One document matched: draft-perkins-rtcweb-rtp-usage-01.txt
Differences from draft-perkins-rtcweb-rtp-usage-00.txt
Network Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Informational M. Westerlund
Expires: December 7, 2011 Ericsson
J. Ott
Aalto University
June 5, 2011
RTP Requirements for RTC-Web
draft-perkins-rtcweb-rtp-usage-01
Abstract
This memo discusses use of RTP in the context of the RTC-Web
activity. It discusses important features of RTP that need to be
considered by other parts of the RTC-Web framework, describes which
RTP profile to use in this environment, and outlines what RTP
extensions should be supported.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on December 7, 2011.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
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include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Expected Topologies . . . . . . . . . . . . . . . . . . . 3
2. Requirements from RTP . . . . . . . . . . . . . . . . . . . . 6
2.1. RTP Multiplexing Points . . . . . . . . . . . . . . . . . 6
2.2. Signalling for RTP sessions . . . . . . . . . . . . . . . 8
2.3. (Lack of) Signalling for Payload Format Changes . . . . . 9
3. RTP Profile . . . . . . . . . . . . . . . . . . . . . . . . . 10
4. RTP and RTCP Guidelines . . . . . . . . . . . . . . . . . . . 10
5. RTP Optimizations . . . . . . . . . . . . . . . . . . . . . . 11
5.1. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 11
5.2. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 11
5.3. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 11
5.4. Generation of the RTCP Canonical Name (CNAME) . . . . . . 12
6. RTP Extensions . . . . . . . . . . . . . . . . . . . . . . . . 12
6.1. RTP Conferencing Extensions . . . . . . . . . . . . . . . 12
6.1.1. RTCP Feedback Message: Full Intra Request . . . . . . 13
6.1.2. RTCP Feedback Message: Picture Loss Indicator . . . . 13
6.1.3. RTCP Feedback Message: Temporary Maximum Media
Stream Bit Rate Request . . . . . . . . . . . . . . . 14
6.2. RTP Header Extensions . . . . . . . . . . . . . . . . . . 14
6.3. Rapid Synchronisation Extensions . . . . . . . . . . . . . 15
7. Improving RTP Transport Robustness . . . . . . . . . . . . . . 15
7.1. RTP Retransmission . . . . . . . . . . . . . . . . . . . . 15
7.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 15
8. RTP Rate Control and Media Adaptation . . . . . . . . . . . . 16
9. RTP Performance Monitoring . . . . . . . . . . . . . . . . . . 16
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 16
11. Security Considerations . . . . . . . . . . . . . . . . . . . 16
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17
13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 17
13.1. Normative References . . . . . . . . . . . . . . . . . . . 17
13.2. Informative References . . . . . . . . . . . . . . . . . . 19
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 19
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1. Introduction
This memo discusses the Real-time Transport Protocol (RTP) [RFC3551]
in the context of the RTC-Web activity. The work in the IETF Audio/
Video Transport Working Group, and it's successors, has been about
providing building blocks for real-time multimedia transport, and has
not specified who should use which building blocks. The selection of
building blocks and functionalities can really only be done in the
context of some application, for example RTC-Web. We have selected a
set of RTP features and extensions that are suitable for a number of
applications that fits the RTC-Web context. Thus applications such
as VoIP, audio and video conferencing, and on-demand multimedia
streaming are considered. Applications that rely on IP multicast
have not been considered likely to be applicable to RTC-Web, thus
extensions related to multicast have been excluded. We believe that
RTC-Web will greatly benefit in interoperability if a reasonable set
of RTP functionalities and extensions are selected. This memo is
intended as a starting point for discussion of those features in the
RTC-Web framework.
This memo is structured into different topics. For each topic, one
or several recommendations from the authors are done. When it comes
to the importance of extensions, or the need for implementation
support, we use three requirement levels to indicate the importance
of the feature to the RTC-Web specification:
REQUIRED: Functionality that is absolutely needed to make the RTC-
Web solution work well, or functionality of low complexity that
provides high value.
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 memo discusses RTP, it includes the RTP Control Protocol
(RTCP) unless explicitly stated otherwise. RTCP is a fundamental and
integral part of the RTP protocol, and is REQUIRED to be implemented.
1.1. Expected Topologies
As RTC-Web is focused on peer to peer connections established from
clients in web browsers the following topologies further discussed in
RTP Topologies [RFC5117] are primarily considered. The topologies
are depicted and briefly explaind here for ease of the reader.
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+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 1: Point to Point
The point to point topology (Figure 1) is going to be very common in
any single user to single user applications.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 2: Multi-unicast
For small multiparty sessions it is practical enough to create RTP
sessions by letting every participant send individual unicast RTP/UDP
flows to each of the other participants. This is called multi-
unicast and is unfortunately not discussed in the RTP Topologies
[RFC5117]. This topology has the benefit of not requiring central
nodes. On the downside is that it increase the used bandwidth by
requiring one copy of the media streams for each participant part of
the same session beyond the sender itself. Thus this is limited to
scenarios with few end-points unless the media is very low bandwidth.
It needs to be noted that if this topology is to be supported by the
RTC-Web framework it needs to be possible to connect one RTP session
to multiple established peer to peer flows that are individually
established.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 3: RTP Mixer with Only Unicast Paths
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An RTP mixer (Figure 3) is a centralized point that selects or mixes
content in a conference to optimize the RTP session so that each end-
point only needs connect to one entity, the mixer. The mixer also
reduces the bit-rate needs as the media sent from the mixer to the
end-point can be optimized in different ways. These optimizations
include methods like only chosing media from the currently most
active speaker or mixing together audio so that only one audio stream
is required in stead of 3 in the depicted scenario. The downside of
the mixer is that someone is required to provide the actual mixer.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 4: RTP Translator (Relay) with Only Unicast Paths
If one wants a less complex central node it is possible to use an
relay (called an Transport Translator) (Figure 4) that takes on the
role of forwarding the media to the other end-points but doesn't
perform any media processing. It simply forwards the media from all
other to all the other. Thus one endpoint A will only need to send a
media once to the relay, but it will still receive 3 RTP streams with
the media if B, C and D all currently transmitts.
+------------+
| |
+---+ | | +---+
| A |<---->| Translator |<---->| B |
+---+ | | +---+
| |
+------------+
Figure 5: Translator towards Legacy end-point
To support legacy end-point (B) that don't fulfill the requiremetns
of RTC-Web it is possible to insert a Translator (Figure 5) that
takes on the role to ensure that from A's perspective B looks like a
fully compliant end-point. Thus it is the combination of the
Translator and B that looks like the end-point B. The intention is
that the presence of the translator is transparant to A, however it
is not certain that is possible. Thus this case is include so that
it can be discussed if any mechanism specified to be used for RTC-Web
results in such issues and how to handle them.
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2. Requirements from RTP
This section discusses some requirements RTP and RTCP [RFC3550] place
on their underlying transport protocol, the signalling channel, etc.
2.1. RTP Multiplexing Points
There are three fundamental points of multiplexing within the RTP
framework:
Use of separate RTP Sessions: The first, and the most important,
multiplexing point is the RTP session. This multiplexing point
does not have an identifier within the RTP protocol itself, but
instead relies on the lower layer to separate the different RTP
sessions. This is most often done by separating different RTP
sessions onto different UDP ports, or by sending to different IP
multicast addresses. The distinguishing feature of an RTP session
is that it has a separate SSRC identifier space; a single RTP
session can span multiple transport connections provided packets
are gatewayed such that participants are known to each other.
Different RTP sessions are used to separate different types of
media within a multimedia session. For example, audio and video
flows are sent on separate RTP sessions.
Multiplexing using the SSRC within an RTP session: The second
multiplexing point is the SSRC that separates different sources of
media within a single RTP session. An example might be different
participants in a multiparty teleconference, or different camera
views of a presentation. In most cases, each participant within
an RTP session has a single SSRC, although this may change over
time if collisions are detected. However, in some more complex
scenarios participants may generate multiple media streams of the
same type simultaneously (e.g., if they have two cameras, and so
send two video streams at once) and so will have more than one
SSRC in use at once. The RTCP CNAME can be used to distinguish
between a single participant using two SSRC values (where the RTCP
CNAME will be the same for each SSRC), and two participants (who
will have different RTCP CNAMEs).
Multiplexing using the Payload Type within an RTP session: If
different media encodings of the same type are to be used at
different times within an RTP session, for example a single
participant that can switch between two different audio codecs,
the payload type is used to identify how the media from that
particular source is encoded. When changing media formats within
an RTP Session, the SSRC of the sender remains unchanged, but the
RTP Payload Type changes to indicate the change in media format.
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These multiplexing points area fundamental part of the design of RTP
and are discussed in Section 5.2 of [RFC3550]. Of special importance
is the need to separate different RTP sessions using a multiplexing
mechanism at some lower layer than RTP, rather than trying to combine
several RTP sessions into one lower layer flow.
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,
as determined by the RTP session on which it arrives. Hence, it is
important to separate such RTP session from each other. This could
of course be achieved by other methods, like tagging SSRC values with
their purpose (this is not defined in any known specification), but
there are reasons why this method isn't defined. First of all it is
not the simple solution, as this require additional signalling, and
possibly synchronization between session peers. In addition,
combining RTP sessions into a single lower-layer flow complicates
quality of service and 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.
There are also various other RTP mechanism that become problematic if
one doesn'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
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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
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.
Due to these design principle implementors of various services or
applications using RTP have not commonly violated this model. If one
choses to violate it today, one fails to achieve interoperability
with a number of existing services, applications and implementations.
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 use 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. Thus the authors considered it REQUIRED that RTP
sessions are multiplexed using a mechanism outside of RTP. The
RECOMMENDED mechanism to accomplish that would be to use unique UDP
flows. If the WG comes to a consensus that due to NAT/Firewall
traversal aspects would be greately simplified with a single flow
between peers and accept that flow based QoS can only be done on the
aggreage of all RTP sessions then the authors RECOMMEND that some
type of multiplexing layer is inserted between UDP flow and the RTP/
RTCP header to separate the RTP sessions.
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:
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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 in addition to any
cryptographic transoformation of the packet content.
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, media formats, and media format parameters: 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).
RTP Extensions: The RTP extensions one intendeds to use needs to be
agreed on, including any parameters for that extension. In some
case just to avoid spending bit-rate on features that the other
end-point will ignore. But for certain mechanisms there is
requirement for this to happen as interoperability failure
otherwise happens.
RTCP Bandwidth: 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.
These parameters are often expressed in SDP messages conveyed within
an offer/answer exchange. RTP does not depend on SDP or on the
offer/answer model, but does require all the necessary parameters to
be negotiated somehow, and provided to the RTP implementation.
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
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endpoint, that endpoint is REQUIRED to 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
during the session setup. This is needed for rapid rate adaptation.
3. RTP Profile
The "Extended Secure RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124] is REQUIRED to
be implemented. This builds on the basic RTP/AVP profile [RFC3551],
the RTP/AVPF feedback profile [RFC4585], and the secure RTP/SAVP
profile [RFC3711].
The RTP/AVPF part of RTP/SAVPF is required to get the improved RTCP
timer model, that allows more flexible transmission of RTCP packets
in 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 on which to send feedback.
This functionality is needed to make use of the RTP conferencing
extensions discussed in Section 6.1.
The RTP/SAVP part of RTP/SAVPF is for support for Secure RTP (SRTP)
[RFC3711]. 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 chosen. 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:
Guidelines for Writers of RTP Payload Format Specifications [RFC2736]
and Guidelines for Extending the RTP Control Protocol [RFC5968].
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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.
Supporting this specification is generally a simplification in code,
since it relaxes the tests in [RFC3550].
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. Supporting
this specification is generally a simplification in code, since it
relaxes the tests in [RFC3550].
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.
The reasons for using symmetric RTP is primarily to avoid issues with
NAT and Firewalls by ensuring that the flow is actually bi-
directional and thus kept alive and registred as flow the intended
recipient actually wants. In addition it saves resources in the form
of ports at the end-points, but also in the network as NAT mappings
or firewall state is not unnecessary bloated. Also the number of QoS
state are reduced.
Using Symmetric RTP and RTCP [RFC4961] is REQURIED.
5.4. Generation of the RTCP Canonical Name (CNAME)
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)" [RFC6222] 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, whether using IP
multicast, multi-unicast, or based on a centralised server. 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
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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:
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)
We note that 3 and 4 are not well utilizing the functions of RTP and
in some cases even violates the RTP specifications. Thus we
recommend that one focus on 1 and 2.
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 the Extended RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
6.1.1. RTCP Feedback Message: Full Intra Request
The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of CCM
[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 AVPF
[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 Full Intra Request above. It is
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RECOMMENDED that this feedback message is supported as a loss
tolerance mechanism.
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 CCM
[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
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.
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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,
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 (FEC)
Support of some type of FEC to combat the effects of packet loss is
beneficial, but is heavily application dependent. However, some FEC
mechanisms are encumbered.
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(tbd: add further discussion here)
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, and maintain the user experience in the presence of network
problems.
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].
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 memo 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 use 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 memo, that needs to be done
taking the RTC-Web communication model into account.
In the context of RTC-Web the actual security properties required
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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-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.
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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.
[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.
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[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, April 2011.
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
(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
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