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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group R. Jesup 3 Internet-Draft Mozilla 4 Intended status: Informational Feb 25, 2013 5 Expires: August 29, 2013 7 Congestion Control Requirements For RMCAT 8 draft-jesup-rmcat-reqs-01 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 by low-delay, semi-reliable data delivery, 16 are different from the requirements for bulk transfer like FTP or 17 bursty transfers like Web pages, and the TCP algorithms are not 18 suitable for this traffic. 20 This document attempts to describe a set of requirements that can be 21 used to evaluate other congestion control mechanisms in order to 22 figure out their fitness for this purpose, and in particular to 23 provide a set of possible requirements for proposals coming out of 24 the RMCAT Working Group. 26 This document is derived from draft-jesup-rtp-congestion-reqs 27 [I-D.jesup-rtp-congestion-reqs]. 29 Requirements Language 31 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 32 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 33 document are to be interpreted as described in RFC 2119 [RFC2119]. 35 Status of this Memo 37 This Internet-Draft is submitted in full conformance with the 38 provisions of BCP 78 and BCP 79. 40 Internet-Drafts are working documents of the Internet Engineering 41 Task Force (IETF). Note that other groups may also distribute 42 working documents as Internet-Drafts. The list of current Internet- 43 Drafts is at http://datatracker.ietf.org/drafts/current/. 45 Internet-Drafts are draft documents valid for a maximum of six months 46 and may be updated, replaced, or obsoleted by other documents at any 47 time. It is inappropriate to use Internet-Drafts as reference 48 material or to cite them other than as "work in progress." 49 This Internet-Draft will expire on August 29, 2013. 51 Copyright Notice 53 Copyright (c) 2013 IETF Trust and the persons identified as the 54 document authors. All rights reserved. 56 This document is subject to BCP 78 and the IETF Trust's Legal 57 Provisions Relating to IETF Documents 58 (http://trustee.ietf.org/license-info) in effect on the date of 59 publication of this document. Please review these documents 60 carefully, as they describe your rights and restrictions with respect 61 to this document. Code Components extracted from this document must 62 include Simplified BSD License text as described in Section 4.e of 63 the Trust Legal Provisions and are provided without warranty as 64 described in the Simplified BSD License. 66 Table of Contents 68 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 69 2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 4 70 3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 7 71 4. Security Considerations . . . . . . . . . . . . . . . . . . . . 8 72 5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 8 73 6. References . . . . . . . . . . . . . . . . . . . . . . . . . . 8 74 6.1. Normative References . . . . . . . . . . . . . . . . . . . 8 75 6.2. Informative References . . . . . . . . . . . . . . . . . . 8 76 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 9 78 1. Introduction 80 The traditional TCP congestion control requirements were developed in 81 order to promote efficient use of the Internet for reliable bulk 82 transfer of non-time-critical data, such as transfer of large files. 83 They have also been used successfully to govern the reliable transfer 84 of smaller chunks of data in "as fast as possible" mode, such as when 85 fetching Web pages. 87 These algorithms have also been used for transfer of media streams 88 that are viewed in a non-interactive manner, such as "streaming" 89 video, where having the data ready when the viewer wants it is 90 important, but the exact timing of the delivery is not. 92 When doing real time interactive media, the requirements are 93 different; one needs to provide the data continuously, within a very 94 limited time window (no more than 100s of milliseconds end-to-end 95 delay), the sources of data may be able to adapt the amount of data 96 that needs sending within fairly wide margins, and may tolerate some 97 amount of packet loss, but since the data is generated in real time, 98 sending "future" data is impossible, and since it's consumed in real 99 time, data delivered late is useless. 101 One particular protocol portofolio being developed for this use case 102 is WebRTC [I-D.ietf-rtcweb-overview], where one envisions sending 103 multiple RTP-based flows between two peers, in conjunction with data 104 flows, all at the same time, without having special arrangements with 105 the intervening service providers. 107 Given that this use case is the focus of this document, use cases 108 involving noninteractive media such as YouTube-like video streaming, 109 and use cases using multicast/broadcast-type technologies, are out of 110 scope. 112 The terminology defined in [I-D.ietf-rtcweb-overview] is used in this 113 memo. 115 2. Requirements 117 1. The congestion control algorithm must attempt to provide as-low- 118 as-possible-delay transit for real-time traffic while still 119 providing a useful amount of bandwidth, even when faced with 120 intermediate bottlenecks and competing flows. There may be 121 lower limits on the amount of bandwidth that is useful, but this 122 is largely application-specific and the application may be able 123 to modify or remove flows in order allow some useful flows to 124 get enough bandwidth. (Example: not enough bandwidth for low- 125 latency video+audio, but enough for audio-only.) 127 A. It should also deal well with routing changes and interface 128 changes (WiFi to 3G data, etc) which may radically change 129 the bandwidth available. 131 2. The algorithm must be fair to other flows, both realtime flows 132 (such as other instances of itself), and TCP flows, both long- 133 lived and bursts such as the traffic generated by a typical web 134 browsing session. Note that 'fair' is a rather hard-to-define 135 term. 137 A. The algorithm must not overreact to short-term bursts (such 138 as web-browsing) which can quickly saturate a local- 139 bottleneck router or link, but also clear quickly, and 140 should recover quickly when the burst ends. 142 B. We will need make some evaluation of fairness, but deciding 143 what is "fair" is a tough question and likely to be 144 partially subjective, but we should specify some of the 145 inputs needed in order to select among algorithms and 146 tunings presented as options. 148 3. The algorithm should where possible merge information across 149 multiple RTP streams between the same endpoints, whether or not 150 they're multiplexed on the same ports, in order to allow 151 congestion control of the set of streams together instead of as 152 multiple independent streams. This allows better overall 153 bandwidth management, faster response to changing conditions, 154 and fairer sharing of bandwidth with other network users. 156 A. If possible, it should also share information and adaptation 157 with other non-RTP flows between the same endpoints, such as 158 a WebRTC data channel 160 4. The algorithm should not require any special support from 161 network elements (ECN, etc). As much as possible, it should 162 leverage existing information about the incoming flows to 163 provide feedback to the sender. Examples of this information 164 are the packet arrival times, acknowledgments and feedback, 165 packet timestamps, packet sizes, packet losses. Extra 166 information could be added to the packets to provide more 167 detailed information on actual send times (as opposed to 168 sampling times), but should not be required. 170 A. When additional input signals such as ECN are available, 171 they should be utilized if possible. 173 5. Since the assumption here is a set of RTP streams, the 174 backchannel typically should be done via RTCP; the alternative 175 would be to include it in a reverse RTP channel using header 176 extensions. 178 A. In order to react sufficiently quickly, the AVPF/SAVPF RTP 179 profile[RFC4585] must be used 181 B. Note that in some cases, backchannel messages may be delayed 182 until the RTCP channel can be allocated enough bandwidth, 183 even under AVPF rules. This may also imply negotiating a 184 higher maximum percentage for RTCP data or allowing RMCAT 185 solutions to violate or modify the rules specified for AVPF. 187 C. Note that RTCP is of course unreliable 189 D. Bandwidth for the feedback messages should be minimized 190 (such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR) 192 E. Header extensions would avoid the RTCP timing rules issues, 193 and allow the application to allocate bandwidth as needed 194 for the congestion algorithm. 196 F. Backchannel data should be minimized to avoid taking too 197 much reverse-channel bandwidth (since this will often be 198 used in a bidirectional set of flows). In areas of 199 stability, backchannel data may be sent more infrequently so 200 long as algorithm stability and fairness are maintained. 201 When the channel is unstable or has not yet reached 202 equilibrium after a change, backchannel feedback may be more 203 frequent and use more reverse-channel bandwidth. This is an 204 area with considerable flexibility of design, and different 205 approaches to backchannel messages and fequency are expected 206 to be evaluated. 208 6. Where possible and helpful, the algorithm should leverage and 209 piggyback on other RTP/RTCP communications, such as SR/RR, 210 rctp-fb PLI, RPSI, SLI or application-specific NACK messages 211 (such as for loss information), and also reverse-direction RTP. 213 7. The algorithm should sense the unexpected lack of backchannel 214 information as a possible indication of a channel overuse 215 problem and react accordingly to avoid burst events causing a 216 congestion collapse. 218 8. It should attempt to avoid bandwidth 'collapse' when facing a 219 long-lived saturating TCP flow or flows. (I.e. a classic delay- 220 sensitive algorithm will reduce bandwidth to keep delay down 221 until the TCP flow has all the bandwidth). See the Cx-TCP 222 algorithm discussed in a recent Transactions On Networking 223 [cx-tcp] for an example of a delay-sensitive congestion-control 224 algorithm that transitions to a loss-based mode when competing 225 with TCP flows - at the cost of increased delay. 227 9. The algorithm should be stable and low-delay when faced with 228 active queue management (AQM) such as RED [RFC2309] or CoDel 229 [I-D.nichols-tsvwg-codel] in the channel. 231 10. The algorithm should quickly adapt to initial network conditions 232 at the start of a flow. This should occur both if the initial 233 bandwidth is above or below the bottleneck bandwidth. 235 A. The startup adaptation may be faster than adaptation later 236 in a flow. It should allow for both slow-start operation 237 (adapt up) and history-based startup (start at a point 238 expected to be at or below channel bandwidth from historical 239 information, which may need to adapt down quickly if the 240 initial guess is wrong). Starting too low and/or adapting 241 up too slowly can cause a critical point in a personal 242 communication to be poor ("Hello!"). 244 B. Starting over-bandwidth causes other problems for user 245 experience, so there's a tension here. 247 C. Alternative methods to help startup like probing during 248 setup with dummy data may be useful in some applications. 250 11. It should be evaluated in how it works both with backbone-router 251 bottlenecks, (asymmetric) local-loop bottlenecks, and local-lan 252 (WiFi/etc) bottlenecks, and in competition with varying numbers 253 and types of streams (TCP, TCP variants in use, LEDBAT 254 [I-D.ietf-ledbat-congestion], inflexible VoIP UDP flows). 256 12. It should be stable if the RTP streams are halted or 257 discontinuous (VAD/DTX). 259 A. After a resumption of RTP data it may adapt more quickly 260 (similar to the start of a flow), and previous bandwidth 261 estimates may need to be aged or thrown away. 263 3. IANA Considerations 265 This document makes no request of IANA. 267 Note to RFC Editor: this section may be removed on publication as an 268 RFC. 270 4. Security Considerations 272 An attacker with the ability to delete, delay or insert messages in 273 the flow can fake congestion signals, unless they are passed on a 274 tamper-proof path. Since some possible algorithms depend on the 275 timing of packet arrival, even a traditional protected channel does 276 not fully mitigate such attacks. 278 An attack that reduces bandwidth is not necessarily significant, 279 since an on-path attacker could break the connection by discarding 280 all packets. Attacks that increase the percieved available bandwidth 281 are concievable, and need to be evaluated. 283 Algorithm designers SHOULD consider the possibility of malicious on- 284 path attackers. 286 5. Acknowledgements 288 This document is the result of discussions in various fora of the 289 WebRTC effort, in particular on the rtp-congestion@alvestrand.no 290 mailing list. Many people contributed their thoughts to this. 292 6. References 294 6.1. Normative References 296 [I-D.ietf-rtcweb-overview] 297 Alvestrand, H., "Overview: Real Time Protocols for Brower- 298 based Applications", draft-ietf-rtcweb-overview-06 (work 299 in progress), February 2013. 301 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 302 Requirement Levels", BCP 14, RFC 2119, March 1997. 304 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 305 "Extended RTP Profile for Real-time Transport Control 306 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, 307 July 2006. 309 6.2. Informative References 311 [I-D.ietf-ledbat-congestion] 312 Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, 313 "Low Extra Delay Background Transport (LEDBAT)", 314 draft-ietf-ledbat-congestion-10 (work in progress), 315 September 2012. 317 [I-D.jesup-rtp-congestion-reqs] 318 Jesup, R. and H. Alvestrand, "Congestion Control 319 Requirements For Real Time Media", 320 draft-jesup-rtp-congestion-reqs-00 (work in progress), 321 March 2012. 323 [I-D.nichols-tsvwg-codel] 324 Nichols, K., "Controlled Delay Active Queue Management", 325 draft-nichols-tsvwg-codel-00 (work in progress), 326 July 2012. 328 [RFC2309] Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering, 329 S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G., 330 Partridge, C., Peterson, L., Ramakrishnan, K., Shenker, 331 S., Wroclawski, J., and L. Zhang, "Recommendations on 332 Queue Management and Congestion Avoidance in the 333 Internet", RFC 2309, April 1998. 335 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size 336 Real-Time Transport Control Protocol (RTCP): Opportunities 337 and Consequences", RFC 5506, April 2009. 339 [cx-tcp] Budzisz, L., Stanojevic, R., Schlote, A., Baker, F., and 340 R. Shorten, "On the Fair Coexistence of Loss- and Delay- 341 Based TCP", December 2011. 343 Author's Address 345 Randell Jesup 346 Mozilla 347 USA 349 Email: randell-ietf@jesup.org