Network Working Group Magnus Westerlund
INTERNET-DRAFT Ericsson
Expires: November 2006 Stephan Wenger
Nokia
May 24, 2006
RTP Topologies
draft-wenger-avt-topologies-00.txt>
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Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
This document disucsses multi-endpoint topologies commonly used in
RTP based environments. In particular, centralized topologies
commonly employed in the video conferencing industry are mapped to
the RTP terminology.
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TABLE OF CONTENTS
1. Introduction....................................................3
2. Definitions.....................................................3
2.1. Glossary...................................................4
2.2. Terminology................................................4
2.3. Topologies.................................................5
2.3.1. Point to Point........................................5
2.3.2. Point to Multi-point using Multicast..................6
2.3.3. Point to Multipoint using the RFC 3550 translator.....7
2.3.4. Point to Multipoint using the RFC 3550 mixer model....9
2.3.5. Point to Multipoint using video switching MCU........11
2.3.6. Point to Multipoint using RTCP-terminating MCU.......12
2.3.7. Combining Topologies.................................13
3. Acknowledgements...............................................13
4. References.....................................................14
4.1. Normative references......................................14
4.2. Informative references....................................14
5. Authors' Addresses.............................................14
6. List of Changes relative to previous drafts....................15
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1. Introduction
When working on the Codec Control Messages [CCM], we noticed a
considerable confusion in the community with respect to terms such as
MCU, mixer, and translator. In the process of writing, we became
increasingly unsure of our own understanding, and therefore added
what became the core of this draft to the CCM draft. Later, it was
found that this information has its own value, and was ''outsourced''
from the CCM draft into the present memo.
It could be argued that this document clarifies and explains sections
of the RTP spec [RFC3550], and is therefore of informational nature.
In this case, the present memo may end up as an informational RFC.
When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was
developed, the main emphasis lied in the efficient support of point-
to-point and small multipoint scenarios without centralized
multipoint control. However, in practice, many small multipoint
conferences operate utilizing devices known as Multipoint Control
Units (MCUs). MCUs comprise mixers and translators (in RTP [RFC3550]
terminology), but also signalling support. Long standing experience
of the conversational video conferencing industry suggests that there
is a need for a few additional feedback messages, to efficiently
support MCU-based multipoint conferencing. Some of the messages have
applications beyond centralized multipoint, and this is indicated in
the description of the message.
Some of the messages defined here are forward only, in that they do
not require an explicit acknowledgement. Other messages require
acknowledgement, leading to a two way communication model that could
suggest to some to be useful for control purposes. It is not the
intention of this memo to open up the use of RTCP to generalized
control protocol functionality. All mentioned messages have
relatively strict real-time constraints and are of transient nature,
which make the use of more traditional control protocol means, such
as SIP re-invites, undesirable. Furthermore, all messages are of a
very simple format that can be easily processed by an RTP/RTCP
sender/receiver. Finally, all messages infer only to the RTP stream
they are related to, and not to any other property of a communication
system.
2. Definitions
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2.1. Glossary
ASM - Asynchronous Multicast
AVPF - The Extended RTP Profile for RTCP-based Feedback
MCU - Multipoint Control Unit
PtM - Point to Multipoint
PtP - Point to Point
2.2. 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 RFC 2119 [RFC2119].
Message:
Codepoint defined by this specification, of one of the
following types:
Request:
Message that requires Acknowledgement
Acknowledgment:
Message that answers a Request
Command:
Message that forces the receiver to an action
Indication:
Message that reports a situation
Notification:
See Indication.
Note that, with the exception of ''Notification'', this terminology
is in alignment with ITU-T Rec. H.245.
Decoder Refresh Point:
A bit string, packetised in one or more RTP packets, which
completely resets the decoder to a known state. Typical
examples of Decoder Refresh Points are H.261 Intra pictures
and H.264 IDR pictures. However, there are also much more
complex decoder refresh points.
Typical examples for "hard" decoder refresh points are Intra
pictures in H.261, H.263, MPEG 1, MPEG 2, and MPEG-4 part 2,
and IDR pictures in H.264. "Gradual" decoder refresh points
may also be used. While both "hard" and "gradual" decoder
refresh points are acceptable in the scope of this
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specification, in most cases the user experience will
benefit from using a "hard" decoder refresh point.
A decoder refresh point also contains all header information
above the picture layer (or equivalent, depending on the
video compression standard) that is conveyed in-band. In
H.264, for example, a decoder refresh point contains
parameter set NAL units that generate parameter sets
necessary for the decoding of the following slice/data
partition NAL units (and that are not conveyed out of band).
To the best of the author's knowledge, the term "Decoder
Refresh Point" has been formally defined only in H.264;
hence we are referring here to this video compression
standard.
Decoding:
The operation of reconstructing the media stream.
Rendering:
The operation of presenting (parts of) the reconstructed
media stream to the user.
Stream thinning:
The operation of removing some of the packets from a media
stream. Stream thinning, preferably, is performed in a
media aware fashion implying that the media packets are
removed in the order of their relevance to the reproductive
quality. However even when employing media-aware stream
thinning, most media streams quickly lose quality when
subject to increasing levels of thinning. Media-unaware
stream thinning leads to even worse quality degradation.
2.3. Topologies
This subsection defines several basic topologies that are relevant
for codec control. The first four relate to the RTP system model
utilizing multicast and/or unicast, as envisioned in RFC 3550. The
last two topologies, in contrast, describe the widely deployed system
model as used in most H.323 video conferences, where both the media
streams and the RTCP control traffic terminate at the MCU. More
topologies can be constructed by combining any of the models, see
Section 2.3.7.
2.3.1. Point to Point
The Point to Point (PtP) topology (Figure 1) consists of two end-
points with unicast capabilities between them. Both RTP and RTCP
traffic are conveyed endpoint to endpoint using unicast traffic only
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(even if this unicast traffic happens to be conveyed over an IP-
multicast address).
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 1 - Point to Point
The main property of this topology is that A sends to B and only B,
while B sends to A and only A. This avoids all complexities of
handling multiple endpoints and combining the requirements from them.
Do note that an endpoint may still use multiple RTP Synchronization
Sources (SSRCs) in an RTP session.
2.3.2. Point to Multi-point using Multicast
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Figure 2 - Point to Multipoint using Multicast
We define Point to Multipoint (PtM) using multicast topology as a
transmission model in which traffic from any participant reaches all
the other participants, except for cases such as
o packet loss occurs,
o a participant participant does not wish to receive the traffic
from a certain other participant, and therefore has not
subscribed to the IP multicast group in question.
In this sense, "traffic" encompasses both RTP and RTCP traffic. The
number of participants can be between one and many -- as RTP and RTCP
scales to very large multicast groups (the theoretical limit of RTP
is approximately two billion participants).
This draft is primarily interested in the subset of multicast session
where the number of participants in the multicast group allows the
participants to use early or immediate feedback as defined in AVPF.
This document refers to those groups as as "small multicast groups".
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2.3.3. Point to Multipoint using the RFC 3550 translator
Two main categories of Translators can be distinguished.
Transport Translators do not modify the media stream itself, but are
concerned with transport parameters. Transport parameters, in the
sense of this section, comprise the transport addresses to bridge
different domains, and the media packetization to allow other
transport protocols to be interconnected to a session (gateways).
Media Translators, in contrast, modify the media stream itself. This
process is commonly known as transcoding. The modification of the
media stream can be as small as removing parts of the stream, and can
go all the way to a full transcoding utilizing a different media
codec. Media translators are commonly used to connect entities
without a common interoperability point.
Stand-alone Media Translators are rare. Most commonly, a combination
of Transport and Media Translators are used to translate both the
media stream and the transport aspects of a stream between two
transport domains (or clouds).
Both Translator types share common attributes that separates them
from mixers. For each media stream that the translator receives, it
generates an individual stream in the other domain. However, a
translator maintains a complete view of all existing participants
between both domains. Therefore, the SSRC space is shared across the
two domains.
The RTCP translation process can be trivial, for example when
Transport translators just need to adjust IP addresses, and can be
quite complex in the case of media translators. See section 7.2 of
[RFC3550].
+-----+
+---+ / \ +------------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Translator |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+
+-----+
Figure 3 - Point to Multipoint using a Translator
Figure 3 depicts an example of a Transport Translator performing at
least IP address translation. It allows the (non multicast capable)
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participants B and D to take part in a multicasted session by having
the translator forward their unicast traffic to the multicast
addresses in use, and vice versa. It must also forward B's traffic
to D and vice versa, to provide each of B and D with a complete view
of the session.
If B were behind a limited link, the translator may perform media
transcoding to allow the traffic received from the other participants
to reach B without overloading the link.
When in the example depicted in Figure 3 the translator acts only as
a Transport Translator, then the RTCP traffic can simply be
forwarded, similar to the media traffic. However, when media
translation occurs, the translator's task becomes substantially more
complex even with respect to the RTCP traffic. In this case, the
translator needs to rewrite B's RTCP receiver report, before
forwarding them to D and the multicast network. The rewriting is
needed as the stream received by B is not the same stream as the
other participants receive. For example, the number of packets
transmitted to B may be lower than what D receives, due to the
different media format. Therefore, if the receiver reports were
forwarded without changes, the extended highest sequence number would
indicate that B were substantially behind in reception -- while it
most likely it would not be. Therefore, the translator must translate
that number to a corresponding sequence number for the stream the
translator received. Similar arguments can be made for most other
fields in the RTCP receiver reports.
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 4 - MCU with RTP Translator (relay) with only unicast links
A common MCU scenario is the one depicted in Figure 4 - MCU with RTP
Translator (relay) with only unicast links. Herein, the MCU connects
multiple users of a conference through unicast. This can be
implemented using a very simple transport translator, which could be
called a relay. The relay forwards all traffic it receives, both RTP
and RTCP, to all other participants. In doing so, a multicast network
is emulated without relying on a multicast capable network structure.
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2.3.4. Point to Multipoint using the RFC 3550 mixer model
A mixer is a middlebox that aggregates multiple RTP streams that are
part of a session, by mixing the media data and generating a new RTP
stream. One common application for a mixer is to allow a participant
to receive a session with a reduced amount of resources.
+-----+
+---+ / \ +-----------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Mixer |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +-----------+ +---+
+-----+
Figure 5 - Point to Multipoint using RFC 3550 mixer model
A mixer can be viewed as a device terminating the media streams
received from other session participants. Using the media data from
the received media streams, a mixer generates a media stream that is
sent to the session participant.
The content that the mixer provides is the mixed aggregate of what
the mixer receives from the PtP or PtM links, which are part of the
same conference session.
The mixer is the content source, as it mixes the content (often in
the uncompressed domain) and then encodes it for transmission to a
participant. The CC and CSRC fields in the RTP header are used to
indicate the contributors of to the newly generated stream. The
SSRCs of the to-be-mixed streams on the mixer input appear as the
CSRCs at the mixer output. That output stream uses a new SSRC that
identifies the Mixer. The CSRC are forwarded between the two domains
to allow for loop detection and identification of sources that are
part of the global session.
The mixer is responsible for generating RTCP packets in accordance
with its role. It is a receiver and should therefore send reception
reports for the media streams it receives. As a media sender itself
it should also generate sender report for those media streams sent.
The content of the SRs created by the mixer may or may not take into
account the situation on its receiving side. Similarly, the content
of RRs created by the mixer may or may not be based on the situation
on the mixer's sending side. This is left open to the
implementation. As specified in Section 7.3 of RFC 3550, a mixer
must not forward RTCP unaltered between the two domains.
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The mixer depicted in Figure 5 has three domains that needs to be
separated; the multicast network, participant B and participant D.
The Mixer produces different mixed streams to B and D, as the one to
B may contain D and vice versa. However the mixer does only need one
SSRC in each domain that is the receiving entity and transmitter of
mixed content.
In the multicast domain the mixer does not provide a mixed view of
the other domains and only forwards media from B and D into the
multicast network using B's and D's SSRC.
The mixer is responsible for receiving the codec control messages and
handles them appropriately. The definition of "appropriate" depends
on the message itself and the context. In some cases, the reception
of a codec control message may result in the generation and
transmission of codec control messages by the mixer to the
participants in the other domain. In other cases, a message is
handled by the mixer itself and therefore not forwarded to any other
domains.
It should be noted that this form of mixing technology is not widely
deployed. Most multipoint video conferences used today employ one of
the models discussed in the next sections.
When replacing the multicast network in Figure 5 (to the left of the
mixer) with individual unicast links as depicted in Figure 6, the
mixer model is very similar to the one discussed in section 2.3.6
below.
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 6 - RTP Mixer with only unicast links
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2.3.5. Point to Multipoint using video switching MCU
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
Figure 7 - Point to Multipoint using relaying MCU
This PtM topology is, today, perhaps the most widely deployed one.
It reflects today's lack of wide deployment of IP multicast
technologies on IP networks and the Internet, as well as the
simplicity of content switching when compared to content mixing. The
technology is commonly implemented in what is known as ''Video
Switching MCUs''.
A video switch MCU forwards to a participant a single media stream,
selected from the available streams. The criteria for selection are
often based on voice activity in the audio-visual conference, but
other conference management mechanisms (like explicit floor control)
are known to exist as well.
The video switching MCU may also perform media translation to modify
the content in bit-rate, encoding, resolution; however it still
indicates the original sender of the content through the SSRC. The
values of the CC and CSRC fields are retained.
RTCP Sender Reports are forwarded for the currently selected sender.
All RTCP receiver reports are freely forward between the
participants. In addition, the MCU may also originate RTCP control
traffic in order to control the session and/or report on status from
its viewpoint.
The video switching MCU has mostly the attributes of a translator.
However its stream selection is a mixing behaviour. This behaviour
has some RTP and RTCP issues associated with it. The suppression of
all but one media stream results in that most participants see only a
subset of the sent media streams at any given time; often a single
stream per conference. Therefore, RTCP receiver reports only report
on these streams. In consequence, the media senders that are not
currently forwarded receive a view of the session that indicates
their media streams disappearing somewhere en route. This makes the
use of RTCP for congestion control very problematic. To avoid these
issues the MCU needs to modify the RTCP RRs.
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2.3.6. Point to Multipoint using RTCP-terminating MCU
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 8 - Point to Multipoint using content modifying MCU
In this PtM scenario, each participant runs an RTP point-to-point
session between itself and the MCU. The content that the MCU provides
to each participant is either:
a) A selection of the content received from the other participants,
or
b) The mixed aggregate of what the MCU receives from the other PtP
links, which are part of the same conference session.
In case a) the MCU may modify the content in bit-rate, encoding,
resolution. No explicit RTP mechanism is used to establish the
relationship between the original media sender and the version the
MCU sends. In other words, the outgoing session typically uses a
different SSRC, and may well use a different PT, even if this
different PT happens to be mapped to the same media type. (This is
the definition of this topology and distinguishes it from the
topologies previously discussed).
In case b) the MCU is the content source as it mixes the content and
then encodes it for transmission to a participant. The participant's
content that is included in the aggregated content is not indicated
through any explicit RTP mechanism. For example, regardless of the
number of streams that are aggregated, in the MCU generated streams
CC is zero and therefore no CSRC fields are present.
The MCU is responsible for receiving the codec control messages and
handle them appropriately. In some cases, the reception of a codec
control message may result in the generation and transmission of
codec control messages by the MCU to some or all of the other
participants.
An MCU may transparently relay some codec control messages and
intercept, modify, and (when appropriate) generate codec control
messages of its own and transmit them to the media senders.
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The main feature that sets this topology apart from what RFC 3550
describes, is the lack of an explicit RTP level indication of all
participants. If one were using the mechanisms available in RTP and
RTCP to signal this explicitly, the topology would follow the
approach of an RTP mixer. The lack of explicit indication has at
least the following potential problems:
1) Loop detection cannot be performed on the RTP level. When
carelessly connecting two misconfigured MCUs, a loop could be
generated.
2) There is no information about active media senders available in
the RTP packet. As this information is missing, receivers
cannot use it. It also deprive the participant's clients
information about who are actively sending in a machine usable
way. Thus preventing clients from doing indication of currently
active speakers in user interfaces, etc.
2.3.7. Combining Topologies
Topologies can be combined and linked to each other using mixers or
translators. Care must however be taken to how the SSRC space is
handled, mixers separate the SSRC space into two parts, while
translators maintain the space across themselves. Any hybrid, like
the video switching MCU, 2.3.5, requires considerable afterthought on
how RTCP is dealt with.
3. Security Considerations
This document does not specify any protocol mechanisms and should not
have any security issues
4. IANA considerations
None
5. Acknowledgements
The authors would like to thank N.N.
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6. References
6.1. Normative references
None.
6.2. Informative references
[CCM] Wenger, S., Chandra, U., Westerlund, M, Burman, B., ''Codec
Control Messages in the Audio-Visual Profile with Feedback
(AVPF)'', draft-wenger-avt-avpf-ext-04.txt, Work in
Progress, May 2006
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., Rey, J.,
''Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)'', RFC 4585, July
2006
Any 3GPP document can be downloaded from the 3GPP web server,
"http://www.3gpp.org/", see specifications.
7. Authors' Addresses
Magnus Westerlund
Ericsson Research
Ericsson AB
SE-164 80 Stockholm, SWEDEN
Phone: +46 8 7190000
EMail: magnus.westerlund@ericsson.com
Stephan Wenger
Nokia Corporation
P.O. Box 100
FIN-33721 Tampere
FINLAND
Phone: +358-50-486-0637
EMail: stewe@stewe.org
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8. List of Changes relative to previous drafts
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RFC Editor Considerations
The RFC editor is requested to replace all occurrences of XXXX with
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