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Jesup 3 Internet-Draft Mozilla 4 Intended status: Informational February 14, 2014 5 Expires: August 18, 2014 7 Congestion Control Requirements For RMCAT 8 draft-ietf-rmcat-cc-requirements-02 10 Abstract 12 Congestion control is needed for all data transported across the 13 Internet, in order to promote fair usage and prevent congestion 14 collapse. The requirements for interactive, point-to-point real time 15 multimedia, which needs low-delay, semi-reliable data delivery, are 16 different from the requirements for bulk transfer like FTP or bursty 17 transfers like Web pages. 19 This document attempts to describe a set of requirements that can be 20 used to evaluate other congestion control mechanisms in order to 21 figure out their fitness for this purpose, and in particular to 22 provide a set of possible requirements for proposals coming out of 23 the RMCAT Working Group. 25 This document is derived from draft-jesup-rtp-congestion-reqs 26 [I-D.jesup-rtp-congestion-reqs]. 28 Requirements Language 30 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 31 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 32 document are to be interpreted as described in RFC 2119 [RFC2119]. 34 Status of This Memo 36 This Internet-Draft is submitted in full conformance with the 37 provisions of BCP 78 and BCP 79. 39 Internet-Drafts are working documents of the Internet Engineering 40 Task Force (IETF). Note that other groups may also distribute 41 working documents as Internet-Drafts. The list of current Internet- 42 Drafts is at http://datatracker.ietf.org/drafts/current/. 44 Internet-Drafts are draft documents valid for a maximum of six months 45 and may be updated, replaced, or obsoleted by other documents at any 46 time. It is inappropriate to use Internet-Drafts as reference 47 material or to cite them other than as "work in progress." 48 This Internet-Draft will expire on August 18, 2014. 50 Copyright Notice 52 Copyright (c) 2014 IETF Trust and the persons identified as the 53 document authors. All rights reserved. 55 This document is subject to BCP 78 and the IETF Trust's Legal 56 Provisions Relating to IETF Documents 57 (http://trustee.ietf.org/license-info) in effect on the date of 58 publication of this document. Please review these documents 59 carefully, as they describe your rights and restrictions with respect 60 to this document. Code Components extracted from this document must 61 include Simplified BSD License text as described in Section 4.e of 62 the Trust Legal Provisions and are provided without warranty as 63 described in the Simplified BSD License. 65 Table of Contents 67 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 68 2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 3 69 3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 70 4. Security Considerations . . . . . . . . . . . . . . . . . . . 7 71 5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 72 6. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 73 6.1. Normative References . . . . . . . . . . . . . . . . . . 8 74 6.2. Informative References . . . . . . . . . . . . . . . . . 8 75 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 9 77 1. Introduction 79 The traditional TCP congestion control requirements were developed in 80 order to promote efficient use of the Internet for reliable bulk 81 transfer of non-time-critical data, such as transfer of large files. 82 They have also been used successfully to govern the reliable transfer 83 of smaller chunks of data in as short a time as possible, such as 84 when fetching Web pages. 86 These algorithms have also been used for transfer of media streams 87 that are viewed in a non-interactive manner, such as "streaming" 88 video, where having the data ready when the viewer wants it is 89 important, but the exact timing of the delivery is not. 91 When doing real time interactive media, the requirements are 92 different; one needs to provide the data continuously, within a very 93 limited time window (no more than 100s of milliseconds end-to-end 94 delay), the sources of data may be able to adapt the amount of data 95 that needs sending within fairly wide margins, and may tolerate some 96 amount of packet loss, but since the data is generated in real time, 97 sending "future" data is impossible, and since it's consumed in real 98 time, data delivered late is useless. 100 While the requirements for RMCAT differ from the requirements for the 101 other flow types, these other flow types will be present in the 102 network. The RMCAT congestion control algorithm must work properly 103 when these other flow types are present as cross traffic on the 104 network. 106 One particular protocol portofolio being developed for this use case 107 is WebRTC [I-D.ietf-rtcweb-overview], where one envisions sending 108 multiple RTP-based flows between two peers, in conjunction with data 109 flows, all at the same time, without having special arrangements with 110 the intervening service providers. 112 Given that this use case is the focus of this document, use cases 113 involving noninteractive media such as YouTube-like video streaming, 114 and use cases using multicast/broadcast-type technologies, are out of 115 scope. 117 The terminology defined in [I-D.ietf-rtcweb-overview] is used in this 118 memo. 120 2. Requirements 122 1. The congestion control algorithm must attempt to provide as-low- 123 as-possible-delay transit for real-time traffic while still 124 providing a useful amount of bandwidth, even when faced with 125 intermediate bottlenecks and competing flows. There may be 126 lower limits on the amount of bandwidth that is useful, but this 127 is largely application-specific and the application may be able 128 to modify or remove flows in order allow some useful flows to 129 get enough bandwidth. (Example: not enough bandwidth for low- 130 latency video+audio, but enough for audio-only.) 132 A. It should also handle routing changes and interface changes 133 (WiFi to 3G data, etc) which may radically change the 134 bandwidth available, and react quickly, especially if there 135 is a reduction in available bandwidth. 137 B. The offered load may be less than the available bandwidth at 138 any given moment, and may vary dramatically over time, 139 including dropping to no load and then resuming a high load, 140 such as in a mute operation. The reaction time between a 141 change in the bandwidth available from the algorithm and a 142 change in the offered load is variable, and may be different 143 when increasing versus decreasing. 145 C. The algorithm must not overreact to short-term bursts (such 146 as web-browsing) which can quickly saturate a local- 147 bottleneck router or link, but also clear quickly, and 148 should recover quickly when the burst ends. This is 149 inherently at odds with the need to react quickly-enough to 150 avoid queue buildup. 152 D. Similarly periodic bursty flows such as DASH or proprietary 153 media streaming algorithms may compete in bursts with the 154 algorithm, and may not be adaptive within a burst. They are 155 often are layered on top of TCP. The algorithm must avoid 156 too much delay buildup during those bursts, and quickly 157 recover. Note that this traffic may on an access link, or 158 may cause a shift in the location of the bottleneck fir the 159 duration of the burst. 161 2. The algorithm must be fair to other flows, both realtime flows 162 (such as other instances of itself), and TCP flows, both long- 163 lived and bursts such as the traffic generated by a typical web 164 browsing session. Note that 'fair' is a rather hard-to-define 165 term. 167 A. Existing flows at a bottleneck must also be fair to new 168 flows to that bottleneck, and must allow new flows to ramp 169 up to a useful share of the bottleneck bandwidth quickly. 171 3. The algorithm should where possible merge information across 172 multiple RTP streams between the same endpoints, whether or not 173 they're multiplexed on the same ports, in order to allow 174 congestion control of the set of streams together instead of as 175 multiple independent streams. This allows better overall 176 bandwidth management, faster response to changing conditions, 177 and fairer sharing of bandwidth with other network users. 178 Alternatively, it should work with an external bandwidth control 179 framework to coordinate bandwidth usage across a bottleneck, 180 such as draft-welzl-rmcat-coupled-cc 181 [I-D.welzl-rmcat-coupled-cc]. 183 A. If possible, it should also share information and adaptation 184 with other non-RTP flows between the same endpoints, such as 185 a WebRTC data channel 187 B. The most correlated bandwidth usage would be with other 188 flows on the same 5-tuple, but there may be use in 189 coordinating measurement and control of the local link(s). 191 C. Use of information about previous flows, especially on the 192 same 5-tuple, may be useful input to the algorithm, 193 especially to startup performance of a new flow. 195 4. The algorithm should not require any special support from 196 network elements (ECN, etc). As much as possible, it should 197 leverage available information about the incoming flow to 198 provide feedback to the sender. Examples of this information 199 are the ECN, packet arrival times, acknowledgments and feedback, 200 packet timestamps, and packet losses; all of these can provide 201 information about the state of the path and any bottlenecks. 203 A. Extra information could be added to the packets to provide 204 more detailed information on actual send times (as opposed 205 to sampling times), but should not be required. 207 B. When additional input signals such as ECN are available, 208 they should be utilized if possible. 210 5. Since the assumption here is a set of RTP streams, the 211 backchannel typically should be done via RTCP; one alternative 212 would be to include it instead in a reverse RTP channel using 213 header extensions. 215 A. In order to react sufficiently quickly when using RTCP for a 216 backchannel, an RTP profile such as AVPF/SAVPF that allows 217 sufficiently frequent feedback [RFC4585] MUST be used. 219 B. Note that in some cases, backchannel messages may be delayed 220 until the RTCP channel can be allocated enough bandwidth, 221 even under AVPF rules. This may also imply negotiating a 222 higher maximum percentage for RTCP data or allowing RMCAT 223 solutions to violate or modify the rules specified for AVPF. 225 C. Bandwidth for the feedback messages should be minimized 226 (such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR) 228 D. Header extensions would avoid the RTCP timing rules issues, 229 and allow the application to allocate bandwidth as needed 230 for the congestion algorithm. 232 E. Backchannel data should be minimized to avoid taking too 233 much reverse-channel bandwidth (since this will often be 234 used in a bidirectional set of flows). In areas of 235 stability, backchannel data may be sent more infrequently so 236 long as algorithm stability and fairness are maintained. 237 When the channel is unstable or has not yet reached 238 equilibrium after a change, backchannel feedback may be more 239 frequent and use more reverse-channel bandwidth. This is an 240 area with considerable flexibility of design, and different 241 approaches to backchannel messages and frequency are 242 expected to be evaluated. 244 6. Flows managed by this algorithm and flows competed against at a 245 bottleneck may have different DSCP markings depending on the 246 type of traffic. A particular bottleneck or section of the 247 network path may or may not honor these markings. 249 A. In WebRTC, a division of packets into 4 classes is 250 envisioned in order of priority: faster-than-audio, audio, 251 video, best-effort, and bulk-transfer. Typically the flows 252 managed by this algorithm would be audio or video in that 253 heirarchy, and feedback flows would be faster-than-audio. 255 7. The algorithm should sense the unexpected lack of backchannel 256 information as a possible indication of a channel overuse 257 problem and react accordingly to avoid burst events causing a 258 congestion collapse. 260 8. The algorithm should not starve competing TCP flows, and should 261 as best as possible avoid starvation by TCP flows. 263 A. An algorithm may be more successful at avoiding starvation 264 from short-lived TCP long-lived/saturating TCP flows. 266 B. In order to avoid starvation other goals may need to be 267 compromised (such as delay). 269 9. The algorithm should be stable and low-delay when faced with 270 active queue management (AQM) algorithms. Also note that these 271 algorithms may apply across multiple queues in the bottleneck, 272 or to a single queue 274 10. The algorithm should quickly adapt to initial network conditions 275 at the start of a flow. This should occur both if the initial 276 bandwidth is above or below the bottleneck bandwidth. 278 A. The startup adaptation may be faster than adaptation later 279 in a flow. It should allow for both slow-start operation 280 (adapt up) and history-based startup (start at a point 281 expected to be at or below channel bandwidth from historical 282 information, which may need to adapt down quickly if the 283 initial guess is wrong). Starting too low and/or adapting 284 up too slowly can cause a critical point in a personal 285 communication to be poor ("Hello!"). Starting over- 286 bandwidth causes other problems for user experience, so 287 there's a tension here. 289 B. Alternative methods to help startup like probing during 290 setup with dummy data may be useful in some applications; in 291 some cases there will be a considerable gap in time between 292 flow creation and the initial flow of data. 294 C. A flow may need to change adaptation rates due to network 295 conditions or changes in the provided flows (such as un- 296 muting or sending data after a gap). 298 11. It should be evaluated in how it works both with backbone-router 299 bottlenecks, (asymmetric) local-loop bottlenecks, and local-lan 300 (WiFi/etc) bottlenecks, and in competition with varying numbers 301 and types of streams (TCP, TCP variants in use, LEDBAT 302 [I-D.ietf-ledbat-congestion], inflexible VoIP UDP flows). 304 12. It should be stable if the RTP streams are halted or 305 discontinuous (VAD/DTX). 307 A. After a resumption of RTP data it may adapt more quickly 308 (similar to the start of a flow), and previous bandwidth 309 estimates may need to be aged or thrown away. 311 3. IANA Considerations 313 This document makes no request of IANA. 315 Note to RFC Editor: this section may be removed on publication as an 316 RFC. 318 4. Security Considerations 320 An attacker with the ability to delete, delay or insert messages in 321 the flow can fake congestion signals, unless they are passed on a 322 tamper-proof path. Since some possible algorithms depend on the 323 timing of packet arrival, even a traditional protected channel does 324 not fully mitigate such attacks. 326 An attack that reduces bandwidth is not necessarily significant, 327 since an on-path attacker could break the connection by discarding 328 all packets. Attacks that increase the percieved available bandwidth 329 are concievable, and need to be evaluated. 331 Algorithm designers SHOULD consider the possibility of malicious on- 332 path attackers. 334 5. Acknowledgements 336 This document is the result of discussions in various fora of the 337 WebRTC effort, in particular on the rtp-congestion@alvestrand.no 338 mailing list. Many people contributed their thoughts to this. 340 6. References 342 6.1. Normative References 344 [I-D.ietf-rtcweb-overview] 345 Alvestrand, H., "Overview: Real Time Protocols for Brower- 346 based Applications", draft-ietf-rtcweb-overview-08 (work 347 in progress), September 2013. 349 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 350 Requirement Levels", BCP 14, RFC 2119, March 1997. 352 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 353 "Extended RTP Profile for Real-time Transport Control 354 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 355 2006. 357 6.2. Informative References 359 [I-D.ietf-ledbat-congestion] 360 Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, 361 "Low Extra Delay Background Transport (LEDBAT)", draft- 362 ietf-ledbat-congestion-10 (work in progress), September 363 2012. 365 [I-D.jesup-rtp-congestion-reqs] 366 Jesup, R. and H. Alvestrand, "Congestion Control 367 Requirements For Real Time Media", draft-jesup-rtp- 368 congestion-reqs-00 (work in progress), March 2012. 370 [I-D.welzl-rmcat-coupled-cc] 371 Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion 372 control for RTP media", draft-welzl-rmcat-coupled-cc-02 373 (work in progress), October 2013. 375 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size 376 Real-Time Transport Control Protocol (RTCP): Opportunities 377 and Consequences", RFC 5506, April 2009. 379 Author's Address 381 Randell Jesup 382 Mozilla 383 USA 385 Email: randell-ietf@jesup.org