One document matched: draft-westerlund-avtcore-rtp-topologies-update-02.xml
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docName="draft-westerlund-avtcore-rtp-topologies-update-02"
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 point to point and 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><xref target="RFC3550">Real-time Transport Protocol (RTP)</xref>
topologies describe methods for interconnecting RTP entities and their
processing behavior of RTP and RTCP. This document tries to address past
and existing confusion, especially with respect to terms not defined in
RTP but in common use in the conversational communication industry, such
as MCU. In doing so, this memo provides 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>
<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 section starts
with point to point cases, without and with middleboxes. Then follows a
number of different methods for establishing point to multipoint
communication. These are structure around the most fundamental enabler,
i.e. multicast, a mesh of connections, translators, mixers and source
projection middlebox, to finally discuss MCUs. The section ends by
discussing de-composed endpoints, asymmetric middlebox behaviors and
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 discuss 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. At
this stage we don't intended to discuss in detail how each and every
feedback messages should be treated in the various topologies.</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. For RTP sessions which use
multiple SSRC per endpoint it can be relevant to implement support for
cross reporting suppression as defined in <xref
target="I-D.lennox-avtcore-rtp-multi-stream">"Real-Time Transport
Protocol (RTP) Considerations for Endpoints Sending Multiple Media
Streams"</xref>.</t>
</section>
<section title="Point to Point via Middlebox">
<t>This section discusses cases where two endpoints communicate but
have one or more middlebox involved in the RTP session.</t>
<section anchor="sec-ptp-translators" title="Translators">
<t>Shortcut name: Topo-PtP-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 keeps the SSRC
for a stream across the translation, whereas a Mixer can select a
single media stream, or send out multiple mixed media streams, but
always under its own SSRC, possibly using the CSRC field to indicate
the source(s) of the content. Mixers are more common in point to
multipoint cases than in PtP. The reason is that in PtP use cases
the primary focus is interoperability, such as transcoding to a
codec the receiver supports, which can be done by a media
translator.</t>
<t>As specified in Section 7.1 of <xref target="RFC3550"/>, the SSRC
space is common for all participants in the RTP session, independent
of on which side of the Translator the session resides. Therefore,
it is the responsibility of the participants to run SSRC collision
detection, and the SSRC is thus a field the Translator cannot
change. Any SDES information associated with a SSRC or CSRC also
needs to be forwarded between the domains for any SSRC/CSRC used in
the different domains.</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 to have
its own SSRC 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 in a translator
configured to detect packet loss of important video packets and
wants to trigger repair by the media sender, by sending feedback
messages. While such feedback could use the SSRC of the target for
the translator, but this in turn would require translation of the
targets RTCP reports to make them consistent. It may be simpler to
expose an additional SSRC in the session, the only concern are
endpoints failing to support the full RTP specification, thus having
issues with multiple SSRCs reporting on the RTP streams sent by that
endpoint.</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>
<section anchor="sec-transport-anchor"
title="Transport Relay/Anchoring">
<t>There exist a number of different types of middleboxes that
might be inserted between two RTP endpoints on the transport
level, e.g. perform changes on the IP/UDP headers, and are,
therefore, basic transport translators. These middleboxes come in
many variations including <xref target="RFC3022">NAT</xref>
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 for audit by employing relaying, or preserving
privacy by hiding each peers transport addresses to the other
party. Other protocols or functionalities that provide this
behavior are <xref target="RFC5766">TURN</xref> servers, Session
Border Gateways and Media Processing Nodes with media anchoring
functionalities.</t>
<figure align="center" anchor="fig-ptp-translator"
title="Point to Point with Translator">
<artwork><![CDATA[
+---+ +---+ +---+
| A |<------>| T |<------->| B |
+---+ +---+ +---+
]]></artwork>
</figure>
<t>What is common for these functions is that they are normally
transparent on RTP level, i.e. they perform no changes on any RTP
or RTCP packet fields, only on the lower layers. However, they may
effect the path the RTP and RTCP packets are routed between the
endpoints in the RTP session, and thereby only indirectly affect
the RTP session. For this reason, one could believe that transport
translator type middleboxes do not need to included in this
document. However, this topology can raise additional requirements
the RTP implementation and its interactions with the signalling
solution. Both in signalling and in certain RTCP field other
network addresses than those of the relay can occur, due to that B
has different network address than the relay (T). However,
implementation not capable of this will neither not work when
endpoints are subject to NAT. </t>
</section>
<section title="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 such unicast to
multicast) 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>
<t>The most basic transport translators that operate below RTP
level was already discussed in <xref
target="sec-transport-anchor"/>.</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
in the media encoding.</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>When media translation occurs, the Translator's task regarding
handling of RTCP traffic becomes substantially more complex. In
this case, the Translator needs to rewrite B's RTCP Receiver
Report before forwarding them to A. 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 A sends, 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.</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. In this case, T needs 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 covered under the media translator
definition.</t>
</section>
</section>
<section title="Back to Back RTP sessions">
<t>There exist middleboxes that interconnect two endpoints through
themselves not by being part of a common RTP session. Instead they
establish two different RTP sessions, one between A and the
middlebox (MB) and another between the MB and B.</t>
<figure anchor="fig-b2b-session"
title="When De-composition is a Translator">
<artwork><![CDATA[
|<--Session A-->| |<--Session B-->|
+------+ +------+ +------+
| A |------->| MB |-------->| B |
+------+ +------+ +------+]]></artwork>
</figure>
<t>The MB acts as a application level gateway and bridges the two
RTP session. This bridging can be as basic as forwarding the RTP
payloads between the sessions, or more complex including media
transcoding. The difference with the single RTP session context is
the handling of the SSRCs and the other session related identifiers,
such as CNAMEs. With two different RTP sessions these can be freely
changed and it becomes the MB task to maintain the right
relations.</t>
<t>The signalling or other above-RTP level functionalities
referencing RTP media streams may be what is most impacted by using
two RTP sessions and changing identifiers. The structure with two
RTP sessions also puts a congestion control requirement on the
middlebox, because it becomes fully responsible for the media stream
it sources into each of the sessions.</t>
<t>This can be solved locally or by bridging also statistics from
the receiving endpoint. However, from an implementation point this
requires the implementation to support dealing with a number of
inconsistencies. First, packet loss must be detected for an RTP flow
sent from A to the MB, and that loss must be reported through a
skipped sequence number in the flow from the MB to B. This coupling
and the resulting inconsistencies is conceptually easier to handle
when considering the two flows as belonging to a single RTP
session.</t>
</section>
</section>
<section title="Point to Multipoint Using Multicast">
<t>Multicast is a IP layer functionality that is available in some
networks. Two main flavors can be distinguished: Any Source Multicast
(ASM) where any multicast group participant can send to the group
address and expect the packet to reach all group participants; and
Source Specific Multicast (SSM), where only a particular IP host sends
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 scenario can occur, for
example, where a multi-media session is distributed using two or
more multicast groups and a participant is subscribed only to a
subset of these sessions.</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 in the range of billions). The above can be realized
using Any Source Multicast (ASM).</t>
<t>For feedback usage, it is useful to define a "small multicast
group" as a group where the number of participants is so low (and
other factors such as the connectivity is so good) that it allows
the participants to use early or immediate feedback, as defined in
<xref target="RFC4585">AVPF</xref>. Even when the environment would
allow for the use of a small multicast group, some applications may
still want to use the more limited options for RTCP feedback
available to large multicast groups, for example when there is a
likelyhood that the threshold of the small multicast group (in terms
of participants) may be exceeded during the lifetime of a
session.</t>
<t>RTCP feedback messages in multicast reach, like media, every
subscriber (subject to packet losses and multicast group
subscription). Therefore, the feedback suppression mechanism
discussed in <xref target="RFC4585"/> is typically required. Each
individual node needs to process every feedback message it receives,
not to determine if it is affected or if the feedback message
applies only to some other participant, but also to derive timing
restriction for the sending of its own feedback messages, if
any.</t>
</section>
<section title="Source Specific Multicast (SSM)">
<t>In Any Source Multicast, any of the participants can send to all
the other participants, by sending a packet to the multicast group.
In contrast, <xref target="RFC4607">Source Specific Multicast</xref>
refers to scenarios 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 a dedicated distribution source, which
then forwards the media streams to the multicast group on behalf of
the original senders. The media streams reach the Receivers (R(1) to
R(n)). The Receivers' RTCP cannot be sent to the multicast group, as
the SSM multicast group by definition has only a single source. To
support RTCP, an <xref target="RFC5760">RTP extension for SSM</xref>
was defined. It uses unicast transmission to send RTCP from each of
the receivers to one or more Feedback Targets (FT). The feedback
targets relay the RTCP unmodified, or provide summary of the
participants RTCP reports towards the whole group by forwarding the
RTCP traffic to the distribution source. <xref
target="fig-multipoint-ssm"/> only shows a single feedback target
integrated in the distribution source, but for scalability the FT
can be many and have responsibility for sub-groups of the receivers.
For summary reports, however, there must be a single feedback
aggregating all the summaries to a common message to the whole
receiver group.</t>
<t>The RTP extension for SSM specifies how feedback (both reception
information and specific feedback events) are handled. The more
general problems associated with the use of multicast, where
everyone receives 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 needs to be a member of most or all of these
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 need to have the same RTP and payload type
configuration. Otherwise, A could, for example, be using payload
type 97 to identify the video codec H.264, while B would identify it
as MPEG-2.</t>
<t>Security solutions for this type of group communications are also
challenging. First, 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.</t>
<figure anchor="fig-rams">
<artwork><![CDATA[ ----------- --------------
| |------------------------------------>| |
| |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| |
| | | |
| Multicast | ---------------- | |
| Source | | Retransmission | | |
| |-------->| Server (RS) | | |
| |.-.-.-.->| | | |
| | | ------------ | | |
----------- | | Feedback | |<.=.=.=.=.| |
| | Target (FT)| |<~~~~~~~~~| RTP Receiver |
PRIMARY MULTICAST | ------------ | | (RTP_Rx) |
RTP SESSION with | | | |
UNICAST FEEDBACK | | | |
| | | |
- - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- -
| | | |
UNICAST BURST | ------------ | | |
(or RETRANSMISSION) | | Burst/ | |<~~~~~~~~>| |
RTP SESSION | | Retrans. | |.........>| |
| |Source (BRS)| |<.=.=.=.=>| |
| ------------ | | |
| | | |
---------------- --------------
-------> Multicast RTP Flow
.-.-.-.> Multicast RTCP Flow
.=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages
.......> Unicast RTP Flow]]></artwork>
</figure>
<t>The Rapid acquisition extension allows an endpoint joining an SSM
multicast session to request media starting with the last sync-point
(from where media can be decoded without prior packets) to be sent
at high speed until such time where, after decoding of these bursted
media packets, the correct media timing is established, i.e. media
packets are received within adequate buffer intervals for this
application. This is accomplished by first establishing an unicast
PtP RTP session between the BRS (<xref target="fig-rams"/>) and the
RTP Receiver. That session is used to transmit cached packets from
the multicast group at higher then nominal speed so to synchronize
the receiver to the ongoing multicast packet flow. Once the RTP
receiver and its decoder have caught up with the multicast session's
current delivery, the receiver switches over to receiving from the
multicast group directly. The (still existing) PtP RTP session can
be used as a repair channel, i.e. for RTP Retransmission traffic of
those packets that were not received from the multicast group.</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>An alternative structure for establishing the above topology is to
use independent RTP sessions between each pair of peers, i.e. three
different RTP sessions. In some scenarios, the same RTP media stream
is being sent from each sending endpoint. In others, some form of
local adaptation takes place in one or more of the RTP media streams,
rendering them non-identical. From a topologies viewpoint, a
difference exists in the behaviours around RTCP. For example, when a
single RTP session spans all three endpoints and their connecting
flows, a RTCP bandwidth is calculated and used for this single one
joint session. In contrast, when there are multiple independent RTP
sessions, each has its local RTCP bandwidth allocation. Also, when
multiple sessions are used, endpoints not directly involved in these
sessions do not have any awareness of the conditions occurring in
sessions not involving that endpoint. For example, in case of the
three endpoint configuration above, endpoint A has no awareness of the
conditions occurring in the session between endpoints B and C
(whereas, if a single RTP session were used, it would have such
awareness). Loop detection is also affected. With independent RTP
sessions, the SSRC/CSRC can't be used to determine when a endpoint
receives its own media stream or a mixed media stream including its
own media stream a condition known as a loop. The identification of
loops and, in most cases, its avoidance, has to be achieved by other
means, for example through signaling, or the use of an RTP external
name space binding SSRC/CSRC among any communicating RTP sessions in
the mesh.</t>
</section>
<section anchor="sec-ptm-translator"
title="Point to Multipoint Using the RFC 3550 Translator">
<t/>
<t>This section discusses some additional usages related to point to
multipoint of Translators compared to the point to point only cases in
<xref target="sec-ptp-translators"/>.</t>
<section title="Relay - Transport Translator">
<t>Shortcut name: Topo-PtM-Trn-Translator</t>
<t>This section discusses Transport Translator only usages to enable
multipoint sessions.</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>
<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 set of considerations
those described in the ASM RTP topology. It also puts some
additional signalling requirements onto the session establishment;
for example, a common configuration of RTP payload types is
required.</t>
</section>
<section title="Media Translator">
<t>In the context of multipoint communications a Media Translator is
not providing new mechanisms to establish a multipoint session. It
is much more an enabler or facilitator that ensures one or some
sub-set of session participants can participate in the session.</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. This transcoding can help the other
participants in the Multicast part of the session, by not requiring
the quality transmitted by A to be lowered to the nitrates that B is
actually capable of receiving.</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 generating a new RTP stream and, in most
cases, by manipulation of the media data. 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 can
be 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 different conference participants 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. This also implies that any SDES
information normally needs to be forwarded across the mixer.</t>
<t>The Mixer is responsible for generating RTCP packets in accordance
with its role. It is a receiver and should therefore send receiver
reports for the media streams it receives. In its role as a media
sender, it should also generate sender reports for those media streams
it sends. 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 (including participants A and C), participant B, and
participant D. Assuming all four participants in the conference are
interested in receiving content from each other participant, the Mixer
produces different mixed streams for 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 by the Mixer may result in the generation and
transmission of RTCP feedback messages by the Mixer to the
participants in the other domain(s). 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 they can affect the RTP and RTCP.</t>
<section title="Media Mixing">
<t>The media mixing mixer is likely the one that most think of when
they hear the term "mixer". Its basic pattern of operation is that
it receives media streams from (typically several) participants. Of
those, it selects (either through static configuration or by
dynamic, content dependent means such as voice activation) the
stream(s) to be included in a media domain mix. Then it creates 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 a
mixture of all the input streams; this needs minimal signalling to
be successfully set up. Audio mixing is relatively straightforward
and commonly possible for a reasonable number of participants. Lets
assume that you want to mix N streams from different participants.
The mixer needs to decode those N streams, typically into the sample
domain. Then it needs to produce N or N+1 mixes, the reasons that
different mixes are needed being that each contributing source get a
mix of all other sources except its own, 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 packetized and sent out. In many cases, audio
level normalization is also required before the actual mixing
process.</t>
<t>Video can't really be "mixed" and produce something particularly
useful for the users, however creating an composition out of the
contributed video streams is possible and known as "tiling". For
example the reconstructed, appropriately scaled down videos can be
spatially arranged in a set of tiles, each tile containing the video
from a participant. Tiles can be of different sizes, so that, for
example, a particularly important participant, or the loudest
speaker, is being shown on in larger tile than other participants. A
self-picture can be included in the tiling, which can either be
locally produced or be a feedback from a received and reconstructed
video image (allowing for confidence monitoring, the participant
sees himself/herself just as other participants see him/her). The
tiling normally operates on reconstructed video in the sample
domain. The tiled image is encoded, packetized, and sent by the
mixer. It is possible that a middlebox with media mixing duties
contains only a single mixer of the aforementioned type, in which
case all participants necessarily see the same tiled video, even if
it is being sent over different RTP streams. More common, however,
are mixing arrangement where an individual mixer is available for
each outgoing port of the middlebox, allowing individual
compositions for each participant.</t>
<t>One problem with media mixing is that it consumes both large
amount of media processing (for the actual mixing process in the
uncompressed domain) and encoding resources (for the encoding of the
mixed signal). Another problem is the quality degradation created by
decoding and re-encoding the media that is encapsulated in the RTP
media stream, which is the result of the lossy nature of most, if
not all, commonly used media codecs. A third problem is the latency
introduced by the media mixing, which can be substantial and
annoyingly noticeable in case of video. The advantage of media
mixing is that it is quite simplistic for the clients to handle the
single media stream (which includes the mixed aggregate of many
sources), as they don't need to handle multiple decodings, local
mixing and composition. In fact, mixers were introduced in pre-RTP
times so that legacy, single stream receiving endpoints can
successfully participate in what a user would recognize as a
multiparty session.</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 straightforward
as can be seen in <xref target="fig-media-mixer"/>. The mixer
presents one SSRC towards the receiving client, e.g. MA1 to Peer A;
the associated stream of which is the media mix of the other
participants. As, in this example, each peer receives a different
version produced by the mixer, there is no actual relation between
the different RTP sessions in the actual media or the transport
level information. There are, however, common relationships between
RTP1-RTP3 namely SSRC space and 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, allowing the receiver
to identify the contributing sources even if this were not possible
through the media itself or other signaling means.</t>
<t>The CSRC has, in turn, utility in RTP extensions, like the <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 become one
joint session as they have a common SSRC space. At this stage, the
mixer also need to consider which RTCP information it needs to
expose in the different legs. In the above scenario, commonly, a
mixer would expose nothing more than the Source Description (SDES)
information and RTCP BYE for CSRC leaving the session. 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>Media switching mixers are commonly used in such limited
functionality scenarios where no, or only very limited, concurrent
presentation of multiple sources is required by the application. An
RTP Mixer based on media switching avoids the media decoding and
encoding cycle in the mixer, as it conceptually forwards the encoded
media stream as it was being sent to the mixer, but not the
decryption and re-encryption cycle as it rewrites RTP headers.
Forwarding media (in contrast to reconstructing-mixing-encoding
media) reduces the amount of computational resources needed in the
mixer and increases the media quality (both in terms of fidelity and
reduced latency) per transmitted bit.</t>
<t>A media switching mixer maintains a pool of SSRCs representing
conceptual or functional streams the mixer can produce. These
streams are created by selecting media from one of RTP media streams
received by the mixer and forwarded to the peer using the mixer's
own SSRCs. The mixer can switch between available sources if that is
required by the concept for the source, like currently active
speaker. Note that the mixer, in most cases, still need to perform a
certain amount of media processing, as many media formats do not
allow to "tune" into the stream at arbitrary points of their
bitstream.</t>
<t>To achieve a coherent RTP media stream from the mixer's SSRC, the
mixer needs 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, and each peer is capable of locally handling two video streams
simultaneously.</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, similarly to the Media Mixing
Mixer, reduce the bit-rate required for media transmission towards
the different peers by selecting and forwarding only a sub-set of
RTP media streams it receives from the conference participants. In
many practical cases, the link capacities of either direction
between peers and mixer are the same, which effectively limits the
subset to a single media stream.</t>
<t>To ensure that a media receiver can correctly decode the RTP
media stream after a switch, a state saving (frame-based) codec
needs to start its decoding from independent refresh points or
similar points in the bitstream. For some codecs, for example frame
based speech and audio codecs, this is easily achieved by starting
the decoding at RTP packet boundaries (proper packetization on the
encoder side assumed), as each packet boundary provides a refresh
point. For other (mostly video-) codecs, refresh points are less
common in the bitstream or may not be present at all without an
explicit request to the respective encoder. For this purpose there
exists the <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"/> need to be taken into consideration and apply
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 middlebox can select which of the potential sources that
are currently actively transmitting media, despite that the middlebox,
in another RTP session, may receive media from that end-point. This is
similar to the media switching Mixer but has some important
differences in RTP details.</t>
<figure align="center" anchor="fig-projecting"
title="Media Projecting Middlebox">
<artwork><![CDATA[+-A---------+ +-Middlebox-----------------+
| +-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---+ | | | |
| | | FV1|------------>|---------+-+------>| | |
| | | |<------------|AV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | |
| | | |<------------|EV1 <----+-+-------| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +---------------------------+
]]></artwork>
</figure>
<t>In the six participant conference depicted above <xref
target="fig-projecting">in</xref> one can see that end-point A is
aware of five incoming SSRCs, BV1-FV1. If this middlebox intends to
have a similar 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 need concurrently
transmit media to A. As the middlebox selects the source in the
different RTP sessions that transmit media to the end-points, each RTP
media stream requires some rewriting of RTP header fields when being
projected from one session into another. In particular, the sequence
number needs to be consecutively incremented based on the packet
actually being transmitted in each RTP session. Therefore, the RTP
sequence number offset will change each time a source is turned on in
a RTP session. The timestamp (possibly offset) stays the same.</t>
<t>As the RTP sessions are independent, the SSRC numbers used can also
be handled independently, thereby bypassing the requirement for SSRC
collision detection and avoidance. On the other hand, tools such as
remapping tables between the RTP sessions are required. For example,
the stream that is being sent by endpoint B to the middlebox (BV1) may
use an SSRC value of 12345678. When that media stream is sent to
endpoint F by the middlebox, it can use any SSRC value, e.g. 87654321.
As a result, each endpoint may have a different view of the
application usage of a particular SSRC. Any RTP level identity
information, such as SDES items also needs to update the SSRC
referenced, if the included SDES items are intended to be global. Thus
the application must not use SSRC as references to RTP media streams
when communicating with other peers directly. This also affects loop
detection which will fail to work, as there is no common namespace and
identities across the different legs in the communication session on
RTP level. Instead this responsibility falls onto higher layers.</t>
<t>The middlebox is also responsible to receive any RTCP codec control
requests coming from an end-point, and decide if it can act on the
request locally or needs to translate the request into the RTP session
that contains the media source. Both end-points and the middlebox need
to implement conference related codec control functionalities to
provide a good experience. Commonly used are Full Intra Request to
request from the media source to provide switching points between the
sources, and Temporary Maximum Media Bit-rate Request (TMMBR) to
enable the middlebox to aggregate congestion control responses towards
the media source so to enable it to adjust its bit-rate (obviously
only in case the limitation is not in the source to middlebox
link).</t>
<t>This version of the middlebox 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 be able to handle
as many decoder instances as the middlebox received, or have efficient
switching of decoder contexts in a more limited set of actual decoder
instances to cope with the switches. The application also gets more
responsibility to update how the media provided 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. It also has
mixer aspects, as the streams it provides are not basically translated
version, but instead they have conceptual property assigned to them.
Thus this topology appears to be some hybrid between the translator
and mixer model.</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 was common before, although the RTCP-terminating
MCUs, as discussed in the next section, where perhaps even more
common. This topology, as well as the following one, was a result of
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>
</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">
<t>The implementation of an application may desire to send a subset of
the application's data to each of multiple devices, each with its 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 scheme is possible is highly dependent on the
RTP session usage. It is not really feasible to decompose one logical
end-point into two different transport nodes in one RTP session. A
third party monitor would report such an attempt as two entities being
two different end-points with a CNAME collision. As a result, a fully
RTP conformant de-composited endpoint is one where the different
decomposed parts use 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 different RTP sessions
are used for audio and video. The audio and video parts, however, use
a common CNAME and also have a common clock to ensure that
synchronization 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 requirement that can be derived from the above usage is that
the transport flows for each RTP session might be under common
control, but still are addressed 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. Therefore, 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 locally generated reports reflecting the situation 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 does not forward RTCP from sources in other domains,
but instead generates 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 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.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.3022'?>
<?rfc include='reference.RFC.4607'?>
<?rfc include="reference.RFC.5760"?>
<?rfc include='reference.RFC.5766'?>
<?rfc include='reference.RFC.6285'?>
<?rfc include='reference.RFC.6465'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-security-options'?>
<?rfc include='reference.I-D.lennox-avtcore-rtp-multi-stream'?>
</references>
</back>
</rfc>
| PAFTECH AB 2003-2026 | 2026-04-23 13:32:53 |