Internet Engineering Task Force (IETF) J. Ott Request for Comments: 5968 Aalto University Category: Informational C. Perkins ISSN: 2070-1721 University of Glasgow September 2010
Guidelines for Extending the RTP Control Protocol (RTCP)
Abstract
The RTP Control Protocol (RTCP) is used along with the Real-time Transport Protocol (RTP) to provide a control channel between media senders and receivers. This allows constructing a feedback loop to enable application adaptation and monitoring, among other uses. The basic reporting mechanisms offered by RTCP are generic, yet quite powerful and suffice to cover a range of uses. This document provides guidelines on extending RTCP if those basic mechanisms prove insufficient.
Status of This Memo
This document is not an Internet Standards Track specification; it is published for informational purposes.
This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at http://www.rfc-editor.org/info/rfc5968.
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Copyright Notice
Copyright (c) 2010 IETF Trust and the persons identified as the document authors. All rights reserved.
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.
The Real-time Transport Protocol (RTP) [RFC3550] is used to carry time-dependent (often continuous) media such as audio or video across a packet network in an RTP session. RTP usually runs on top of an unreliable transport such as UDP, Datagram Transport Layer Security (DTLS), or the Datagram Congestion Control Protocol (DCCP), so that RTP packets are susceptible to loss, re-ordering, or duplication. Associated with RTP is the RTP Control Protocol (RTCP), which provides a control channel for each session: media senders provide information about their current sending activities ("feed forward"), and media receivers report on their reception statistics ("feedback") in terms of received packets, losses, and jitter. Senders and receivers provide self-descriptions allowing them to disambiguate all entities in an RTP session and correlate synchronisation source (SSRC) identifiers with specific application instances. RTCP is carried over the same transport as RTP and is inherently best-effort; hence the RTCP reports are designed for such an unreliable environment, e.g., by making them "for information only".
The RTCP control channel provides coarse-grained information about the session in two respects: 1) the RTCP sender report (SR) and receiver report (RR) packets contain only cumulative information or means over a certain period of time and 2) the time period is in the order of seconds and thus neither has a high resolution nor does the feedback come back instantaneously. Both these restrictions have their origin in RTP being scalable and generic. Even these basic mechanisms (which are still not implemented everywhere despite their simplicity and very precise specification, including sample code) offer substantial information for designing adaptive applications and for monitoring purposes, among others.
Recently, numerous extensions have been proposed in different contexts to RTCP that significantly increase the complexity of the protocol and the reported values, mutate it toward a command channel, and/or attempt turning it into a reliable messaging protocol. While the reasons for such extensions may be legitimate, many of the resulting designs appear ill-advised in the light of the RTP architecture. Moreover, extensions are often badly motivated and thus appear unnecessary given what can be achieved with the RTCP mechanisms in place today.
This document is intended to provide some guidelines for designing RTCP extensions. It is particularly intended to avoid an extension creep for corner cases that can only harm interoperability and future evolution of the protocol at large. We first outline the basic operation of RTCP and constructing feedback loops using the basic RTCP mechanisms. Subsequently, we outline categories of extensions
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proposed (and partly already accepted) for RTCP and discuss issues and alternative ways of thinking by example. Finally, we provide some guidelines and highlight a number of questions to ask (and answer!) before writing up an RTCP extension.
The terminology defined in "RTP: A Transport Protocol for Real-Time Applications" [RFC3550], "RTP Profile for Audio and Video Conferences with Minimal Control" [RFC3551], and "Extended RTP Profile for Real- time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)" [RFC4585] apply.
One of the twelve networking truths in [RFC1925] states: "In protocol design, perfection has been reached not when there is nothing left to add, but when there is nothing left to take away". Despite (or because of) this being an April 1st RFC, this specific truth is very valid, and it applies to RTCP as well.
In this section, we will briefly review what is available from the basic RTP/RTCP specifications. As specifications, we include those that are generic, i.e., do not have dependencies on particular media types. This includes the RTP base specification [RFC3550] and profile [RFC3551], the RTCP bandwidth modifiers for session descriptions [RFC3556], the timely feedback extensions (RFC 4585), and the extensions to run RTCP over source-specific multicast (SSM) networks [RFC5760]. RTCP extended reports (XRs) [RFC3611] provide extended reporting mechanisms that are partly generic in nature, and partly specific to a certain media stream.
We do not discuss RTP-related documents that are orthogonal to RTCP. The Secure RTP Profile [RFC3711] can be used to secure RTCP in much the same way it secures RTP data, but otherwise does not affect the behaviour of RTCP. The transport protocol used also has little impact, since RTCP remains a group communication protocol even when running over a unicast transport (such as TCP [RFC4571] or DCCP [RFC5762]), and is little affected by congestion control due to its low rate relative to the media. The description of RTP topologies [RFC5117] is useful knowledge, but is functionally not relevant here. The various RTP error correction mechanisms (e.g., [RFC2198], [RFC4588], [RFC5109]) are useful for protecting RTP media streams, and may be enabled as a result of RTCP feedback, but do not directly affect RTCP behaviour. Finally, RTP and RTCP may be multiplexed inside the same transport connection or using the same port number [RFC5761], but this does not affect the operation of RTCP itself; distinguishing RTP and RTCP packets is achieved because the code
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points for RTCP and the payload types for RTP use disjoint number spaces.
The RTP/RTCP specifications quoted above provide feedback mechanisms with the following properties, which can be considered as "building blocks" for adaptive real-time applications for IP networks.
o Sender reports (SRs) indicate to the receivers the total number of packets and octets that have been sent (since the beginning of the session or the last change of the sender's SSRC). These values allow deducing the mean data rate and mean packet size for both the entire session and, if continuously monitored, for every transmission interval. They also allow a receiver to distinguish between breaks in reception caused by network problems, and those due to pauses in transmission.
o Receiver reports (RRs) and SRs indicate reception statistics from each receiver for every sender. These statistics include:
* The packet loss rate since the last SR or RR was sent.
* The total number of packets lost since the beginning of the session, which may again be broken down to each reporting period.
* The highest sequence number received so far -- which allows a sender to roughly estimate how much data is in flight when used together with the SR and RR timestamps (and also allows observing whether the path still works and at which rate packets are delivered to the receiver).
* The moving average of the inter-arrival jitter of media packets. This gives the sender an indirect view of the size of any adaptive playout buffer used at the receiver ([RFC3611] gives precise figures for Voice over IP (VoIP) sessions).
o Sender reports also contain NTP and RTP format timestamps. These allow receivers to synchronise multiple RTP streams, and (when used in conjunction with receiver reports) allow the sender to calculate the current round-trip time (RTT) to each receiver. This value can be monitored over time and thus may be used to infer trends at coarse granularity. A similar mechanism is provided by [RFC3611] to allow receivers to calculate the RTT to senders.
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RTCP sender reports and receiver reports are sent, and the statistics are sampled, at random intervals chosen uniformly in the range from 0.5 to 1.5 times the deterministic calculated interval, T. The interval T is calculated based on the media bitrate, the mean RTCP packet size, whether the sampling node is a sender or a receiver, and the number of participants in the session, and will remain constant while the number of participants in the session remains constant. The lower bound on the base inter-report interval, T, is five seconds, or 360 seconds divided by the session bandwidth in kilobits/ second (giving an interval smaller than 5 seconds for bandwidths greater than 72 kbits/s) [RFC3550].
This lower limit can be eliminated, allowing more frequent feedback, when using the early feedback profile for RTCP [RFC4585]. In this case, the RTCP frequency is only limited by the available bitrate (usually 5% of the media stream bitrate is allocated for RTCP). If this fraction is insufficient, the RTCP bitrate may be increased in the session description to enable more frequent feedback [RFC3556]. The considerations in [RFC5506] may be used to reduce the mean RTCP packet size, further increasing feedback frequency.
The mechanisms defined in [RFC4585] even allow -- statistically -- a receiver to provide close-to-instant feedback to a sender about observed events in the media stream (e.g., picture or slice loss).
RTCP is suitable for unicast and multicast communications. All basic functions are designed with group communications in mind. While traditional (any-source) multicast (ASM) is clearly not available in the Internet at large, source-specific multicast (SSM) and overlay multicast are -- and both are commercially relevant. RTCP extensions have been defined to operate over SSM, and complex topologies may be created by interconnecting RTP mixers and translators. The group communication nature of RTP and RTCP is also essential for the operation of Multipoint Control Units.
These mechanisms can be used to implement a quite flexible feedback loop and enable short-term reaction to observed events as well as long-term adaptation to changes in the networking environment. Adaptation mechanisms available on the sender side include (but are not limited to) choosing different codecs, different parameters for codecs (spatial or temporal resolution for video, audible quality for audio and voice), and different packet sizes to adjust the bitrate. Furthermore, various forward error correction (FEC) mechanisms and, if RTTs are short and the application permits extra delays, even reactive error control such as retransmissions can be used. Long- term feedback can be provided in regular RTCP reports at configurable
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intervals, whereas (close-to-)instant feedback is available by means of the early feedback profile. Figure 1 below outlines this idea graphically.
Long-term adaptation: RTCP sender reports Media processing: - Codec+parameter choice - Data rate, pkt count - De-jittering - Packet size - Timing and sync info - Synchronisation - FEC, interleaving - Traffic characteristics - Error concealment --------------------------------> - Playout +---------------+/ \+---------------+ | | RTP media stream (codec, repair) | | | Media sender |=================================>| Media receiver | | | | | +---------------+\ RTCP receiver reports /+---------------+ <-------------------------------- Short-term reaction: - long-term statistics Control functions: - Retransmissions - event information - RTP monitoring - Retroactive FEC - media-specific info and reporting - Adaptive source coding - "congestion info"(*) - Instant event - Congestion control(*) notifications
(*) RTCP feedback is insufficient for the purposes of TCP-friendly congestion control due to the infrequent nature of reporting (which should be in the order of once per RTT), but can still be used to adapt to the available bandwidth on slower time-scales.
Figure 1: Outline of an RTCP Feedback Loop
It is important to note that not all information needs to be signalled explicitly -- ever, or upon every RTCP packet -- but can be derived locally from other pieces of information and from the evolution of the information over time.
The design of RTP limits what can meaningfully be done (and hence should be done) with RTCP. In particular, the design favours scalability and loose coupling over tightly controlled feedback loops. Some of these limitations are listed below (they need to be taken into account when designing extensions):
o RTCP is designed to provide occasional feedback, which is unlike, e.g., TCP ACKs, which can be sent in response to every (other) packet. It does not offer per-packet feedback (even when using [RFC4585] with increased RTCP bandwidth fraction, the feedback guarantees are only statistical in nature).
o RTCP is not capable of providing truly instant feedback.
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o RTCP is inherently unreliable and does not guarantee any consistency between the observed state at multiple members of a group.
It is important to note that these features of RTCP are intentional design choices, and are essential for it to scale to large groups.
3.3. Interactions with Network- and Transport-Layer Mechanisms
As discussed above, RTCP flows are used to measure, infer, and convey information about the performance of an RTP media stream.
Inference in baseline RTCP is mainly limited to determining the path RTT from pairs of RTCP SR and RR packets. This inference makes the implicit assumption that RTP and RTCP are treated equally: they are routed along the same path, mapped to the same (DiffServ) traffic classes, and treated as part of the same fair queuing classification. This is true in many cases; however, since RTP and RTCP are generally sent using different ports, any flow classification based upon the 5-tuple (of source and destination IP addresses, source and destination port numbers, and the transport protocol) could lead to a differentiation between RTP and RTCP flows, disrupting the statistics.
While some networks may wish to intentionally prioritise RTCP over RTP (to provide quicker feedback) or RTP over RTCP (since the media is considered more important than control), we recommend that they be treated identically where possible, to enable this inference of network performance, and hence support application adaptation.
When using reliable transport connections for (RTP and) RTCP [RFC2326] [RFC4571], retransmissions and head-of-line blocking may similarly lead to inaccurate RTT estimates derived by RTCP. (These may, nevertheless, properly reflect the mean RTT for a media packet, including retransmissions.)
The conveyance of information in RTCP is affected by the above only as soon as the prioritisation leads to a disproportionately high number of RTCP packets being dropped.
All of this emphasises the unreliable nature of RTCP. Multiplexing on the same port number [RFC5761] or inside the same transport connection might help mitigate some of these effects, but this is limited to speculation at this point and should not be relied upon.
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Issues that have come up in the past with extensions to RTP and RTCP include (but are probably not limited to) the following:
o Defining RTP or RTCP extensions only or primarily for unicast two- party sessions. RTP is inherently a group communication protocol, even when operating on a unicast connection. Extensions may become useful in the future well outside their originally intended area of application, and should consider this. Stating that something works for unicast only is not acceptable, particularly since various flavours of multicast have become relevant again, and as middleboxes such as repair servers, mixers, and RTCP- supporting Multipoint Control Units (MCUs) [RFC5117] become more widely used.
o Assuming reliable (instant) state synchronisation. RTCP reports are sent irregularly and may be lost. Hence, there may be a significant time lag (several seconds) between intending to send a state update to the RTP peer(s) and the packet being received; in some cases, the packet may not be received at all.
o Requiring reliable delivery of RTCP reports. While reliability can be implemented on top of RTCP using acknowledgements, this will come at the cost of significant additional delay, which may defeat the purpose of providing the feedback in the first place. Moreover, for scalability reasons due to the group-based nature of RTCP, these ACKs need to be adaptively rate limited or targeted to a subgroup or individual entity to avoid implosion as group sizes increase. RTCP is not intended or suitable for use as a reliable control channel.
o Issuing commands, rather than giving hints. RTCP is about reporting observations -- in a best-effort manner -- between RTP entities. Causing actions on the remote side requires some form of reliability (see above), and adherence cannot be verified.
o Expanding RTCP reporting, to use it as a network management tool. RTCP is sensitive to the size of RTCP reports as the latter determines the mean reporting interval given a certain bitrate share for RTCP (yet, RTCP may also be used to report information that has fine-grained temporal characteristics, if summarisation or data reduction by the endpoint would lose essential resolution). The information going into RTCP reports should primarily target the peer(s) (and thus include information that can be meaningfully reacted upon); nevertheless, such reports may
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provide useful information to augment other network management tools. Gathering and reporting statistics beyond this is not an RTCP task and should be addressed by out-of-band protocols.
o Creating serious complexity. Related to the previous item, RTCP reports that convey all kinds of data need to gather and calculate/infer this information to begin with (which requires very precise specifications). Given that it already seems to be difficult to even implement baseline RTCP, any added complexity can only discourage implementers, may lead to buggy implementations (in which case the reports do not serve their intended purpose), and hinder interoperability.
o Introducing architectural issues. Extensions are written without considering the architectural concepts of RTP. For example, point-to-point communication is assumed, yet third-party monitors are expected to listen in. Besides being a bad idea to rely on eavesdropping entities on the path, this is obviously not possible if Secure RTP (SRTP) is being used with encrypted SRTCP packets.
This list is surely not exhaustive. Also, the authors do not claim that the suggested extensions (even if using acknowledgements) would not serve a legitimate purpose. We rather want to draw attention to the fact that the same results may be achievable in a way that is architecturally cleaner and conceptually more RTP/RTCP-compliant. The following section contains a first attempt to provide some guidelines on what to consider when thinking about extensions to RTP and RTCP.
Designing RTCP extensions requires consideration of a number of issues, as well as in-depth understanding of the operation of RTP mechanisms. While it is expected that there are many aspects not yet covered by RTCP reporting and operation, quite a bit of functionality is readily available for use. Other mechanisms should probably never become part of the RTP family of specifications, despite the existence of their equivalents in other environments. In the following, we provide some guidance to consider when (and before!) developing an extension to RTCP.
We begin with a short checklist concerning the applicability of RTCP in the first place:
o Check what can be done with the existing mechanisms, exploiting the information that is already available in RTCP. Is the need for an extension only perceived (e.g., due to lazy implementers, or artificial constraints in endpoints), or is the function or
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data really not available (or derivable from existing reports)? It is worthwhile remembering that redundant information supplied by a protocol runs the risk of being inconsistent at some point, and various implementations may handle such situations differently (e.g., give precedence to different values). Similarly, there should be exactly one (well-specified) way of performing every function and operation of the protocol.
o Is the extension applicable to RTP entities running anywhere in the Internet, or is it a link- or environment-specific extension? In the latter cases, local extensions (e.g., header compression, or non-RTP protocols) may be preferable. RTCP should not be used to carry information specific to a particular (access) link.
o Is the extension applicable in a group communication environment, or is it specific to point-to-point communications? RTP and RTCP are inherently group communication protocols, and extensions must scale gracefully with increasing group sizes.
From a conceptual viewpoint, the designer of every RTCP extension should ask -- and answer(!) -- at least the following questions:
o How will this new building block complement and work with the other components of RTCP? Are all interactions fully specified?
o Will this extension work with all different profiles (e.g., the Secure RTP profile [RFC3711], and the extended RTP profile for RTCP-based feedback [RFC4585])? Are any feature interactions expected?
o Should this extension be kept in-line with baseline RTP and its existing profiles, or does it deviate so much from the base RTP operation that an incompatible new profile must be defined? Use and definition of incompatible profiles are strongly discouraged, but if they prove necessary, how do nodes using the different profiles interact? What are the failure modes, and how is it ensured that the system fails in a safe manner?
o How does this extension interoperate with other nodes when the extension is not understood by the peer(s)?
o How will the extension deal with different networking conditions (e.g., how does performance degrade with increases in losses and latency, possibly across orders of magnitude)?
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o How will this extension work with group communication scenarios, such as multicast? Will the extensions degrade gracefully with increasing group sizes? What will be the impact on the RTCP report frequency and bitrate allocation?
For the specific design, the following considerations should be taken into account (they're a mixture of common protocol design guidelines, and specifics for RTCP):
o First of all, if there is (and for RTCP this applies quite often) a mechanism from a different networking environment, don't try to directly recreate this mechanism in RTP/RTCP. The Internet environment is extremely heterogeneous, and will often have drastically different properties and behaviour to other network environments. Instead, ask what the actual semantics and the result required to be perceived by the application or the user are. Then, design a mechanism that achieves this result in a way that is compatible with RTP/RTCP. (And do not forget that every mechanism will break when no packets get through -- the Internet does not guarantee connectivity or performance.)
o Target re-usability of the specification. That is, think broader than a specific use case, and try to solve the general problem in cases where it makes sense to do so. Point solutions need a very good motivation to be dealt with in the IETF in the first place. This essentially suggests developing building blocks whenever possible, allowing them to be combined in different environments than initially considered. Where possible, avoid mechanisms that are specific to particular payload formats, media types, link or network types, etc.
o For everything (packet format, value, procedure, timer, etc.) being defined, make sure that it is defined properly, so that independent interoperable implementation can be built. It is not sufficient that you can implement the feature: it has to be implemented in several years by someone unfamiliar with the working group discussion and industry context. Remember that fields need to be both generated and reacted upon, that mechanisms need to be implemented, etc., and that all of this increases the complexity of an implementation. Features that are too complex won't get implemented (correctly) in the first place.
o Extensions defining new metrics and parameters should reference existing standards whenever possible, rather than try to invent something new and/or proprietary.
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o Remember that not every bit or every action must be represented or signalled explicitly. It may be possible to infer the necessary pieces of information from other values or their evolution (a very prominent example is TCP congestion control). As a result, it may be possible to de-couple bits on the wire from local actions and reduce the overhead.
o Particularly with media streams, reliability can often be "soft". Rather than implementing explicit acknowledgements, receipt of a hint may also be observed from the altered behaviour (e.g., the reception of a requested intra-frame, or changing the reference frame for video, changing the codec, etc.). The semantics of messages should be idempotent so that the respective message may be sent repeatedly. Requiring hard reliability does not scale with increasing group sizes, and does not degrade gracefully as network performance reduces.
o Choose the appropriate extension point. Depending on the type of RTCP extension being developed, new data items can be transported in several different ways:
* A new RTCP Source Description (SDES) item is appropriate for transporting data that describes the source, or the user represented by the source, rather than the ongoing media transmission. New SDES items may be registered to transport source description information of general interest (see [RFC3550], Section 15), or the PRIV item ([RFC3550], Section 6.5.8) may be used for proprietary extensions.
* A new RTCP XR block type is appropriate for transporting new metrics regarding media transmission or reception quality (see [RFC3611], Section 6.2).
* New RTP profiles may define a profile-specific extension to RTCP SR and/or RR packets, to give additional feedback (see [RFC3550], Section 6.4.3). It is important to note that while extensions using this mechanism have low overhead, they are not backwards compatible with other profiles. Where compatibility is needed, it's generally more appropriate to define a new RTCP XR block or a new RTCP packet type instead.
* New RTCP AVPF (Audio-Visual Profile with Feedback) transport- layer feedback messages should be used to transmit general- purpose feedback information that will be generated and processed by the RTP transport. Examples include (negative)
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acknowledgements for particular packets, or requests to limit the transmission rate. This information is intended to be independent of the codec or application in use (see [RFC4585], Sections 6.2 and 9).
* New RTCP AVPF payload-specific feedback messages should be used to convey feedback information that is specific to a particular media codec, RTP payload format, or category of RTP payload formats. Examples include video picture loss indication or reference picture selection, which are useful for many video codecs (see [RFC4585], Sections 6.3 and 9).
* New RTCP AVPF application layer feedback messages should be used to convey higher-level feedback, from one application to another, above the level of codecs or transport (see [RFC4585], Sections 6.4 and 9).
* A new RTCP application-defined, or APP, packet is appropriate for private use by applications that don't need to interoperate with others, or for experimentation before registering a new RTCP packet type ([RFC3550], Section 6.7). It is not appropriate to define a new RTCP APP packet in a standards document: use one of the other extension points, or define a new RTCP packet type instead.
* Finally, new RTCP packet types may be registered with IANA if none of the other RTCP extension points are appropriate (see [RFC3550], Section 15).
The RTP framework was designed following the principle of application level framing with integrated layer processing, proposed by Clark and Tennenhouse [ALF]. Effective use of RTP requires that extensions and implementations be designed and built following the same philosophy. That philosophy differs markedly from many previous systems in this space, and making effective use of RTP requires an understanding of those differences.
This memo does not specify any new protocol mechanisms or procedures, and so raises no explicit security considerations. When designing RTCP extensions, it is important to consider the following points:
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o Privacy: RTCP extensions, in particular new Source Description (SDES) items, can potentially reveal information considered to be sensitive by end users. Extensions should carefully consider the uses to which information they release could be put, and should be designed to reveal the minimum amount of additional information needed for their correct operation.
o Congestion control: RTCP transmission timers have been carefully designed such that the total amount of traffic generated by RTCP is a small fraction of the media data rate. One consequence of this is that the individual RTCP reporting interval scales with both the media data rate and the group size. The RTCP timing algorithms have been shown to scale from two-party unicast sessions to groups with tens of thousands of participants, and to gracefully handle flash crowds and sudden departures [TimerRecon]. Proposals that modify the RTCP timer algorithms must be careful to avoid congestion, potentially leading to denial of service, across the full range of environments where RTCP is used.
o Denial of service: RTCP extensions that change the location where feedback is sent must be carefully designed to prevent denial of service attacks against third-party nodes. When such extensions are signalled, for example in the Session Description Protocol (SDP), this typically requires some form of authentication of the signalling messages (e.g., see the security considerations of [RFC5760]).
The security considerations of the RTP specification [RFC3550] apply, along with any applicable profile (e.g., [RFC3551]).
This document has been motivated by many discussions in the AVT WG. The authors would like to acknowledge the active members in the group for providing the inspiration.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, September 1997.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998.
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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556, July 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, November 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection-Oriented Transport", RFC 4571, July 2006.
[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)", RFC 4585, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, July 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error Correction", RFC 5109, December 2007.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced- Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, April 2009.
[RFC1925] Callon, R., "The Twelve Networking Truths", RFC 1925, April 1996.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008.
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[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback", RFC 5760, February 2010.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5762] Perkins, C., "RTP and the Datagram Congestion Control Protocol (DCCP)", RFC 5762, April 2010.
[ALF] Clark, D. and D. Tennenhouse, "Architectural Considerations for a New Generation of Protocols", Proceedings of ACM SIGCOMM 1990, September 1990.
[TimerRecon] Schulzrinne, H. and J. Rosenberg, "Timer Reconsideration for Enhanced RTP Scalability", Proceedings of IEEE Infocom 1998, March 1998.
Authors' Addresses
Joerg Ott Aalto University School of Science and Technology Otakaari 5 A Espoo, FIN 02150 Finland
EMail: jo@netlab.tkk.fi
Colin Perkins University of Glasgow Department of Computing Science Glasgow G12 8QQ United Kingdom