One document matched: draft-westerlund-avtcore-rtp-topologies-update-01.xml
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docName="draft-westerlund-avtcore-rtp-topologies-update-01"
ipr="trust200902" obsoletes="5117">
<front>
<title abbrev="RTP Topologies">RTP Topologies</title>
<author fullname="Magnus Westerlund" initials="M.W" surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<author fullname="Stephan Wenger" initials="S.W" surname="Wenger">
<organization>Vidyo</organization>
<address>
<postal>
<street>433 Hackensack Ave</street>
<city>Hackensack</city>
<region>NJ</region>
<code>07601</code>
<country>USA</country>
</postal>
<email>stewe@stewe.org</email>
</address>
</author>
<date/>
<abstract>
<t>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.</t>
<t>This document is updated with additional topologies and are intended
to replace RFC 5117.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>When working on the <xref target="RFC5104">Codec Control
Messages</xref>, 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 <xref target="RFC3550">Real-time Transport
Protocol (RTP) spec</xref> in an informal way. It is not intended to
update or change what is normatively specified within RFC 3550.</t>
<t>When the <xref target="RFC4585">Audio-Visual Profile with Feedback
(AVPF)</xref> 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 <xref
target="RFC3550">RTP</xref> terminology) 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.</t>
</section>
<section title="Definitions">
<t/>
<section title="Glossary">
<t><list style="hanging">
<t hangText="ASM:">Any Source Multicast</t>
<t hangText="AVPF:">The Extended RTP Profile for RTCP-based
Feedback</t>
<t hangText="CSRC:">Contributing Source</t>
<t hangText="Link:">The data transport to the next IP hop</t>
<t hangText="MCU:">Multipoint Control Unit</t>
<t hangText="Path:">The concatenation of multiple links, resulting
in an end-to-end data transfer.</t>
<t hangText="PtM:">Point to Multipoint</t>
<t hangText="PtP:">Point to Point</t>
<t hangText="SSM:">Source-Specific Multicast</t>
<t hangText="SSRC:">Synchronization Source</t>
</list></t>
</section>
<section title="Indicating Requirement Levels">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.</t>
<t>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.</t>
</section>
</section>
<section anchor="sec-topologies" title="Topologies">
<t>This subsection defines several topologies that are relevant for
codec control but also RTP usage in other contexts. The first relate to
the RTP system model utilizing multicast and/or unicast, as envisioned
in RFC 3550. Two later topologies (MCU and RTCP terminating), in
contrast, describe the deployed system models as used in many <xref
target="H323">H.323</xref> 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 <xref
target="sec-combining-topologies"/>.</t>
<t>The topologies may be referenced in other documents by a shortcut
name, indicated by the prefix "Topo-".</t>
<t>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 <xref
target="RFC4585"/> and <xref target="RFC5104"/>. Any important
differences between the two will be illuminated in the discussion.</t>
<section title="Point to Point">
<t>Shortcut name: Topo-Point-to-Point</t>
<t>The <xref target="fig-point-to-point">Point to Point (PtP)
topology</xref> 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).</t>
<figure align="center" anchor="fig-point-to-point"
title="Point to Point">
<artwork><![CDATA[
+---+ +---+
| A |<------->| B |
+---+ +---+
]]></artwork>
</figure>
<t>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. The number of RTP sessions in use
between A and B can also be of any number.</t>
<t>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.</t>
</section>
<section title="Point to Multipoint Using Multicast">
<t>Multicast is a IP layer functionality that is available in some
networks. It comes in two main flavors, Any Source Multicast (ASM)
where any multicast group participant can send to the group address
and expect the packet to reach all group participants. The other model
is Source Specific Multicast (SSM) where only a particular IP host is
allowed to send to the multicast group. Both these models are
discussed below in their respective section.</t>
<section title="Any Source Multicast (ASM)">
<t>Shortcut name: Topo-ASM (was Topo-Multicast)</t>
<figure align="center" anchor="fig-ptm-multicast"
title="Point to Multipoint Using Multicast ">
<artwork><![CDATA[
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
]]></artwork>
</figure>
<t>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:<list style="symbols">
<t>packet loss, or</t>
<t>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.</t>
</list></t>
<t>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).</t>
<t>For feedback usage it is relevant to make distinction of 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 <xref
target="RFC4585">AVPF</xref>. This document refers to those groups
as "small multicast groups". Some applications may still want to use
larger multicast groups where the RTCP feedback possibilities are
more limited.</t>
<t>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 <xref target="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.</t>
</section>
<section title="Source Specific Multicast (SSM)">
<t>In Any Source Multicast, any of the participants can send to all
the other participants, simply by sending a packet to the multicast
group. That is not possible in <xref target="RFC4607">Source
Specific Multicast</xref> where only a single source (Distribution
Source) can send to the multicast group, creating a topology that
looks like the one below:</t>
<figure align="center" anchor="fig-multipoint-ssm"
title="Point to Multipoint using Source Specific Multicast">
<artwork><![CDATA[
+--------+ +-----+
|Media | | | Source-specific
|Sender 1|<----->| D S | Multicast
+--------+ | I O | +--+----------------> R(1)
| S U | | | |
+--------+ | T R | | +-----------> R(2) |
|Media |<----->| R C |->+ | : | |
|Sender 2| | I E | | +------> R(n-1) | |
+--------+ | B | | | | | |
: | U | +--+--> R(n) | | |
: | T +-| | | | |
: | I | |<---------+ | | |
+--------+ | O |F|<---------------+ | |
|Media | | N |T|<--------------------+ |
|Sender M|<----->| | |<-------------------------+
+--------+ +-----+ RTCP Unicast
FT = Feedback Target
Transport from the Feedback Target to the Distribution
Source is via unicast or multicast RTCP if they are not
co-located.
]]></artwork>
</figure>
<t>In the <xref target="fig-multipoint-ssm">SSM topology</xref> a
number of RTP sources (1 to M) are allowed to send media to the SSM
group. These send media to the distribution source which then
forwards the media streams to the multicast group. The media streams
reach the Receivers (R(1) to R(n)). The Receivers' RTCP cannot be
sent to the multicast group. To support RTCP, an <xref
target="RFC5760">RTP extension for SSM</xref> was defined to use
unicast transmission to send RTCP from the receivers to one or more
Feedback Targets (FT).</t>
<t>The RTP extension for SSM deals with how feedback both general
reception information and specific feedback events are generally
handled. The general problems of multicast that everyone will
receive what the distribution source sends needs to be accounted
for.</t>
<t>The result of this is some common behaviours for RTP
multicast:<list style="numbers">
<t>Multicast applications often use a group of RTP sessions, not
one. Each endpoint will need to be a member of a number of RTP
sessions in order to perform well.</t>
<t>Within each RTP session, the number of media sinks is likely
to be much larger than the number of RTP sources.</t>
<t>Multicast applications need signalling functions to identify
the relationships between RTP sessions.</t>
<t>Multicast applications need signalling functions to identify
the relationships between SSRCs in different RTP sessions.</t>
</list></t>
<t>All multicast configurations share a signalling requirement; all
of the participants will need to have the same RTP and payload type
configuration. Otherwise, A could for example be using payload type
97 as the video codec H.264 while B thinks it is MPEG-2.</t>
<t>Security solutions for this type of group communications are also
challenging. First of all the key-management and the security
protocol must support group communication. Source authentication
becomes more difficult and requires special solutions. For more
discussion on this please review <xref
target="I-D.ietf-avtcore-rtp-security-options">Options for Securing
RTP Sessions</xref>.</t>
</section>
<section title="SSM with Local Unicast Resources">
<t>[RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP
Sessions" results in additional extensions to SSM Topology. Should
be described.</t>
</section>
</section>
<section title="Point to Multipoint Using Mesh">
<t>Shortcut name: Topo-Mesh</t>
<figure align="center" anchor="fig-mesh"
title="Point to Multi-Point using Mesh">
<artwork><![CDATA[
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
]]></artwork>
</figure>
<t>Based on the RTP session definition, it is clearly possible to have
a joint RTP session over multiple unicast transport flows like the
above three endpoint joint session. In this case, A needs to send its'
media streams and RTCP packets to both B and C over their respective
transport flows. As long as all participants do the same, everyone
will have a joint view of the RTP session.</t>
<t>This doesn't create any additional requirements beyond the need to
have multiple transport flows associated with a single RTP session.
Note that an endpoint may use a single local port to receive all these
transport flows, or it might have separate local reception ports for
each of the endpoints.</t>
<t>There exists an alternative structure for establishing the above
topology which uses independent RTP sessions between each pair of
peers, i.e. three different RTP sessions. Unless independently adapted
the same RTP media stream could be sent in both of the RTP sessions an
endpoint has. The difference exists in the behaviours around RTCP, for
example common RTCP bandwidth for one joint session, rather than three
independent pools, and the awareness based on RTCP reports between the
peers of how that third leg is doing.</t>
</section>
<section anchor="sec-ptm-translator"
title="Point to Multipoint Using the RFC 3550 Translator">
<t>Shortcut name: Topo-Translator</t>
<t>Two main categories of Translators can be distinguished; Transport
Translators and Media translators. 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.</t>
<!--MW: Should the above Translator definition be slightly expanded to allow any 1-to-1
SSRC mappings instead of the current formualtion "keeps the SSRC for a stream across
the translation,"?-->
<t>As specified in Section 7.1 of <xref target="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.</t>
<t>A Translator commonly does not use an SSRC of its own, and is not
visible as an active participant in the session. One reason is 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. This can be done using the SSRC of the target for
the translator, but this requires translation of the targets RTCP
reports to make them consistent, so it is likely simpler to expose an
additional SSRC in the session.</t>
<t>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.</t>
<t>The RTCP translation process can be trivial, for example, when
Transport Translators just need to adjust IP addresses and transport
protocol ports, or they can be quite complex as in the case of media
Translators. See Section 7.2 of <xref target="RFC3550"/>.</t>
<section title="Relay - Transport Translator">
<t>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.</t>
<figure align="center" anchor="fig-ptm-multicast-translator"
title="Point to Multipoint Using Multicast ">
<artwork><![CDATA[
+-----+
+---+ / \ +------------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Translator |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+
+-----+
]]></artwork>
</figure>
<t><xref target="fig-ptm-multicast-translator"/> 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 an any source 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.</t>
<t>Also a point to point communication can end up in a situation
when the peer it is communicating with needs basic transport
translators functions. This include NAT traversal by pinning the
media path to a public address domain relay, network topologies
where the media flow is required to pass a particular point by
employing relaying or preserving privacy by hiding each peers
transport addresses to the other party.</t>
<figure align="center" title="Point to Point with Translator">
<artwork><![CDATA[
+---+ +---+ +---+
| A |<------>| T |<------->| B |
+---+ +---+ +---+
]]></artwork>
</figure>
<t>This type of very basic relay service should in most case need to
have no RTP functionality. Thus one can believe that they do not
need to included in this document. However, due to that the network
level addressing and the RTP identifier view of the RTP session and
who the peer is doesn't match as in the PtP unicast scenario
depicted above this topology can raise additional requirements.</t>
<t/>
<figure align="center" anchor="fig-translator-unicast"
title="RTP Translator (Relay) with Only Unicast Paths">
<artwork><![CDATA[
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
]]></artwork>
</figure>
<t>Another Translator scenario is depicted in <xref
target="fig-translator-unicast"/>. 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.</t>
<t>For RTCP feedback this results in a similar considerations that
arise for the ASM RTP topology. It also puts some special signalling
requirements where common configuration of RTP payload types for
example are required.</t>
</section>
<section title="Media Translator">
<t>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.</t>
<t>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).</t>
<t>If B in <xref target="fig-ptm-multicast-translator"/> 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.</t>
<t>When, in the example depicted in <xref
target="fig-ptm-multicast-translator"/>, 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 and data rate. 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.</t>
<t>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.</t>
<t>A variant of translator behaviour worth pointing out is the one
depicted in <xref target="fig-de-composite-translator"/> of an
endpoint A sends a media flow to B. On the path there is a device T
that on A's behalf does something with the media streams, for
example adds an RTP session with FEC information for A's media
streams. T will in this case need to bind the new FEC streams to A's
media stream, for example by using the same CNAME as A.</t>
<figure anchor="fig-de-composite-translator"
title="When De-composition is a Translator">
<artwork><![CDATA[
+------+ +------+ +------+
| | | | | |
| A |------->| T |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+]]></artwork>
</figure>
<t>This type of functionality where T does something with the media
stream on behalf of A is clearly covered under the media translator
definition.</t>
</section>
</section>
<section anchor="sec-ptm-mixer"
title="Point to Multipoint Using the RFC 3550 Mixer Model">
<t>Shortcut name: Topo-Mixer</t>
<t>A Mixer is a middlebox that aggregates multiple RTP streams, which
are part of a session, by manipulation of 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.</t>
<figure align="center" anchor="fig-ptm-mixer"
title="Point to Multipoint Using the RFC 3550 Mixer Model">
<artwork><![CDATA[
+-----+
+---+ / \ +-----------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Mixer |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +-----------+ +---+
+-----+
]]></artwork>
</figure>
<t>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.</t>
<t>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.</t>
<t>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 should be 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.</t>
<t>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.</t>
<t>The Mixer depicted in <xref target="fig-ptm-mixer"/> is involved in
three domains that need to be separated: the any source 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 may only
need one SSRC per media type in each domain that is the receiving
entity and transmitter of mixed content.</t>
<t>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 <xref target="sec-asymmetric"/> for more
discussion on this topic. However, the mixing operation in each domain
could potentially be different.</t>
<t>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.</t>
<t>When replacing the multicast network in <xref
target="fig-ptm-mixer"/> (to the left of the Mixer) with individual
unicast paths as depicted in <xref target="fig-mixer-unicast"/>, the
Mixer model is very similar to the one discussed in <xref
target="sec-ptm-mcu"/> below. Please see the discussion in <xref
target="sec-ptm-mcu"/> about the differences between these two
models.</t>
<figure align="center" anchor="fig-mixer-unicast"
title="RTP Mixer with Only Unicast Paths ">
<artwork><![CDATA[
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
]]></artwork>
</figure>
<t>Lets now discuss in more detail different mixing operations that a
mixer can perform and how that can affect the RTP and RTCP.</t>
<section title="Media Mixing">
<t>The media mixing mixer is likely the one that most thinks of when
they hear the term mixer. Its basic pattern of operation is that it
will receive the different participants RTP media stream. Select
which that are to be included in a media domain mix of the incoming
RTP media streams. Then create a single outgoing stream from this
mix.</t>
<t>The most commonly deployed media mixer is probably the audio
mixer, used in voice conferencing, where the output consists of some
mixture of all the input streams; this needs minimal signalling to
be successful. Audio mixing is straight forward and commonly
possible to do for a number of participants. Lets assume that you
want to mix N number of streams from different participants. Then
the mixer need to perform N decodings. Then it needs to produce N or
N+1 mixes, the reasons that different mixes are needed are so that
each contributing source get a mix which don't contain themselves,
as this would result in an echo. When N is lower than the number of
all participants one may produce a Mix of all N streams for the
group that are currently not included in the mix, thus N+1 mixes.
These audio streams are then encoded again, RTP packetised and sent
out.</t>
<t>Video can't really be "mixed" and produce something particular
useful for the users, however creating an composition out of the
contributed video streams can be done. In fact it can be done in a
number of ways, tiling the different streams creating a chessboard,
selecting someone as more important and showing them large and a
number of other sources as smaller overlays is another. Also here
one commonly need to produce a number of different compositions so
that the contributing part doesn't need to see themselves. Then the
mixer re-encodes the created video stream, RTP packetise it and send
it out</t>
<t>The problem with media mixing is that it both consume large
amount of media processing and encoding resources. The second is the
quality degradation created by decoding and re-encoding the RTP
media stream. Its advantage is that it is quite simplistic for the
clients to handle as they don't need to handle local mixing and
composition.</t>
<figure align="center" anchor="fig-media-mixer"
title="Session and SSRC details for Media Mixer">
<artwork><![CDATA[+-A---------+ +-MIXER----------------------+
| +-RTP1----| |-RTP1------+ +-----+ |
| | +-Audio-| |-Audio---+ | +---+ | | |
| | | AA1|--------->|---------+-+-|DEC|->| | |
| | | |<---------|MA1 <----+ | +---+ | | |
| | | | |(BA1+CA1)|\| +---+ | | |
| | +-------| |---------+ +-|ENC|<-| B+C | |
| +---------| |-----------+ +---+ | | |
+-----------+ | | | |
| | M | |
+-B---------+ | | E | |
| +-RTP2----| |-RTP2------+ | D | |
| | +-Audio-| |-Audio---+ | +---+ | I | |
| | | BA1|--------->|---------+-+-|DEC|->| A | |
| | | |<---------|MA2 <----+ | +---+ | | |
| | +-------| |(BA1+CA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+C | |
+-----------+ |-----------+ +---+ | | |
| | M | |
+-C---------+ | | I | |
| +-RTP3----| |-RTP3------+ | X | |
| | +-Audio-| |-Audio---+ | +---+ | E | |
| | | CA1|--------->|---------+-+-|DEC|->| R | |
| | | |<---------|MA3 <----+ | +---+ | | |
| | +-------| |(BA1+CA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+B | |
+-----------+ |-----------+ +---+ +-----+ |
+----------------------------+
]]></artwork>
</figure>
<t>From an RTP perspective media mixing can be very straight forward
as can be seen in <xref target="fig-media-mixer"/>. The mixer
present one SSRC towards the peer client, e.g. MA1 to Peer A, which
is the media mix of the other participants. As each peer receives a
different version produced by the mixer there are no actual relation
between the different RTP sessions in the actual media or the
transport level information. There is however one connection between
RTP1-RTP3 in this figure. It has to do with the SSRC space and the
identity information. When A receives the MA1 stream which is a
combination of BA1 and CA1 streams, the mixer may include CSRC
information in the MA1 stream to identify the contributing source
BA1 and CA1.</t>
<t>The CSRC has in its turn utility in RTP extensions, like the in
<xref target="RFC6465">Mixer to Client audio levels RTP header
extension</xref>. If the SSRC from endpoint to mixer leg are used as
CSRC in another RTP session then RTP1, RTP2 and RTP3 becomes one
joint session as they have a common SSRC space. At this stage the
mixer also need to consider which RTCP information it need to expose
in the different legs. For the above situation commonly nothing more
than the Source Description (SDES) information and RTCP BYE for CSRC
need to be exposed. The main goal would be to enable the correct
binding against the application logic and other information sources.
This also enables loop detection in the RTP session.</t>
</section>
<section anchor="sec-media-switching" title="Media Switching">
<t>An RTP Mixer based on media switching avoids the media decoding
and encoding cycle in the mixer, but not the decryption and
re-encryption cycle as it rewrites RTP headers. This both reduces
the amount of computational resources needed in the mixer and
increases the media quality per transmitted bit. This is achieve by
letting the mixer have a number of SSRCs that represents conceptual
or functional streams the mixer produces. These streams are created
by selecting media from one of the by the mixer received RTP media
streams and forward the media using the mixers own SSRCs. The mixer
can then switch between available sources if that is required by the
concept for the source, like currently active speaker.</t>
<t>To achieve a coherent RTP media stream from the mixer's SSRC the
mixer is forced to rewrite the incoming RTP packet's header. First
the SSRC field must be set to the value of the Mixer's SSRC.
Secondly, the sequence number must be the next in the sequence of
outgoing packets it sent. Thirdly the RTP timestamp value needs to
be adjusted using an offset that changes each time one switch media
source. Finally depending on the negotiation the RTP payload type
value representing this particular RTP payload configuration may
have to be changed if the different endpoint mixer legs have not
arrived on the same numbering for a given configuration. This also
requires that the different end-points do support a common set of
codecs, otherwise media transcoding for codec compatibility is still
required.</t>
<t>Lets consider the operation of media switching mixer that
supports a video conference with six participants (A-F) where the
two latest speakers in the conference are shown to each
participants. Thus the mixer has two SSRCs sending video to each
peer.</t>
<figure align="center" anchor="fig-media-switching"
title="Media Switching RTP Mixer">
<artwork><![CDATA[+-A---------+ +-MIXER----------------------+
| +-RTP1----| |-RTP1------+ +-----+ |
| | +-Video-| |-Video---+ | | | |
| | | AV1|------------>|---------+-+------->| S | |
| | | |<------------|MV1 <----+-+-BV1----| W | |
| | | |<------------|MV2 <----+-+-EV1----| I | |
| | +-------| |---------+ | | T | |
| +---------| |-----------+ | C | |
+-----------+ | | H | |
| | | |
+-B---------+ | | M | |
| +-RTP2----| |-RTP2------+ | A | |
| | +-Video-| |-Video---+ | | T | |
| | | BV1|------------>|---------+-+------->| R | |
| | | |<------------|MV3 <----+-+-AV1----| I | |
| | | |<------------|MV4 <----+-+-EV1----| X | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ | | |
+-----------+ | | | |
: : : :
: : : :
+-F---------+ | | | |
| +-RTP6----| |-RTP6------+ | | |
| | +-Video-| |-Video---+ | | | |
| | | CV1|------------>|---------+-+------->| | |
| | | |<------------|MV11 <---+-+-AV1----| | |
| | | |<------------|MV12 <---+-+-EV1----| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +----------------------------+
]]></artwork>
</figure>
<t>The Media Switching RTP mixer can similar to the Media Mixing one
reduce the bit-rate needed towards the different peers by selecting
and switching in a sub-set of RTP media streams out of the ones it
receives from the conference participants.</t>
<t>To ensure that a media receiver can correctly decode the RTP
media stream after a switch, it becomes necessary to ensure for
state saving codecs that they start from default state at the point
of switching. Thus one common tool for video is to request that the
encoding creates an intra picture, something that isn't dependent on
earlier state. This can be done using <xref target="RFC5104">Full
Intra Request</xref> RTCP codec control message.</t>
<t>Also in this type of mixer one could consider to terminate the
RTP sessions fully between the different end-point and mixer legs.
The same arguments and considerations as discussed in <xref
target="sec-ptm-mcu"/> applies here.</t>
</section>
</section>
<section title="Source Projecting Middlebox">
<t>Another method for handling media in the RTP mixer is to project
all potential RTP sources (SSRCs) into a per end-point independent RTP
session. The mixer can then select which of the potential sources that
are currently actively transmitting media, despite that the mixer in
another RTP session receives media from that end-point. This is
similar to the media switching Mixer but have some important
differences in RTP details.</t>
<figure align="center" anchor="fig-projecting"
title="Media Projecting Mixer">
<artwork><![CDATA[+-A---------+ +-MIXER---------------------+
| +-RTP1----| |-RTP1------+ +-----+ |
| | +-Video-| |-Video---+ | | | |
| | | AV1|------------>|---------+-+------>| | |
| | | |<------------|BV1 <----+-+-------| S | |
| | | |<------------|CV1 <----+-+-------| W | |
| | | |<------------|DV1 <----+-+-------| I | |
| | | |<------------|EV1 <----+-+-------| T | |
| | | |<------------|FV1 <----+-+-------| C | |
| | +-------| |---------+ | | H | |
| +---------| |-----------+ | | |
+-----------+ | | M | |
| | A | |
+-B---------+ | | T | |
| +-RTP2----| |-RTP2------+ | R | |
| | +-Video-| |-Video---+ | | I | |
| | | BV1|------------>|---------+-+------>| X | |
| | | |<------------|AV1 <----+-+-------| | |
| | | |<------------|CV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | |
| | | |<------------|FV1 <----+-+-------| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ | | |
+-----------+ | | | |
: : : :
: : : :
+-F---------+ | | | |
| +-RTP6----| |-RTP6------+ | | |
| | +-Video-| |-Video---+ | | | |
| | | CV1|------------>|---------+-+------>| | |
| | | |<------------|AV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | |
| | | |<------------|EV1 <----+-+-------| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +---------------------------+
]]></artwork>
</figure>
<t>So in this six participant conference depicted above <xref
target="fig-projecting">in</xref> one can see that end-point A will in
this case be aware of 5 incoming SSRCs, BV1-FV1. If this mixer intend
to have the same behaviour as in <xref target="sec-media-switching"/>
where the mixer provides the end-points with the two latest speaking
end-points, then only two out of these five SSRCs will concurrently
transmit media to A. As the mixer selects which source in the
different RTP sessions that transmit media to the end-points each RTP
media stream will require some rewriting when being projected from one
session into another. The main thing is that the sequence number will
need to be consecutively incremented based on the packet actually
being transmitted in each RTP session. Thus the RTP sequence number
offset will change each time a source is turned on in a RTP
session.</t>
<t>As the RTP sessions are independent the SSRC numbers used can be
handled independently also thus working around any SSRC collisions by
having remapping tables between the RTP sessions. This will result
that each endpoint may have a different view of the application usage
of a particular SSRC. Thus the application must not use SSRC as
references to RTP media streams when communicating with other peers
directly.</t>
<t>The mixer will also be responsible to act on any RTCP codec control
requests coming from an end-point and decide if it can act on it
locally or needs to translate the request into the RTP session that
contains the media source. Both end-points and the mixer will need to
implement conference related codec control functionalities to provide
a good experience. Full Intra Request to request from the media source
to provide switching points between the sources, Temporary Maximum
Media Bit-rate Request (TMMBR) to enable the mixer to aggregate
congestion control response towards the media source and have it
adjust its bit-rate in case the limitation is not in the source to
mixer link.</t>
<t>This version of the mixer also puts different requirements on the
end-point when it comes to decoder instances and handling of the RTP
media streams providing media. As each projected SSRC can at any time
provide media the end-point either needs to handle having thus many
allocated decoder instances or have efficient switching of decoder
contexts in a more limited set of actual decoder instances to cope
with the switches. The WebRTC application also gets more
responsibility to update how the media provides is to be presented to
the user.</t>
<t>Note, this could potentially be seen as a media translator which
include an on/off logic as part of its media translation. The main
difference would be a common global SSRC space in the case of the
Media Translator and the mapped one used in the above.</t>
</section>
<section anchor="sec-ptm-switch-mcu"
title="Point to Multipoint Using Video Switching MCUs ">
<t>Shortcut name: Topo-Video-switch-MCU</t>
<figure align="center" anchor="fig-ptm-switching-mcu"
title="Point to Multipoint Using a Video Switching MCU">
<artwork><![CDATA[
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
]]></artwork>
</figure>
<t>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".</t>
<t>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.</t>
<t>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.</t>
<t>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.</t>
<t>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 their media
streams disappear somewhere en route. This makes the use of RTCP for
congestion control, or any type of quality reporting, very
problematic.</t>
<t>To avoid the aforementioned issues, the MCU needs to implement two
features. First, it needs to act as a Mixer (see <xref
target="sec-ptm-mixer"/>) 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.</t>
<t/>
</section>
<section anchor="sec-ptm-mcu"
title="Point to Multipoint Using RTCP-Terminating MCU">
<t>Shortcut name: Topo-RTCP-terminating-MCU</t>
<figure align="center" anchor="fig-ptm-terminating-mcu"
title="Point to Multipoint Using Content Modifying MCUs ">
<artwork><![CDATA[
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
]]></artwork>
</figure>
<t>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:<list style="letters">
<t>a selection of the content received from the other
participants, or</t>
<t>the mixed aggregate of what the MCU receives from the other PtP
paths, which are part of the same conference session.</t>
</list></t>
<t>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.</t>
<t>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
<xref target="RFC3550">RTP</xref>, 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.</t>
<t>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:<list style="numbers">
<t>Loop detection cannot be performed on the RTP level. When
carelessly connecting two misconfigured MCUs, a loop could be
generated.</t>
<t>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.</t>
</list></t>
<t>Note that deployed MCUs (and endpoints) rely on signalling layer
mechanisms for the identification of the contributing sources, for
example, a <xref target="RFC4575">SIP conferencing package</xref>.
This alleviates, to some extent, the aforementioned issues resulting
from ignoring RTP's CSRC mechanism.</t>
<t>As a result of the shortcomings of this topology, it is RECOMMENDED
to instead implement the Mixer concept as specified by RFC 3550.</t>
</section>
<section title="De-composite Endpoint">
<!--MW: Harald, is this clearer. If not you better explain why the de-composite that uses different
RTP sessions can't work.-->
<t>The implementation of an application may desire to send a subset of
the application's data to each of multiple devices, each with their
own network address. A very basic use case for this would be to
separate audio and video processing for a particular endpoint, like a
conference room, into one device handling the audio and another
handling the video, being interconnected by some control functions
allowing them to behave as a single endpoint in all aspects except for
transport <xref target="fig-de-composite"/>.</t>
<t>Which decomposition that is possible is highly dependent on the RTP
session usage. It is not really feasible to decomposed one logical
end-point into two different transport node in one RTP session. From a
third party monitor of such an attempt the two entities would look
like two different end-points with a CNAME collision. This put a
requirement on that the only type of de-composited endpoint that RTP
really supports is one where the different parts have separate RTP
sessions to send and/or receive media streams intended for them.</t>
<figure align="center" anchor="fig-de-composite"
title="De-composite End-Point">
<artwork><![CDATA[
+---------------------+
| Endpoint A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+-RTP---\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+-RTP-------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+-SIP---/
| +------------+ |
+---------------------+
]]></artwork>
</figure>
<t>In the above usage, let us assume that the RTP sessions are
different for audio and video. The audio and video parts will use a
common CNAME and also have a common clock to ensure that
synchronisation and clock drift handling works despite the
decomposition. Also the RTCP handling works correctly as long as only
one part of the de-composite is part of each RTP session. That way any
differences in the path between A's audio entity and B and A's video
and B are related to different SSRCs in different RTP sessions.</t>
<t>The requirements that can derived from the above usage is that the
transport flows for each RTP session might be under common control but
still go to what looks like different endpoints based on addresses and
ports. This geometry cannot be accomplished using one RTP session, so
in this case, multiple RTP sessions are needed.</t>
</section>
<section anchor="sec-asymmetric" title="Non-Symmetric Mixer/Translators">
<t>Shortcut name: Topo-Asymmetric</t>
<t>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 <xref
target="fig-ptm-mixer"/>, 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 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.</t>
<t>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.</t>
<t>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.</t>
</section>
<section anchor="sec-combining-topologies" title="Combining Topologies">
<t>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.</t>
</section>
</section>
<section title="Comparing Topologies">
<t>The topologies discussed in <xref target="sec-topologies"/> 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.</t>
<t>Note: This section has not yet been updated with the new additions of
topologies.</t>
<section title="Topology Properties">
<t/>
<section title="All to All Media Transmission">
<t>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.</t>
<t>The transport Translator function called "relay", in <xref
target="sec-ptm-translator"/>, 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.</t>
<t>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.</t>
<t>The transmission of RTCP automatically adapts to any changes in
the number of participants due to the transmission algorithm,
defined in the <xref target="RFC3550">RTP specification</xref>, and
the extensions in <xref target="RFC4585">AVPF</xref> (when
applicable). That way, the resources utilized for RTCP stay within
the bounds configured for the session.</t>
</section>
<section title="Transport or Media Interoperability">
<t>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.</t>
</section>
<section title="Per Domain Bit-Rate Adaptation">
<t>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.</t>
</section>
<section title="Aggregation of Media">
<t>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.</t>
</section>
<section title="View of All Session Participants">
<t>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 <xref
target="sec-ptm-switch-mcu"/> does not entirely fulfill this. The
one described in <xref target="sec-ptm-mcu"/> does not support this
at all.</t>
</section>
<section title="Loop Detection">
<t>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 <xref target="sec-ptm-switch-mcu"/>, at least as
long as it forwards the RTCP between the participants. However, the
MCU in <xref target="sec-ptm-mcu"/> will definitely break the loop
detection mechanism.</t>
</section>
</section>
<section title="Comparison of Topologies">
<t>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.</t>
<figure>
<artwork><![CDATA[
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
]]></artwork>
</figure>
<t>Please note that the Media Translator also includes the transport
Translator functionality.</t>
</section>
</section>
<section title="Security Considerations">
<t>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 <xref
target="RFC3711">Secure Realtime Transport Protocol (SRTP) and Secure
Realtime Transport Control Protocol (SRTCP)</xref> 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
port numbers in a media stream using <xref target="RFC3711">SRTP</xref>.
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.</t>
<t>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.</t>
<t>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.</t>
<t>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.</t>
<t>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.)</t>
<t>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:</t>
<t>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" 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.</t>
<t>Second, the number of crypto contexts (keys and security related
state) needed (for example, in <xref target="RFC3711">SRTP</xref>) 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 <xref
target="fig-ptm-multicast-translator"/>, 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.</t>
<t>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. Please
review <xref target="I-D.ietf-avtcore-rtp-security-options">RTP Security
Options</xref> to get a better understanding of most of the available
options.</t>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This document makes no request of IANA.</t>
<t>Note to RFC Editor: this section may be removed on publication as an
RFC.</t>
</section>
<section title="Acknowledgements">
<t>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.</t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include='reference.RFC.3550'?>
<?rfc include='reference.RFC.3711'?>
<?rfc include='reference.RFC.4575'?>
<?rfc include='reference.RFC.4585'?>
</references>
<references title="Informative References">
<?rfc include="reference.RFC.5104"?>
<?rfc include='reference.RFC.4607'?>
<reference anchor="H323">
<front>
<title>Packet-based multimedia communications systems</title>
<author fullname="ITU-T Recommendation H.323"
surname="ITU-T Recommendation H.323">
<organization/>
</author>
<date month="June" year="2006"/>
</front>
</reference>
<?rfc include="reference.RFC.5760"?>
<?rfc include='reference.RFC.6285'?>
<?rfc include='reference.RFC.6465'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-security-options'?>
</references>
</back>
</rfc>
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