Network Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Informational M. Westerlund
Expires: September 8, 2011 Ericsson
J. Ott
Aalto University
March 7, 2011
RTP Requirements for RTC-Web
draft-perkins-rtcweb-rtp-usage-00
Abstract
This document discusses usage of RTP in the context of RTC-WEB work.
It discusses important factors of RTP to consider by other parts of
the solution, it also discusses which RTP profile to support and
which RTP extensions that should be supported.
Status of this Memo
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This Internet-Draft will expire on September 8, 2011.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Requirements from RTP . . . . . . . . . . . . . . . . . . . . 3
2.1. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 4
2.2. Signalling for RTP sessions . . . . . . . . . . . . . . . 6
2.3. (Lack of) Signalling for Payload Format Changes . . . . . 7
3. RTP Profile . . . . . . . . . . . . . . . . . . . . . . . . . 7
4. RTP and RTCP Guidelines . . . . . . . . . . . . . . . . . . . 7
5. RTP Optimizations . . . . . . . . . . . . . . . . . . . . . . 8
5.1. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 8
5.2. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 8
5.3. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 8
5.4. CNAME generation . . . . . . . . . . . . . . . . . . . . . 9
6. RTP Extensions . . . . . . . . . . . . . . . . . . . . . . . . 9
6.1. RTP Conferencing Extensions . . . . . . . . . . . . . . . 9
6.1.1. RTCP Feedback Message: Full Intra Request . . . . . . 10
6.1.2. RTCP Feedback Message: Picture Loss Indicator . . . . 10
6.1.3. RTCP Feedback Message: Temporary Maximum Media
Stream Bit Rate Request . . . . . . . . . . . . . . . 10
6.2. RTP Header Extensions . . . . . . . . . . . . . . . . . . 11
6.3. Rapid Synchronisation Extensions . . . . . . . . . . . . . 11
7. Improving RTP Transport Robustness . . . . . . . . . . . . . . 12
7.1. RTP Retransmission . . . . . . . . . . . . . . . . . . . . 12
7.2. Forward Error Correction . . . . . . . . . . . . . . . . . 12
8. RTP Rate Control and Media Adaptation . . . . . . . . . . . . 12
9. RTP Performance Monitoring . . . . . . . . . . . . . . . . . . 13
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
11. Security Considerations . . . . . . . . . . . . . . . . . . . 13
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 13
13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13
13.1. Normative References . . . . . . . . . . . . . . . . . . . 13
13.2. Informative References . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16
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1. Introduction
This document discusses RTP in the context of RTC-WEB. This include
RTP's requirement on underlying transport, for example when it comes
to provide multiplexing. It discusses which RTP profile that should
be supported and what RTP extensions that should be supported. The
importance of congestion control and media adaptation is also
discussed. This document is intended as a starting point for
discussing RTP features in RTC-WEB.
The work in the AVT WG has all been about providing building blocks
and not specify who should use which building blocks. Selection of
building blocks and functionalities can really only be done in the
context of some application(s). RTC-WEB will greatly benefit in
interoperability if a reasonable set of RTP functionalities and
extensions are selected. For RTC-WEB we have selected RTP extensions
that are suitable for a number of applications that fits the context.
Thus applications such as VoIP, audio and video conferencing, and on-
demand multi-media streaming are considered. Applications that rely
on multicast transport has not been considered likely to be
applicable to RTC-WEB, thus extensions related to multicast have been
excluded.
The document is structured into different topics. For each topic one
or several recommendations from the authors are done. When it comes
to extensions or need for implementation support we use three levels
to indicate the importance for it to be included in an RTC-WEB
specification. We see it as likely that everything that is included
is in fact mandated to be implemented.
REQUIRED: Absolutely needed functionality to make the solution work
well. Also functionality of low complexity that provide high
value has also been categorized as required.
RECOMMENDED: Should be included as its brings significant benefit,
but the solution can potentially work without it.
OPTIONAL: Something that is useful in some cases, but not always a
benefit.
When this documents discusses RTP it always include RTCP unless
explicitly stated otherwise. This as RTCP is a fundamental and
integral part of the protocol.
2. Requirements from RTP
This section discusses some requirements RTP/RTCP [RFC3550] puts its
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underlying transport, the signalling etc.
2.1. RTP Session Multiplexing
RTP has three fundamental points of multiplexing. The first one is
the RTP session, which is used to separate media of different kind or
purpose. Such as Audio and Video, or the document camera and the
speaker camera in video conference. This multiplexing point does not
have an identifier within the RTP protocol, instead it relies on the
lower layer to separate the different RTP session. Thus the most
common RTP session separation is different UDP port numbers, but also
IP address or other identifiers maybe used to achieve this
separation. The second multiplexing point is the SSRC that separates
different sources of media within a single RTP session. The third is
the RTP Payload type, which identifies how the media from a
particular source is encoded.
These multiplexing points area fundamental part of the design of RTP
and is discussed in Section 5.2 of [RFC3550]. From that list the
ones that are directly related to the importance of the RTP session
as concept are 4 and 5 (from RFC 3550):
"4. An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream."
"5. Carrying multiple media in one RTP session precludes: the use of
different network paths or network resource allocations if
appropriate; reception of a subset of the media if desired, for
example just audio if video would exceed the available bandwidth;
and receiver implementations that use separate processes for the
different media, whereas using separate RTP sessions permits
either single- or multiple-process implementations."
Point 4, has to do with media of different kind or purpose. The
processing that can happen in an RTP mixer, translator or in an end-
point is dependent on the purpose and media type of the stream. Thus
there is an importance of separating such streams from each other.
This could of course be achieved by other methods, like tagging SSRC
values with their purpose, however there are reasons why this was not
chosen. First of all it is not the simple solution, as this require
additional signalling, and possibly synchronization between session
peers. In addition there is the issue point 5 raises.
Point 5 has to do with enabling quality of service or traffic
engineering between the media flows in different RTP sessions. By
using different transport layer ports, QoS mechanism that are capable
of operating on the 5-tuple (Source address, port, destination
address, port, and protocol) can be used without modification on RTP.
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Due to these design principle implementors of various services or
applications using RTP has commonly not violated this model. If one
choses to violate it today one fails to achieve interoperability with
a number of existing services, applications and implementations.
Lets assume one overloads multiple RTP sessions into one by tagging
the SSRC to belong to different purposes. If one would gateway that
design into a legacy system, then there would be a significant issue
with SSRC collision. This as the legacy system would not know about
the need to avoid using the same SSRC in the different RTP sessions.
There are also various RTP mechanism that has the potential for
issues if one don't have a clear separation of RTP sessions:
Scalabilty: RTP was built with media scalability in consideration.
The simplest way of achieving separation between different
scalability layers are placing them in different RTP sessions, and
using the same SSRC and CNAME in each session to bind them
together. This is most commonly done in multicast, and not
particular applicable to RTC-WEB, but gatewaying of such a session
would then require more alterations and likely stateful
translation.
RTP Retransmission in Session Multiplexing mode: RTP Retransmission
[RFC4588] does have a mode for session multiplexing. This would
not be the main mode used in RTC-WEB, but for interoperability and
reduced cost in translation support for different RTP Sessions are
required.
Forward Error Correction: The "An RTP Payload Format for Generic
Forward Error Correction" [RFC2733] and its update [RFC5109] can
only be used on media formats that produce RTP packets that are
smaller than half the MTU if the FEC flow and media flow being
protected are to be sent in the same RTP session, this is due to
"RTP Payload for Redundant Audio Data" [RFC2198]. This is because
the SSRC value of the original flow is recovered from the FEC
packets SSRC field. So for anything that desires to use these
format with RTP payloads that are close to MTU needs to put the
FEC data in a separate RTP session compared to the original
transmissions.
RTCP behavior also becomes a factor in why overloading RTP sessions
is problematic. The extension mechanisms used in RTCP depends on the
media streams. For example the Extended RTCP report block for VoIP
is of suitable for conversational audio, but clearly not useful for
Video. This has three impacts, either one get unusable reports if
they are generated for streams where there are little purpose. This
is maybe less likely for the VoIP report, but for example the more
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detailed media agnostic reports it may occur. It otherwise makes the
implementation of RTCP more complex as the SSRC purpose tagging needs
not only to be one the media side, but also on the RTCP reporting.
Also the RTCP reporting interval and transmission scheduling will be
affected.
As a conclusion not ensuring that RTP sessions are used for its
intended purpose as a multiplexing point does violate the RTP design
philosophy. It prevents the usage of certain RTP extensions. It
will require additional extensions to function and will significantly
increase the complexity of the implementation. At the same time it
will significantly reduce the interoperability with current
implementations.
2.2. Signalling for RTP sessions
RTP is built with the assumption of an external to RTP/RTCP
signalling channel to configure the RTP sessions and its functions.
The basic configuration of an RTP session consists of the following
parameters:
RTP Profile: The name of the RTP profile to be used in session. The
RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
on basic level, as can their secure variants RTP/SAVP [RFC3711]
and RTP/SAVPF [RFC5124]. The secure variants of the profiles do
not directly interoperate with the non-secure variants, due to the
presence of additional header fields.
Transport Information: Source and destination address(s) and ports
for RTP and RTCP must be signalled for each RTP session. If RTP
and RTCP multiplexing [RFC5761] is to be used, such that a single
port is used for RTP and RTCP flows, this must be signalled.
RTP Payload Types and Media formats: The mapping between media type
names (and hence the RTP payload formats to be used) and the RTP
payload type numbers must be signalled. Each media type may also
have a number of media type parameters that must also be signalled
to configure the codec and RTP payload format (the "a=fmtp:" line
from SDP).
Support for exchanging RTCP Bandwidth values to the end-points will
be necessary, as described in "Session Description Protocol (SDP)
Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth"
[RFC3556], or something semantically equivalent. This also ensures
that the end-points have a common view of the RTCP bandwidth, this is
important as too different view of the bandwidths may lead to failure
to interoperate.
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2.3. (Lack of) Signalling for Payload Format Changes
As discussed in Section 2.2, the mapping between media type name, and
its associated RTP payload format, and the RTP payload type number to
be used for that format must be signalled as part of the session
setup. An endpoint may signal support for multiple media formats, or
multiple configurations of a single format, each using a different
RTP payload type number. If multiple formats are signalled by an
endpoint, that endpoint must be prepared to receive data encoded in
any of those formats at any time. RTP does not require advance
signalling for changes between formats that were signalled as part of
the session setup. This is necessary for rapid rate adaptation.
3. RTP Profile
The RTP profile REQUIRED to implement is "Extended Secure RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
SAVPF)" [RFC5124]. Which will mean implicit support for AVPF
[RFC4585], AVP [RFC3551] and SAVP [RFC3711].
The AVPF part of SAVPF is required to get the improved RTCP timer
model, that allows more flexible transmission of RTCP packets as
response to events, rather than strictly according to bandwidth.
This also saves RTCP bandwidth and will commonly only utilize the
full amount when there is a lot of events to send feedback on.
The S part of SAVPF is for support of SRTP. This provides media
encryption, integrity protection, replay protection and a limited
form of source authentication. It does not contain a specific keying
mechanism. So that and the set of security transforms will be
required to be selected. It is possible that a security mechanism
operating on a lower layer than RTP can be used instead and that
should be evaluated. However, the reasons for the design of SRTP
should be taken into consideration in that discussion.
4. RTP and RTCP Guidelines
RTP and RTCP are two flexible and extensible protocols that allow, on
the one hand, choosing from a variety of building blocks and
combining those to meet application needs, and on the other hand,
create extensions where existing mechanisms are not sufficient: from
new payload formats to RTP extension headers to additional RTCP
control packets.
Different informational documents provide guidelines to the use and
particularly the extension of RTP and RTCP, including the following:
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Guidelines for Writers of RTP Payload Format Specifications [RFC2736]
and Guidelines for Extending the RTP Control Protocol [RFC5968].
5. RTP Optimizations
This section discusses some optimizations that makes RTP/RTCP work
better and more efficient and therefore are considered.
5.1. RTP and RTCP Multiplexing
Historically, RTP and RTCP have been run on separate UDP ports. With
the increased use of Network Address Port Translation (NAPT) this has
become problematic, since maintaining multiple NAT bindings can be
costly. It also complicates firewall administration, since multiple
ports must be opened to allow RTP traffic. To reduce these costs and
session setup times, support for multiplexing RTP data packets and
RTCP control packets on a single port [RFC5761] is REQUIRED.
Note that the use of RTP and RTCP multiplexed on a single port
ensures that there is occasional traffic sent on that port, even if
there is no active media traffic. This may be useful to keep-alive
NAT bindings.
5.2. Reduced Size RTCP
RTCP packets are usually sent as compound RTCP packets; and RFC 3550
demands that those compound packets always start with an SR or RR
packet. However, especially when using frequent feedback messages,
these general statistics are not needed in every packet and
unnecessarily increase the mean RTCP packet size and thus limit the
frequency at which RTCP packets can be sent within the RTCP bandwidth
share.
RFC5506 "Support for Reduced-Size Real-Time Transport Control
Protocol (RTCP): Opportunities and Consequences" [RFC5506] specifies
how to reduce the mean RTCP message and allow for more frequent
feedback. Frequent feedback, in turn, is essential to make real-time
application quickly aware of changing network conditions and allow
them to adapt their transmission and encoding behavior.
Support for RFC5506 is REQUIRED.
5.3. Symmetric RTP/RTCP
RTP entities choose the RTP and RTCP transport addresses, i.e., IP
addresses and port numbers, to receive packets on and bind their
respective sockets to those. When sending RTP packets, however, they
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may use a different IP address or port number for RTP, RTCP, or both;
e.g., when using a different socket instance for sending and for
receiving. Symmetric RTP/RTCP requires that the IP address and port
number for sending and receiving RTP/RTCP packets are identical.
Using Symmetric RTP and RTCP [RFC4961] is REQURIED.
5.4. CNAME generation
The RTCP Canonical Name (CNAME) provides a persistent transport-level
identifier for an RTP endpoint. While the Synchronization Source
(SSRC) identifier for an RTP endpoint may change if a collision is
detected, or when the RTP application is restarted, it's RTCP CNAME
is meant to stay unchanged, so that RTP endpoints can be uniquely
identified and associated with their RTP media streams. For proper
functionality, RTCP CNAMEs should be unique within the participants
of an RTP session.
The RTP specification [RFC3550] includes guidelines for choosing a
unique RTP CNAME, but these are not sufficient in the presence of NAT
devices. In addition, some may find long-term persistent identifiers
problematic from a privacy viewpoint. Accordingly, support for
generating the RTP CNAME as specified in "Guidelines for Choosing RTP
Control Protocol (RTCP) Canonical Names (CNAMEs)"
[I-D.ietf-avt-rtp-cnames] is RECOMMENDED, since this addresses both
concerns.
6. RTP Extensions
There are a number of RTP extensions that could be very useful in the
RTC-WEB context. One set is related to conferencing, others are more
generic in nature.
6.1. RTP Conferencing Extensions
RTP is inherently defined for group communications, originally
assuming the availability of IP multicast. In today's practice,
however, overlay-based conferencing dominates, typically using one or
a few so-called conference bridges or servers to connect endpoints in
a star or flat tree topology. Quite diverse conferencing topologies
can be created using the basic elements of RTP mixers and translators
as defined in RFC 3550.
An number of conferencing topologies are defined in [RFC5117] out of
the which the following ones are the more common (and most likely in
practice workable) ones:
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1) RTP Translator (Relay) with Only Unicast Paths (RFC 5117, section
3.3)
2) RTP Mixer with Only Unicast Paths (RFC 5117, section 3.4)
3) Point to Multipoint Using a Video Switching MCU (RFC 5117, section
3.5)
4) Point to Multipoint Using Content Modifying MCUs (RFC 5117,
section 3.6)
RTP protocol extensions to be used with conferencing are included
because they are important in the context of centralized
conferencing, where one RTP Mixer (Conference Focus) receives a
participants media streams and distribute them to the other
participants. These messages are defined in AVPF [RFC4585] or in
"Codec Control Messages in the RTP Audio-Visual Profile with Feedback
(AVPF)" [RFC5104].
6.1.1. RTCP Feedback Message: Full Intra Request
The Full Intra Request is defined in Section 3.51 and 4.3.1 of
[RFC5104]. It is used to have the mixer request from the currently
distributed session participants a new Intra picture. This is used
when switching between sources to ensure that the receivers can
decode the video or other predicted media encoding with long
prediction chains. It is RECOMMENDED that this feedback message is
supported.
6.1.2. RTCP Feedback Message: Picture Loss Indicator
The Picture Loss Indicator is defined in Section 6.3.1 of [RFC4585].
It is used by a receiver to tell the encoder that it lost the decoder
context and would like to have it repaired somehow. This is
semantically different from the the Full Intra Request above. It is
RECOMMENDED that this feedback message is supported.
6.1.3. RTCP Feedback Message: Temporary Maximum Media Stream Bit Rate
Request
This feedback message is defined in Section 3.5.4 and 4.2.1 in
[RFC5104]. This message and its notification message is used by a
media receiver, to inform the sending party that there is a current
limitation on the amount of bandwidth available to this receiver.
This can be for various reasons, and can for example be used by an
RTP mixer to limit the media sender being forwarded by the mixer
(without doing media transcoding) to fit the bottlenecks existing
towards the other session participants. It is RECOMMENDED that this
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feedback message is supported.
6.2. RTP Header Extensions
The RTP specification [RFC3550] provides a capability to extend the
RTP header with in-band data, but the format and semantics of the
extensions are poorly specified. Accordingly, if header extensions
are to be used, it is REQUIRED that they be formatted and signalled
according to the general mechanism of RTP header extensions defined
in [RFC5285].
As noted in [RFC5285], the requirement from the RTP specification
that header extensions are "designed so that the header extension may
be ignored" [RFC3550] stands. To be specific, header extensions must
only be used for data that can safely be ignored by the recipient
without affecting interoperability, and must not be used when the
presence of the extension has changed the form or nature of the rest
of the packet in a way that is not compatible with the way the stream
is signaled (e.g., as defined by the payload type). Valid examples
might include metadata that is additional to the usual RTP
information.
The RTP rapid synchronisation header extension is recommended, as
discussed in Section 6.3.
Currently no other header extensions are recommended. But we do
include a list of the available ones for consideration below:
Transmission Time offsets: [RFC5450] defines a format for including
an RTP timestamp offset of the actual transmission time of the RTP
packet in relation to capture/display timestamp present in the RTP
header. This can be used to improve jitter determination and
buffer management.
Associating Time-Codes with RTP Streams: [RFC5484] defines how to
associate SMPTE times codes with the RTP streams.
Audio Levels indications: There is ongoing work to define RTP header
extensions for providing audio levels both from a media sender to
an mixer [I-D.ietf-avtext-client-to-mixer-audio-level], and from a
mixer to a receiver[I-D.ietf-avtext-mixer-to-client-audio-level].
6.3. Rapid Synchronisation Extensions
Many RTP sessions require synchronisation between audio, video, and
other content. This synchronisation is performed by receivers, using
information contained in RTCP SR packets, as described in the RTP
specification [RFC3550]. This basic mechanism can be slow, however,
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so it is RECOMMENDED that the rapid RTP synchronisation extensions
described in [RFC6051] be implemented. The rapid synchronisation
extensions use the general RTP header extension mechanism [RFC5285],
which requires signalling, but are otherwise backwards compatible.
7. Improving RTP Transport Robustness
There are some tools that can robustify RTP flows against Packet loss
and reduce the impact on media quality. However they all add extra
bits compared to a non-robustified stream. These extra bits needs to
be considered and the aggregate bit-rate needs to be rate controlled.
Thus robustification might require a lower base encoding quality but
has the potential to give that quality with fewer errors in it.
7.1. RTP Retransmission
Support for RTP retransmission as defined by "RTP Retransmission
Payload Format" [RFC4588] is RECOMMENDED.
The retransmission scheme in RTP allows flexible application of
retransmissions. Only selected missing packets can be requested by
the receiver. It also allows for the sender to prioritize between
missing packets based on senders knowledge about their content.
Compared to TCP, RTP retransmission also allows one to give up on a
packet that despite retransmission(s) still has not been received
within a time window.
7.2. Forward Error Correction
Support of some type of FEC scheme to combat packet loss is
beneficial, but is application dependent and also claimed to be
mostly encumbered. For further discussion.
8. RTP Rate Control and Media Adaptation
It is REQUIRED to have an RTP Rate Control mechanism using Media
adaptation to ensure that the generated RTP flows are network
friendly.
The biggest issue is that there are no standardized and ready to use
mechanism that can simply be included in RTC-WEB. Thus there will be
need for the IETF to produce such a specification. A potential
starting point for defining a solution is "RTP with TCP Friendly Rate
Control"[rtp-tfrc].
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9. RTP Performance Monitoring
RTCP does contains a basic set of RTP flow monitoring points like
packet loss and jitter. There exist a number of extensions that
could be included in the set to be supported. However, in most cases
which RTP monitoring that is needed depends on the application, which
makes it difficult to select which to include when the set of
applications is very large.
10. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
11. Security Considerations
RTP and its various extensions each have their own security
considerations. These should be taken into account when considering
the security properties of the complete suite. We currently don't
think this suite creates any additional security issues or
properties. The usage of SRTP will provide protection or mitigation
against all the fundamental issues by offering confidentiality,
integrity and partial source authentication. We don't discuss the
key-management aspect of SRTP in this document, that needs to be done
taking the RTC-WEB communication model into account.
In the context of RTC-WEB the actual security properties required
from RTP are currently not fully understood. Until security goals
and requirements are specified it will be difficult to determine what
security features in addition to SRTP and a suitable key-management,
if any, that are needed.
12. Acknowledgements
13. References
13.1. Normative References
[I-D.ietf-avt-rtp-cnames]
Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", draft-ietf-avt-rtp-cnames-05 (work in
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progress), January 2011.
[I-D.ietf-avtext-client-to-mixer-audio-level]
Lennox, J., Ivov, E., and E. Marocco, "A Real-Time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication",
draft-ietf-avtext-client-to-mixer-audio-level-00 (work in
progress), February 2011.
[I-D.ietf-avtext-mixer-to-client-audio-level]
Ivov, E., Marocco, E., and J. Lennox, "A Real-Time
Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication",
draft-ietf-avtext-mixer-to-client-audio-level-00 (work in
progress), February 2011.
[RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
for Generic Forward Error Correction", RFC 2733,
December 1999.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736,
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.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[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.
Perkins, et al. Expires September 8, 2011 [Page 14]
Internet-Draft RTP for RTC-Web March 2011
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, July 2007.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[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.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, March 2009.
[RFC5484] Singer, D., "Associating Time-Codes with RTP Streams",
RFC 5484, March 2009.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP
Control Protocol (RTCP)", RFC 5968, September 2010.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
13.2. Informative 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.
[rtp-tfrc]
Gharai, L., "RTP with TCP Friendly Rate Control
Perkins, et al. Expires September 8, 2011 [Page 15]
Internet-Draft RTP for RTC-Web March 2011
(draft-gharai-avtcore-rtp-tfrc-00)", March 2011.
Authors' Addresses
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Joerg Ott
Aalto University
School of Electrical Engineering
Espoo 02150
Finland
Email: jorg.ott@aalto.fi
Perkins, et al. Expires September 8, 2011 [Page 16]