AVT Working Group J. Ott
Internet-Draft Helsinki University of Technology
Intended status: Informational C. Perkins
Expires: September 10, 2009 University of Glasgow
March 9, 2009
Guidelines for Extending the RTP Control Protocol (RTCP)
draft-ietf-avt-rtcp-guidelines-01.txt
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Abstract
The RTP Control Protocol (RTCP) is used along with the Real-time
Transport Protocol (RTP) to provide a control channel between media
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senders and receivers. This allows constructing a feedback loop to
enable application adaptivity 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.
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 . . . . . . . . . . . . . . . . . 8
5. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 10
6. Security Considerations . . . . . . . . . . . . . . . . . . . 13
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
9.1. Normative References . . . . . . . . . . . . . . . . . . . 15
9.2. Informative References . . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17
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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, DTLS, or 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 to disambiguate all entities in an RTP
session and correlate SSRC identifiers with specific application
instances. RTCP is carried over the same transport as RTP and is
hence inherently best-effort and 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 SR and 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 which significantly increase the complexity of the
protocol and the reported values, mutate it toward an 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 which 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 proposed (and partly already accepted) for RTCP and
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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 key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14, RFC 2119
[RFC2119] and indicate requirement levels for compliant
implementations.
The terminology defined in RTP [RFC3550], the RTP Profile for Audio
and Video Conferences with Minimal Control [RFC3551], and the
Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF) [RFC4585]
apply.
3. RTP and RTCP Operation Overview
One of the twelve networking truths 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" [RFC1925]. 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
which 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 SSM networks. RTCP XR [RFC3611]
provides extended reporting mechanisms which are partly generic in
nature, 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
[I-D.ietf-dccp-rtp]), 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], [RFC2733], [RFC4588], [RFC5109]) are useful for protecting
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RTP media streams, and may be enabled as a result of RTCP feedback,
but do not directly affect RTCP behaviour.
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 (SR) indicate to the receivers the total number of
packets and octets 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 (RR) 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 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 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.
RTCP sender reports and receiver reports are sent, and the statistics
are sampled, at random intervals chosen uniformly in the range 0.5
... 1.5 times the deterministic calculated interval, T. The interval
T is calculated based on the media bit rate, 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
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360 seconds divided by the session bandwidth in kilobits/second
(giving an interval smaller than 5 seconds for bandwidths greater
than 72 kb/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 bit rate 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].
Ongoing work [I-D.ietf-avt-rtcp-non-compound] may 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 Conference Units.
These mechanisms can 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 bit rate.
Furthermore, various forward error correction mechanisms and, if RTTs
are short and the application permits extra delays, even reactive
error control such as retransmissions. Long-term feedback can be
provided in regular RTCP reports at configurable intervals, whereas
(close-to-)instant feedback is available by means of the early
feedback profile. Figure 1 below outlines this idea graphically.
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Long-term adaptation: RTCP Sender Reports Media processing:
- Codec+parameter choice - Data rate, pkt count - Dejittering
- Packet size - Timing and sync info - Synchronization
- 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
- Retro-active FEC - media-specific info and reporting
- Adaptive source coding - "congestion info"(*) - Instant event
- Congestion control(*) notifications
(*) RTCP feedback is insufficient for TCP-Friendly congestion control
purposes 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 timescales.
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.
o RTCP is inherently unreliable, and does not guarantee any
consistency between the observed state at multiple members of a
group.
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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
quintuple (source and destination IP address and port number,
transport protocol) could lead to a differentiation between RTP and
RTCP flows, disrupting the statistics.
While some networks may wish to intentionally prioritize 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 prioritization leads to RTCP packets being dropped
overproportionally.
All of this emphasizes the unreliable nature of RTCP. Multiplexing
on the same port number [I-D.ietf-avt-rtp-and-rtcp-mux] or inside the
same transport connection might help mitigating some of these
effects; but this is limited to speculation at this point and should
not be relied upon.
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:
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o Defined 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 MCUs [RFC5117] become
more widely used.
o Assuming reliable (instant) state synchronization. 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 Commands are issued, rather than hints given. 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 RTCP reporting is expanded to become a network management tool.
RTCP is sensitive to the size of RTCP reports as the latter
determines the mean reporting interval given a certain bit rate
share for RTCP. The amount of information going into RTCP reports
should primarily target the peer (and thus include information
that can be meaningfully reacted upon). Gathering and reporting
statistics beyond this is not an RTCP task and should be addressed
by out-of-band protocols.
o Serious complexity is created. Related to the previous item, RTCP
reports that convey all kinds of data first 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 the
purpose they were intended to), and hinder interoperability.
o 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 SRTP is
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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 which 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 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 existance 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 check list 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
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 implementation 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.
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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 is 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)?
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)
an old-style mechanisms 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 buildings blocks whenever
possible, allowing them to be combined in different environments
than initially considered. Where possible, avoid mechanisms that
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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
implementable 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 which 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.
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 decouple 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 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
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is needed, it's generally more appropriate to define a new RTCP
XR block or a new RTCP packet type instead.
* New RTCP AVPF transport layer feedback messages should be used
to transmit general purpose feedback information, to be
generated and processed by the RTP transport. Examples include
(negative) 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, that are useful for many video
codecs (see [RFC4585] sections 6.3 and 9).
* New RTCP AVPF application layer feedback messages should be
used to convery 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 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 mechansims or procedures,
and so raises no explicit security considerations. When designing
RTCP extensions, it is important to consider the following points:
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
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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 group 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 SDP, this typically requires some
form of authentication of the signalling messages (e.g. see the
security considerations of [I-D.ietf-avt-rtcpssm]).
At the time of this writing there are several proposals for in-band
signalling of secure RTP sessions, where the signalling information
is conveyed on the media path. These proposals were discussed in the
Audio/Video Transport working group session at the 67th IETF meeting,
with the consensus being that such signalling is not to be conveyed
within RTP data packets, but should instead be sent within some form
of control packet, and that it is acceptable to multiplex control and
data packets on the same port, provided the packet types can be
clearly distinguished. There was no consensus in the working group
on the question of whether keying information should be conveyed in
RTCP packets multiplexed on the RTP port, or in some other protocol
multiplexed on the RTP port. The opinion of the working group chairs
and area director was, however, that both approaches are workable,
and fit within the RTP architecture, but that it may be cleaner to
use a separate keying protocol (modelled after STUN), than to try to
fit keying within RTCP
The security considerations of the RTP specification [RFC3550] apply,
along with any applicable profile (e.g. [RFC3551]).
7. IANA Considerations
No IANA actions are necessary.
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8. Acknowledgements
This draft 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.
9. References
9.1. Normative References
[RFC1925] Callon, R., "The Twelve Networking Truths", RFC 1925,
April 1996.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[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.
[RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
for Generic Forward Error Correction", RFC 2733,
December 1999.
[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.
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[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.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008.
9.2. Informative References
[I-D.ietf-avt-rtcpssm]
Ott, J., "RTCP Extensions for Single-Source Multicast
Sessions with Unicast Feedback", draft-ietf-avt-rtcpssm-17
(work in progress), January 2008.
[I-D.ietf-avt-rtcp-non-compound]
Johansson, I. and M. Westerlund, "Support for non-compound
RTCP, opportunities and consequences",
draft-ietf-avt-rtcp-non-compound-02 (work in progress),
February 2008.
[I-D.ietf-dccp-rtp]
Perkins, C., "RTP and the Datagram Congestion Control
Protocol (DCCP)", draft-ietf-dccp-rtp-07 (work in
progress), June 2007.
[I-D.ietf-avt-rtp-and-rtcp-mux]
Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port",
draft-ietf-avt-rtp-and-rtcp-mux-07 (work in progress),
August 2007.
[ALF] Clark, D. and D. Tennenhouse, "Architectural
Considerations for a New Generation of Protocols",
Proceedings of ACM SIGCOMM 1990, September 1990.
[TimerRecon]
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Schulzrinne, H. and J. Rosenberg, "Timer Reconsideration
for Enhanced RTP Scalability", Proceedings of IEEE
Infocom 1998, March 1998.
Authors' Addresses
Joerg Ott
Helsinki University of Technology
Otakaari 5 A
Espoo, FIN 02150
Finland
Email: jo@netlab.hut.fi
Colin Perkins
University of Glasgow
Department of Computing Science
Lilybank Gardens
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
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