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.
Ott & Perkins Informational [Page 1]RFC 5968 Guidelines for RTCP Extensions September 2010
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Table of Contents
1. Introduction ....................................................3
2. Terminology .....................................................4
3. RTP and RTCP Operation Overview .................................4
3.1. RTCP Capabilities ..........................................5
3.2. RTCP Limitations ...........................................7
3.3. Interactions with Network- and Transport-Layer Mechanisms ..8
4. Issues with RTCP Extensions .....................................9
5. Guidelines .....................................................10
6. Security Considerations ........................................14
7. Acknowledgements ...............................................15
8. References .....................................................15
8.1. Normative References ......................................15
8.2. Informative References ....................................16
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1. 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 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.
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 code
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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.
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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.
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.
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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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4. 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 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.
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 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.
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:
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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]).
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. References
8.1. 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.org
Ott & Perkins Informational [Page 17]
