Internet DRAFT - draft-ietf-avt-topologies
draft-ietf-avt-topologies
Network Working Group Magnus Westerlund
INTERNET-DRAFT Ericsson
Expires: April 2008 Stephan Wenger
Intended Status: Informational Nokia
26 October, 2007
RTP Topologies
draft-ietf-avt-topologies-07.txt>
Status of this Memo
By submitting this Internet-Draft, each author represents that any
applicable patent or other IPR claims of which he or she is aware
have been or will be disclosed, and any of which he or she becomes
aware will be disclosed, in accordance with Section 6 of BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.
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."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt.
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
Copyright Notice
Copyright (C) The IETF Trust (2007).
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.
Wenger, et al. [Page 1]
INTERNET-DRAFT RTP Topologies October 26, 2007
TABLE OF CONTENTS
1. Introduction....................................................3
2. Definitions.....................................................3
2.1. Glossary...................................................3
2.2. Indicating Requirement leves...............................3
3. Topologies......................................................4
3.1. Point to Point.............................................4
3.2. Point to Multi-point 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 MCU.............12
3.6. Point to Multipoint using RTCP-terminating MCU............13
3.7. Non-Symmetric Mixer/Translators...........................14
3.8. Combining Topologies......................................15
4. Comparing Topologies...........................................15
4.1. Topology Proporties.......................................16
4.1.1. All to All media transmission........................16
4.1.2. Transport or Media Interoperability..................16
4.1.3. Per Domain Bit-rate Adaptation.......................16
4.1.4. Aggregation of Media.................................17
4.1.5. View of all session participants.....................17
4.1.6. Loop Detection.......................................17
4.2. Comparision of topologies.................................18
5. Security Considerations........................................18
6. Acknowledgements...............................................20
7. IANA Considerations............................................20
8. References.....................................................21
8.1. Normative References......................................21
8.2. Informative References....................................21
9. Authors' Addresses.............................................22
Wenger, et al. Informational [Page 2]
INTERNET-DRAFT RTP Topologies October 26, 2007
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 mixers and translators (in RTP
[RFC3550] terminology), but also 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 a end-
to-end data transfer.
PtM - Point to Multipoint
PtP - Point to Point
SSM - Source-Specific Multicast
SSRC - Synchronization Source
2.2. Indicating Requirement leves
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
Wenger, et al. Informational [Page 3]
INTERNET-DRAFT RTP Topologies October 26, 2007
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 interprete based on
Synchronization Source (SSRC) values), and therefore has no full
information about all the endpoint's situation as reported in RTCP-
Receiver Reports (RR)s. More topologies can be constructed by
combining any of the models; see Section 3.8.
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 message 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 end-
points, 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).
Wenger, et al. Informational [Page 4]
INTERNET-DRAFT RTP Topologies October 26, 2007
+---+ +---+
| 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.
3.2. Point to Multi-point 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 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 scales to very large multicast groups (the
Wenger, et al. Informational [Page 5]
INTERNET-DRAFT RTP Topologies October 26, 2007
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 draft 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] 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 which 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 consists in how to handle
different sources and their identity. 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 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.
Wenger, et al. Informational [Page 6]
INTERNET-DRAFT RTP Topologies October 26, 2007
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, allways under its own SSRC, using the CSRC field to
indicate the source(s) of the content.
The RTCP translation process can be trivial---for example, when
Transport translators just need to adjust IP addresses---or 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) 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
Wenger, et al. Informational [Page 7]
INTERNET-DRAFT RTP Topologies October 26, 2007
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.
As specified in Section 7.1 of [RFC3550], the SSRC space is common
for all participants in the session, independent of 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 it 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.
Wenger, et al. Informational [Page 8]
INTERNET-DRAFT RTP Topologies October 26, 2007
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.
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 translate also 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 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.
Wenger, et al. Informational [Page 9]
INTERNET-DRAFT RTP Topologies October 26, 2007
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 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.
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.
Wenger, et al. Informational [Page 10]
INTERNET-DRAFT RTP Topologies October 26, 2007
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 3.6
about the differences between these two models.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 6 - RTP Mixer with only unicast paths
Wenger, et al. Informational [Page 11]
INTERNET-DRAFT RTP Topologies October 26, 2007
3.5. Point to Multipoint using video switching MCU
Shortcut name: Topo-Video-switch-MCU
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
Figure 7 - Point to Multipoint using relaying 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 mostly 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
Wenger, et al. Informational [Page 12]
INTERNET-DRAFT RTP Topologies October 26, 2007
are not currently forwarded receive a view of the session that
indicates 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 MCU
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 uses 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 session to each participant is
negotiated individually.
Wenger, et al. Informational [Page 13]
INTERNET-DRAFT RTP Topologies October 26, 2007
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
Wenger, et al. Informational [Page 14]
INTERNET-DRAFT RTP Topologies October 26, 2007
streams from 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 does 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 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's 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. This as B and D only has 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.
4. Comparing Topologies
The topologies discussed in section 3 have different properties.
This section first lists these properties and then maps the
Wenger, et al. Informational [Page 15]
INTERNET-DRAFT RTP Topologies October 26, 2007
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 what 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 least capable
receiver at this time instance, 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.
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
Wenger, et al. Informational [Page 16]
INTERNET-DRAFT RTP Topologies October 26, 2007
Participants are most likely to be connected to each other with a
heterogenous set of paths. This makes congestion control in a point
to multi-point 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 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 need 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 fully fulfill this. The one described in Section 3.6
does not support this at all.
4.1.6. Loop Detection
In complex topologies with multiple domains interconnected, 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.
Wenger, et al. Informational [Page 17]
INTERNET-DRAFT RTP Topologies October 26, 2007
4.2. Comparision of topologies
The table below attempts to summarize the properties of the
different topologies. The legend to the topology abbrevations 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 below table Y indicates Y or full support, N
indicates No support, (Y) indicates partial support. N/A is 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 SRTP
and SRTCP [RFC3711] packets to session endpoints. If encryption is
employed, the media translator and mixer needs 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 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
Wenger, et al. Informational [Page 18]
INTERNET-DRAFT RTP Topologies October 26, 2007
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 creates 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 in
one domain a weaker algorithm or key than in another. Therefore,
care should be taken that appropriatly strong security parameters
are negotiated in all domains. In many cases, "appropriate"
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, 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
Wenger, et al. Informational [Page 19]
INTERNET-DRAFT RTP Topologies October 26, 2007
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 to avoid
two-time pad. 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 what 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. IANA Considerations
This document specifies no actions for IANA.
Wenger, et al. Informational [Page 20]
INTERNET-DRAFT RTP Topologies October 26, 2007
8. References
8.1. 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.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4575] J. Rosenberg, H. Schulzrinne, O. Levin, "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.
8.2. Informative References
[CCM] Wenger, S., Chandra, U., Westerlund, M., Burman, B.,
"Codec Control Messages in the Audio-Visual Profile with
Feedback (AVPF)", Internet Draft, Work in Progress,
draft-ietf-avt-avpf-ccm-08.txt>, 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," draft-ietf-avt-rtcpssm-13, work in progress,
March 2007.
Wenger, et al. Informational [Page 21]
INTERNET-DRAFT RTP Topologies October 26, 2007
9. 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
Wenger, et al. Informational [Page 22]
INTERNET-DRAFT RTP Topologies October 26, 2007
Full Copyright Statement
Copyright (C) The IETF Trust (2007).
This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors
retain all their rights.
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST
AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES,
EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT
THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY
IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR
PURPOSE.
Intellectual Property
The IETF takes no position regarding the validity or scope of any
Intellectual Property Rights or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; nor does it represent that it has
made any independent effort to identify any such rights. Information
on the procedures with respect to rights in RFC documents can be
found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
attempt made to obtain a general license or permission for the use of
such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository at
http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at
ietf-ipr@ietf.org.
Acknowledgement
Funding for the RFC Editor function is provided by the IETF
Administrative Support Activity (IASA).
RFC Editor Considerations
None
Wenger, et al. Informational [Page 23]
