Network Working Group M. Westerlund
Request for Comments: 5117 Ericsson
Category: Informational S. Wenger
Nokia
January 2008
RTP Topologies
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Abstract
This document discusses multi-endpoint topologies used in Real-time
Transport Protocol (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 ....................................................2
2. Definitions .....................................................3
2.1. Glossary ...................................................3
2.2. Indicating Requirement Levels ..............................3
3. Topologies ......................................................3
3.1. Point to Point .............................................4
3.2. Point to Multipoint Using Multicast ........................5
3.3. Point to Multipoint Using the RFC 3550 Translator ..........6
3.4. Point to Multipoint Using the RFC 3550 Mixer Model .........9
3.5. Point to Multipoint Using Video Switching MCUs ............11
3.6. Point to Multipoint Using RTCP-Terminating MCU ............12
3.7. Non-Symmetric Mixer/Translators ...........................13
3.8. Combining Topologies ......................................14
4. Comparing Topologies ...........................................15
4.1. Topology Properties .......................................15
4.1.1. All to All Media Transmission ......................15
4.1.2. Transport or Media Interoperability ................16
4.1.3. Per Domain Bit-Rate Adaptation .....................16
4.1.4. Aggregation of Media ...............................16
4.1.5. View of All Session Participants ...................16
4.1.6. Loop Detection .....................................17
4.2. Comparison of Topologies ..................................17
5. Security Considerations ........................................17
6. Acknowledgements ...............................................19
7. References .....................................................19
7.1. Normative References ......................................19
7.2. Informative References ....................................20
1. Introduction
When working on the Codec Control Messages [CCM], considerable
confusion was noticed in the community with respect to terms such as
Multipoint Control Unit (MCU), Mixer, and Translator, and their usage
in various topologies. This document tries to address this confusion
by providing a common information basis for future discussion and
specification work. It attempts to clarify and explain sections of
the Real-time Transport Protocol (RTP) spec [RFC3550] in an informal
way. It is not intended to update or change what is normatively
specified within RFC 3550.
When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was
developed the main emphasis lay 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 may implement Mixer or Translator (in RTP [RFC3550] terminology)
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functionality and signalling support. They may also contain
additional application functionality. This document focuses on the
media transport aspects of the MCU that can be realized using RTP, as
discussed below. Further considered are the properties of Mixers and
Translators, and how some types of deployed MCUs deviate from these
properties.
2. Definitions
2.1. Glossary
ASM - Any Source Multicast
AVPF - The Extended RTP Profile for RTCP-based Feedback
CSRC - Contributing Source
Link - The data transport to the next IP hop
MCU - Multipoint Control Unit
Path - The concatenation of multiple links, resulting in an
end-to-end data transfer.
PtM - Point to Multipoint
PtP - Point to Point
SSM - Source-Specific Multicast
SSRC - Synchronization Source
2.2. Indicating Requirement Levels
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].
The RFC 2119 language is used in this document to highlight those
important requirements and/or resulting solutions that are necessary
to address the issues raised in this document.
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 deployed system models
as used in many H.323 [H323] video conferences, where both the media
streams and the RTP Control Protocol (RTCP) control traffic terminate
at the MCU. In these two cases, the media sender does not receive
the (unmodified or Translator-modified) Receiver Reports from all
sources (which it needs to interpret based on Synchronization Source
(SSRC) values) and therefore has no full information about all the
endpoint's situation as reported in RTCP Receiver Reports (RRs).
More topologies can be constructed by combining any of the models;
see Section 3.8.
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The topologies may be referenced in other documents by a shortcut
name, indicated by the prefix "Topo-".
For each of the RTP-defined topologies, we discuss how RTP, RTCP, and
the carried media are handled. With respect to RTCP, we also
introduce the handling of RTCP feedback messages as defined in
[RFC4585] and [CCM]. Any important differences between the two will
be illuminated in the discussion.
3.1. Point to Point
Shortcut name: Topo-Point-to-Point
The Point to Point (PtP) topology (Figure 1) consists of two
endpoints, communicating using unicast. Both RTP and RTCP traffic
are conveyed endpoint-to-endpoint, using unicast traffic only (even
if, in exotic cases, 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.
Note that an endpoint can still use multiple RTP Synchronization
Sources (SSRCs) in an RTP session.
RTCP feedback messages for the indicated SSRCs are communicated
directly between the endpoints. Therefore, this topology poses
minimal (if any) issues for any feedback messages.
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3.2. Point to Multipoint Using Multicast
Shortcut name: Topo-Multicast
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Figure 2 - Point to Multipoint Using Multicast
Point to Multipoint (PtM) is defined here as using a 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, or
o when a participant does not wish to receive the traffic for a
specific multicast group and therefore has not subscribed to the
IP-multicast group in question. This is for the cases where a
multi-media session is distributed using two or more multicast
groups.
In the above context, "traffic" encompasses both RTP and RTCP
traffic. The number of participants can vary between one and many,
as RTP and RTCP scale to very large multicast groups (the theoretical
limit of the number of participants in a single RTP session is
approximately two billion). The above can be realized using Any
Source Multicast (ASM). Source-Specific Multicast (SSM) may be also
be used with RTP. However, then only the designated source may reach
all receivers. Please review [RTCP-SSM] for how RTCP can be made to
work in combination with SSM.
This document is primarily interested in that subset of multicast
sessions wherein the number of participants in the multicast group is
so low that it allows the participants to use early or immediate
feedback, as defined in AVPF [RFC4585]. This document refers to
those groups as "small multicast groups".
RTCP feedback messages in multicast will, like media, reach everyone
(subject to packet losses and multicast group subscription).
Therefore, the feedback suppression mechanism discussed in [RFC4585]
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is required. Each individual node needs to process every feedback
message it receives to determine if it is affected or if the feedback
message applies only to some other participant.
3.3. Point to Multipoint Using the RFC 3550 Translator
Shortcut name: Topo-Translator
Two main categories of Translators can be distinguished:
Transport Translators (Topo-Trn-Translator) 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 (in gateways). Of the transport Translators, this memo
is primarily interested in those that use RTP on both sides, and this
is assumed henceforth. Translators that bridge between different
protocol worlds need to be concerned about the mapping of the
SSRC/CSRC (Contributing Source) concept to the non-RTP protocol.
When designing a Translator to a non-RTP-based media transport, one
crucial factor lies in how to handle different sources and their
identities. This problem space is not discussed henceforth.
Media Translators (Topo-Media-Translator), 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 it can go all the way to a full transcoding
(down to the sample level or equivalent) 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 separate them from
Mixers. For each media stream that the Translator receives, it
generates an individual stream in the other domain. A Translator
always keeps the SSRC for a stream across the translation, where a
Mixer can select a media stream, or send them out mixed, always under
its own SSRC, using the CSRC field to indicate the source(s) of the
content.
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The RTCP translation process can be trivial, for example, when
Transport Translators just need to adjust IP addresses, or they can
be quite complex as 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)
participants B and D to take part in a multicast 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 network path, the Translator may perform
media transcoding to allow the traffic received from the other
participants to reach B without overloading the path.
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.
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As specified in Section 7.1 of [RFC3550], the SSRC space is common
for all participants in the session, independent of on which side
they are of the Translator. Therefore, it is the responsibility of
the participants to run SSRC collision detection, and the SSRC is a
field the Translator should not change.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 4 - RTP Translator (Relay) with Only Unicast Paths
Another Translator scenario is depicted in Figure 4. Herein, the
Translator connects multiple users of a conference through unicast.
This can be implemented using a very simple transport Translator,
which in this document is 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 infrastructure.
A Translator normally does not use an SSRC of its own, and is not
visible as an active participant in the session. One exception can
be conceived when a Translator acts as a quality monitor that sends
RTCP reports and therefore is required to have an SSRC. Another
example is the case when a Translator is prepared to use RTCP
feedback messages. This may, for example, occur when it suffers
packet loss of important video packets and wants to trigger repair by
the media sender, by sending feedback messages. To be able to do
this it needs to have a unique SSRC.
A media Translator may in some cases act on behalf of the "real"
source and respond to RTCP feedback messages. This may occur, for
example, when a receiver requests a bandwidth reduction, and the
media Translator has not detected any congestion or other reasons for
bandwidth reduction between the media source and itself. In that
case, it is sensible that the media Translator reacts to the codec
control messages itself, for example, by transcoding to a lower media
rate. If it were not reacting, the media quality in the media
sender's domain may suffer, as a result of the media sender adjusting
its media rate (and quality) according to the needs of the slow
past-Translator endpoint, at the expense of the rate and quality of
all other session participants.
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In general, a Translator implementation should consider which RTCP
feedback messages or codec-control messages it needs to understand in
relation to the functionality of the Translator itself. This is
completely in line with the requirement to also translate RTCP
messages between the domains.
3.4. Point to Multipoint Using the RFC 3550 Mixer Model
Shortcut name: Topo-Mixer
A Mixer is a middlebox that aggregates multiple RTP streams, which
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 the 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 over the PtP or PtM paths, 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 CSRC Count (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
unique SSRC that identifies the Mixer's stream. The CSRC are
forwarded between the two domains to allow for loop detection and
identification of sources that are part of the global session. Note
that Section 7.1 of RFC 3550 requires the SSRC space to be shared
between domains for these reasons.
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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. In its role as a media
sender, it should also generate Sender Reports for those media
streams sent. As specified in Section 7.3 of RFC 3550, a Mixer must
not forward RTCP unaltered between the two domains.
The Mixer depicted in Figure 5 is involved in three domains that need
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 content received from D, and vice
versa. However, the Mixer only needs one SSRC in each domain that is
the receiving entity and transmitter of mixed content.
In the multicast domain, a Mixer still needs to provide a mixed view
of the other domains. This makes the Mixer simpler to implement and
avoids any issues with advanced RTCP handling or loop detection,
which would be problematic if the Mixer were providing non-symmetric
behavior. Please see Section 3.7 for more discussion on this topic.
A Mixer is responsible for receiving RTCP feedback messages and
handling 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 RTCP feedback 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
domain.
When replacing the multicast network in Figure 5 (to the left of the
Mixer) with individual unicast paths as depicted in Figure 6, the
Mixer model is very similar to the one discussed in Section 3.6
below. Please see the discussion in Section 3.6 about the
differences between these two models.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 6 - RTP Mixer with Only Unicast Paths
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3.5. Point to Multipoint Using Video Switching MCUs
Shortcut name: Topo-Video-switch-MCU
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
Figure 7 - Point to Multipoint Using a Video Switching MCU
This PtM topology is still deployed today, although the
RTCP-terminating MCUs, as discussed in the next section, are perhaps
more common. This topology, as well as the following one, reflect
today's lack of wide availability of IP multicast technologies, 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 switching 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
presentation mode or 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, or resolution. However, it still
may indicate the original sender of the content through the SSRC. In
this case, the values of the CC and CSRC fields are retained.
If not terminating RTP, the RTCP Sender Reports are forwarded for the
currently selected sender. All RTCP Receiver Reports are freely
forwarded 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 most of the attributes of a Translator.
However, its stream selection is a mixing behavior. This behavior
has some RTP and RTCP issues associated with it. The suppression of
all but one media stream results in most participants seeing 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. Consequently, the media senders that are not
currently forwarded receive a view of the session that indicates
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their media streams disappear somewhere en route. This makes the use
of RTCP for congestion control, or any type of quality reporting,
very problematic.
To avoid the aforementioned issues, the MCU needs to implement two
features. First, it needs to act as a Mixer (see Section 3.4) and
forward the selected media stream under its own SSRC and with the
appropriate CSRC values. Second, the MCU needs to modify the RTCP
RRs it forwards between the domains. As a result, it is RECOMMENDED
that one implement a centralized video switching conference using a
Mixer according to RFC 3550, instead of the shortcut implementation
described here.
3.6. Point to Multipoint Using RTCP-Terminating MCU
Shortcut name: Topo-RTCP-terminating-MCU
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 8 - Point to Multipoint Using Content Modifying MCUs
In this PtM scenario, each participant runs an RTP point-to-point
session between itself and the MCU. This is a very commonly deployed
topology in multipoint video conferencing. 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
paths, which are part of the same conference session.
In case a), the MCU may modify the content in bit-rate, encoding, or
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 sessions typically use a
different SSRC, and may well use a different payload type (PT), even
if this different PT happens to be mapped to the same media type.
This is a result of the individually negotiated session for each
participant.
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In case b), the MCU is the content source as it mixes the content and
then encodes it for transmission to a participant. According to RTP
[RFC3550], the SSRC of the contributors are to be signalled using the
CSRC/CC mechanism. In practice, today, most deployed MCUs do not
implement this feature. Instead, the identification of the
participants whose content is included in the Mixer's output is not
indicated through any explicit RTP mechanism. That is, most deployed
MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby
indicating no available CSRC information, even if they could identify
the content sources as suggested in RTP.
The main feature that sets this topology apart from what RFC 3550
describes is the breaking of the common RTP session across the
centralized device, such as the MCU. This results in the loss of
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 deprives the client of information related to
currently active senders in a machine-usable way, thus preventing
clients from indicating currently active speakers in user
interfaces, etc.
Note that deployed MCUs (and endpoints) rely on signalling layer
mechanisms for the identification of the contributing sources, for
example, a SIP conferencing package [RFC4575]. This alleviates, to
some extent, the aforementioned issues resulting from ignoring RTP's
CSRC mechanism.
As a result of the shortcomings of this topology, it is RECOMMENDED
to instead implement the Mixer concept as specified by RFC 3550.
3.7. Non-Symmetric Mixer/Translators
Shortcut name: Topo-Asymmetric
It is theoretically possible to construct an MCU that is a Mixer in
one direction and a Translator in another. The main reason to
consider this would be to allow topologies similar to Figure 5, where
the Mixer does not need to mix in the direction from B or D towards
the multicast domains with A and C. Instead, the media streams from
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B and D are forwarded without changes. Avoiding this mixing would
save media processing resources that perform the mixing in cases
where it isn't needed. However, there would still be a need to mix
B's stream towards D. Only in the direction B -> multicast domain or
D -> multicast domain would it be possible to work as a Translator.
In all other directions, it would function as a Mixer.
The Mixer/Translator would still need to process and change the RTCP
before forwarding it in the directions of B or D to the multicast
domain. One issue is that A and C do not know about the mixed-media
stream the Mixer sends to either B or D. Thus, any reports related
to these streams must be removed. Also, receiver reports related to
A and C's media stream would be missing. To avoid A and C thinking
that B and D aren't receiving A and C at all, the Mixer needs to
insert its Receiver Reports for the streams from A and C into B and
D's Sender Reports. In the opposite direction, the Receiver Reports
from A and C about B's and D's stream also need to be aggregated into
the Mixer's Receiver Reports sent to B and D. Since B and D only
have the Mixer as source for the stream, all RTCP from A and C must
be suppressed by the Mixer.
This topology is so problematic and it is so easy to get the RTCP
processing wrong, that it is NOT RECOMMENDED to implement this
topology.
3.8. Combining Topologies
Topologies can be combined and linked to each other using Mixers or
Translators. However, care must be taken in handling the SSRC/CSRC
space. A Mixer will not forward RTCP from sources in other domains,
but will instead generate its own RTCP packets for each domain it
mixes into, including the necessary Source Description (SDES)
information for both the CSRCs and the SSRCs. Thus, in a mixed
domain, the only SSRCs seen will be the ones present in the domain,
while there can be CSRCs from all the domains connected together with
a combination of Mixers and Translators. The combined SSRC and CSRC
space is common over any Translator or Mixer. This is important to
facilitate loop detection, something that is likely to be even more
important in combined topologies due to the mixed behavior between
the domains. Any hybrid, like the Topo-Video-switch-MCU or
Topo-Asymmetric, requires considerable thought on how RTCP is dealt
with.
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4. Comparing Topologies
The topologies discussed in Section 3 have different properties.
This section first lists these properties and then maps the different
topologies to them. Please note that even if a certain property is
supported within a particular topology concept, the necessary
functionality may, in many cases, be optional to implement.
4.1. Topology Properties
4.1.1. All to All Media Transmission
Multicast, at least Any Source Multicast (ASM), provides the
functionality that everyone may send to, or receive from, everyone
else within the session. MCUs, Mixers, and Translators may all
provide that functionality at least on some basic level. However,
there are some differences in which type of reachability they
provide.
The transport Translator function called "relay", in Section 3.3, is
the one that provides the emulation of ASM that is closest to true
IP-multicast-based, all to all transmission. Media Translators,
Mixers, and the MCU variants do not provide a fully meshed forwarding
on the transport level; instead, they only allow limited forwarding
of content from the other session participants.
The "all to all media transmission" requires that any media
transmitting entity considers the path to the least capable receiver.
Otherwise, the media transmissions may overload that path.
Therefore, a media sender needs to monitor the path from itself to
any of the participants, to detect the currently least capable
receiver, and adapt its sending rate accordingly. As multiple
participants may send simultaneously, the available resources may
vary. RTCP's Receiver Reports help performing this monitoring, at
least on a medium time scale.
The transmission of RTCP automatically adapts to any changes in the
number of participants due to the transmission algorithm, defined in
the RTP specification [RFC3550], and the extensions in AVPF [RFC4585]
(when applicable). That way, the resources utilized for RTCP stay
within the bounds configured for the session.
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4.1.2. Transport or Media Interoperability
Translators, Mixers, and RTCP-terminating MCU all allow changing the
media encoding or the transport to other properties of the other
domain, thereby providing extended interoperability in cases where
the participants lack a common set of media codecs and/or transport
protocols.
4.1.3. Per Domain Bit-Rate Adaptation
Participants are most likely to be connected to each other with a
heterogeneous set of paths. This makes congestion control in a Point
to Multipoint set problematic. For the ASM and "relay" scenario,
each individual sender has to adapt to the receiver with the least
capable path. This is no longer necessary when Media Translators,
Mixers, or MCUs are involved, as each participant only needs to adapt
to the slowest path within its own domain. The Translator, Mixer, or
MCU topologies all require their respective outgoing streams to
adjust the bit-rate, packet-rate, etc., to adapt to the least capable
path in each of the other domains. That way one can avoid lowering
the quality to the least-capable participant in all the domains at
the cost (complexity, delay, equipment) of the Mixer or Translator.
4.1.4. Aggregation of Media
In the all to all media property mentioned above and provided by ASM,
all simultaneous media transmissions share the available bit-rate.
For participants with limited reception capabilities, this may result
in a situation where even a minimal acceptable media quality cannot
be accomplished. This is the result of multiple media streams
needing to share the available resources. The solution to this
problem is to provide for a Mixer or MCU to aggregate the multiple
streams into a single one. This aggregation can be performed
according to different methods. Mixing or selection are two common
methods.
4.1.5. View of All Session Participants
The RTP protocol includes functionality to identify the session
participants through the use of the SSRC and CSRC fields. In
addition, it is capable of carrying some further identity information
about these participants using the RTCP Source Descriptors (SDES).
To maintain this functionality, it is necessary that RTCP is handled
correctly in domain bridging function. This is specified for
Translators and Mixers. The MCU described in Section 3.5 does not
entirely fulfill this. The one described in Section 3.6 does not
support this at all.
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4.1.6. Loop Detection
In complex topologies with multiple interconnected domains, it is
possible to form media loops. RTP and RTCP support detecting such
loops, as long as the SSRC and CSRC identities are correctly set in
forwarded packets. It is likely that loop detection works for the
MCU, described in Section 3.5, at least as long as it forwards the
RTCP between the participants. However, the MCU in Section 3.6 will
definitely break the loop detection mechanism.
4.2. Comparison of Topologies
The table below attempts to summarize the properties of the different
topologies. The legend to the topology abbreviations are:
Topo-Point-to-Point (PtP), Topo-Multicast (Multic),
Topo-Trns-Translator (TTrn), Topo-Media-Translator (including
Transport Translator) (MTrn), Topo-Mixer (Mixer), Topo-Asymmetric
(ASY), Topo-Video-switch-MCU (MCUs), and Topo-RTCP-terminating-MCU
(MCUt). In the table below, Y indicates Yes or full support, N
indicates No support, (Y) indicates partial support, and N/A
indicates not applicable.
Property PtP Multic TTrn MTrn Mixer ASY MCUs MCUt
------------------------------------------------------------------
All to All media N Y Y Y (Y) (Y) (Y) (Y)
Interoperability N/A N Y Y Y Y N Y
Per Domain Adaptation N/A N N Y Y Y N Y
Aggregation of media N N N N Y (Y) Y Y
Full Session View Y Y Y Y Y Y (Y) N
Loop Detection Y Y Y Y Y Y (Y) N
Please note that the Media Translator also includes the transport
Translator functionality.
5. Security Considerations
The use of Mixers and Translators has impact on security and the
security functions used. The primary issue is that both Mixers and
Translators modify packets, thus preventing the use of integrity and
source authentication, unless they are trusted devices that take part
in the security context, e.g., the device can send Secure Realtime
Transport Protocol (SRTP) and Secure Realtime Transport Control
Protocol (SRTCP) [RFC3711] packets to session endpoints. If
encryption is employed, the media Translator and Mixer need to be
able to decrypt the media to perform its function. A transport
Translator may be used without access to the encrypted payload in
cases where it translates parts that are not included in the
encryption and integrity protection, for example, IP address and UDP
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port numbers in a media stream using SRTP [RFC3711]. However, in
general, the Translator or Mixer needs to be part of the signalling
context and get the necessary security associations (e.g., SRTP
crypto contexts) established with its RTP session participants.
Including the Mixer and Translator in the security context allows the
entity, if subverted or misbehaving, to perform a number of very
serious attacks as it has full access. It can perform all the
attacks possible (see RFC 3550 and any applicable profiles) as if the
media session were not protected at all, while giving the impression
to the session participants that they are protected.
Transport Translators have no interactions with cryptography that
works above the transport layer, such as SRTP, since that sort of
Translator leaves the RTP header and payload unaltered. Media
Translators, on the other hand, have strong interactions with
cryptography, since they alter the RTP payload. A media Translator
in a session that uses cryptographic protection needs to perform
cryptographic processing to both inbound and outbound packets.
A media Translator may need to use different cryptographic keys for
the inbound and outbound processing. For SRTP, different keys are
required, because an RFC 3550 media Translator leaves the SSRC
unchanged during its packet processing, and SRTP key sharing is only
allowed when distinct SSRCs can be used to protect distinct packet
streams.
When the media Translator uses different keys to process inbound and
outbound packets, each session participant needs to be provided with
the appropriate key, depending on whether they are listening to the
Translator or the original source. (Note that there is an
architectural difference between RTP media translation, in which
participants can rely on the RTP Payload Type field of a packet to
determine appropriate processing, and cryptographically protected
media translation, in which participants must use information that is
not carried in the packet.)
When using security mechanisms with Translators and Mixers, it is
possible that the Translator or Mixer could create different security
associations for the different domains they are working in. Doing so
has some implications:
First, it might weaken security if the Mixer/Translator accepts a
weaker algorithm or key in one domain than in another. Therefore,
care should be taken that appropriately strong security parameters
are negotiated in all domains. In many cases, "appropriate"
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translates to "similar" strength. If a key management system does
allow the negotiation of security parameters resulting in a different
strength of the security, then this system SHOULD notify the
participants in the other domains about this.
Second, the number of crypto contexts (keys and security related
state) needed (for example, in SRTP [RFC3711]) may vary between
Mixers and Translators. A Mixer normally needs to represent only a
single SSRC per domain and therefore needs to create only one
security association (SRTP crypto context) per domain. In contrast,
a Translator needs one security association per participant it
translates towards, in the opposite domain. Considering Figure 3,
the Translator needs two security associations towards the multicast
domain, one for B and one for D. It may be forced to maintain a set
of totally independent security associations between itself and B and
D respectively, so as to avoid two-time pad occurrences. These
contexts must also be capable of handling all the sources present in
the other domains. Hence, using completely independent security
associations (for certain keying mechanisms) may force a Translator
to handle N*DM keys and related state; where N is the total number of
SSRCs used over all domains and DM is the total number of domains.
There exist a number of different mechanisms to provide keys to the
different participants. One example is the choice between group keys
and unique keys per SSRC. The appropriate keying model is impacted
by the topologies one intends to use. The final security properties
are dependent on both the topologies in use and the keying
mechanisms' properties, and need to be considered by the application.
Exactly which mechanisms are used is outside of the scope of this
document.
6. Acknowledgements
The authors would like to thank Bo Burman, Umesh Chandra, Roni Even,
Keith Lantz, Ladan Gharai, Geoff Hunt, and Mark Baugher for their
help in reviewing this document.
7. References
7.1. Normative References
[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.
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[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol
(SRTP)", RFC 3711, March 2004.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, August 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J.
Rey, "Extended RTP Profile for Real-time Transport
Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC
4585, July 2006.
7.2. Informative References
[CCM] Wenger, S., Chandra, U., Westerlund, M., Burman, B.,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", Work in Progress, July 2007.
[H323] ITU-T Recommendation H.323, "Packet-based multimedia
communications systems", June 2006.
[RTCP-SSM] J. Ott, J. Chesterfield, E. Schooler, "RTCP Extensions
for Single-Source Multicast Sessions with Unicast
Feedback," Work in Progress, March 2007.
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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