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Singh 5 Intended status: Standards Track Aalto University 6 Expires: September 07, 2015 March 06, 2015 8 Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions 9 draft-ietf-avtcore-rtp-circuit-breakers-09 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 September 07, 2015. 45 Copyright Notice 47 Copyright (c) 2015 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 . . . . . . . . 8 67 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . 8 68 4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 10 69 4.4. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 13 70 4.5. Choice of Circuit Breaker Interval . . . . . . . . . . . 14 71 4.6. Ceasing Transmission . . . . . . . . . . . . . . . . . . 15 72 5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles 16 73 6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 17 74 7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 17 75 8. Impact of Explicit Congestion Notification (ECN) . . . . . . 18 76 9. Impact of Bundled Media and Layered Coding . . . . . . . . . 18 77 10. Security Considerations . . . . . . . . . . . . . . . . . . . 18 78 11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19 79 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 19 80 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 19 81 13.1. Normative References . . . . . . . . . . . . . . . . . . 19 82 13.2. Informative References . . . . . . . . . . . . . . . . . 20 83 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 22 85 1. Introduction 87 The Real-time Transport Protocol (RTP) [RFC3550] is widely used in 88 voice-over-IP, video teleconferencing, and telepresence systems. 89 Many of these systems run over best-effort UDP/IP networks, and can 90 suffer from packet loss and increased latency if network congestion 91 occurs. Designing effective RTP congestion control algorithms, to 92 adapt the transmission of RTP-based media to match the available 93 network capacity, while also maintaining the user experience, is a 94 difficult but important problem. Many such congestion control and 95 media adaptation algorithms have been proposed, but to date there is 96 no consensus on the correct approach, or even that a single standard 97 algorithm is desirable. 99 This memo does not attempt to propose a new RTP congestion control 100 algorithm. Rather, it proposes a minimal set of "circuit breakers"; 101 conditions under which there is general agreement that an RTP flow is 102 causing serious congestion, and ought to cease transmission. It is 103 expected that future standards-track congestion control algorithms 104 for RTP will operate within the envelope defined by this memo. 106 2. Terminology 108 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 109 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 110 document are to be interpreted as described in RFC 2119 [RFC2119]. 111 This interpretation of these key words applies only when written in 112 ALL CAPS. Mixed- or lower-case uses of these key words are not to be 113 interpreted as carrying special significance in this memo. 115 3. Background 117 We consider congestion control for unicast RTP traffic flows. This 118 is the problem of adapting the transmission of an audio/visual data 119 flow, encapsulated within an RTP transport session, from one sender 120 to one receiver, so that it matches the available network bandwidth. 121 Such adaptation needs to be done in a way that limits the disruption 122 to the user experience caused by both packet loss and excessive rate 123 changes. Congestion control for multicast flows is outside the scope 124 of this memo. Multicast traffic needs different solutions, since the 125 available bandwidth estimator for a group of receivers will differ 126 from that for a single receiver, and because multicast congestion 127 control has to consider issues of fairness across groups of receivers 128 that do not apply to unicast flows. 130 Congestion control for unicast RTP traffic can be implemented in one 131 of two places in the protocol stack. One approach is to run the RTP 132 traffic over a congestion controlled transport protocol, for example 133 over TCP, and to adapt the media encoding to match the dictates of 134 the transport-layer congestion control algorithm. This is safe for 135 the network, but can be suboptimal for the media quality unless the 136 transport protocol is designed to support real-time media flows. We 137 do not consider this class of applications further in this memo, as 138 their network safety is guaranteed by the underlying transport. 140 Alternatively, RTP flows can be run over a non-congestion controlled 141 transport protocol, for example UDP, performing rate adaptation at 142 the application layer based on RTP Control Protocol (RTCP) feedback. 143 With a well-designed, network-aware, application, this allows highly 144 effective media quality adaptation, but there is potential to disrupt 145 the network's operation if the application does not adapt its sending 146 rate in a timely and effective manner. We consider this class of 147 applications in this memo. 149 Congestion control relies on monitoring the delivery of a media flow, 150 and responding to adapt the transmission of that flow when there are 151 signs that the network path is congested. Network congestion can be 152 detected in one of three ways: 1) a receiver can infer the onset of 153 congestion by observing an increase in one-way delay caused by queue 154 build-up within the network; 2) if Explicit Congestion Notification 155 (ECN) [RFC3168] is supported, the network can signal the presence of 156 congestion by marking packets using ECN Congestion Experienced (CE) 157 marks; or 3) in the extreme case, congestion will cause packet loss 158 that can be detected by observing a gap in the received RTP sequence 159 numbers. Once the onset of congestion is observed, the receiver has 160 to send feedback to the sender to indicate that the transmission rate 161 needs to be reduced. How the sender reduces the transmission rate is 162 highly dependent on the media codec being used, and is outside the 163 scope of this memo. 165 There are several ways in which a receiver can send feedback to a 166 media sender within the RTP framework: 168 o The base RTP specification [RFC3550] defines RTCP Reception Report 169 (RR) packets to convey reception quality feedback information, and 170 Sender Report (SR) packets to convey information about the media 171 transmission. RTCP SR packets contain data that can be used to 172 reconstruct media timing at a receiver, along with a count of the 173 total number of octets and packets sent. RTCP RR packets report 174 on the fraction of packets lost in the last reporting interval, 175 the cumulative number of packets lost, the highest sequence number 176 received, and the inter-arrival jitter. The RTCP RR packets also 177 contain timing information that allows the sender to estimate the 178 network round trip time (RTT) to the receivers. RTCP reports are 179 sent periodically, with the reporting interval being determined by 180 the number of SSRCs used in the session and a configured session 181 bandwidth estimate (the number of SSRCs used is usually two in a 182 unicast session, one for each participant, but can be greater if 183 the participants send multiple media streams). The interval 184 between reports sent from each receiver tends to be on the order 185 of a few seconds on average, although it varies with the session 186 bandwidth, and sub-second reporting intervals are possible in high 187 bandwidth sessions, and it is randomised to avoid synchronisation 188 of reports from multiple receivers. RTCP RR packets allow a 189 receiver to report ongoing network congestion to the sender. 190 However, if a receiver detects the onset of congestion part way 191 through a reporting interval, the base RTP specification contains 192 no provision for sending the RTCP RR packet early, and the 193 receiver has to wait until the next scheduled reporting interval. 195 o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more 196 complex and sophisticated reception quality metrics, but do not 197 change the RTCP timing rules. RTCP extended reports of potential 198 interest for congestion control purposes are the extended packet 199 loss, discard, and burst metrics [RFC3611], [RFC7002], [RFC7097], 200 [RFC7003], [RFC6958]; and the extended delay metrics [RFC6843], 201 [RFC6798]. Other RTCP Extended Reports that could be helpful for 202 congestion control purposes might be 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] (or its secure variant, RTP/SAVPF [RFC5124]) 207 in place of the RTP/AVP profile [RFC3551]. This modifies the RTCP 208 timing rules to allow RTCP reports to be sent early, in some cases 209 immediately, provided the RTCP transmission rate keeps within its 210 bandwidth allocation. It also defines transport-layer feedback 211 messages, including negative acknowledgements (NACKs), that can be 212 used to report on specific congestion events. RTP Codec Control 213 Messages [RFC5104] extend the RTP/AVPF profile with additional 214 feedback messages that can be used to influence that way in which 215 rate adaptation occurs, but do not further change the dynamics of 216 how rapidly feedback can be sent. Use of the RTP/AVPF profile is 217 dependent on signalling. 219 o Finally, Explicit Congestion Notification (ECN) for RTP over UDP 220 [RFC6679] can be used to provide feedback on the number of packets 221 that received an ECN Congestion Experienced (CE) mark. This RTCP 222 extension builds on the RTP/AVPF profile to allow rapid congestion 223 feedback when ECN is supported. 225 In addition to these mechanisms for providing feedback, the sender 226 can include an RTP header extension in each packet to record packet 227 transmission times. There are two methods: [RFC5450] represents the 228 transmission time in terms of a time-offset from the RTP timestamp of 229 the packet, while [RFC6051] includes an explicit NTP-format sending 230 timestamp (potentially more accurate, but a higher header overhead). 231 Accurate sending timestamps can be helpful for estimating queuing 232 delays, to get an early indication of the onset of congestion. 234 Taken together, these various mechanisms allow receivers to provide 235 feedback on the senders when congestion events occur, with varying 236 degrees of timeliness and accuracy. The key distinction is between 237 systems that use only the basic RTCP mechanisms, without RTP/AVPF 238 rapid feedback, and those that use the RTP/AVPF extensions to respond 239 to congestion more rapidly. 241 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile 243 The feedback mechanisms defined in [RFC3550] and available under the 244 RTP/AVP profile [RFC3551] are the minimum that can be assumed for a 245 baseline circuit breaker mechanism that is suitable for all unicast 246 applications of RTP. Accordingly, for an RTP circuit breaker to be 247 useful, it needs to be able to detect that an RTP flow is causing 248 excessive congestion using only basic RTCP features, without needing 249 RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports. 251 RTCP is a fundamental part of the RTP protocol, and the mechanisms 252 described here rely on the implementation of RTCP. Implementations 253 that claim to support RTP, but that do not implement RTCP, cannot use 254 the circuit breaker mechanisms described in this memo. Such 255 implementations SHOULD NOT be used on networks that might be subject 256 to congestion unless equivalent mechanisms are defined using some 257 non-RTCP feedback channel to report congestion and signal circuit 258 breaker conditions. 260 Three potential congestion signals are available from the basic RTCP 261 SR/RR packets and are reported for each synchronisation source (SSRC) 262 in the RTP session: 264 1. The sender can estimate the network round-trip time once per RTCP 265 reporting interval, based on the contents and timing of RTCP SR 266 and RR packets. 268 2. Receivers report a jitter estimate (the statistical variance of 269 the RTP data packet inter-arrival time) calculated over the RTCP 270 reporting interval. Due to the nature of the jitter calculation 271 ([RFC3550], section 6.4.4), the jitter is only meaningful for RTP 272 flows that send a single data packet for each RTP timestamp value 273 (i.e., audio flows, or video flows where each packet comprises 274 one video frame). 276 3. Receivers report the fraction of RTP data packets lost during the 277 RTCP reporting interval, and the cumulative number of RTP packets 278 lost over the entire RTP session. 280 These congestion signals limit the possible circuit breakers, since 281 they give only limited visibility into the behaviour of the network. 283 RTT estimates are widely used in congestion control algorithms, as a 284 proxy for queuing delay measures in delay-based congestion control or 285 to determine connection timeouts. RTT estimates derived from RTCP SR 286 and RR packets sent according to the RTP/AVP timing rules are too 287 infrequent to be useful though, and don't give enough information to 288 distinguish a delay change due to routing updates from queuing delay 289 caused by congestion. Accordingly, we cannot use the RTT estimate 290 alone as an RTP circuit breaker. 292 Increased jitter can be a signal of transient network congestion, but 293 in the highly aggregated form reported in RTCP RR packets, it offers 294 insufficient information to estimate the extent or persistence of 295 congestion. Jitter reports are a useful early warning of potential 296 network congestion, but provide an insufficiently strong signal to be 297 used as a circuit breaker. 299 The remaining congestion signals are the packet loss fraction and the 300 cumulative number of packets lost. If considered carefully, these 301 can be effective indicators that congestion is occurring in networks 302 where packet loss is primarily due to queue overflows, although loss 303 caused by non-congestive packet corruption can distort the result in 304 some networks. TCP congestion control [RFC5681] intentionally tries 305 to fill the router queues, and uses the resulting packet loss as 306 congestion feedback. An RTP flow competing with TCP traffic will 307 therefore expect to see a non-zero packet loss fraction that has to 308 be related to TCP dynamics to estimate available capacity. This 309 behaviour of TCP is reflected in the congestion circuit breaker 310 below, and will affect the design of any RTP congestion control 311 protocol. 313 Two packet loss regimes can be observed: 1) RTCP RR packets show a 314 non-zero packet loss fraction, while the extended highest sequence 315 number received continues to increment; and 2) RR packets show a loss 316 fraction of zero, but the extended highest sequence number received 317 does not increment even though the sender has been transmitting RTP 318 data packets. The former corresponds to the TCP congestion avoidance 319 state, and indicates a congested path that is still delivering data; 320 the latter corresponds to a TCP timeout, and is most likely due to a 321 path failure. A third condition is that data is being sent but no 322 RTCP feedback is received at all, corresponding to a failure of the 323 reverse path. We derive circuit breaker conditions for these loss 324 regimes in the following. 326 4.1. RTP/AVP Circuit Breaker #1: Media Timeout 328 If RTP data packets are being sent, but the RTCP SR or RR packets 329 reporting on that SSRC indicate a non-increasing extended highest 330 sequence number received, this is an indication that those RTP data 331 packets are not reaching the receiver. This could be a short-term 332 issue affecting only a few packets, perhaps caused by a slow-to-open 333 firewall or a transient connectivity problem, but if the issue 334 persists, it is a sign of a more ongoing and significant problem. 335 Accordingly, if a sender of RTP data packets receives CB_INTERVAL or 336 more consecutive RTCP SR or RR packets from the same receiver (see 337 Section 4.5), and those packets correspond to its transmission and 338 have a non-increasing extended highest sequence number received 339 field, then that sender SHOULD cease transmission (see Section 4.6). 340 The extended highest sequence number received field is non-increasing 341 if the sender receives at least CB_INTERVAL consecutive RTCP SR or RR 342 packets that report the same value for this field, but it has sent 343 RTP data packets, at a rate of at least one per RTT, that would have 344 caused an increase in the reported value if they had reached the 345 receiver. 347 The rationale for waiting for CB_INTERVAL or more consecutive RTCP 348 packets with a non-increasing extended highest sequence number is to 349 give enough time for transient reception problems to resolve 350 themselves, but to stop problem flows quickly enough to avoid causing 351 serious ongoing network congestion. A single RTCP report showing no 352 reception could be caused by a transient fault, and so will not cease 353 transmission. Waiting for more than CB_INTERVAL consecutive RTCP 354 reports before stopping a flow might avoid some false positives, but 355 could lead to problematic flows running for a long time period 356 (potentially tens of seconds, depending on the RTCP reporting 357 interval) before being cut off. Equally, an application that sends 358 few packets when the packet loss rate is high runs the risk that the 359 media timeout circuit breaker triggers inadvertently. The chosen 360 timeout interval is a trade-off between these extremes. 362 The rationale for enforcing a minimum sending rate below which the 363 media timeout circuit breaker will not trigger is to avoid spurious 364 circuit breaker triggers when the number of packets sent per RTCP 365 reporting interval is small (e.g., a telephony application sends only 366 two RTP comfort noise packets during a five second RTCP reporting 367 interval, and both are lost; this is 100% packet loss, but it seems 368 extreme to terminate the RTP session). The one packet per RTT bound 369 derives from [RFC5405]. 371 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout 372 In addition to media timeouts, as were discussed in Section 4.1, an 373 RTP session has the possibility of an RTCP timeout. This can occur 374 when RTP data packets are being sent, but there are no RTCP reports 375 returned from the receiver. This is either due to a failure of the 376 receiver to send RTCP reports, or a failure of the return path that 377 is preventing those RTCP reporting from being delivered. In either 378 case, it is not safe to continue transmission, since the sender has 379 no way of knowing if it is causing congestion. Accordingly, an RTP 380 sender that has not received any RTCP SR or RTCP RR packets reporting 381 on the SSRC it is using for three or more of its RTCP reporting 382 intervals SHOULD cease transmission (see Section 4.6). When 383 calculating the timeout, the deterministic RTCP reporting interval, 384 Td, without the randomization factor, and using the fixed minimum 385 interval of Tmin=5 seconds, MUST be used. The rationale for this 386 choice of timeout is as described in Section 6.2 of [RFC3550] ("so 387 that implementations which do not use the reduced value for 388 transmitting RTCP packets are not timed out by other participants 389 prematurely"), as updated by Section 6.1.4 of 390 [I-D.ietf-avtcore-rtp-multi-stream] to account for the use of the RTP 391 /AVPF profile [RFC4585] or the RTP/SAVPF profile [RFC5124]. 393 To reduce the risk of premature timeout, implementations SHOULD NOT 394 configure the RTCP bandwidth such that Td is larger than 5 seconds. 395 Similarly, implementations that use the RTP/AVPF profile [RFC4585] or 396 the RTP/SAVPF profile [RFC5124] SHOULD NOT configure T_rr_interval to 397 values larger than 4 seconds (the reduced limit for T_rr_interval 398 follows Section 6.1.3 of [I-D.ietf-avtcore-rtp-multi-stream]). 400 The choice of three RTCP reporting intervals as the timeout is made 401 following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that 402 participants in an RTP session will timeout and remove an RTP sender 403 from the list of active RTP senders if no RTP data packets have been 404 received from that RTP sender within the last two RTCP reporting 405 intervals. Using a timeout of three RTCP reporting intervals is 406 therefore large enough that the other participants will have timed 407 out the sender if a network problem stops the data packets it is 408 sending from reaching the receivers, even allowing for loss of some 409 RTCP packets. 411 If a sender is transmitting a large number of RTP media streams, such 412 that the corresponding RTCP SR or RR packets are too large to fit 413 into the network MTU, the receiver will generate RTCP SR or RR 414 packets in a round-robin manner. In this case, the sender SHOULD 415 treat receipt of an RTCP SR or RR packet corresponding to any SSRC it 416 sent on the same 5-tuple of source and destination IP address, port, 417 and protocol, as an indication that the receiver and return path are 418 working, preventing the RTCP timeout circuit breaker from triggering. 420 4.3. RTP/AVP Circuit Breaker #3: Congestion 422 If RTP data packets are being sent, and the corresponding RTCP SR or 423 RR packets show non-zero packet loss fraction and increasing extended 424 highest sequence number received, then those RTP data packets are 425 arriving at the receiver, but some degree of congestion is occurring. 426 The RTP/AVP profile [RFC3551] states that: 428 If best-effort service is being used, RTP receivers SHOULD monitor 429 packet loss to ensure that the packet loss rate is within 430 acceptable parameters. Packet loss is considered acceptable if a 431 TCP flow across the same network path and experiencing the same 432 network conditions would achieve an average throughput, measured 433 on a reasonable time scale, that is not less than the RTP flow is 434 achieving. This condition can be satisfied by implementing 435 congestion control mechanisms to adapt the transmission rate (or 436 the number of layers subscribed for a layered multicast session), 437 or by arranging for a receiver to leave the session if the loss 438 rate is unacceptably high. 440 The comparison to TCP cannot be specified exactly, but is intended 441 as an "order-of-magnitude" comparison in time scale and 442 throughput. The time scale on which TCP throughput is measured is 443 the round-trip time of the connection. In essence, this 444 requirement states that it is not acceptable to deploy an 445 application (using RTP or any other transport protocol) on the 446 best-effort Internet which consumes bandwidth arbitrarily and does 447 not compete fairly with TCP within an order of magnitude. 449 The phase "order of magnitude" in the above means within a factor of 450 ten, approximately. In order to implement this, it is necessary to 451 estimate the throughput a TCP connection would achieve over the path. 452 For a long-lived TCP Reno connection, it has been shown that the TCP 453 throughput can be estimated using the following equation [Padhye]: 455 s 456 X = -------------------------------------------------------------- 457 R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2))) 459 where: 461 X is the transmit rate in bytes/second. 463 s is the packet size in bytes. If data packets vary in size, then 464 the average size is to be used. 466 R is the round trip time in seconds. 468 p is the loss event rate, between 0 and 1.0, of the number of loss 469 events as a fraction of the number of packets transmitted. 471 t_RTO is the TCP retransmission timeout value in seconds, generally 472 approximated by setting t_RTO = 4*R. 474 b is the number of packets that are acknowledged by a single TCP 475 acknowledgement; [RFC3448] recommends the use of b=1 since many 476 TCP implementations do not use delayed acknowledgements. 478 This is the same approach to estimated TCP throughput that is used in 479 [RFC3448]. Under conditions of low packet loss the second term on 480 the denominator is small, so this formula can be approximated with 481 reasonable accuracy as follows [Mathis]: 483 s 484 X = ----------------- 485 R * sqrt(2*b*p/3) 487 It is RECOMMENDED that this simplified throughout equation be used, 488 since the reduction in accuracy is small, and it is much simpler to 489 calculate than the full equation. Measurements have shown that the 490 simplified TCP throughput equation is effective as an RTP circuit 491 breaker for multimedia flows sent to hosts on residential networks 492 using ADSL and cable modem links [Singh]. The data shows that the 493 full TCP throughput equation tends to be more sensitive to packet 494 loss and triggers the RTP circuit breaker earlier than the simplified 495 equation. Implementations that desire this extra sensitivity MAY use 496 the full TCP throughput equation in the RTP circuit breaker. Initial 497 measurements in LTE networks have shown that the extra sensitivity is 498 helpful in that environment, with the full TCP throughput equation 499 giving a more balanced circuit breaker response than the simplified 500 TCP equation [Sarker]; other networks might see similar behaviour. 502 No matter what TCP throughput equation is chosen, two parameters need 503 to be estimated and reported to the sender in order to calculate the 504 throughput: the round trip time, R, and the loss event rate, p (the 505 packet size, s, is known to the sender). The round trip time can be 506 estimated from RTCP SR and RR packets. This is done too infrequently 507 for accurate statistics, but is the best that can be done with the 508 standard RTCP mechanisms. 510 Report blocks in RTCP SR or RR packets contain the packet loss 511 fraction, rather than the loss event rate, so p cannot be reported 512 (TCP typically treats the loss of multiple packets within a single 513 RTT as one loss event, but RTCP RR packets report the overall 514 fraction of packets lost, and does not report when the packet losses 515 occurred). Using the loss fraction in place of the loss event rate 516 can overestimate the loss. We believe that this overestimate will 517 not be significant, given that we are only interested in order of 518 magnitude comparison ([Floyd] section 3.2.1 shows that the difference 519 is small for steady-state conditions and random loss, but using the 520 loss fraction is more conservative in the case of bursty loss). 522 The congestion circuit breaker is therefore: when a sender that is 523 transmitting more than one RTP packet per RTT receives an RTCP SR or 524 RR packet that contains a report block for an SSRC it is using, the 525 sender MUST record the value of the fraction lost field in the report 526 block and the time since the last report block was received for that 527 SSRC. If more than CB_INTERVAL (see Section 4.5) report blocks have 528 been received for that SSRC, the sender MUST calculate the average 529 fraction lost over the last CB_INTERVAL reporting intervals, and then 530 estimate the TCP throughput that would be achieved over the path 531 using the chosen TCP throughput equation and the measured values of 532 the round-trip time, R, the loss event rate, p (as approximated by 533 the average fraction lost), and the packet size, s. This estimate of 534 the TCP throughput is then compared with the actual sending rate. If 535 the actual sending rate is more than ten times the TCP throughput 536 estimate, then the congestion circuit breaker is triggered. 538 The average fraction lost is calculated based on the sum, over the 539 last CB_INTERVAL reporting intervals, of the fraction lost in each 540 reporting interval multiplied by the duration of the corresponding 541 reporting interval, divided by the total duration of the last 542 CB_INTERVAL reporting intervals. 544 The rationale for enforcing a minimum sending rate below which the 545 congestion circuit breaker will not trigger is to avoid spurious 546 circuit breaker triggers when the number of packets sent per RTCP 547 reporting interval is small, and hence the fraction lost samples are 548 subject to measurement artefacts. The one packet per RTT bound 549 derives from [RFC5405]. 551 When the congestion circuit breaker is triggered, the sender SHOULD 552 cease transmission (see Section 4.6). However, if the sender is able 553 to reduce its sending rate by a factor of (approximately) ten, then 554 it MAY first reduce its sending rate by this factor (or some larger 555 amount) to see if that resolves the congestion. If the sending rate 556 is reduced in this way and the congestion circuit breaker triggers 557 again after the next CB_INTERVAL RTCP reporting intervals, the sender 558 MUST then cease transmission. An example of such a rate reduction 559 might be a video conferencing system that backs off to sending audio 560 only, before completely dropping the call. If such a reduction in 561 sending rate resolves the congestion problem, the sender MAY 562 gradually increase the rate at which it sends data after a reasonable 563 amount of time has passed, provided it takes care not to cause the 564 problem to recur ("reasonable" is intentionally not defined here). 566 The RTCP reporting interval of the media sender does not affect how 567 quickly congestion circuit breaker can trigger. The timing is based 568 on the RTCP reporting interval of the receiver that generates the SR/ 569 RR packets from which the loss rate and RTT estimate are derived 570 (note that RTCP requires all participants in a session to have 571 similar reporting intervals, else the participant timeout rules in 572 [RFC3550] will not work, so this interval is likely similar to that 573 of the sender). If the incoming RTCP SR or RR packets are using a 574 reduced minimum RTCP reporting interval (as specified in Section 6.2 575 of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that 576 reduced RTCP reporting interval is used when determining if the 577 circuit breaker is triggered. 579 As in Section 4.1 and Section 4.2, we use CB_INTERVAL reporting 580 intervals to avoid triggering the circuit breaker on transient 581 failures. This circuit breaker is a worst-case condition, and 582 congestion control needs to be performed to keep well within this 583 bound. It is expected that the circuit breaker will only be 584 triggered if the usual congestion control fails for some reason. 586 If there are more media streams that can be reported in a single RTCP 587 SR or RR packet, or if the size of a complete RTCP SR or RR packet 588 exceeds the network MTU, then the receiver will report on a subset of 589 sources in each reporting interval, with the subsets selected round- 590 robin across multiple intervals so that all sources are eventually 591 reported [RFC3550]. When generating such round-robin RTCP reports, 592 priority SHOULD be given to reports on sources that have high packet 593 loss rates, to ensure that senders are aware of network congestion 594 they are causing (this is an update to [RFC3550]). 596 4.4. RTP/AVP Circuit Breaker #4: Media Usability 598 Applications that use RTP are generally tolerant to some amount of 599 packet loss. How much packet loss can be tolerated will depend on 600 the application, media codec, and the amount of error correction and 601 packet loss concealment that is applied. There is an upper bound on 602 the amount of loss can be corrected, however, beyond which the media 603 becomes unusable. Similarly, many applications have some upper bound 604 on the media capture to play-out latency that can be tolerated before 605 the application becomes unusable. The latency bound will depend on 606 the application, but typical values can range from the order of a few 607 hundred milliseconds for voice telephony and interactive conferencing 608 applications, up to several seconds for some video-on-demand systems. 610 As a final circuit breaker, RTP senders SHOULD monitor the reported 611 packet loss and delay to estimate whether the media is likely to be 612 suitable for the intended purpose. If the packet loss rate and/or 613 latency is such that the media has become unusable, and has remained 614 unusable for a significant time period, then the application SHOULD 615 cease transmission. Similarly, receivers SHOULD monitor the quality 616 of the media they receive, and if the quality is unusable for a 617 significant time period, they SHOULD terminate the session. This 618 memo intentionally does not define a bound on the packet loss rate or 619 latency that will result in unusable media, nor does it specify what 620 time period is deemed significant, as these are highly application 621 dependent. 623 Sending media that suffers from such high packet loss or latency that 624 it is unusable at the receiver is both wasteful of resources, and of 625 no benefit to the user of the application. It also is highly likely 626 to be congesting the network, and disrupting other applications. As 627 such, the congestion circuit breaker will almost certainly trigger to 628 stop flows where the media would be unusable due to high packet loss 629 or latency. However, in pathological scenarios where the congestion 630 circuit breaker does not stop the flow, it is desirable that the RTP 631 application cease sending useless traffic. The role of the media 632 usability circuit breaker is to protect the network in such cases. 634 4.5. Choice of Circuit Breaker Interval 636 The CB_INTERVAL parameter determines the number of consecutive RTCP 637 reporting intervals that need to suffer congestion before the media 638 timeout circuit breaker (see Section 4.1) or the congestion circuit 639 breaker (see Section 4.3) triggers. It determines the sensitivity 640 and responsiveness of these circuit breakers. 642 The CB_INTERVAL parameter is set to min(floor(3+(2.5/Td)), 30) RTCP 643 reporting intervals, where Td is the deterministic calculated RTCP 644 interval described in section 6.3.1 of [RFC3550]. This expression 645 gives an CB_INTERVAL that varies as follows: 647 Td | CB_INTERVAL | Time to trigger 648 --------------+------------------------------+----------------- 649 0.016 seconds | 30 RTCP reporting intervals | 0.48 seconds 650 0.033 seconds | 30 RTCP reporting intervals | 0.99 seconds 651 0.1 seconds | 28 RTCP reporting intervals | 2.8 seconds 652 0.5 seconds | 8 RTCP reporting intervals | 4.0 seconds 653 1.0 seconds | 5 RTCP reporting intervals | 5.5 seconds 654 2.0 seconds | 4 RTCP reporting intervals | 8.5 seconds 655 5.0 seconds | 5 RTCP reporting intervals | 17.5 seconds 656 10.0 seconds | 3 RTCP reporting intervals | 32.5 seconds 658 If the RTP/AVPF profile [RFC4585] or the RTP/SAVPF [RFC5124] is used, 659 and the T_rr_interval parameter is used to reduce the frequency of 660 regular RTCP reports, then the value Td in the above expression for 661 the CB_INTERVAL parameter MUST be replaced by T_rr_interval. 663 The CB_INTERVAL parameter is calculated on joining the session, and 664 recalculated on receipt of each RTCP packet, after checking whether 665 the media timeout circuit breaker or the congestion circuit breaker 666 has been triggered. 668 To ensure a timely response to persistent congestion, implementations 669 SHOULD NOT configure the RTCP bandwidth such that Td is larger than 5 670 seconds. Similarly, implementations that use the RTP/AVPF profile 671 [RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure 672 T_rr_interval to values larger than 4 seconds (the reduced limit for 673 T_rr_interval follows Section 6.1.3 of 674 [I-D.ietf-avtcore-rtp-multi-stream]). 676 Rationale: If the CB_INTERVAL was always set to the same number of 677 RTCP reporting intervals, this would cause higher rate RTP sessions 678 to trigger the RTP circuit breaker after a shorter time interval than 679 lower rate sessions, because the RTCP reporting interval scales based 680 on the RTP session bandwidth. This is felt to penalise high rate RTP 681 sessions too aggressively. Conversely, scaling CB_INTERVAL according 682 to the inverse of the RTCP reporting interval, so the RTP circuit 683 breaker triggers after a constant time interval, doesn't sufficiently 684 protect the network from congestion caused by high-rate flows. The 685 chosen expression for CB_INTERVAL seeks a balance between these two 686 extremes. It causes higher rate RTP sessions subject to persistent 687 congestion to trigger the RTP circuit breaker after a shorter time 688 interval than do lower rate RTP sessions, while also making the RTP 689 circuit breaker for such sessions less sensitive by requiring the 690 congestion to persist for longer numbers of RTCP reporting intervals. 692 4.6. Ceasing Transmission 694 What it means to cease transmission depends on the application, but 695 the intention is that the application will stop sending RTP data 696 packets to a particular destination 3-tuple (transport protocol, 697 destination port, IP address), until the user makes an explicit 698 attempt to restart the call. It is important that a human user is 699 involved in the decision to try to restart the call, since that user 700 will eventually give up if the calls repeatedly trigger the circuit 701 breaker. This will help avoid problems with automatic redial systems 702 from congesting the network. Accordingly, RTP flows halted by the 703 circuit breaker SHOULD NOT be restarted automatically unless the 704 sender has received information that the congestion has dissipated. 706 It is recognised that the RTP implementation in some systems might 707 not be able to determine if a call set-up request was initiated by a 708 human user, or automatically by some scripted higher-level component 709 of the system. These implementations SHOULD rate limit attempts to 710 restart a call to the same destination 3-tuple as used by a previous 711 call that was recently halted by the circuit breaker. The chosen 712 rate limit ought to not exceed the rate at which an annoyed human 713 caller might redial a misbehaving phone. 715 5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles 717 Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF) 718 [RFC4585] allows receivers to send early RTCP reports in some cases, 719 to inform the sender about particular events in the media stream. 720 There are several use cases for such early RTCP reports, including 721 providing rapid feedback to a sender about the onset of congestion. 722 The RTP/SAVPF Profile [RFC5124] is a secure variant of the RTP/AVPF 723 profile, that is treated the same in the context of the RTP circuit 724 breaker. These feedback profiles are often used with non-compound 725 RTCP reports [RFC5506] to reduce the reporting overhead. 727 Receiving rapid feedback about congestion events potentially allows 728 congestion control algorithms to be more responsive, and to better 729 adapt the media transmission to the limitations of the network. It 730 is expected that many RTP congestion control algorithms will adopt 731 the RTP/AVPF profile or the RTP/SAVPF profile for this reason, 732 defining new transport layer feedback reports that suit their 733 requirements. Since these reports are not yet defined, and likely 734 very specific to the details of the congestion control algorithm 735 chosen, they cannot be used as part of the generic RTP circuit 736 breaker. 738 Reduced-size RTCP reports sent under the RTP/AVPF early feedback 739 rules that do not contain an RTCP SR or RR packet MUST be ignored by 740 the congestion circuit breaker (they do not contain the information 741 needed by the congestion circuit breaker algorithm), but MUST be 742 counted as received packets for the RTCP timeout circuit breaker. 743 Reduced-size RTCP reports sent under the RTP/AVPF early feedback 744 rules that contain RTCP SR or RR packets MUST be processed by the 745 congestion circuit breaker as if they were sent as regular RTCP 746 reports, and counted towards the circuit breaker conditions specified 747 in Section 4 of this memo. This will potentially make the RTP 748 circuit breaker trigger earlier than it would if the RTP/AVPF profile 749 was not used. 751 When using ECN with RTP (see Section 8), early RTCP feedback packets 752 can contain ECN feedback reports. The count of ECN-CE marked packets 753 contained in those ECN feedback reports is counted towards the number 754 of lost packets reported if the ECN Feedback Report report is sent in 755 an compound RTCP packet along with an RTCP SR/RR report packet. 756 Reports of ECN-CE packets sent as reduced-size RTCP ECN feedback 757 packets without an RTCP SR/RR packet MUST be ignored. 759 These rules are intended to allow the use of low-overhead RTP/AVPF 760 feedback for generic NACK messages without triggering the RTP circuit 761 breaker. This is expected to make such feedback suitable for RTP 762 congestion control algorithms that need to quickly report loss events 763 in between regular RTCP reports. The reaction to reduced-size RTCP 764 SR/RR packets is to allow such algorithms to send feedback that can 765 trigger the circuit breaker, when desired. 767 The RTP/AVPF and RTP/SAVPF profiles include the T_rr_interval 768 parameter that can be used to adjust the regular RTCP reporting 769 interval. The use of the T_rr_interval parameter changes the 770 behaviour of the RTP circuit breaker, as described in Section 4. 772 6. Impact of RTCP Extended Reports (XR) 774 RTCP Extended Report (XR) blocks provide additional reception quality 775 metrics, but do not change the RTCP timing rules. Some of the RTCP 776 XR blocks provide information that might be useful for congestion 777 control purposes, others provided non-congestion-related metrics. 778 With the exception of RTCP XR ECN Summary Reports (see Section 8), 779 the presence of RTCP XR blocks in a compound RTCP packet does not 780 affect the RTP circuit breaker algorithm. For consistency and ease 781 of implementation, only the reception report blocks contained in RTCP 782 SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets, 783 are used by the RTP circuit breaker algorithm. 785 7. Impact of RTCP Reporting Groups 787 An optimisation for grouping RTCP reception statistics and other 788 feedback in RTP sessions with large numbers of participants is given 789 in [I-D.ietf-avtcore-rtp-multi-stream-optimisation]. This allows one 790 SSRC to act as a representative that sends reports on behalf of other 791 SSRCs that are co-located in the same endpoint and see identical 792 reception quality. When running the circuit breaker algorithms, an 793 endpoint MUST treat a reception report from the representative of the 794 reporting group as if a reception report was received from all 795 members of that group. 797 8. Impact of Explicit Congestion Notification (ECN) 799 The use of ECN for RTP flows does not affect the media timeout RTP 800 circuit breaker (Section 4.1) or the RTCP timeout circuit breaker 801 (Section 4.2), since these are both connectivity checks that simply 802 determinate if any packets are being received. 804 ECN-CE marked packets SHOULD be treated as if it were lost for the 805 purposes of congestion control, when determining the optimal media 806 sending rate for an RTP flow. If an RTP sender has negotiated ECN 807 support for an RTP session, and has successfully initiated ECN use on 808 the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD 809 be treated as if they were lost when calculating if the congestion- 810 based RTP circuit breaker (Section 4.3) has been met. The count of 811 ECN-CE marked RTP packets is returned in RTCP XR ECN summary report 812 packets if support for ECN has been initiated for an RTP session. 814 9. Impact of Bundled Media and Layered Coding 816 The RTP circuit breaker operates on a per-RTP session basis. An RTP 817 sender that participates in several RTP sessions MUST treat each RTP 818 session independently with regards to the RTP circuit breaker. 820 An RTP sender can generate several media streams within a single RTP 821 session, with each stream using a different SSRC. This can happen if 822 bundled media are in use, when using simulcast, or when using layered 823 media coding. By default, each SSRC will be treated independently by 824 the RTP circuit breaker. However, the sender MAY choose to treat the 825 flows (or a subset thereof) as a group, such that a circuit breaker 826 trigger for one flow applies to the group of flows as a whole, and 827 either causes the entire group to cease transmission, or the sending 828 rate of the group to reduce by a factor of ten, depending on the RTP 829 circuit breaker triggered. Grouping flows in this way is expected to 830 be especially useful for layered flows sent using multiple SSRCs, as 831 it allows the layered flow to react as a whole, ceasing transmission 832 on the enhancement layers first to reduce sending rate if necessary, 833 rather than treating each layer independently. 835 10. Security Considerations 837 The security considerations of [RFC3550] apply. 839 If the RTP/AVPF profile is used to provide rapid RTCP feedback, the 840 security considerations of [RFC4585] apply. If ECN feedback for RTP 841 over UDP/IP is used, the security considerations of [RFC6679] apply. 843 If non-authenticated RTCP reports are used, an on-path attacker can 844 trivially generate fake RTCP packets that indicate high packet loss 845 rates, causing the circuit breaker to trigger and disrupting an RTP 846 session. This is somewhat more difficult for an off-path attacker, 847 due to the need to guess the randomly chosen RTP SSRC value and the 848 RTP sequence number. This attack can be avoided if RTCP packets are 849 authenticated; authentication options are discussed in [RFC7201]. 851 Timely operation of the RTP circuit breaker depends on the choice of 852 RTCP reporting interval. If the receiver has a reporting interval 853 that is overly long, then the responsiveness of the circuit breaker 854 decreases. In the limit, the RTP circuit breaker can be disabled for 855 all practical purposes by configuring an RTCP reporting interval that 856 is many minutes duration. This issue is not specific to the circuit 857 breaker: long RTCP reporting intervals also prevent reception quality 858 reports, feedback messages, codec control messages, etc., from being 859 used. Implementations are expected to impose an upper limit on the 860 RTCP reporting interval they are willing to negotiate (based on the 861 session bandwidth and RTCP bandwidth fraction) when using the RTP 862 circuit breaker, as discussed in Section 4.5. 864 11. IANA Considerations 866 There are no actions for IANA. 868 12. Acknowledgements 870 The authors would like to thank Bernard Aboba, Harald Alvestrand, 871 Gorry Fairhurst, Nazila Fough, Kevin Gross, Cullen Jennings, Randell 872 Jesup, Jonathan Lennox, Matt Mathis, Stephen McQuistin, Eric 873 Rescorla, Abheek Saha, Fabio Verdicchio, and Magnus Westerlund for 874 their valuable feedback. 876 13. References 878 13.1. Normative References 880 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 881 Requirement Levels", BCP 14, RFC 2119, March 1997. 883 [RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP 884 Friendly Rate Control (TFRC): Protocol Specification", RFC 885 3448, January 2003. 887 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 888 Jacobson, "RTP: A Transport Protocol for Real-Time 889 Applications", STD 64, RFC 3550, July 2003. 891 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 892 Video Conferences with Minimal Control", STD 65, RFC 3551, 893 July 2003. 895 [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control 896 Protocol Extended Reports (RTCP XR)", RFC 3611, November 897 2003. 899 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 900 "Extended RTP Profile for Real-time Transport Control 901 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 902 2006. 904 13.2. Informative References 906 [Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer, 907 "Equation-Based Congestion Control for Unicast 908 Applications", Proceedings of the ACM SIGCOMM conference, 909 2000, DOI 10.1145/347059.347397, August 2000. 911 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] 912 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 913 "Sending Multiple Media Streams in a Single RTP Session: 914 Grouping RTCP Reception Statistics and Other Feedback", 915 draft-ietf-avtcore-rtp-multi-stream-optimisation-05 (work 916 in progress), February 2015. 918 [I-D.ietf-avtcore-rtp-multi-stream] 919 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 920 "Sending Multiple Media Streams in a Single RTP Session", 921 draft-ietf-avtcore-rtp-multi-stream-06 (work in progress), 922 October 2014. 924 [Mathis] Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The 925 macroscopic behavior of the TCP congestion avoidance 926 algorithm", ACM SIGCOMM Computer Communication Review 927 27(3), DOI 10.1145/263932.264023, July 1997. 929 [Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose, 930 "Modeling TCP Throughput: A Simple Model and its Empirical 931 Validation", Proceedings of the ACM SIGCOMM conference, 932 1998, DOI 10.1145/285237.285291, August 1998. 934 [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition 935 of Explicit Congestion Notification (ECN) to IP", RFC 936 3168, September 2001. 938 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 939 "Codec Control Messages in the RTP Audio-Visual Profile 940 with Feedback (AVPF)", RFC 5104, February 2008. 942 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for 943 Real-time Transport Control Protocol (RTCP)-Based Feedback 944 (RTP/SAVPF)", RFC 5124, February 2008. 946 [RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines 947 for Application Designers", BCP 145, RFC 5405, November 948 2008. 950 [RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in 951 RTP Streams", RFC 5450, March 2009. 953 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size 954 Real-Time Transport Control Protocol (RTCP): Opportunities 955 and Consequences", RFC 5506, April 2009. 957 [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion 958 Control", RFC 5681, September 2009. 960 [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP 961 Flows", RFC 6051, November 2010. 963 [RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., 964 and K. Carlberg, "Explicit Congestion Notification (ECN) 965 for RTP over UDP", RFC 6679, August 2012. 967 [RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended 968 Report (XR) Block for Packet Delay Variation Metric 969 Reporting", RFC 6798, November 2012. 971 [RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol 972 (RTCP) Extended Report (XR) Block for Delay Metric 973 Reporting", RFC 6843, January 2013. 975 [RFC6958] Clark, A., Zhang, S., Zhao, J., and Q. Wu, "RTP Control 976 Protocol (RTCP) Extended Report (XR) Block for Burst/Gap 977 Loss Metric Reporting", RFC 6958, May 2013. 979 [RFC7002] Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol 980 (RTCP) Extended Report (XR) Block for Discard Count Metric 981 Reporting", RFC 7002, September 2013. 983 [RFC7003] Clark, A., Huang, R., and Q. Wu, "RTP Control Protocol 984 (RTCP) Extended Report (XR) Block for Burst/Gap Discard 985 Metric Reporting", RFC 7003, September 2013. 987 [RFC7097] Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol 988 (RTCP) Extended Report (XR) for RLE of Discarded Packets", 989 RFC 7097, January 2014. 991 [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP 992 Sessions", RFC 7201, April 2014. 994 [Sarker] Sarker, Z., Singh, V., and C.S. Perkins, "An Evaluation of 995 RTP Circuit Breaker Performance on LTE Networks", 996 Proceedings of the IEEE Infocom workshop on Communication 997 and Networking Techniques for Contemporary Video, 2014, 998 April 2014. 1000 [Singh] Singh, V., McQuistin, S., Ellis, M., and C.S. Perkins, 1001 "Circuit Breakers for Multimedia Congestion Control", 1002 Proceedings of the International Packet Video Workshop, 1003 2013, DOI 10.1109/PV.2013.6691439, December 2013. 1005 Authors' Addresses 1007 Colin Perkins 1008 University of Glasgow 1009 School of Computing Science 1010 Glasgow G12 8QQ 1011 United Kingdom 1013 Email: csp@csperkins.org 1015 Varun Singh 1016 Aalto University 1017 School of Electrical Engineering 1018 Otakaari 5 A 1019 Espoo, FIN 02150 1020 Finland 1022 Email: varun@comnet.tkk.fi 1023 URI: http://www.netlab.tkk.fi/~varun/