Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
Author(s): Varun Singh, Colin Perkins
The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in the applications, then network congestion will deteriorate...
AVTCORE Working Group C. S. Perkins Internet-Draft University of Glasgow Updates: 3550 (if approved) V. Singh Intended status: Standards Track Aalto University Expires: January 16, 2014 July 15, 2013 Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions draft-ietf-avtcore-rtp-circuit-breakers-03 Abstract The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in the applications, then network congestion will deteriorate the user's multimedia experience. This document does not propose a congestion control algorithm; instead, it defines a minimal set of RTP "circuit-breakers". Circuit-breakers are conditions under which an RTP sender needs to stop transmitting media data in order to protect the network from excessive congestion. It is expected that, in the absence of severe congestion, all RTP applications running on best-effort IP networks will be able to run without triggering these circuit breakers. Any future RTP congestion control specification will be expected to operate within the constraints defined by these circuit breakers. Status of This Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." This Internet-Draft will expire on January 16, 2014. Copyright Notice Copyright (c) 2013 IETF Trust and the persons identified as the document authors. All rights reserved. Perkins & Singh Expires January 16, 2014 [Page 1] Internet-Draft RTP Circuit Breakers July 2013 This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 6 4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 7 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . 8 4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 9 4.4. Ceasing Transmission . . . . . . . . . . . . . . . . . . 12 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 12 6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 13 7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 14 8. Impact of Explicit Congestion Notification (ECN) . . . . . . 14 9. Security Considerations . . . . . . . . . . . . . . . . . . . 14 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 15 12.1. Normative References . . . . . . . . . . . . . . . . . . 15 12.2. Informative References . . . . . . . . . . . . . . . . . 15 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17 1. Introduction The Real-time Transport Protocol (RTP) [RFC3550] is widely used in voice-over-IP, video teleconferencing, and telepresence systems. Many of these systems run over best-effort UDP/IP networks, and can suffer from packet loss and increased latency if network congestion occurs. Designing effective RTP congestion control algorithms, to adapt the transmission of RTP-based media to match the available network capacity, while also maintaining the user experience, is a difficult but important problem. Many such congestion control and media adaptation algorithms have been proposed, but to date there is no consensus on the correct approach, or even that a single standard algorithm is desirable. This memo does not attempt to propose a new RTP congestion control algorithm. Rather, it proposes a minimal set of "circuit breakers"; Perkins & Singh Expires January 16, 2014 [Page 2] Internet-Draft RTP Circuit Breakers July 2013 conditions under which there is general agreement that an RTP flow is causing serious congestion, and ought to cease transmission. It is expected that future standards-track congestion control algorithms for RTP will operate within the envelope defined by this memo. 2. Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC 2119]. This interpretation of these key words applies only when written in ALL CAPS. Mixed- or lower-case uses of these key words are not to be interpreted as carrying special significance in this memo. 3. Background We consider congestion control for unicast RTP traffic flows. This is the problem of adapting the transmission of an audio/visual data flow, encapsulated within an RTP transport session, from one sender to one receiver, so that it matches the available network bandwidth. Such adaptation needs to be done in a way that limits the disruption to the user experience caused by both packet loss and excessive rate changes. Congestion control for multicast flows is outside the scope of this memo. Multicast traffic needs different solutions, since the available bandwidth estimator for a group of receivers will differ from that for a single receiver, and because multicast congestion control has to consider issues of fairness across groups of receivers that do not apply to unicast flows. Congestion control for unicast RTP traffic can be implemented in one of two places in the protocol stack. One approach is to run the RTP traffic over a congestion controlled transport protocol, for example over TCP, and to adapt the media encoding to match the dictates of the transport-layer congestion control algorithm. This is safe for the network, but can be suboptimal for the media quality unless the transport protocol is designed to support real-time media flows. We do not consider this class of applications further in this memo, as their network safety is guaranteed by the underlying transport. Alternatively, RTP flows can be run over a non-congestion controlled transport protocol, for example UDP, performing rate adaptation at the application layer based on RTP Control Protocol (RTCP) feedback. With a well-designed, network-aware, application, this allows highly effective media quality adaptation, but there is potential to disrupt the network's operation if the application does not adapt its sending rate in a timely and effective manner. We consider this class of applications in this memo. Perkins & Singh Expires January 16, 2014 [Page 3] Internet-Draft RTP Circuit Breakers July 2013 Congestion control relies on monitoring the delivery of a media flow, and responding to adapt the transmission of that flow when there are signs that the network path is congested. Network congestion can be detected in one of three ways: 1) a receiver can infer the onset of congestion by observing an increase in one-way delay caused by queue build-up within the network; 2) if Explicit Congestion Notification (ECN) [RFC3168] is supported, the network can signal the presence of congestion by marking packets using ECN Congestion Experienced (CE) marks; or 3) in the extreme case, congestion will cause packet loss that can be detected by observing a gap in the received RTP sequence numbers. Once the onset of congestion is observed, the receiver has to send feedback to the sender to indicate that the transmission rate needs to be reduced. How the sender reduces the transmission rate is highly dependent on the media codec being used, and is outside the scope of this memo. There are several ways in which a receiver can send feedback to a media sender within the RTP framework: o The base RTP specification [RFC3550] defines RTCP Reception Report (RR) packets to convey reception quality feedback information, and Sender Report (SR) packets to convey information about the media transmission. RTCP SR packets contain data that can be used to reconstruct media timing at a receiver, along with a count of the total number of octets and packets sent. RTCP RR packets report on the fraction of packets lost in the last reporting interval, the cumulative number of packets lost, the highest sequence number received, and the inter-arrival jitter. The RTCP RR packets also contain timing information that allows the sender to estimate the network round trip time (RTT) to the receivers. RTCP reports are sent periodically, with the reporting interval being determined by the number of SSRCs used in the session and a configured session bandwidth estimate (the number of SSRCs used is usually two in a unicast session, one for each participant, but can be greater if the participants send multiple media streams). The interval between reports sent from each receiver tends to be on the order of a few seconds on average, and it is randomised to avoid synchronisation of reports from multiple receivers. RTCP RR packets allow a receiver to report ongoing network congestion to the sender. However, if a receiver detects the onset of congestion partway through a reporting interval, the base RTP specification contains no provision for sending the RTCP RR packet early, and the receiver has to wait until the next scheduled reporting interval. o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more complex and sophisticated reception quality metrics, but do not change the RTCP timing rules. RTCP extended reports of potential Perkins & Singh Expires January 16, 2014 [Page 4] Internet-Draft RTP Circuit Breakers July 2013 interest for congestion control purposes are the extended packet loss, discard, and burst metrics [RFC3611], [I-D.ietf-xrblock-rtcp-xr-discard], [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics], [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard], [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]; and the extended delay metrics [RFC6843], [RFC6798]. Other RTCP Extended Reports that could be helpful for congestion control purposes might be developed in future. o Rapid feedback about the occurrence of congestion events can be achieved using the Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF) [RFC4585] in place of the more common RTP/AVP profile [RFC3551]. This modifies the RTCP timing rules to allow RTCP reports to be sent early, in some cases immediately, provided the average RTCP reporting interval remains unchanged. It also defines new transport-layer feedback messages, including negative acknowledgements (NACKs), that can be used to report on specific congestion events. The use of the RTP/AVPF profile is dependent on signalling, but is otherwise generally backwards compatible with the RTP/AVP profile, as it keeps the same average RTCP reporting interval as the base RTP specification. The RTP Codec Control Messages [RFC5104] extend the RTP/AVPF profile with additional feedback messages that can be used to influence that way in which rate adaptation occurs. The dynamics of how rapidly feedback can be sent are unchanged. o Finally, Explicit Congestion Notification (ECN) for RTP over UDP [RFC6679] can be used to provide feedback on the number of packets that received an ECN Congestion Experienced (CE) mark. This RTCP extension builds on the RTP/AVPF profile to allow rapid congestion feedback when ECN is supported. In addition to these mechanisms for providing feedback, the sender can include an RTP header extension in each packet to record packet transmission times. There are two methods: [RFC5450] represents the transmission time in terms of a time-offset from the RTP timestamp of the packet, while [RFC6051] includes an explicit NTP-format sending timestamp (potentially more accurate, but a higher header overhead). Accurate sending timestamps can be helpful for estimating queuing delays, to get an early indication of the onset of congestion. Taken together, these various mechanisms allow receivers to provide feedback on the senders when congestion events occur, with varying degrees of timeliness and accuracy. The key distinction is between systems that use only the basic RTCP mechanisms, without RTP/AVPF rapid feedback, and those that use the RTP/AVPF extensions to respond to congestion more rapidly. Perkins & Singh Expires January 16, 2014 [Page 5] Internet-Draft RTP Circuit Breakers July 2013 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile The feedback mechanisms defined in [RFC3550] and available under the RTP/AVP profile [RFC3551] are the minimum that can be assumed for a baseline circuit breaker mechanism that is suitable for all unicast applications of RTP. Accordingly, for an RTP circuit breaker to be useful, it needs to be able to detect that an RTP flow is causing excessive congestion using only basic RTCP features, without needing RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports. RTCP is a fundamental part of the RTP protocol, and the mechanisms described here rely on the implementation of RTCP. Implementations which claim to support RTP, but that do not implement RTCP, cannot use the circuit breaker mechanisms described in this memo. Such implementations SHOULD NOT be used on networks that might be subject to congestion unless equivalent mechanisms are defined using some non-RTCP feedback channel to report congestion and signal circuit breaker conditions. Three potential congestion signals are available from the basic RTCP SR/RR packets and are reported for each synchronisation source (SSRC) in the RTP session: 1. The sender can estimate the network round-trip time once per RTCP reporting interval, based on the contents and timing of RTCP SR and RR packets. 2. Receivers report a jitter estimate (the statistical variance of the RTP data packet inter-arrival time) calculated over the RTCP reporting interval. Due to the nature of the jitter calculation ([RFC3550], section 6.4.4), the jitter is only meaningful for RTP flows that send a single data packet for each RTP timestamp value (i.e., audio flows, or video flows where each packet comprises one video frame). 3. Receivers report the fraction of RTP data packets lost during the RTCP reporting interval, and the cumulative number of RTP packets lost over the entire RTP session. These congestion signals limit the possible circuit breakers, since they give only limited visibility into the behaviour of the network. RTT estimates are widely used in congestion control algorithms, as a proxy for queuing delay measures in delay-based congestion control or to determine connection timeouts. RTT estimates derived from RTCP SR and RR packets sent according to the RTP/AVP timing rules are far too infrequent to be useful though, and don't give enough information to distinguish a delay change due to routing updates from queuing delay Perkins & Singh Expires January 16, 2014 [Page 6] Internet-Draft RTP Circuit Breakers July 2013 caused by congestion. Accordingly, we cannot use the RTT estimate alone as an RTP circuit breaker. Increased jitter can be a signal of transient network congestion, but in the highly aggregated form reported in RTCP RR packets, it offers insufficient information to estimate the extent or persistence of congestion. Jitter reports are a useful early warning of potential network congestion, but provide an insufficiently strong signal to be used as a circuit breaker. The remaining congestion signals are the packet loss fraction and the cumulative number of packets lost. If considered carefully, these can be effective indicators that congestion is occurring in networks where packet loss is primarily due to queue overflows, although loss caused by non-congestive packet corruption can distort the result in some networks. TCP congestion control intentionally tries to fill the router queues, and uses the resulting packet loss as congestion feedback. An RTP flow competing with TCP traffic will therefore expect to see a non-zero packet loss fraction that has to be related to TCP dynamics to estimate available capacity. This behaviour of TCP is reflected in the congestion circuit breaker below, and will affect the design of any RTP congestion control protocol. Two packet loss regimes can be observed: 1) RTCP RR packets show a non-zero packet loss fraction, while the extended highest sequence number received continues to increment; and 2) RR packets show a loss fraction of zero, but the extended highest sequence number received does not increment even though the sender has been transmitting RTP data packets. The former corresponds to the TCP congestion avoidance state, and indicates a congested path that is still delivering data; the latter corresponds to a TCP timeout, and is most likely due to a path failure. A third condition is that data is being sent but no RTCP feedback is received at all, corresponding to a failure of the reverse path. We derive circuit breaker conditions for these loss regimes in the following. 4.1. RTP/AVP Circuit Breaker #1: Media Timeout If RTP data packets are being sent, but the RTCP SR or RR packets reporting on that SSRC indicate a non-increasing extended highest sequence number received, this is an indication that those RTP data packets are not reaching the receiver. This could be a short-term issue affecting only a few packets, perhaps caused by a slow-to-open firewall or a transient connectivity problem, but if the issue persists, it is a sign of a more ongoing and significant problem. Accordingly, if a sender of RTP data packets receives two or more consecutive RTCP SR or RR packets from the same receiver, and those packets correspond to its transmission and have a non-increasing Perkins & Singh Expires January 16, 2014 [Page 7] Internet-Draft RTP Circuit Breakers July 2013 extended highest sequence number received field (i.e., the sender receivers at least three RTCP SR or RR packets that report the same value in the extended highest sequence number received field for an SSRC, but the sender has sent RTP data packets for that SSRC that would have caused an increase in the reported value of the extended highest sequence number received if they had reached the receiver), then that sender SHOULD cease transmission (see Section 4.4). The reason for waiting for two or more consecutive RTCP packets with a non-increasing extended highest sequence number is to give enough time for transient reception problems to resolve themselves, but to stop problem flows quickly enough to avoid causing serious ongoing network congestion. A single RTCP report showing no reception could be caused by a transient fault, and so will not cease transmission. Waiting for more than two consecutive RTCP reports before stopping a flow might avoid some false positives, but could lead to problematic flows running for a long time period (potentially tens of seconds, depending on the RTCP reporting interval) before being cut off. 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout In addition to media timeouts, as were discussed in Section 4.1, an RTP session has the possibility of an RTCP timeout. This can occur when RTP data packets are being sent, but there are no RTCP reports returned from the receiver. This is either due to a failure of the receiver to send RTCP reports, or a failure of the return path that is preventing those RTCP reporting from being delivered. In either case, it is not safe to continue transmission, since the sender has no way of knowing if it is causing congestion. Accordingly, an RTP sender that has not received any RTCP SR or RTCP RR packets reporting on the SSRC it is using for three or more RTCP reporting intervals SHOULD cease transmission (see Section 4.4). When calculating the timeout, the fixed minimum RTCP reporting interval SHOULD be used (based on the rationale in Section 6.2 of RFC3550 [RFC3550]). The choice of three RTCP reporting intervals as the timeout is made following Section 6.3.5 of RFC3550 [RFC3550]. This specifies that participants in an RTP session will timeout and remove an RTP sender from the list of active RTP senders if no RTP data packets have been received from that RTP sender within the last two RTCP reporting intervals. Using a timeout of three RTCP reporting intervals is therefore large enough that the other participants will have timed out the sender if a network problem stops the data packets it is sending from reaching the receivers, even allowing for loss of some RTCP packets. If a sender is transmitting a large number of RTP media streams, such that the corresponding RTCP SR or RR packets are too large to fit Perkins & Singh Expires January 16, 2014 [Page 8] Internet-Draft RTP Circuit Breakers July 2013 into the network MTU, this will force the receiver to generate RTCP SR or RR packets in a round-robin manner. In this case, the sender MAY treat receipt of an RTCP SR or RR packet corresponding to an SSRC it sent using the same 5-tuple of source and destination IP address, port, and protocol, as an indication that the receiver and return path are working to prevent the RTCP timeout circuit breaker from triggering. 4.3. RTP/AVP Circuit Breaker #3: Congestion If RTP data packets are being sent, and the corresponding RTCP SR or RR packets show non-zero packet loss fraction and increasing extended highest sequence number received, then those RTP data packets are arriving at the receiver, but some degree of congestion is occurring. The RTP/AVP profile [RFC3551] states that: If best-effort service is being used, RTP receivers SHOULD monitor packet loss to ensure that the packet loss rate is within acceptable parameters. Packet loss is considered acceptable if a TCP flow across the same network path and experiencing the same network conditions would achieve an average throughput, measured on a reasonable time scale, that is not less than the RTP flow is achieving. This condition can be satisfied by implementing congestion control mechanisms to adapt the transmission rate (or the number of layers subscribed for a layered multicast session), or by arranging for a receiver to leave the session if the loss rate is unacceptably high. The comparison to TCP cannot be specified exactly, but is intended as an "order-of-magnitude" comparison in time scale and throughput. The time scale on which TCP throughput is measured is the round-trip time of the connection. In essence, this requirement states that it is not acceptable to deploy an application (using RTP or any other transport protocol) on the best-effort Internet which consumes bandwidth arbitrarily and does not compete fairly with TCP within an order of magnitude. The phase "order of magnitude" in the above means within a factor of ten, approximately. In order to implement this, it is necessary to estimate the throughput a TCP connection would achieve over the path. For a long-lived TCP Reno connection, Padhye et al. [Padhye] showed that the throughput can be estimated using the following equation: s X = -------------------------------------------------------------- R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2))) Perkins & Singh Expires January 16, 2014 [Page 9] Internet-Draft RTP Circuit Breakers July 2013 where: X is the transmit rate in bytes/second. s is the packet size in bytes. If data packets vary in size, then the average size is to be used. R is the round trip time in seconds. p is the loss event rate, between 0 and 1.0, of the number of loss events as a fraction of the number of packets transmitted. t_RTO is the TCP retransmission timeout value in seconds, approximated by setting t_RTO = 4*R. b is the number of packets acknowledged by a single TCP acknowledgement ([RFC3448] recommends the use of b=1 since many TCP implementations do not use delayed acknowledgements). This is the same approach to estimated TCP throughput that is used in [RFC3448]. Under conditions of low packet loss, this formula can be approximated as follows with reasonable accuracy: s X = --------------- R * sqrt(p*2/3) It is RECOMMENDED that this simplified throughout equation be used, since the reduction in accuracy is small, and it is much simpler to calculate than the full equation. Given this TCP equation, two parameters need to be estimated and reported to the sender in order to calculate the throughput: the round trip time, R, and the loss event rate, p (the packet size, s, is known to the sender). The round trip time can be estimated from RTCP SR and RR packets. This is done too infrequently for accurate statistics, but is the best that can be done with the standard RTCP mechanisms. Report blocks in RTCP SR or RR packets contain the packet loss fraction, rather than the loss event rate, so p cannot be reported (TCP typically treats the loss of multiple packets within a single RTT as one loss event, but RTCP RR packets report the overall fraction of packets lost, not caring about when the losses occurred). Using the loss fraction in place of the loss event rate can overestimate the loss. We believe that this overestimate will not be significant, given that we are only interested in order of magnitude Perkins & Singh Expires January 16, 2014 [Page 10] Internet-Draft RTP Circuit Breakers July 2013 comparison ([Floyd] section 3.2.1 shows that the difference is small for steady-state conditions and random loss, but using the loss fraction is more conservative in the case of bursty loss). The congestion circuit breaker is therefore: when a sender receives an RTCP SR or RR packet that contains a report block for an SSRC it is using, that sender has to check the fraction lost field in that report block to determine if there is a non-zero packet loss rate. If the fraction lost field is zero, then continue sending as normal. If the fraction lost is greater than zero, then estimate the TCP throughput using the simplified equation above, and the measured R, p (approximated by the fraction lost), and s. Compare this with the actual sending rate. If the actual sending rate is more than ten times the estimated sending rate derived from the TCP throughput equation for two consecutive RTCP reporting intervals, the sender SHOULD cease transmission (see Section 4.4). Systems that usually send at a high data rate, but that can reduce their data rate significantly (i.e., by at least a factor of ten), MAY first reduce their sending rate to this lower value to see if this resolves the congestion, but MUST then cease transmission if the problem does not resolve itself within a further two RTCP reporting intervals (see Section 4.4). An example of this might be a video conferencing system that backs off to sending audio only, before completely dropping the call. If such a reduction in sending rate resolves the congestion problem, the sender MAY gradually increase the rate at which it sends data after a reasonable amount of time has passed, provided it takes care not to cause the problem to recur ("reasonable" is intentionally not defined here). If the incoming RTCP SR or RR packets are using a reduced minimum RTCP reporting interval (as specified in Section 6.2 of RFC3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that reduced RTCP reporting interval is used when determining if the circuit breaker is triggered. The RTCP reporting interval of the media sender does not affect how quickly congestion circuit breaker can trigger. The timing is based on the RTCP reporting interval of the receiver that matters (note that RTCP requires all participants in a session to have similar reporting intervals, else the participant timeout rules in [RFC3550] will not work). As in Section 4.1, we use two reporting intervals to avoid triggering the circuit breaker on transient failures. This circuit breaker is a worst-case condition, and congestion control needs to be performed to keep well within this bound. It is expected that the circuit breaker will only be triggered if the usual congestion control fails for some reason. Perkins & Singh Expires January 16, 2014 [Page 11] Internet-Draft RTP Circuit Breakers July 2013 If there are more media streams that can be reported in a single RTCP SR or RR packet, or if the size of a complete RTCP SR or RR packet exceeds the network MTU, then the receiver will report on a subset of sources in each reporting interval, with the subsets selected round- robin across multiple intervals so that all sources are eventually reported [RFC3550]. When generating such round-robin RTCP reports, priority SHOULD be given to reports on sources that have high packet loss rates, to ensure that senders are aware of network congestion they are causing (this is an update to [RFC3550]). 4.4. Ceasing Transmission What it means to cease transmission depends on the application, but the intention is that the application will stop sending RTP data packets to a particular destination 3-tuple (transport protocol, destination port, IP address), until the user makes an explicit attempt to restart the call. It is important that a human user is involved in the decision to try to restart the call, since that user will eventually give up if the calls repeatedly trigger the circuit breaker. This will help avoid problems with automatic redial systems from congesting the network. Accordingly, RTP flows halted by the circuit breaker SHOULD NOT be restarted automatically unless the sender has received information that the congestion has dissipated. It is recognised that the RTP implementation in some systems might not be able to determine if a call set-up request was initiated by a human user, or automatically by some scripted higher-level component of the system. These implementations SHOULD rate limit attempts to restart a call to the same destination 3-tuple as used by a previous call that was recently halted by the circuit breaker. The chosen rate limit ought to not exceed the rate at which an annoyed human caller might redial a misbehaving phone. 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF) [RFC4585] allows receivers to send early RTCP reports in some cases, to inform the sender about particular events in the media stream. There are several use cases for such early RTCP reports, including providing rapid feedback to a sender about the onset of congestion. Receiving rapid feedback about congestion events potentially allows congestion control algorithms to be more responsive, and to better adapt the media transmission to the limitations of the network. It is expected that many RTP congestion control algorithms will adopt the RTP/AVPF profile for this reason, defining new transport layer feedback reports that suit their requirements. Since these reports are not yet defined, and likely very specific to the details of the Perkins & Singh Expires January 16, 2014 [Page 12] Internet-Draft RTP Circuit Breakers July 2013 congestion control algorithm chosen, they cannot be used as part of the generic RTP circuit breaker. If the extension for Reduced-Size RTCP [RFC5506] is not used, early RTCP feedback packets sent according to the RTP/AVPF profile will be compound RTCP packets that include an RTCP SR/RR packet. That RTCP SR/RR packet MUST be processed as if it were sent as a regular RTCP report and counted towards the circuit breaker conditions specified in Section 4 of this memo. This will potentially make the RTP circuit breaker fire earlier than it would if the RTP/AVPF profile was not used. Reduced-size RTCP reports sent under the RTP/AVPF early feedback rules that do not contain an RTCP SR or RR packet MUST be ignored by the RTP circuit breaker (they do not contain the information used by the circuit breaker algorithm). Reduced-size RTCP reports sent under the RTP/AVPF early feedback rules that contain RTCP SR or RR packets MUST be processed as if they were sent as regular RTCP reports, and counted towards the circuit breaker conditions specified in Section 4 of this memo. This will potentially make the RTP circuit breaker fire earlier than it would if the RTP/AVPF profile was not used. When using ECN with RTP (see Section 8), early RTCP feedback packets can contain ECN feedback reports. The count of ECN-CE marked packets contained in those ECN feedback reports is counted towards the number of lost packets reported if the ECN Feedback Report report is sent in an compound RTCP packet along with an RTCP SR/RR report packet. Reports of ECN-CE packets sent as reduced-size RTCP ECN feedback packets without an RTCP SR/RR packet MUST be ignored. These rules are intended to allow the use of low-overhead early RTP/ AVPF feedback for generic NACK messages without triggering the RTP circuit breaker. This is expected to make such feedback suitable for RTP congestion control algorithms that need to quickly report loss events in between regular RTCP reports. The reaction to reduced-size RTCP SR/RR packets is to allow such algorithms to send feedback that can trigger the circuit breaker, when desired. 6. Impact of RTCP XR RTCP Extended Report (XR) blocks provide additional reception quality metrics, but do not change the RTCP timing rules. Some of the RTCP XR blocks provide information that might be useful for congestion control purposes, others provided non-congestion-related metrics. With the exception of RTCP XR ECN Summary Reports (see Section 8), the presence of RTCP XR blocks in a compound RTCP packet does not affect the RTP circuit breaker algorithm. For consistency and ease of implementation, only the reception report blocks contained in RTCP Perkins & Singh Expires January 16, 2014 [Page 13] Internet-Draft RTP Circuit Breakers July 2013 SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets, are used by the RTP circuit breaker algorithm. 7. Impact of RTCP Reporting Groups An optimisation for grouping RTCP reception statistics and other feedback in RTP sessions with large numbers of participants is given in [I-D.ietf-avtcore-rtp-multi-stream-optimisation]. This allows one SSRC to act as a representative that sends reports on behalf of other SSRCs that are co-located in the same endpoint and see identical reception quality. When running the circuit breaker algorithms, an endpoint MUST treat a reception report from the representative of the reporting group as if a reception report was received from all members of that group. 8. Impact of Explicit Congestion Notification (ECN) The use of ECN for RTP flows does not affect the media timeout RTP circuit breaker (Section 4.1) or the RTCP timeout circuit breaker (Section 4.2), since these are both connectivity checks that simply determinate if any packets are being received. ECN-CE marked packets SHOULD be treated as if it were lost for the purposes of congestion control, when determining the optimal media sending rate for an RTP flow. If an RTP sender has negotiated ECN support for an RTP session, and has successfully initiated ECN use on the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD be treated as if they were lost when calculating if the congestion- based RTP circuit breaker (Section 4.3) has been met. The count of ECN-CE marked RTP packets is returned in RTCP XR ECN summary report packets if support for ECN has been initiated for an RTP session. 9. Security Considerations The security considerations of [RFC3550] apply. If the RTP/AVPF profile is used to provide rapid RTCP feedback, the security considerations of [RFC4585] apply. If ECN feedback for RTP over UDP/IP is used, the security considerations of [RFC6679] apply. If non-authenticated RTCP reports are used, an on-path attacker can trivially generate fake RTCP packets that indicate high packet loss rates, causing the circuit breaker to trigger and disrupting an RTP session. This is somewhat more difficult for an off-path attacker, due to the need to guess the randomly chosen RTP SSRC value and the RTP sequence number. This attack can be avoided if RTCP packets are authenticated, for example using the Secure RTP profile [RFC3711]. Perkins & Singh Expires January 16, 2014 [Page 14] Internet-Draft RTP Circuit Breakers July 2013 10. IANA Considerations There are no actions for IANA. 11. Acknowledgements The authors would like to thank Bernard Aboba, Harald Alvestrand, Kevin Gross, Cullen Jennings, Randell Jesup, Jonathan Lennox, Matt Mathis, Stephen McQuistin, Eric Rescorla, and Abheek Saha for their valuable feedback. 12. References 12.1. Normative References [RFC 2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP Friendly Rate Control (TFRC): Protocol Specification", RFC 3448, January 2003. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC3550, July 2003. [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC3551, July 2003. [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC3611, November 2003. [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC4585, July 2006. 12.2. Informative References [Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "Equation-Based Congestion Control for Unicast Applications", Proc. ACM SIGCOMM 2000, DOI 10.1145/ 347059.347397, August 2000. [I-D.ietf-avtcore-rtp-multi-stream-optimisation] Perkins & Singh Expires January 16, 2014 [Page 15] Internet-Draft RTP Circuit Breakers July 2013 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, "Sending Multiple Media Streams in a Single RTP Session: Grouping RTCP Reception Statistics and Other Feedback", draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work in progress), July 2013. [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard] Clark, A., Huang, R., and W. Wu, "RTP Control Protocol(RTCP) Extended Report (XR) Block for Burst/Gap Discard metric Reporting", draft-ietf-xrblock-rtcp-xr- burst-gap-discard-14 (work in progress), April 2013. [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss] Clark, A., Zhang, S., Zhao, J., and W. Wu, "RTP Control Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Loss metric Reporting", draft-ietf-xrblock-rtcp-xr-burst- gap-loss-12 (work in progress), April 2013. [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics] Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol (RTCP) Extended Reports (XR) for Run Length Encoding (RLE) of Discarded Packets", draft-ietf-xrblock-rtcp-xr-discard- rle-metrics-06 (work in progress), July 2013. [I-D.ietf-xrblock-rtcp-xr-discard] Clark, A., Zorn, G., and W. Wu, "RTP Control Protocol (RTCP) Extended Report (XR) Block for Discard Count metric Reporting", draft-ietf-xrblock-rtcp-xr-discard-15 (work in progress), June 2013. [Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose, "Modeling TCP Throughput: A Simple Model and its Empirical Validation", Proc. ACM SIGCOMM 1998, DOI 10.1145/ 285237.285291, August 1998. [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, September 2001. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC3711, March 2004. [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC5104, February 2008. Perkins & Singh Expires January 16, 2014 [Page 16] Internet-Draft RTP Circuit Breakers July 2013 [RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in RTP Streams", RFC5450, March 2009. [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC5506, April 2009. [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP Flows", RFC6051, November 2010. [RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., and K. Carlberg, "Explicit Congestion Notification (ECN) for RTP over UDP", RFC6679, August 2012. [RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended Report (XR) Block for Packet Delay Variation Metric Reporting", RFC6798, November 2012. [RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol (RTCP) Extended Report (XR) Block for Delay Metric Reporting", RFC6843, January 2013. Authors' Addresses Colin Perkins University of Glasgow School of Computing Science Glasgow G12 8QQ United Kingdom Email: firstname.lastname@example.org Varun Singh Aalto University School of Electrical Engineering Otakaari 5 A Espoo, FIN 02150 Finland Email: email@example.com URI: http://www.netlab.tkk.fi/~varun/ Perkins & Singh Expires January 16, 2014 [Page 17]