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Internet Engineering Task Force (IETF)                            J. OttRequest for Comments: 5968                              Aalto UniversityCategory: Informational                                       C. PerkinsISSN: 2070-1721                                    University of Glasgow                                                          September 2010Guidelines 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; seeSection 2 of RFC 5741.   Information about the current status of this document, any errata,   and how to provide feedback on it may be obtained athttp://www.rfc-editor.org/info/rfc5968.Ott & Perkins                 Informational                     [Page 1]

RFC 5968             Guidelines for RTCP Extensions       September 2010Copyright Notice   Copyright (c) 2010 IETF Trust and the persons identified as the   document authors.  All rights reserved.   This document is subject toBCP 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 Contents1. Introduction ....................................................32. Terminology .....................................................43. RTP and RTCP Operation Overview .................................43.1. RTCP Capabilities ..........................................53.2. RTCP Limitations ...........................................73.3. Interactions with Network- and Transport-Layer Mechanisms ..84. Issues with RTCP Extensions .....................................95. Guidelines .....................................................106. Security Considerations ........................................147. Acknowledgements ...............................................158. References .....................................................158.1. Normative References ......................................158.2. Informative References ....................................16Ott & Perkins                 Informational                     [Page 2]

RFC 5968             Guidelines for RTCP Extensions       September 20101.  Introduction   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 extensionsOtt & Perkins                 Informational                     [Page 3]

RFC 5968             Guidelines for RTCP Extensions       September 2010   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.2.  Terminology   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.3.  RTP and RTCP Operation Overview   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 codeOtt & Perkins                 Informational                     [Page 4]

RFC 5968             Guidelines for RTCP Extensions       September 2010   points for RTCP and the payload types for RTP use disjoint number   spaces.3.1.  RTCP Capabilities   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.Ott & Perkins                 Informational                     [Page 5]

RFC 5968             Guidelines for RTCP Extensions       September 2010   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 configurableOtt & Perkins                 Informational                     [Page 6]

RFC 5968             Guidelines for RTCP Extensions       September 2010   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.3.2.  RTCP Limitations   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.Ott & Perkins                 Informational                     [Page 7]

RFC 5968             Guidelines for RTCP Extensions       September 2010   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.Ott & Perkins                 Informational                     [Page 8]

RFC 5968             Guidelines for RTCP Extensions       September 20104.  Issues with RTCP Extensions   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 mayOtt & Perkins                 Informational                     [Page 9]

RFC 5968             Guidelines for RTCP Extensions       September 2010      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.5.  Guidelines   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 orOtt & Perkins                 Informational                    [Page 10]

RFC 5968             Guidelines for RTCP Extensions       September 2010      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)?Ott & Perkins                 Informational                    [Page 11]

RFC 5968             Guidelines for RTCP Extensions       September 2010   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.Ott & Perkins                 Informational                    [Page 12]

RFC 5968             Guidelines for RTCP Extensions       September 2010   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)Ott & Perkins                 Informational                    [Page 13]

RFC 5968             Guidelines for RTCP Extensions       September 2010         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],         Sections6.2 and9).      *  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], Sections6.3 and9).      *  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],         Sections6.4 and9).      *  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.6.  Security Considerations   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:Ott & Perkins                 Informational                    [Page 14]

RFC 5968             Guidelines for RTCP Extensions       September 2010   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]).7.  Acknowledgements   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.8.  References8.1.  Normative References   [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.Ott & Perkins                 Informational                    [Page 15]

RFC 5968             Guidelines for RTCP Extensions       September 2010   [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 2009.8.2.  Informative References   [RFC1925]      Callon, R., "The Twelve Networking Truths",RFC 1925,                  April 1996.   [RFC5117]      Westerlund, M. and S. Wenger, "RTP Topologies",RFC 5117, January 2008.Ott & Perkins                 Informational                    [Page 16]

RFC 5968             Guidelines for RTCP Extensions       September 2010   [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   EMail: csp@csperkins.orgOtt & Perkins                 Informational                    [Page 17]