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Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) ** Obsolete normative reference: RFC 3448 (Obsoleted by RFC 5348) == Outdated reference: A later version (-12) exists of draft-ietf-avtcore-rtp-multi-stream-optimisation-00 == Outdated reference: A later version (-09) exists of draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-06 Summary: 1 error (**), 0 flaws (~~), 3 warnings (==), 3 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 AVTCORE Working Group C. S. Perkins 3 Internet-Draft University of Glasgow 4 Updates: 3550 (if approved) V. Singh 5 Intended status: Standards Track Aalto University 6 Expires: January 16, 2014 July 15, 2013 8 Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions 9 draft-ietf-avtcore-rtp-circuit-breakers-03 11 Abstract 13 The Real-time Transport Protocol (RTP) is widely used in telephony, 14 video conferencing, and telepresence applications. Such applications 15 are often run on best-effort UDP/IP networks. If congestion control 16 is not implemented in the applications, then network congestion will 17 deteriorate the user's multimedia experience. This document does not 18 propose a congestion control algorithm; instead, it defines a minimal 19 set of RTP "circuit-breakers". Circuit-breakers are conditions under 20 which an RTP sender needs to stop transmitting media data in order to 21 protect the network from excessive congestion. It is expected that, 22 in the absence of severe congestion, all RTP applications running on 23 best-effort IP networks will be able to run without triggering these 24 circuit breakers. Any future RTP congestion control specification 25 will be expected to operate within the constraints defined by these 26 circuit breakers. 28 Status of This Memo 30 This Internet-Draft is submitted in full conformance with the 31 provisions of BCP 78 and BCP 79. 33 Internet-Drafts are working documents of the Internet Engineering 34 Task Force (IETF). Note that other groups may also distribute 35 working documents as Internet-Drafts. The list of current Internet- 36 Drafts is at http://datatracker.ietf.org/drafts/current/. 38 Internet-Drafts are draft documents valid for a maximum of six months 39 and may be updated, replaced, or obsoleted by other documents at any 40 time. It is inappropriate to use Internet-Drafts as reference 41 material or to cite them other than as "work in progress." 43 This Internet-Draft will expire on January 16, 2014. 45 Copyright Notice 47 Copyright (c) 2013 IETF Trust and the persons identified as the 48 document authors. All rights reserved. 50 This document is subject to BCP 78 and the IETF Trust's Legal 51 Provisions Relating to IETF Documents 52 (http://trustee.ietf.org/license-info) in effect on the date of 53 publication of this document. Please review these documents 54 carefully, as they describe your rights and restrictions with respect 55 to this document. Code Components extracted from this document must 56 include Simplified BSD License text as described in Section 4.e of 57 the Trust Legal Provisions and are provided without warranty as 58 described in the Simplified BSD License. 60 Table of Contents 62 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 63 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 64 3. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3 65 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 6 66 4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 7 67 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . 8 68 4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 9 69 4.4. Ceasing Transmission . . . . . . . . . . . . . . . . . . 12 70 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 12 71 6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 13 72 7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 14 73 8. Impact of Explicit Congestion Notification (ECN) . . . . . . 14 74 9. Security Considerations . . . . . . . . . . . . . . . . . . . 14 75 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 76 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15 77 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 15 78 12.1. Normative References . . . . . . . . . . . . . . . . . . 15 79 12.2. Informative References . . . . . . . . . . . . . . . . . 15 80 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17 82 1. Introduction 84 The Real-time Transport Protocol (RTP) [RFC3550] is widely used in 85 voice-over-IP, video teleconferencing, and telepresence systems. 86 Many of these systems run over best-effort UDP/IP networks, and can 87 suffer from packet loss and increased latency if network congestion 88 occurs. Designing effective RTP congestion control algorithms, to 89 adapt the transmission of RTP-based media to match the available 90 network capacity, while also maintaining the user experience, is a 91 difficult but important problem. Many such congestion control and 92 media adaptation algorithms have been proposed, but to date there is 93 no consensus on the correct approach, or even that a single standard 94 algorithm is desirable. 96 This memo does not attempt to propose a new RTP congestion control 97 algorithm. Rather, it proposes a minimal set of "circuit breakers"; 98 conditions under which there is general agreement that an RTP flow is 99 causing serious congestion, and ought to cease transmission. It is 100 expected that future standards-track congestion control algorithms 101 for RTP will operate within the envelope defined by this memo. 103 2. Terminology 105 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 106 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 107 document are to be interpreted as described in RFC 2119 [RFC2119]. 108 This interpretation of these key words applies only when written in 109 ALL CAPS. Mixed- or lower-case uses of these key words are not to be 110 interpreted as carrying special significance in this memo. 112 3. Background 114 We consider congestion control for unicast RTP traffic flows. This 115 is the problem of adapting the transmission of an audio/visual data 116 flow, encapsulated within an RTP transport session, from one sender 117 to one receiver, so that it matches the available network bandwidth. 118 Such adaptation needs to be done in a way that limits the disruption 119 to the user experience caused by both packet loss and excessive rate 120 changes. Congestion control for multicast flows is outside the scope 121 of this memo. Multicast traffic needs different solutions, since the 122 available bandwidth estimator for a group of receivers will differ 123 from that for a single receiver, and because multicast congestion 124 control has to consider issues of fairness across groups of receivers 125 that do not apply to unicast flows. 127 Congestion control for unicast RTP traffic can be implemented in one 128 of two places in the protocol stack. One approach is to run the RTP 129 traffic over a congestion controlled transport protocol, for example 130 over TCP, and to adapt the media encoding to match the dictates of 131 the transport-layer congestion control algorithm. This is safe for 132 the network, but can be suboptimal for the media quality unless the 133 transport protocol is designed to support real-time media flows. We 134 do not consider this class of applications further in this memo, as 135 their network safety is guaranteed by the underlying transport. 137 Alternatively, RTP flows can be run over a non-congestion controlled 138 transport protocol, for example UDP, performing rate adaptation at 139 the application layer based on RTP Control Protocol (RTCP) feedback. 140 With a well-designed, network-aware, application, this allows highly 141 effective media quality adaptation, but there is potential to disrupt 142 the network's operation if the application does not adapt its sending 143 rate in a timely and effective manner. We consider this class of 144 applications in this memo. 146 Congestion control relies on monitoring the delivery of a media flow, 147 and responding to adapt the transmission of that flow when there are 148 signs that the network path is congested. Network congestion can be 149 detected in one of three ways: 1) a receiver can infer the onset of 150 congestion by observing an increase in one-way delay caused by queue 151 build-up within the network; 2) if Explicit Congestion Notification 152 (ECN) [RFC3168] is supported, the network can signal the presence of 153 congestion by marking packets using ECN Congestion Experienced (CE) 154 marks; or 3) in the extreme case, congestion will cause packet loss 155 that can be detected by observing a gap in the received RTP sequence 156 numbers. Once the onset of congestion is observed, the receiver has 157 to send feedback to the sender to indicate that the transmission rate 158 needs to be reduced. How the sender reduces the transmission rate is 159 highly dependent on the media codec being used, and is outside the 160 scope of this memo. 162 There are several ways in which a receiver can send feedback to a 163 media sender within the RTP framework: 165 o The base RTP specification [RFC3550] defines RTCP Reception Report 166 (RR) packets to convey reception quality feedback information, and 167 Sender Report (SR) packets to convey information about the media 168 transmission. RTCP SR packets contain data that can be used to 169 reconstruct media timing at a receiver, along with a count of the 170 total number of octets and packets sent. RTCP RR packets report 171 on the fraction of packets lost in the last reporting interval, 172 the cumulative number of packets lost, the highest sequence number 173 received, and the inter-arrival jitter. The RTCP RR packets also 174 contain timing information that allows the sender to estimate the 175 network round trip time (RTT) to the receivers. RTCP reports are 176 sent periodically, with the reporting interval being determined by 177 the number of SSRCs used in the session and a configured session 178 bandwidth estimate (the number of SSRCs used is usually two in a 179 unicast session, one for each participant, but can be greater if 180 the participants send multiple media streams). The interval 181 between reports sent from each receiver tends to be on the order 182 of a few seconds on average, and it is randomised to avoid 183 synchronisation of reports from multiple receivers. RTCP RR 184 packets allow a receiver to report ongoing network congestion to 185 the sender. However, if a receiver detects the onset of 186 congestion partway through a reporting interval, the base RTP 187 specification contains no provision for sending the RTCP RR packet 188 early, and the receiver has to wait until the next scheduled 189 reporting interval. 191 o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more 192 complex and sophisticated reception quality metrics, but do not 193 change the RTCP timing rules. RTCP extended reports of potential 194 interest for congestion control purposes are the extended packet 195 loss, discard, and burst metrics [RFC3611], 196 [I-D.ietf-xrblock-rtcp-xr-discard], 197 [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics], 198 [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard], 199 [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]; and the extended delay 200 metrics [RFC6843], [RFC6798]. Other RTCP Extended Reports that 201 could be helpful for congestion control purposes might be 202 developed in future. 204 o Rapid feedback about the occurrence of congestion events can be 205 achieved using the Extended RTP Profile for RTCP-Based Feedback 206 (RTP/AVPF) [RFC4585] in place of the more common RTP/AVP profile 207 [RFC3551]. This modifies the RTCP timing rules to allow RTCP 208 reports to be sent early, in some cases immediately, provided the 209 average RTCP reporting interval remains unchanged. It also 210 defines new transport-layer feedback messages, including negative 211 acknowledgements (NACKs), that can be used to report on specific 212 congestion events. The use of the RTP/AVPF profile is dependent 213 on signalling, but is otherwise generally backwards compatible 214 with the RTP/AVP profile, as it keeps the same average RTCP 215 reporting interval as the base RTP specification. The RTP Codec 216 Control Messages [RFC5104] extend the RTP/AVPF profile with 217 additional feedback messages that can be used to influence that 218 way in which rate adaptation occurs. The dynamics of how rapidly 219 feedback can be sent are unchanged. 221 o Finally, Explicit Congestion Notification (ECN) for RTP over UDP 222 [RFC6679] can be used to provide feedback on the number of packets 223 that received an ECN Congestion Experienced (CE) mark. This RTCP 224 extension builds on the RTP/AVPF profile to allow rapid congestion 225 feedback when ECN is supported. 227 In addition to these mechanisms for providing feedback, the sender 228 can include an RTP header extension in each packet to record packet 229 transmission times. There are two methods: [RFC5450] represents the 230 transmission time in terms of a time-offset from the RTP timestamp of 231 the packet, while [RFC6051] includes an explicit NTP-format sending 232 timestamp (potentially more accurate, but a higher header overhead). 233 Accurate sending timestamps can be helpful for estimating queuing 234 delays, to get an early indication of the onset of congestion. 236 Taken together, these various mechanisms allow receivers to provide 237 feedback on the senders when congestion events occur, with varying 238 degrees of timeliness and accuracy. The key distinction is between 239 systems that use only the basic RTCP mechanisms, without RTP/AVPF 240 rapid feedback, and those that use the RTP/AVPF extensions to respond 241 to congestion more rapidly. 243 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile 245 The feedback mechanisms defined in [RFC3550] and available under the 246 RTP/AVP profile [RFC3551] are the minimum that can be assumed for a 247 baseline circuit breaker mechanism that is suitable for all unicast 248 applications of RTP. Accordingly, for an RTP circuit breaker to be 249 useful, it needs to be able to detect that an RTP flow is causing 250 excessive congestion using only basic RTCP features, without needing 251 RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports. 253 RTCP is a fundamental part of the RTP protocol, and the mechanisms 254 described here rely on the implementation of RTCP. Implementations 255 which claim to support RTP, but that do not implement RTCP, cannot 256 use the circuit breaker mechanisms described in this memo. Such 257 implementations SHOULD NOT be used on networks that might be subject 258 to congestion unless equivalent mechanisms are defined using some 259 non-RTCP feedback channel to report congestion and signal circuit 260 breaker conditions. 262 Three potential congestion signals are available from the basic RTCP 263 SR/RR packets and are reported for each synchronisation source (SSRC) 264 in the RTP session: 266 1. The sender can estimate the network round-trip time once per RTCP 267 reporting interval, based on the contents and timing of RTCP SR 268 and RR packets. 270 2. Receivers report a jitter estimate (the statistical variance of 271 the RTP data packet inter-arrival time) calculated over the RTCP 272 reporting interval. Due to the nature of the jitter calculation 273 ([RFC3550], section 6.4.4), the jitter is only meaningful for RTP 274 flows that send a single data packet for each RTP timestamp value 275 (i.e., audio flows, or video flows where each packet comprises 276 one video frame). 278 3. Receivers report the fraction of RTP data packets lost during the 279 RTCP reporting interval, and the cumulative number of RTP packets 280 lost over the entire RTP session. 282 These congestion signals limit the possible circuit breakers, since 283 they give only limited visibility into the behaviour of the network. 285 RTT estimates are widely used in congestion control algorithms, as a 286 proxy for queuing delay measures in delay-based congestion control or 287 to determine connection timeouts. RTT estimates derived from RTCP SR 288 and RR packets sent according to the RTP/AVP timing rules are far too 289 infrequent to be useful though, and don't give enough information to 290 distinguish a delay change due to routing updates from queuing delay 291 caused by congestion. Accordingly, we cannot use the RTT estimate 292 alone as an RTP circuit breaker. 294 Increased jitter can be a signal of transient network congestion, but 295 in the highly aggregated form reported in RTCP RR packets, it offers 296 insufficient information to estimate the extent or persistence of 297 congestion. Jitter reports are a useful early warning of potential 298 network congestion, but provide an insufficiently strong signal to be 299 used as a circuit breaker. 301 The remaining congestion signals are the packet loss fraction and the 302 cumulative number of packets lost. If considered carefully, these 303 can be effective indicators that congestion is occurring in networks 304 where packet loss is primarily due to queue overflows, although loss 305 caused by non-congestive packet corruption can distort the result in 306 some networks. TCP congestion control intentionally tries to fill 307 the router queues, and uses the resulting packet loss as congestion 308 feedback. An RTP flow competing with TCP traffic will therefore 309 expect to see a non-zero packet loss fraction that has to be related 310 to TCP dynamics to estimate available capacity. This behaviour of 311 TCP is reflected in the congestion circuit breaker below, and will 312 affect the design of any RTP congestion control protocol. 314 Two packet loss regimes can be observed: 1) RTCP RR packets show a 315 non-zero packet loss fraction, while the extended highest sequence 316 number received continues to increment; and 2) RR packets show a loss 317 fraction of zero, but the extended highest sequence number received 318 does not increment even though the sender has been transmitting RTP 319 data packets. The former corresponds to the TCP congestion avoidance 320 state, and indicates a congested path that is still delivering data; 321 the latter corresponds to a TCP timeout, and is most likely due to a 322 path failure. A third condition is that data is being sent but no 323 RTCP feedback is received at all, corresponding to a failure of the 324 reverse path. We derive circuit breaker conditions for these loss 325 regimes in the following. 327 4.1. RTP/AVP Circuit Breaker #1: Media Timeout 329 If RTP data packets are being sent, but the RTCP SR or RR packets 330 reporting on that SSRC indicate a non-increasing extended highest 331 sequence number received, this is an indication that those RTP data 332 packets are not reaching the receiver. This could be a short-term 333 issue affecting only a few packets, perhaps caused by a slow-to-open 334 firewall or a transient connectivity problem, but if the issue 335 persists, it is a sign of a more ongoing and significant problem. 336 Accordingly, if a sender of RTP data packets receives two or more 337 consecutive RTCP SR or RR packets from the same receiver, and those 338 packets correspond to its transmission and have a non-increasing 339 extended highest sequence number received field (i.e., the sender 340 receivers at least three RTCP SR or RR packets that report the same 341 value in the extended highest sequence number received field for an 342 SSRC, but the sender has sent RTP data packets for that SSRC that 343 would have caused an increase in the reported value of the extended 344 highest sequence number received if they had reached the receiver), 345 then that sender SHOULD cease transmission (see Section 4.4). 347 The reason for waiting for two or more consecutive RTCP packets with 348 a non-increasing extended highest sequence number is to give enough 349 time for transient reception problems to resolve themselves, but to 350 stop problem flows quickly enough to avoid causing serious ongoing 351 network congestion. A single RTCP report showing no reception could 352 be caused by a transient fault, and so will not cease transmission. 353 Waiting for more than two consecutive RTCP reports before stopping a 354 flow might avoid some false positives, but could lead to problematic 355 flows running for a long time period (potentially tens of seconds, 356 depending on the RTCP reporting interval) before being cut off. 358 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout 360 In addition to media timeouts, as were discussed in Section 4.1, an 361 RTP session has the possibility of an RTCP timeout. This can occur 362 when RTP data packets are being sent, but there are no RTCP reports 363 returned from the receiver. This is either due to a failure of the 364 receiver to send RTCP reports, or a failure of the return path that 365 is preventing those RTCP reporting from being delivered. In either 366 case, it is not safe to continue transmission, since the sender has 367 no way of knowing if it is causing congestion. Accordingly, an RTP 368 sender that has not received any RTCP SR or RTCP RR packets reporting 369 on the SSRC it is using for three or more RTCP reporting intervals 370 SHOULD cease transmission (see Section 4.4). When calculating the 371 timeout, the fixed minimum RTCP reporting interval SHOULD be used 372 (based on the rationale in Section 6.2 of RFC 3550 [RFC3550]). 374 The choice of three RTCP reporting intervals as the timeout is made 375 following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that 376 participants in an RTP session will timeout and remove an RTP sender 377 from the list of active RTP senders if no RTP data packets have been 378 received from that RTP sender within the last two RTCP reporting 379 intervals. Using a timeout of three RTCP reporting intervals is 380 therefore large enough that the other participants will have timed 381 out the sender if a network problem stops the data packets it is 382 sending from reaching the receivers, even allowing for loss of some 383 RTCP packets. 385 If a sender is transmitting a large number of RTP media streams, such 386 that the corresponding RTCP SR or RR packets are too large to fit 387 into the network MTU, this will force the receiver to generate RTCP 388 SR or RR packets in a round-robin manner. In this case, the sender 389 MAY treat receipt of an RTCP SR or RR packet corresponding to an SSRC 390 it sent using the same 5-tuple of source and destination IP address, 391 port, and protocol, as an indication that the receiver and return 392 path are working to prevent the RTCP timeout circuit breaker from 393 triggering. 395 4.3. RTP/AVP Circuit Breaker #3: Congestion 397 If RTP data packets are being sent, and the corresponding RTCP SR or 398 RR packets show non-zero packet loss fraction and increasing extended 399 highest sequence number received, then those RTP data packets are 400 arriving at the receiver, but some degree of congestion is occurring. 401 The RTP/AVP profile [RFC3551] states that: 403 If best-effort service is being used, RTP receivers SHOULD monitor 404 packet loss to ensure that the packet loss rate is within 405 acceptable parameters. Packet loss is considered acceptable if a 406 TCP flow across the same network path and experiencing the same 407 network conditions would achieve an average throughput, measured 408 on a reasonable time scale, that is not less than the RTP flow is 409 achieving. This condition can be satisfied by implementing 410 congestion control mechanisms to adapt the transmission rate (or 411 the number of layers subscribed for a layered multicast session), 412 or by arranging for a receiver to leave the session if the loss 413 rate is unacceptably high. 415 The comparison to TCP cannot be specified exactly, but is intended 416 as an "order-of-magnitude" comparison in time scale and 417 throughput. The time scale on which TCP throughput is measured is 418 the round-trip time of the connection. In essence, this 419 requirement states that it is not acceptable to deploy an 420 application (using RTP or any other transport protocol) on the 421 best-effort Internet which consumes bandwidth arbitrarily and does 422 not compete fairly with TCP within an order of magnitude. 424 The phase "order of magnitude" in the above means within a factor of 425 ten, approximately. In order to implement this, it is necessary to 426 estimate the throughput a TCP connection would achieve over the path. 427 For a long-lived TCP Reno connection, Padhye et al. [Padhye] showed 428 that the throughput can be estimated using the following equation: 430 s 431 X = -------------------------------------------------------------- 432 R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2))) 434 where: 436 X is the transmit rate in bytes/second. 438 s is the packet size in bytes. If data packets vary in size, then 439 the average size is to be used. 441 R is the round trip time in seconds. 443 p is the loss event rate, between 0 and 1.0, of the number of loss 444 events as a fraction of the number of packets transmitted. 446 t_RTO is the TCP retransmission timeout value in seconds, 447 approximated by setting t_RTO = 4*R. 449 b is the number of packets acknowledged by a single TCP 450 acknowledgement ([RFC3448] recommends the use of b=1 since many 451 TCP implementations do not use delayed acknowledgements). 453 This is the same approach to estimated TCP throughput that is used in 454 [RFC3448]. Under conditions of low packet loss, this formula can be 455 approximated as follows with reasonable accuracy: 457 s 458 X = --------------- 459 R * sqrt(p*2/3) 461 It is RECOMMENDED that this simplified throughout equation be used, 462 since the reduction in accuracy is small, and it is much simpler to 463 calculate than the full equation. 465 Given this TCP equation, two parameters need to be estimated and 466 reported to the sender in order to calculate the throughput: the 467 round trip time, R, and the loss event rate, p (the packet size, s, 468 is known to the sender). The round trip time can be estimated from 469 RTCP SR and RR packets. This is done too infrequently for accurate 470 statistics, but is the best that can be done with the standard RTCP 471 mechanisms. 473 Report blocks in RTCP SR or RR packets contain the packet loss 474 fraction, rather than the loss event rate, so p cannot be reported 475 (TCP typically treats the loss of multiple packets within a single 476 RTT as one loss event, but RTCP RR packets report the overall 477 fraction of packets lost, not caring about when the losses occurred). 478 Using the loss fraction in place of the loss event rate can 479 overestimate the loss. We believe that this overestimate will not be 480 significant, given that we are only interested in order of magnitude 481 comparison ([Floyd] section 3.2.1 shows that the difference is small 482 for steady-state conditions and random loss, but using the loss 483 fraction is more conservative in the case of bursty loss). 485 The congestion circuit breaker is therefore: when a sender receives 486 an RTCP SR or RR packet that contains a report block for an SSRC it 487 is using, that sender has to check the fraction lost field in that 488 report block to determine if there is a non-zero packet loss rate. 489 If the fraction lost field is zero, then continue sending as normal. 490 If the fraction lost is greater than zero, then estimate the TCP 491 throughput using the simplified equation above, and the measured R, p 492 (approximated by the fraction lost), and s. Compare this with the 493 actual sending rate. If the actual sending rate is more than ten 494 times the estimated sending rate derived from the TCP throughput 495 equation for two consecutive RTCP reporting intervals, the sender 496 SHOULD cease transmission (see Section 4.4). Systems that usually 497 send at a high data rate, but that can reduce their data rate 498 significantly (i.e., by at least a factor of ten), MAY first reduce 499 their sending rate to this lower value to see if this resolves the 500 congestion, but MUST then cease transmission if the problem does not 501 resolve itself within a further two RTCP reporting intervals (see 502 Section 4.4). An example of this might be a video conferencing 503 system that backs off to sending audio only, before completely 504 dropping the call. If such a reduction in sending rate resolves the 505 congestion problem, the sender MAY gradually increase the rate at 506 which it sends data after a reasonable amount of time has passed, 507 provided it takes care not to cause the problem to recur 508 ("reasonable" is intentionally not defined here). 510 If the incoming RTCP SR or RR packets are using a reduced minimum 511 RTCP reporting interval (as specified in Section 6.2 of RFC 3550 512 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that reduced RTCP 513 reporting interval is used when determining if the circuit breaker is 514 triggered. The RTCP reporting interval of the media sender does not 515 affect how quickly congestion circuit breaker can trigger. The 516 timing is based on the RTCP reporting interval of the receiver that 517 matters (note that RTCP requires all participants in a session to 518 have similar reporting intervals, else the participant timeout rules 519 in [RFC3550] will not work). 521 As in Section 4.1, we use two reporting intervals to avoid triggering 522 the circuit breaker on transient failures. This circuit breaker is a 523 worst-case condition, and congestion control needs to be performed to 524 keep well within this bound. It is expected that the circuit breaker 525 will only be triggered if the usual congestion control fails for some 526 reason. 528 If there are more media streams that can be reported in a single RTCP 529 SR or RR packet, or if the size of a complete RTCP SR or RR packet 530 exceeds the network MTU, then the receiver will report on a subset of 531 sources in each reporting interval, with the subsets selected round- 532 robin across multiple intervals so that all sources are eventually 533 reported [RFC3550]. When generating such round-robin RTCP reports, 534 priority SHOULD be given to reports on sources that have high packet 535 loss rates, to ensure that senders are aware of network congestion 536 they are causing (this is an update to [RFC3550]). 538 4.4. Ceasing Transmission 540 What it means to cease transmission depends on the application, but 541 the intention is that the application will stop sending RTP data 542 packets to a particular destination 3-tuple (transport protocol, 543 destination port, IP address), until the user makes an explicit 544 attempt to restart the call. It is important that a human user is 545 involved in the decision to try to restart the call, since that user 546 will eventually give up if the calls repeatedly trigger the circuit 547 breaker. This will help avoid problems with automatic redial systems 548 from congesting the network. Accordingly, RTP flows halted by the 549 circuit breaker SHOULD NOT be restarted automatically unless the 550 sender has received information that the congestion has dissipated. 552 It is recognised that the RTP implementation in some systems might 553 not be able to determine if a call set-up request was initiated by a 554 human user, or automatically by some scripted higher-level component 555 of the system. These implementations SHOULD rate limit attempts to 556 restart a call to the same destination 3-tuple as used by a previous 557 call that was recently halted by the circuit breaker. The chosen 558 rate limit ought to not exceed the rate at which an annoyed human 559 caller might redial a misbehaving phone. 561 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile 563 Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF) 564 [RFC4585] allows receivers to send early RTCP reports in some cases, 565 to inform the sender about particular events in the media stream. 566 There are several use cases for such early RTCP reports, including 567 providing rapid feedback to a sender about the onset of congestion. 569 Receiving rapid feedback about congestion events potentially allows 570 congestion control algorithms to be more responsive, and to better 571 adapt the media transmission to the limitations of the network. It 572 is expected that many RTP congestion control algorithms will adopt 573 the RTP/AVPF profile for this reason, defining new transport layer 574 feedback reports that suit their requirements. Since these reports 575 are not yet defined, and likely very specific to the details of the 576 congestion control algorithm chosen, they cannot be used as part of 577 the generic RTP circuit breaker. 579 If the extension for Reduced-Size RTCP [RFC5506] is not used, early 580 RTCP feedback packets sent according to the RTP/AVPF profile will be 581 compound RTCP packets that include an RTCP SR/RR packet. That RTCP 582 SR/RR packet MUST be processed as if it were sent as a regular RTCP 583 report and counted towards the circuit breaker conditions specified 584 in Section 4 of this memo. This will potentially make the RTP 585 circuit breaker fire earlier than it would if the RTP/AVPF profile 586 was not used. 588 Reduced-size RTCP reports sent under the RTP/AVPF early feedback 589 rules that do not contain an RTCP SR or RR packet MUST be ignored by 590 the RTP circuit breaker (they do not contain the information used by 591 the circuit breaker algorithm). Reduced-size RTCP reports sent under 592 the RTP/AVPF early feedback rules that contain RTCP SR or RR packets 593 MUST be processed as if they were sent as regular RTCP reports, and 594 counted towards the circuit breaker conditions specified in Section 4 595 of this memo. This will potentially make the RTP circuit breaker 596 fire earlier than it would if the RTP/AVPF profile was not used. 598 When using ECN with RTP (see Section 8), early RTCP feedback packets 599 can contain ECN feedback reports. The count of ECN-CE marked packets 600 contained in those ECN feedback reports is counted towards the number 601 of lost packets reported if the ECN Feedback Report report is sent in 602 an compound RTCP packet along with an RTCP SR/RR report packet. 603 Reports of ECN-CE packets sent as reduced-size RTCP ECN feedback 604 packets without an RTCP SR/RR packet MUST be ignored. 606 These rules are intended to allow the use of low-overhead early RTP/ 607 AVPF feedback for generic NACK messages without triggering the RTP 608 circuit breaker. This is expected to make such feedback suitable for 609 RTP congestion control algorithms that need to quickly report loss 610 events in between regular RTCP reports. The reaction to reduced-size 611 RTCP SR/RR packets is to allow such algorithms to send feedback that 612 can trigger the circuit breaker, when desired. 614 6. Impact of RTCP XR 616 RTCP Extended Report (XR) blocks provide additional reception quality 617 metrics, but do not change the RTCP timing rules. Some of the RTCP 618 XR blocks provide information that might be useful for congestion 619 control purposes, others provided non-congestion-related metrics. 620 With the exception of RTCP XR ECN Summary Reports (see Section 8), 621 the presence of RTCP XR blocks in a compound RTCP packet does not 622 affect the RTP circuit breaker algorithm. For consistency and ease 623 of implementation, only the reception report blocks contained in RTCP 624 SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets, 625 are used by the RTP circuit breaker algorithm. 627 7. Impact of RTCP Reporting Groups 629 An optimisation for grouping RTCP reception statistics and other 630 feedback in RTP sessions with large numbers of participants is given 631 in [I-D.ietf-avtcore-rtp-multi-stream-optimisation]. This allows one 632 SSRC to act as a representative that sends reports on behalf of other 633 SSRCs that are co-located in the same endpoint and see identical 634 reception quality. When running the circuit breaker algorithms, an 635 endpoint MUST treat a reception report from the representative of the 636 reporting group as if a reception report was received from all 637 members of that group. 639 8. Impact of Explicit Congestion Notification (ECN) 641 The use of ECN for RTP flows does not affect the media timeout RTP 642 circuit breaker (Section 4.1) or the RTCP timeout circuit breaker 643 (Section 4.2), since these are both connectivity checks that simply 644 determinate if any packets are being received. 646 ECN-CE marked packets SHOULD be treated as if it were lost for the 647 purposes of congestion control, when determining the optimal media 648 sending rate for an RTP flow. If an RTP sender has negotiated ECN 649 support for an RTP session, and has successfully initiated ECN use on 650 the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD 651 be treated as if they were lost when calculating if the congestion- 652 based RTP circuit breaker (Section 4.3) has been met. The count of 653 ECN-CE marked RTP packets is returned in RTCP XR ECN summary report 654 packets if support for ECN has been initiated for an RTP session. 656 9. Security Considerations 658 The security considerations of [RFC3550] apply. 660 If the RTP/AVPF profile is used to provide rapid RTCP feedback, the 661 security considerations of [RFC4585] apply. If ECN feedback for RTP 662 over UDP/IP is used, the security considerations of [RFC6679] apply. 664 If non-authenticated RTCP reports are used, an on-path attacker can 665 trivially generate fake RTCP packets that indicate high packet loss 666 rates, causing the circuit breaker to trigger and disrupting an RTP 667 session. This is somewhat more difficult for an off-path attacker, 668 due to the need to guess the randomly chosen RTP SSRC value and the 669 RTP sequence number. This attack can be avoided if RTCP packets are 670 authenticated, for example using the Secure RTP profile [RFC3711]. 672 10. IANA Considerations 674 There are no actions for IANA. 676 11. Acknowledgements 678 The authors would like to thank Bernard Aboba, Harald Alvestrand, 679 Kevin Gross, Cullen Jennings, Randell Jesup, Jonathan Lennox, Matt 680 Mathis, Stephen McQuistin, Eric Rescorla, and Abheek Saha for their 681 valuable feedback. 683 12. References 685 12.1. Normative References 687 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 688 Requirement Levels", BCP 14, RFC 2119, March 1997. 690 [RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP 691 Friendly Rate Control (TFRC): Protocol Specification", RFC 692 3448, January 2003. 694 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 695 Jacobson, "RTP: A Transport Protocol for Real-Time 696 Applications", STD 64, RFC 3550, July 2003. 698 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 699 Video Conferences with Minimal Control", STD 65, RFC 3551, 700 July 2003. 702 [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control 703 Protocol Extended Reports (RTCP XR)", RFC 3611, November 704 2003. 706 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 707 "Extended RTP Profile for Real-time Transport Control 708 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 709 2006. 711 12.2. Informative References 713 [Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer, 714 "Equation-Based Congestion Control for Unicast 715 Applications", Proc. ACM SIGCOMM 2000, DOI 10.1145/ 716 347059.347397, August 2000. 718 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] 719 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 720 "Sending Multiple Media Streams in a Single RTP Session: 721 Grouping RTCP Reception Statistics and Other Feedback", 722 draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work 723 in progress), July 2013. 725 [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard] 726 Clark, A., Huang, R., and W. Wu, "RTP Control 727 Protocol(RTCP) Extended Report (XR) Block for Burst/Gap 728 Discard metric Reporting", draft-ietf-xrblock-rtcp-xr- 729 burst-gap-discard-14 (work in progress), April 2013. 731 [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss] 732 Clark, A., Zhang, S., Zhao, J., and W. Wu, "RTP Control 733 Protocol (RTCP) Extended Report (XR) Block for Burst/Gap 734 Loss metric Reporting", draft-ietf-xrblock-rtcp-xr-burst- 735 gap-loss-12 (work in progress), April 2013. 737 [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics] 738 Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol 739 (RTCP) Extended Reports (XR) for Run Length Encoding (RLE) 740 of Discarded Packets", draft-ietf-xrblock-rtcp-xr-discard- 741 rle-metrics-06 (work in progress), July 2013. 743 [I-D.ietf-xrblock-rtcp-xr-discard] 744 Clark, A., Zorn, G., and W. Wu, "RTP Control Protocol 745 (RTCP) Extended Report (XR) Block for Discard Count metric 746 Reporting", draft-ietf-xrblock-rtcp-xr-discard-15 (work in 747 progress), June 2013. 749 [Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose, 750 "Modeling TCP Throughput: A Simple Model and its Empirical 751 Validation", Proc. ACM SIGCOMM 1998, DOI 10.1145/ 752 285237.285291, August 1998. 754 [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition 755 of Explicit Congestion Notification (ECN) to IP", RFC 756 3168, September 2001. 758 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 759 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 760 RFC 3711, March 2004. 762 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 763 "Codec Control Messages in the RTP Audio-Visual Profile 764 with Feedback (AVPF)", RFC 5104, February 2008. 766 [RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in 767 RTP Streams", RFC 5450, March 2009. 769 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size 770 Real-Time Transport Control Protocol (RTCP): Opportunities 771 and Consequences", RFC 5506, April 2009. 773 [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP 774 Flows", RFC 6051, November 2010. 776 [RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., 777 and K. Carlberg, "Explicit Congestion Notification (ECN) 778 for RTP over UDP", RFC 6679, August 2012. 780 [RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended 781 Report (XR) Block for Packet Delay Variation Metric 782 Reporting", RFC 6798, November 2012. 784 [RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol 785 (RTCP) Extended Report (XR) Block for Delay Metric 786 Reporting", RFC 6843, January 2013. 788 Authors' Addresses 790 Colin Perkins 791 University of Glasgow 792 School of Computing Science 793 Glasgow G12 8QQ 794 United Kingdom 796 Email: csp@csperkins.org 798 Varun Singh 799 Aalto University 800 School of Electrical Engineering 801 Otakaari 5 A 802 Espoo, FIN 02150 803 Finland 805 Email: varun@comnet.tkk.fi 806 URI: http://www.netlab.tkk.fi/~varun/