. Loss generation model 289 Many models for generating packet loss are available, some yield 290 correlated, others independent losses; losses can also be extracted 291 from packet traces. As a (simple) minimum loss model with minimal 292 parameterization (i.e., the loss rate), independent random losses 293 must be used in the evaluation. 295 It is known that independent loss models may reflect reality poorly 296 and hence more sophisticated loss models could be considered. 297 Suitable models for correlated losses includes the Gilbert-Elliot 298 model and losses generated by modeling a queue including its 299 (different) drop behaviors. 301 4.5. Jitter models 303 This section defines jitter models for the purposes of this document. 304 When jitter is to be applied to both the congestion controlled RTP 305 flow and any competing flow (such as a TCP competing flow), the 306 competing flow will use the jitter definition below that does not 307 allow for re-ordering of packets on the competing flow (see NR-RBPDV 308 definition below). 310 Jitter is an overloaded term in communications. Its meaning is 311 typically associated with the variation of a metric (e.g., delay) 312 with respect to some reference metric (e.g., average delay or minimum 313 delay). For example, RFC 3550 jitter is a smoothed estimate of 314 jitter which is particularly meaningful if the underlying packet 315 delay variation was caused by a Gaussian random process. 317 Because jitter is an overloaded term, we instead use the term Packet 318 Delay Variation (PDV) to describe the variation of delay of 319 individual packets in the same sense as the IETF IPPM WG has defined 320 PDV in their documents (e.g., RFC 3393) and as the ITU-T SG16 has 321 defined IP Packet Delay Variation (IPDV) in their documents (e.g., 322 Y.1540). 324 Most PDV distributions in packet network systems are one-sided 325 distributions (the measurement of which with a finite number of 326 measurement samples result in one-sided histograms). In the usual 327 packet network transport case there is typically one packet that 328 transited the network with the minimum delay, then a majority of 329 packets also transit the system within some variation from this 330 minimum delay, and then a minority of the packets transit the network 331 with delays higher than the median or average transit time (these are 332 outliers). Although infrequent, outliers can cause significant 333 deleterious operation in adaptive systems and should be considered in 334 rate adaptation designs for RTP congestion control. 336 In this section we define two different bounded PDV characteristics, 337 1) Random Bounded PDV and 2) Approximately Random Subject to No- 338 Reordering Bounded PDV. 340 The former, 1) Random Bounded PDV is presented for information only, 341 while the latter, 2) Approximately Random Subject to No-Reordering 342 Bounded PDV, must be used in the evaluation. 344 4.5.1. Random Bounded PDV (RBPDV) 346 The RBPDV probability distribution function (PDF) is specified to be 347 of some mathematically describable function which includes some 348 practical minimum and maximum discrete values suitable for testing. 349 For example, the minimum value, x_min, might be specified as the 350 minimum transit time packet and the maximum value, x_max, might be 351 defined to be two standard deviations higher than the mean. 353 Since we are typically interested in the distribution relative to the 354 mean delay packet, we define the zero mean PDV sample, z(n), to be 355 z(n) = x(n) - x_mean, where x(n) is a sample of the RBPDV random 356 variable x and x_mean is the mean of x. 358 We assume here that s(n) is the original source time of packet n and 359 the post-jitter induced emission time, j(n), for packet n is j(n) = 360 {[z(n) + x_mean] + s(n)}. It follows that the separation in the post- 361 jitter time of packets n and n+1 is {[s(n+1)-s(n)] - [z(n)-z(n+1)]}. 362 Since the first term is always a positive quantity, we note that 363 packet reordering at the receiver is possible whenever the second 364 term is greater than the first. Said another way, whenever the 365 difference in possible zero mean PDV sample delays (i.e., [x_max- 366 x_min]) exceeds the inter-departure time of any two sent packets, we 367 have the possibility of packet re-ordering. 369 There are important use cases in real networks where packets can 370 become re-ordered such as in load balancing topologies and during 371 route changes. However, for the vast majority of cases there is no 372 packet re-ordering because most of the time packets follow the same 373 path. Due to this, if a packet becomes overly delayed, the packets 374 after it on that flow are also delayed. This is especially true for 375 mobile wireless links where there are per-flow queues prior to base 376 station scheduling. Owing to this important use case, we define 377 another PDV profile similar to the above, but one that does not allow 378 for re-ordering within a flow. 380 4.5.2. Approximately Random Subject to No-Reordering Bounded PDV (NR- 381 RPVD) 383 No Reordering RPDV, NR-RPVD, is defined similarly to the above with 384 one important exception. Let serial(n) be defined as the 385 serialization delay of packet n at the lowest bottleneck link rate 386 (or other appropriate rate) in a given test. Then we produce all the 387 post-jitter values for j(n) for n = 1, 2, ... N, where N is the 388 length of the source sequence s to be offset-ed. The exception can 389 be stated as follows: We revisit all j(n) beginning from index n=2, 390 and if j(n) is determined to be less than [j(n-1)+serial(n-1)], we 391 redefine j(n) to be equal to [j(n-1)+serial(n-1)] and continue for 392 all remaining n (i.e., n = 3, 4, .. N). This models the case where 393 the packet n is sent immediately after packet (n-1) at the bottleneck 394 link rate. Although this is generally the theoretical minimum in 395 that it assumes that no other packets from other flows are in-between 396 packet n and n+1 at the bottleneck link, it is a reasonable 397 assumption for per flow queuing. 399 We note that this assumption holds for some important exception 400 cases, such as packets immediately following outliers. There are a 401 multitude of software controlled elements common on end-to-end 402 Internet paths (such as firewalls, ALGs and other middleboxes) which 403 stop processing packets while servicing other functions (e.g., 404 garbage collection). Often these devices do not drop packets, but 405 rather queue them for later processing and cause many of the 406 outliers. Thus NR-RPVD models this particular use case (assuming 407 serial(n+1) is defined appropriately for the device causing the 408 outlier) and thus is believed to be important for adaptation 409 development for congestion controlled RTP streams. 411 4.5.3. Recommended distribution 413 Whether Random Bounded PDV or Approximately Random Subject to No- 414 Reordering Bounded PDV, it is recommended that z(n) is distributed 415 according to a truncated Gaussian for the above jitter models: 417 z(n) ~ |max(min(N(0, std^2), N_STD * std), -N_STD * std)| 419 where N(0, std^2) is the Gaussian distribution with zero mean and 420 standard deviation std. Recommended values: 422 o std = 5 ms 424 o N_STD = 3 426 5. WiFi or Cellular Links 428 [I-D.ietf-rmcat-wireless-tests] describes the test cases to simulate 429 networks with wireless links. The document describes mechanism to 430 simulate both cellular and WiFi networks. 432 6. Traffic Models 434 6.1. TCP traffic model 436 Long-lived TCP flows will download data throughout the session and 437 are expected to have infinite amount of data to send or receive. 438 This roughly applies, for example, when downloading software 439 distributions. 441 Each short TCP flow is modeled as a sequence of file downloads 442 interleaved with idle periods. Not all short TCP flows start at the 443 same time, i.e., some start in the ON state while others start in the 444 OFF state. 446 The short TCP flows can be modeled as follows: 30 connections start 447 simultaneously fetching small (30-50 KB) amounts of data. This 448 covers the case where the short TCP flows are not fetching a video 449 file. 451 The idle period between bursts of starting a group of TCP flows is 452 typically derived from an exponential distribution with the mean 453 value of 10 seconds. 455 [These values were picked based on the data available at 456 http://httparchive.org/interesting.php as of October 2015]. 458 Many different TCP congestion control schemes are deployed today. 459 Therefore, experimentation with a range of different schemes, 460 especially including CUBIC, is encouraged. Experiments MUST document 461 in detail which congestion control schemes they tested against and 462 which parameters were used. 464 6.2. RTP Video model 466 [I-D.ietf-rmcat-video-traffic-model] describes two types of video 467 traffic models for evaluating candidate algorithms for RTP congestion 468 control. The first model statistically characterizes the behavior of 469 a video encoder. Whereas the second model uses video traces. 471 For example, test sequences are available at: [xiph-seq] and 472 [HEVC-seq]. The currently chosen video streams are: Foreman and 473 FourPeople. 475 6.3. Background UDP 477 Background UDP flow is modeled as a constant bit rate (CBR) flow. It 478 will download data at a particular CBR rate for the complete session, 479 or will change to particular CBR rate at predefined intervals. The 480 inter packet interval is calculated based on the CBR and the packet 481 size (is typically set to the path MTU size, the default value can be 482 1500 bytes). 484 Note that new transport protocols such as QUIC may use UDP but, due 485 to their congestion control algorithms, will exhibit behavior 486 conceptually similar in nature to TCP flows above and can thus be 487 subsumed by the above, including the division into short- and long- 488 lived flows. As QUIC evolves independently of TCP congestion control 489 algorithms, its future congestion control SHOULD be considered as 490 competing traffic as appropriate. 492 7. Security Considerations 494 Security issues have not been discussed in this memo. 496 8. IANA Considerations 498 There are no IANA impacts in this memo. 500 9. Contributors 502 The content and concepts within this document are a product of the 503 discussion carried out in the Design Team. 505 Michael Ramalho provided the text for the Jitter model. 507 10. Acknowledgments 509 Much of this document is derived from previous work on congestion 510 control at the IETF. 512 The authors would like to thank Harald Alvestrand, Anna Brunstrom, 513 Luca De Cicco, Wesley Eddy, Lars Eggert, Kevin Gross, Vinayak Hegde, 514 Randell Jesup, Mirja Kuehlewind, Karen Nielsen, Piers O'Hanlon, Colin 515 Perkins, Michael Ramalho, Zaheduzzaman Sarker, Timothy B. 516 Terriberry, Michael Welzl, Mo Zanaty, and Xiaoqing Zhu for providing 517 valuable feedback on earlier versions of this draft. Additionally, 518 also thank the participants of the design team for their comments and 519 discussion related to the evaluation criteria. 521 11. References 523 11.1. Normative References 525 [I-D.ietf-rmcat-cc-requirements] 526 Jesup, R. and Z. Sarker, "Congestion Control Requirements 527 for Interactive Real-Time Media", draft-ietf-rmcat-cc- 528 requirements-09 (work in progress), December 2014. 530 [I-D.ietf-rmcat-wireless-tests] 531 Sarker, Z., Johansson, I., Zhu, X., Fu, J., Tan, W., and 532 M. Ramalho, "Evaluation Test Cases for Interactive Real- 533 Time Media over Wireless Networks", draft-ietf-rmcat- 534 wireless-tests-05 (work in progress), June 2018. 536 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 537 Jacobson, "RTP: A Transport Protocol for Real-Time 538 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 539 July 2003, . 541 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 542 Video Conferences with Minimal Control", STD 65, RFC 3551, 543 DOI 10.17487/RFC3551, July 2003, . 546 [RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed., 547 "RTP Control Protocol Extended Reports (RTCP XR)", 548 RFC 3611, DOI 10.17487/RFC3611, November 2003, 549 . 551 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 552 "Extended RTP Profile for Real-time Transport Control 553 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, 554 DOI 10.17487/RFC4585, July 2006, . 557 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size 558 Real-Time Transport Control Protocol (RTCP): Opportunities 559 and Consequences", RFC 5506, DOI 10.17487/RFC5506, April 560 2009, . 562 [RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control: 563 Circuit Breakers for Unicast RTP Sessions", RFC 8083, 564 DOI 10.17487/RFC8083, March 2017, . 567 11.2. Informative References 569 [HEVC-seq] 570 HEVC, "Test Sequences", 571 http://www.netlab.tkk.fi/~varun/test_sequences/ . 573 [I-D.ietf-netvc-testing] 574 Daede, T., Norkin, A., and I. Brailovskiy, "Video Codec 575 Testing and Quality Measurement", draft-ietf-netvc- 576 testing-07 (work in progress), July 2018. 578 [I-D.ietf-rmcat-eval-test] 579 Sarker, Z., Singh, V., Zhu, X., and M. Ramalho, "Test 580 Cases for Evaluating RMCAT Proposals", draft-ietf-rmcat- 581 eval-test-07 (work in progress), October 2018. 583 [I-D.ietf-rmcat-video-traffic-model] 584 Zhu, X., Cruz, S., and Z. Sarker, "Video Traffic Models 585 for RTP Congestion Control Evaluations", draft-ietf-rmcat- 586 video-traffic-model-06 (work in progress), November 2018. 588 [RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion 589 Control Algorithms", BCP 133, RFC 5033, 590 DOI 10.17487/RFC5033, August 2007, . 593 [RFC5166] Floyd, S., Ed., "Metrics for the Evaluation of Congestion 594 Control Mechanisms", RFC 5166, DOI 10.17487/RFC5166, March 595 2008, . 597 [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion 598 Control", RFC 5681, DOI 10.17487/RFC5681, September 2009, 599 . 601 [SA4-LR] S4-050560, 3GPP., "Error Patterns for MBMS Streaming over 602 UTRAN and GERAN", 3GPP S4-050560, 5 2008. 604 [TCP-eval-suite] 605 Lachlan, A., Marcondes, C., Floyd, S., Dunn, L., Guillier, 606 R., Gang, W., Eggert, L., Ha, S., and I. Rhee, "Towards a 607 Common TCP Evaluation Suite", Proc. PFLDnet. 2008, August 608 2008. 610 [xiph-seq] 611 Daede, T., "Video Test Media Set", 612 https://people.xiph.org/~tdaede/sets/ . 614 Appendix A. Application Trade-off 616 Application trade-off is yet to be defined. see RMCAT requirements 617 [I-D.ietf-rmcat-cc-requirements] document. Perhaps each experiment 618 should define the application's expectation or trade-off. 620 A.1. Measuring Quality 622 No quality metric is defined for performance evaluation, it is 623 currently an open issue. However, there is consensus that congestion 624 control algorithm should be able to show that it is useful for 625 interactive video by performing analysis using a real codec and video 626 sequences. 628 Appendix B. Change Log 630 Note to the RFC-Editor: please remove this section prior to 631 publication as an RFC. 633 B.1. Changes in draft-ietf-rmcat-eval-criteria-07 635 Updated the draft according to the discussion at IETF-101. 637 o Updated the discussion on fairness. Thanks to Xiaoqing Zhu for 638 providing text. 640 o Fixed a simple loss model and provided pointers to more 641 sophisticated ones. 643 o Fixed the choice of the jitter model. 645 B.2. Changes in draft-ietf-rmcat-eval-criteria-06 647 o Updated Jitter. 649 B.3. Changes in draft-ietf-rmcat-eval-criteria-05 651 o Improved text surrounding wireless tests, video sequences, and 652 short-TCP model. 654 B.4. Changes in draft-ietf-rmcat-eval-criteria-04 656 o Removed the guidelines section, as most of the sections are now 657 covered: wireless tests, video model, etc. 659 o Improved Short TCP model based on the suggestion to use 660 httparchive.org. 662 B.5. Changes in draft-ietf-rmcat-eval-criteria-03 664 o Keep-alive version. 666 o Moved link parameters and traffic models from eval-test 668 B.6. Changes in draft-ietf-rmcat-eval-criteria-02 670 o Incorporated fairness test as a working test. 672 o Updated text on mimimum evaluation requirements. 674 B.7. Changes in draft-ietf-rmcat-eval-criteria-01 676 o Removed Appendix B. 678 o Removed Section on Evaluation Parameters. 680 B.8. Changes in draft-ietf-rmcat-eval-criteria-00 682 o Updated references. 684 o Resubmitted as WG draft. 686 B.9. Changes in draft-singh-rmcat-cc-eval-04 688 o Incorporate feedback from IETF 87, Berlin. 690 o Clarified metrics: convergence time, bandwidth utilization. 692 o Changed fairness criteria to fairness test. 694 o Added measuring pre- and post-repair loss. 696 o Added open issue of measuring video quality to appendix. 698 o clarified use of DropTail and AQM. 700 o Updated text in "Minimum Requirements for Evaluation" 702 B.10. Changes in draft-singh-rmcat-cc-eval-03 704 o Incorporate the discussion within the design team. 706 o Added a section on evaluation parameters, it describes the flow 707 and network characteristics. 709 o Added Appendix with self-fairness experiment. 711 o Changed bottleneck parameters from a proposal to an example set. 713 o 715 B.11. Changes in draft-singh-rmcat-cc-eval-02 717 o Added scenario descriptions. 719 B.12. Changes in draft-singh-rmcat-cc-eval-01 721 o Removed QoE metrics. 723 o Changed stability to steady-state. 725 o Added measuring impact against few and many flows. 727 o Added guideline for idle and data-limited periods. 729 o Added reference to TCP evaluation suite in example evaluation 730 scenarios. 732 Authors' Addresses 734 Varun Singh 735 CALLSTATS I/O Oy 736 Runeberginkatu 4c A 4 737 Helsinki 00100 738 Finland 740 Email: varun@callstats.io 741 URI: https://www.callstats.io/about 743 Joerg Ott 744 Technical University of Munich 745 Faculty of Informatics 746 Boltzmannstrasse 3 747 Garching bei Muenchen, DE 85748 748 Germany 750 Email: ott@in.tum.de 751 Stefan Holmer 752 Google 753 Kungsbron 2 754 Stockholm 11122 755 Sweden 757 Email: holmer@google.com