One document matched: draft-ietf-avtcore-rtp-topologies-update-04.xml
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<rfc category="info" docName="draft-ietf-avtcore-rtp-topologies-update-04"
ipr="trust200902" obsoletes="5117">
<front>
<title abbrev="RTP Topologies">RTP Topologies</title>
<author fullname="Magnus Westerlund" initials="M." 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." 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 is 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 the Multipoint Control Unit or MCU.</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. In practice, however, most multipoint
conferences operate utilizing centralized units referred to as MCUs.
MCUs may implement Mixer or Translator functionality (in <xref
target="RFC3550">RTP</xref> terminology), and signalling support. They
may also contain additional application layer 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>
<t>This document also codifies new multipoint architectures that have
recently been introduced and which were not anticipated in RFC 5117.
These architectures use scalable video coding and simulcasting, and
their associated centralized units are referred to as Selective
Forwarding Units (SFU). This codification provides a common information
basis for future discussion and specification work.</t>
<t>The document's attempt to clarify and explain sections of the <xref
target="RFC3550">Real-time Transport Protocol (RTP) spec</xref> is
informal. It is not intended to update or change what is normatively
specified within RFC 3550.</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="Middlebox:">A device that is on the Path that media
travel between two Endpoints</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="SFU:">Selective Forwarding Unit</t>
<t hangText="SSM:">Source-Specific Multicast</t>
<t hangText="SSRC:">Synchronization Source</t>
</list></t>
</section>
<section title="Definitions related to RTP grouping taxonomy">
<t> [Note to RFC editor: The following definitions have been taken from
draft-ietf-avtext-rtp-grouping-taxonomy-02 (taxonomy draft henceforth).
It is avtcore working group agreement to not delay the publication of
the topologies-update document through a dependency to the taxonomy
draft. If, however, the taxonomy draft and this draft are in your
work queue at the same time and there would be no significant additional
delay (through your schedule, normative reference citations, or similar)
in publishing both documents roughly in parallel, it would be preferable
to replace the definition language with something like "as in [RFC YYYY]"
where YYYY would be the RFC number of the published taxonomy draft.] </t>
<t> The following definitions have been taken from
draft-ietf-avtext-rtp-grouping-taxonomy-02, and are used in capitalized form
throughout the document. </t>
<t><list style="hanging">
<t hangText="Communication Session:">A Communication Session is an association
among group of participants communicating with each other via a set of
Multimedia Sessions.</t>
<t hangText="End Point:">A single addressable entity sending or receiving RTP
packets. It may be decomposed into several functional blocks, but as
long as it behaves as a single RTP stack entity it is classified as
a single "End Point".
</t>
<t hangText="Media Source:">A Media Source is the logical source of a reference clock
synchronized, time progressing, digital media stream, called a Source
Stream.
</t>
<t hangText="Multimedia Session: ">A multimedia session is an association among a group of participants
engaged in the communication via one or more RTP Sessions.
</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, with or without middleboxes. Then follows a
number of different methods for establishing point to multipoint
communication. These are structured around the most fundamental enabler,
i.e., multicast, a mesh of connections, translators, mixers and finally
MCUs and SFUs. The section ends by discussing de-composited terminals,
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"/>.</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 End Points, communicating using
unicast. Both RTP and RTCP traffic are conveyed endpoint-to-endpoint,
using unicast traffic only (even if, in exotic cases, this unicast
traffic happens to be conveyed over an IP-multicast address).</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 End Points and combining the requirements stemming
from them. Note that an End Point 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, subject
only to system level limitations like the number range of ports.</t>
<t>RTCP feedback messages for the indicated SSRCs are communicated
directly between the End Points. Therefore, this topology poses minimal
(if any) issues for any feedback messages. For RTP sessions which use
multiple SSRC per End Point it can be relevant to implement support for
cross-reporting suppression as defined in <xref
target="I-D.ietf-avtcore-rtp-multi-stream-optimisation">"Sending
Multiple Media Streams in a Single RTP Session"</xref>.</t>
</section>
<section title="Point to Point via Middlebox">
<t>This section discusses cases where two End Points communicate but
have one or more middleboxes 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
RTP stream that the Translator receives, it generates an
individual RTP stream in the other domain. A translator keeps the SSRC
for an RTP stream across the translation, whereas a Mixer can select a
single RTP stream from multiple received RTP streams (in cases like
audio/video switching), or send out an RTP stream composed of multiple mixed
media received in multiple RTP streams (in cases like audio mixing or
video tiling), 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 of a middlebox is enabling interoperability,
between otherwise non-interoperable End Points, 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 End Points (as the RTP session 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 RTP 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 sending End Point, by sending feedback
messages. While such feedback could use the SSRC of the target for
the translator (the receiving End Point), 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 is
End Points failing to support the full RTP specification may have
issues with multiple SSRCs reporting on the RTP streams sent by that
End Point, as this use case may be viewed as excotic by implementers.</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 End Points on the transport
level, e.g., to 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 RTP stream is required to pass
a particular point for audit by employing relaying, or preserving
privacy by hiding each peer's 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>A common element in these functions is that they are normally
transparent at the RTP level, i.e., they perform no changes on any
RTP or RTCP packet fields and only affect the lower layers. They
may affect, however, the path the RTP and RTCP packets are routed
between the End Points in the RTP session, and thereby
indirectly affect the RTP session. For this reason, one could
believe that transport translator-type middleboxes do not need to
be included in this document. This topology, however, can raise
additional requirements in the RTP implementation and its
interactions with the signalling solution. Both in signalling and
in certain RTCP fields, network addresses other than those of the
relay can occur since B has a different network address than the
relay (T). Implementations that cannot support this will also not
work correctly when End Points are subject to NAT.</t>
<t>The transport relay implementations also have to take into account security
considerations. In particular, source address filtering of incoming packets is usually
important in relays, to prevent attackers to inject traffic into a
session, which one peer may, in the absence fo adequate security in the
relay, think it comes from the other peer.</t>
</section>
<section title="Transport Translator">
<t>Transport Translators (Topo-Trn-Translator) do not modify the
RTP 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). </t>
<t>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, an important
consideration is how to handle different sources and their
identities. This problem space is not discussed henceforth.</t>
<t> Of the
transport Translators, this memo is primarily interested in those
that use RTP on both sides, and this is assumed henceforth.</t>
<t>The most basic transport translators that operate below the RTP
level were already discussed in <xref
target="sec-transport-anchor"/>.</t>
</section>
<section title="Media Translator">
<t>Media Translators (Topo-Media-Translator) modify the media inside
the RTP stream. This process is commonly known as transcoding. The
modification of the media can be as small as removing parts
of the stream, and it can go all the way to a full decoding and
re-encoding (down to the sample level or equivalent) utilizing a
different media codec. Media Translators are commonly used to
connect End Points 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 Translator is used to translate
both the media and the transport aspects of the RTP stream carrying
the media between two transport domains.</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 End Point B's RTCP Receiver
Report before forwarding them to End Point A. The rewriting is needed as the
RTP stream received by B is not the same RTP 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 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 requirements exists
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 (the End Point originally sending the media to the Translator)
and respond to RTCP feedback messages. This may
occur, for example, when a receiving End Point requests a bandwidth
reduction, and the media Translator has not detected any
congestion or other reasons for bandwidth reduction between the
sending End Point and itself. In that case, it is sensible that the
media Translator reacts to 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
End Point A sending a RTP stream containing media (only) to B.
On the path there is a device
T that on A's behalf manipulates the RTP streams. One common example is
that T adds a second RTP stream containing Forward Error Correction (FEC)
information in order to protect A's (non FEC-protected) RTP stream.
In this case, T needs to semantically bind the new FEC RTP stream to A's
media-carrying RTP stream, for example by using the same CNAME as A.</t>
<figure align="center" anchor="fig-de-composite-translator"
title="Media Translator adding FEC">
<artwork><![CDATA[
+------+ +------+ +------+
| | | | | |
| A |------->| T |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+]]></artwork>
</figure>
<t>there may also be cases where information is added into the original
RTP stream, while leaving most or all of the original RTP packets intact
(with the exception of certain RTP header fields, such as the sequence
number). One example is the injection of meta-data into the RTP stream,
carried in their own RTP packets.</t>
<t>Similarly, a Media Translator can sometimes remove information from
the RTP stream, while otherwise leaving teh remaining RTP packets
unchanged (again with the exception of certain RTP header fields). </t>
<t>Either type of functionality where T manipulates the RTP stream,
or adds an accompanying RTP stream, on behalf of A is also covered
under the media translator definition.</t>
</section>
</section>
<section title="Back to Back RTP sessions">
<t>There exist middleboxes that interconnect two End Points A and B through
themselves (MB), but not by being part of a common RTP session. They
establish instead two different RTP sessions, one between A and the
middlebox and another between the middlebox and B. This topology is
called Topo-Back-To-Back</t>
<figure align="center" anchor="fig-b2b-session"
title="Back-to-back RTP sessions through Middlebox">
<artwork><![CDATA[
|<--Session A-->| |<--Session B-->|
+------+ +------+ +------+
| A |------->| MB |-------->| B |
+------+ +------+ +------+]]></artwork>
</figure>
<t>The middlebox acts as an application-level gateway and bridges
the two RTP sessions. This bridging can be as basic as forwarding
the RTP payloads between the sessions, or more complex including
media transcoding. The difference of this topology relative to 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 middlebox's respnsibility to
maintain the correct relations.</t>
<t>The signalling or other above-RTP level functionalities
referencing RTP 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>Adherence to congestion control can be solved locally on each of
the two segments, or by
bridging statistics from the receiving End Point through the middlebox
to the sending End Point. From an
implementation point, however, the latter requires dealing with a number
of inconsistencies. First, packet loss must be detected for an RTP
stream sent from A to the middlebox, and that loss must be reported
through a skipped sequence number in the RTP stream from the middlebox to
B. This coupling and the resulting inconsistencies are conceptually
easier to handle when considering the two RTP streams as belonging to a
single RTP session.</t>
</section>
</section>
<section title="Point to Multipoint Using Multicast">
<t>Multicast is an IP layer functionality that is available in some
networks. Two main flavors can be distinguished: <xref
target="RFC1112">Any Source
Multicast (ASM)</xref> where any multicast
group participant can send to the group address and expect the packet
to reach all group participants; and <xref target="RFC3569">Source
Specific Multicast (SSM)</xref>, where only a particular IP host sends
to the multicast group. Both these models are discussed below in their
respective sections.</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
multicast group participant reaches all the other multicast group
participants, except for cases
such as:<list style="symbols">
<t>packet loss, or</t>
<t>when a multicast group 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 multimedia session is distributed using two or
more multicast groups and a multicast group 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 multicast group 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 multicast group 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
likelihood that the threshold of the small multicast group (in terms
of multicast group participants) may be exceeded during the lifetime of a
session.</t>
<t>RTCP feedback messages in multicast reach, like media data, 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 End Point that is a multicast group participant needs
to process every feedback message it receives,
not only to determine if it is affected or if the feedback message
applies only to some other End Point, but also to derive timing
restrictions 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 multicast group participants can send to all
the other multicast group participants, by sending a packet to the multicast group.
In contrast, <xref target="RFC3569">Source Specific
Multicast</xref><xref target="RFC4607"/> 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 sending End Points (RTP sources henceforth) (1 to M) are allowed to send media to the SSM
group. These sources send media to a dedicated distribution source,
which forwards the RTP streams to the multicast group on behalf of
the original RTP sources. The RTP streams reach the receiving End Points (Receivers henceforth) (R(1) to
R(n)). The Receivers' RTCP messages cannot be sent to the multicast
group, as the SSM multicast group by definition has only a single IP
sender. 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 a 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 distributed and each instance can have responsibility for sub-groups of the receivers.
For summary reports, however, there typically must be a single feedback target
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>Aforementioned situation results in common behavior for RTP
multicast:<list style="numbers">
<t>Multicast applications often use a group of RTP sessions, not
one. Each End Point 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 End Points need to have the same RTP and payload type
configuration. Otherwise, End Point A could, for example, be using payload
type 97 to identify the video codec H.264, while End Point B would identify it
as MPEG-2, with unpredicatble but almost certainly not visually pleasing results.</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 specialized solutions. For more discussion on
this please review <xref target="RFC7201">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 Stream
.-.-.-.> Multicast RTCP Stream
.=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages
.......> Unicast RTP Stream]]></artwork>
</figure>
<t>The Rapid acquisition extension allows an End Point joining an SSM
multicast session to request media starting with the last sync-point
(from where media can be decoded without requiring context
established by the decoding of prior packets) to be sent at high
speed until such time where, after decoding of these burst-delivered
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 a unicast
PtP RTP session between the Burst/Retransmission Source (BRS, <xref
target="fig-rams"/>) and the RTP Receiver. The unicast session is
used to transmit cached packets from the multicast group at higher
then normal speed in order to synchronize the receiver to the
ongoing multicast RTP stream. Once the RTP receiver and its decoder
have caught up with the multicast session's current delivery, the
receiver switches over to receiving directly from the multicast
group. The (still existing) PtP RTP session is, in many deployed
applications, 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 involving three or more End Points over multiple unicast transport flows, like the
joint three End point session depicted above. In this case, A needs to send its
RTP streams and RTCP packets to both B and C over their respective
transport flows. As long as all End Points do the same, everyone
will have a joint view of the RTP session.</t>
<t>This topology does not create any additional requirements beyond the need to
have multiple transport flows associated with a single RTP session.
Note that an End Point may use a single local port to receive all these
transport flows (in which case the sending port, IP address, or SSRC can
be used to demultiplex), or it might have separate local reception ports for
each of the End Points.</t>
<figure anchor="fig-mesh-joint-session"
title="An Multi-unicast Mesh with a joint RTP session">
<artwork><![CDATA[
+-A--------------------+
|+---+ |
||CAM| | +-B-----------+
|+---+ +-UDP1------| |-UDP1------+ |
| | | +-RTP1----| |-RTP1----+ | |
| V | | +-Video-| |-Video-+ | | |
|+----+ | | | |<----------------|BV1 | | | |
||ENC |----+-+-+--->AV1|---------------->| | | | |
|+----+ | | +-------| |-------+ | | |
| | | +---------| |---------+ | |
| | +-----------| |-----------+ |
| | | +-------------+
| | |
| | | +-C-----------+
| | +-UDP2------| |-UDP2------+ |
| | | +-RTP1----| |-RTP1----+ | |
| | | | +-Video-| |-Video-+ | | |
| +-------+-+-+--->AV1|---------------->| | | | |
| | | | |<----------------|CV1 | | | |
| | | +-------| |-------+ | | |
| | +---------| |---------+ | |
| +-----------| |-----------+ |
+----------------------+ +-------------+
]]></artwork>
</figure>
<t>A joint RTP session from End Point A's perspective for the Mesh depicted in
<xref target="fig-mesh"/> with a joint RTP session have multiple
transport flows, here enumerated as UDP1 and UDP2. However, there is
only one RTP session (RTP1). The Media Source (CAM) is encoded and
transmitted over the SSRC (AV1) across both transport layers. However,
as this is a joint RTP session, the two streams must be the same.
Thus, an congestion control adaptation needed for the paths A to B and
A to C needs to use the most restricting path's properties.</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 stream
may be sent from the transmitting End Point, however it also supports local
adaptation taking place in one or more of the RTP streams,
rendering them non-identical.</t>
<figure anchor="fig-mesh-diff-session"
title="An Multi-unicast Mesh with independent RTP session">
<artwork><![CDATA[
+-A----------------------+ +-B-----------+
|+---+ | | |
||MIC| +-UDP1------| |-UDP1------+ |
|+---+ | +-RTP1----| |-RTP1----+ | |
| | +----+ | | +-Audio-| |-Audio-+ | | |
| +->|ENC1|--+-+-+--->AA1|------------->| | | | |
| | +----+ | | | |<-------------|BA1 | | | |
| | | | +-------| |-------+ | | |
| | | +---------| |---------+ | |
| | +-----------| |-----------+ |
| | ------------| |-------------|
| | | |-------------+
| | |
| | | +-C-----------+
| | | | |
| | +-UDP2------| |-UDP2------+ |
| | | +-RTP2----| |-RTP2----+ | |
| | +----+ | | +-Audio-| |-Audio-+ | | |
| +->|ENC2|--+-+-+--->AA2|------------->| | | | |
| +----+ | | | |<-------------|CA1 | | | |
| | | +-------| |-------+ | | |
| | +---------| |---------+ | |
| +-----------| |-----------+ |
+------------------------+ +-------------+
]]></artwork>
</figure>
<t>Lets review the topology when independent RTP sessions are used,
from A's perspective in <xref target="fig-mesh"/> by considering both
how the media is a handled and the RTP sessions that are set-up in
<xref target="fig-mesh-diff-session"/>. A's microphone is captured and
the digital audio can then be fed into two different encoder
instances, as each being associated with two independent RTP sessions
(RTP1 and RTP2). The SSRCs (AA1 and AA2) in each RTP session are
completely independent and the media bit-rate produced by the encoders
can also be tuned differently to address any congestion control
requirements differing for the paths A to B compared to A to C.</t>
<t>From a topologies viewpoint, an important difference exists in the
behavior around RTCP. First, when a single RTP session spans all three
End Points A, B, and C, and their connecting RTP streams, a common RTCP bandwidth is
calculated and used for this single joint session. In contrast, when
there are multiple independent RTP sessions, each RTP session has its
local RTCP bandwidth allocation.</t>
<t>Further, when multiple sessions are used, End Points not directly
involved in a session do not have any awareness of the conditions in
those sessions. For example, in the case of the three End Point
configuration in <xref target="fig-mesh"/>, End Point A has no
awareness of the conditions occurring in the session between End Points
B and C (whereas, if a single RTP session were used, it would have
such awareness).</t>
<t>Loop detection is also affected. With independent RTP sessions, the
SSRC/CSRC cannot be used to determine when an End Point 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, their 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) End Points B
and D to take part in an any source multicast session involving End Points A and C, 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"/>. The Translator in this case
connects multiple End Points 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 End Points. 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
to 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>
<t>Transport translators and relays should always consider implementing
source address filtering, to prevent attackers to inject traffic
using the listening ports on the translator. The translator can,
however, go one step further, and especially if explicit SSRC
signalling is used, prevent End points to send SSRCs other than its own
(that are, for example, used by other participants in the session). This can improve
the security properties of the session, despite the use of group
keys that on cryptographic level allows anyone to impersonate
another in the same RTP session.</t>
<t>A Translator that doesn't change the RTP/RTCP packets content can
be operated without the requiring it to have access to
the security contexts used to protect the RTP/RTCP traffic between
the participants.</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 more of an enabler, or facilitator, that ensures a given End Point or
a defined sub-set of End Points can participate in the session.</t>
<t>If End Point 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 End Points to reach B
without overloading the path. This transcoding can help the other
End Points in the multicast part of the session, by not requiring
the quality transmitted by A to be lowered to the bitrates that B is
actually capable of receiving (and vice versa).</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 that are
part of a session by generating one or more new RTP streams and, in most cases,
by manipulating 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 RTP streams
received from other End Points in the same RTP session. Using the media data carried in
the received RTP streams, a Mixer generates derived RTP streams that are
sent to the receiving End Points.</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 Communication Session.</t>
<t>The Mixer creates the Media Source and the source RTP stream just like
an End Point, as it mixes the content (often in
the uncompressed domain) and then encodes and packetizes it for transmission to a
receiving endpoint. The CSRC Count (CC) and CSRC fields in the RTP header can
be used to indicate the contributors to the newly generated RTP 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 End Points to allow for loop detection and
identification of sources that are part of the Communication 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 an RTP receiver and should therefore send RTCP receiver
reports for the RTP streams it receives and terminates. In its role as an RTP
sender, it should also generate RTCP sender reports for those RTP 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 End Points A and C), End Point B, and
End Point D. Assuming all four End Points in the conference are
interested in receiving content from each other End Point, the Mixer
produces different mixed RTP 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 where it 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. The mixing operation, however, 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
End Points in the other domain(s). In other cases, a message is
handled by the Mixer locally 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>We now discuss in more detail the different mixing operations that
a mixer can perform and how they can affect RTP and RTCP behavior.</t>
<section title="Media Mixing Mixer">
<t>The media mixing mixer is likely the one that most think of when
they hear the term "mixer". Its basic mode of operation is that it
receives RTP streams from several End Points and selects the
stream(s) to be included in a media-domain mix. The selection can be
through static configuration or by dynamic, content dependent means
such as voice activation. The mixer then creates a single outgoing
RTP 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 audio signals; this needs minimal signalling to
be successfully set up. From a signal processing viewpoint, audio mixing is relatively straightforward
and commonly possible for a reasonable number of End Points.
Assume, for example, that one wants to mix N streams from N different
End Points. The mixer needs to decode those N streams, typically
into the sample domain, and then produce N or N+1 mixes. Different
mixes are needed so that each contributing source gets 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 End points, 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, noise suppression, and similar signal processing steps
are also required or desirable before the actual mixing process commences.</t>
<t>In video, the term "mixing" has a different interpretation than
audio. It is commonly used to refer to the process of spatially
combining contributed video streams, which is also known as "tiling". The
reconstructed, appropriately scaled down videos can be spatially
arranged in a set of tiles, each tile containing the video from an
End Point (typically showing a human 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-view
picture can be included in the tiling, which can either be locally
produced or be a feedback from a mixer-received and reconstructed video
image. Such remote loopback allows for confidence monitoring, i.e.,
it enables the participant to see himself/herself in the same quality 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 to the receiving End Points. 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 receiving End Point (a
feature commonly referred to as personalized layout).</t>
<t>One problem with media mixing is that it consumes both large
amounts of media processing resources (for the decoding and 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, which is the result of the lossy nature of most
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, or in case of audio if that
mixed audio is lip-sychronized with high latency video. The
advantage of media mixing is that it is straightforward for the
End Points 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 (that, in some protocol environments, actually didn't need to
be aware of the multipoint nature of teh conference) could successfully
participate in what a user would
recognize as a multiparty video conference.</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 a very simple
process, as can be seen in <xref target="fig-media-mixer"/>. The
mixer presents one SSRC towards the receiving End Point, e.g., MA1 to
Peer A, where the associated stream is the media mix of the other
End Points. As each peer, in this example, receives a different
version of a mix from the mixer, there is no actual relation between
the different RTP sessions in terms of actual media or 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 through 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 SSRCs from the End Point to mixer paths are
used as CSRCs 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 needs to consider which RTCP information it
needs to expose in the different paths. In the above scenario, a
mixer would normally expose nothing more than the Source Description
(SDES) information and RTCP BYE for a 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 used in limited functionality
scenarios where no, or only very limited, concurrent presentation of
multiple sources is required by the application, to more complex
multi-stream usages with receiver mixing or tiling, including
combined with simulcast and/or scalability between source and mixer.
An RTP Mixer based on media switching avoids the media decoding and
encoding operations in the mixer, as it conceptually forwards the
encoded media stream as it was being sent to the mixer. It does not
avoid, however, 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).</t>
<t>A media switching mixer maintains a pool of SSRCs representing
conceptual or functional RTP streams that the mixer can produce. These
RTP streams are created by selecting media from one of the RTP
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 the currently
active speaker. Note that the mixer, in most cases, still needs to
perform a certain amount of media processing, as many media formats
do not allow to "tune into" the stream at arbitrary points in their
bitstream.</t>
<t>To achieve a coherent RTP 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. Second, the
sequence number must be the next in the sequence of outgoing packets
it sent. Third, the RTP timestamp value needs to be adjusted using
an offset that changes each time one switches media source. Finally,
depending on the negotiation of the RTP payload type, the value
representing this particular RTP payload configuration may have to
be changed if the different End Point-to-mixer paths have not arrived on
the same numbering for a given configuration. This also requires
that the different End Points support a common set of codecs,
otherwise media transcoding for codec compatibility would still be
required.</t>
<t>We now consider the operation of a media switching mixer that
supports a video conference with six participating End Points (A-F) where the
two most recent speakers in the conference are shown to each
receiving End Point. The mixer has thus 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 streams it receives from the sending End Points. In
cases the mixer receives simulcast transmissions or a scalable
encoding of the media source, the mixer has more degrees of freedom
to select streams or sub-sets of stream to forward to a receiving End Point,
both based on transport or End Point restrictions as well as
application logic.</t>
<t>To ensure that a media receiver in an End Point can correctly decode the media in the RTP
stream after a switch, a codec that uses temporal prediction
needs to start its decoding from independent refresh points, or
points in the bitstream offering similar functionality (like "dirty refresh points"). For some codecs, for example frame
based speech and audio codecs, this is easily achieved by starting
the decoding at RTP packet boundaries, as each packet boundary
provides a refresh point (assuming proper packetization on the
encoder side). For other codecs, particularly in video, refresh
points are less common in the bitstream or may not be present at all
without an explicit request to the respective encoder. The <xref
target="RFC5104">Full Intra Request</xref> RTCP codec control
message has been defined for this purpose.</t>
<t>In this type of mixer one could consider to fully terminate the
RTP sessions between the different End Point and mixer paths. 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="Selective Forwarding Middlebox">
<t>Another method for handling media in the RTP mixer is to "project",
or make available, 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 will be sent to each of the End Points. This is similar to the
media switching Mixer but has some important differences in RTP
details.</t>
<figure align="center" anchor="fig-projecting"
title="Selective Forwarding 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 End Point 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 behavior 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
stream requires rewriting of certain 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 RTP stream that is being sent by End Point B to the middlebox (BV1) may
use an SSRC value of 12345678. When that RTP stream is sent to
End Point F by the middlebox, it can use any SSRC value, e.g. 87654321.
As a result, each End Point 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 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>The selective forwarding middlebox has been introduced in recently
developed videoconferencing systems in conjunction with, and to
capitalize on, scalable video coding as well as simulcasting. An
example of scalable video coding is Annex G of H.264, but other
codecs, including H.264 AVC and VP8 also exhibit scalability, albeit
only in the temporal dimension. In both scalable coding and simulcast
cases the video signal is represented by a set of two or more
bitstreams, providing a corresponding number of distinct fidelity
points. The middlebox selects which parts of a scalable bitstream (or
which bitstream, in the case of simulcasting) to forward to each of
the receiving End Points. The decision may be driven by a number of
factors, such as available bit rate, desired layout, etc. Contrary to
transcoding MCUs, these "Selective Forwarding Units" (SFUs) have
extremely low delay, and provide features that are typically
associated with high-end systems (personalized layout, error
localization) without any signal processing at the middlebox. They are
also capable of scaling to a large number of concurrent users,
and--due to their very low delay--can also be cascaded.</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 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 that this topology 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>
<t>The differences between selective forwarding middlebox and a <xref
target="sec-media-switching">switching mixer</xref> are minor, and
they share most properties. The above requirement on having a large
number of decoding instances or requiring efficient switching of
decoder contexts, are one point of difference. The other is how the
identification is performed, where the Mixer uses CSRC to provide information on
what is included in a particular RTP stream that represent a
particular concept. Selective forwarding gets the source information
through the SSRC, and instead have to use other mechanism to make
clear the streams current purpose.</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 popular in early implementations of
multipoint videoconferencing systems due to its simplicity, and the
corresponding middlebox design has been known as a "video switching
MCU". The more complex RTCP-terminating MCUs, discussed in the next
section, became the norm, however, when technology allowed
implementations at acceptable costs.</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 End points. 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 RTP stream results in most participants seeing only a subset
of the sent RTP streams at any given time, often a single RTP stream per
conference. Therefore, RTCP Receiver Reports only report on these
RTP streams. Consequently, the End Points emitting RTP streams that are not currently
forwarded receive a view of the session that indicates their RTP
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 RTP 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 End Point 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
End Points, or</t>
<t>the mixed aggregate of what the MCU receives from the other PtP
paths, which are part of the same Communication Session.</t>
</list></t>
<t>In case (a), the MCU may modify the content in terms of bit-rate,
encoding format, or resolution. No explicit RTP mechanism is used to
establish the relationship between the original RTP stream of the media being sent
RTP stream the MCU sends. In other words, the outgoing RTP streams 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 RTP session for each
End Point.</t>
<t>In case (b), the MCU is the Media Source and generates the Source RTP
Stream as it mixes the received content
and then encodes and packetizes it for transmission to an End Point. 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 End Points 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 original sending End Points 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 many/most deployed MCUs (and video conferencing 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>
</section>
<section title="Split Component Terminal">
<t>Shortcut name: Topo-Split-Terminal</t>
<t>In some applications, for example in some telepresence systems,
terminals may be not integrated into a single functional unit, but
composed of more than one subunits. For example, a telepresence room
terminal employing multiple cameras and monitors may consist of
multiple video conferencing subunits, each capable of handling a
single camera and monitor. Another example would be a video
conferencing terminal in which audio is handled by one subunit, and
video by another. Each of these subunits uses its own physical network
interface (for example: Ethernet jack) and network address.
</t>
<t>
The various (media processing) subunits need (logically and
physically) to be interconnected by control functionality, but their
media plane functionality may be split. This type of terminals is
referred to as split component terminals. Historically, the earliest split
component terminals were perhaps the (independent) audio and video conference
software tools used over the MBONE in the late 1990s.
</t>
<t>An example for such a split component terminal is depicted in <xref
target="fig-de-composite"/>. Within split component terminal A,
at least audio and video subunits are addressed by their own
network addresses. In some of these systems, the control stack
subunit may also have its own network address. </t>
<t>From an RTP viewpoint, each of the subunits terminates RTP, and acts
as an End Point in the sense that each subunit includes its own,
independent RTP stack. However, as the subunits are semantically
part of the same terminal, it is appropriate that this semantic
relationship is expressed in RTCP protocol elements, namely in
the CNAME.</t>
<figure align="center" anchor="fig-de-composite"
title="Split Component Terminal">
<artwork><![CDATA[
+---------------------+
| Endpoint A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+-RTP---\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+-RTP-------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+-SIP---/
| +------------+ |
+---------------------+
]]></artwork>
</figure>
<t>
It is further sensible that the subunits share a common clock from
which RTP and RTCP clocks are derived, to facilitate synchronization
and avoid clock drift.
</t>
<t>
To indicate that audio and video Source Streams generated by different
sub-units share a common clock, and can be synchronized, the RTP streams
generated from those Source Streams need to include the same CNAME in
their RTCP SDES packets. The use of a common CNAME for RTP flows carried
in different transport-layer flows is entirely normal for RTP and RTCP
senders, and fully compliant RTP End Points, middle-boxes, and other
tools should have no problem with this.
</t>
<t>
However, outside of the split component terminal scenario (and perhaps
a multi-homed End Point scenario, which is not further discussed herein),
the use of a common CNAME in RTP streams sent from separate endpoints
(as opposed to a common CNAME for RTP streams sent on different transport
layer flows between two endpoints) is rare. It has been reported that
at least some third party tools like some network monitors do not handle
endpoints that use of a common CNAME across multiple transport layer
flows gracefully: they report an error condition that two separate
End Points are using the same CNAME. Depending on the sophistication
of the support staff, such erroneous reports can lead to support issues.
</t>
<t>
Aforementioned support issue can sometimes be avoided if each of the
subunits of a split component terminal is configured to use a different
CNAME, with the synchronization between the RTP streams being indicated
by some non-RTP signaling channel rather than using a common CNAME sent
in RTCP. This complicates the signaling, especially in cases where
there are multiple SSRCs in use with complex synchronization requirements,
as is the same in many current telepresence systems. Unless one uses RTCP
terminating topologies such as Topo-RTCP-terminating-MCU, sessions
involving more than one video subunit with a common CNAME are close to
unavoidable.
</t>
<t>
The different RTP streams comprising a split terminal system can form a
single RTP session or they can form multiple RTP sessions, depending on
the visibility of their SSRC values in RTCP reports. If the receiver of
the RTP streams sent by the split terminal sends reports relating to all
of the RTP flows (i.e., to each SSRC) in each RTCP report then a single
RTP session is formed. Alternatively, if the receiver of the RTP streams
sent by the split terminal does not send cross-reports in RTCP, then the
audio and video form separate RTP sessions.
</t>
<t>
For example, in the <xref target="fig-de-composite"/>, B will send RTCP reports to each of the
sub-units of A. If the RTCP packets that B sends to the audio sub-unit
of A include reports on the reception quality of the video as well as
the audio, and similarly if the RTCP packets that B sends to the video
sub-unit of A include reports on the reception quality of the audio as
well as video, then a single RTP session is formed. However, if the
RTCP packets B sends to the audio sub-unit of A only report on the
received audio, and the RTCP packet B sends to the video sub-unit of
A only report on the received video, then there are two separate RTP
sessions.
</t>
<t>
Forming a single RTP session across the RTP streams sent by the different
sub-units of a split terminal gives each sub-unit visibility into reception
quality of RTP streams sent by the other sub-units. This information can
help diagnose reception quality problems, but at the cost of increased RTCP
bandwidth use.
</t>
<t>
RTP streams sent by the sub-units of a split terminal need to use the
same CNAME in their RTCP packets if they are to be synchronized,
irrespective of whether a single RTP session is formed or not.
</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 RTP 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 media 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 RTP streams 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 for implementation.</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. It 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 describes these properties and
then analyzes how these properties are supported by the different
topologies. Note that, even if a certain property is supported within a
particular topology concept, the necessary functionality may be optional
to implement.</t>
<section title="Topology Properties">
<section title="All to All Media Transmission">
<t> To recapitulate, multicast, and in particular Any Source Multicast (ASM), provides the
functionality that everyone may send to, or receive from, everyone
else within the session. Source-specific Multicast (SSM) can provide a similar functionality by having anyone intending to participate as sender to
send its media to the SSM distribution source. The SSM distribution source forwards the media
to all receivers subscribed to the multicast group. Mesh, MCUs, Mixers, SFMs 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>Closest to true IP-multicast-based, all-to-all transmission comes perhaps the transport Translator function called "relay" in
in <xref
target="sec-ptm-translator"/>, as
well as the Mesh with joint RTP sessions. Media Translators, Mesh with independent RTP
Sessions, Mixers, SFUs 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 End Point considers the path to the least capable receiving End Point.
Otherwise, the media transmissions may overload that path.
Therefore, a sending End Point needs to monitor the path from itself to
any of the receiving End Points, to detect the currently least capable
receiver, and adapt its sending rate accordingly. As multiple
End Points 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 resource consumption for performing all to all transmission
varies depending with the topology. Both ASM and SSM have the benefit that only one copy of
each packet traverses a particular link. Using a relay causes the transmission of one copy of a packet
per End Point-to-relay path and packet transmitted. However, in
most cases the links carrying the multiple copies will be the ones close
to the relay (which can be assumed to be part of the network infrastructure with
good connectivity to the backbone), rather than the End Points (which may be behind
slower access links).
The Mesh causes N-1 streams of transmitted packets to traverse the
first hop link from the End Point, in an N End Point mesh. How long the
different paths are common, is highly situation dependent.
</t>
<t>The transmission of RTCP by design 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>All Translators, Mixers, and RTCP-terminating MCU, and Mesh with
individual RTP sessions, allow changing the media encoding or the
transport to other properties of the other domain, thereby providing
extended interoperability in cases where the End Points lack a
common set of media codecs and/or transport protocols. Selective
Forwarding Middleboxes can adopt the transport, and (at least) selectively
forward the encoded streams that match a receiving End Point's capability.
It requires an additional translator to change the media
encoding if the encoded streams do not match the receiving End Point's capabilities.
</t>
</section>
<section title="Per Domain Bit-Rate Adaptation">
<t> End Points are often connected to each other with a
heterogeneous set of paths. This makes congestion control in a Point
to Multipoint set problematic. For the ASM, SSM, Mesh with common
RTP session, and Transport Relay scenario, each individual sending End Point has
to adapt to the receiving End Point behind the least capable path, yielding
suboptimal quality for the End Points behind the more capable paths. This is no
longer an issue when Media Translators, Mixers, SFM or MCUs are
involved, as each End Point only needs to adapt to the slowest path
within its own domain. The Translator, Mixer, SFM, or MCU topologies
all require their respective outgoing RTP 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 End Point in all the domains at the cost (complexity,
delay, equipment) of the Mixer, SFM or Translator, and potentially
media sender (multicast/layered encoding and sending the different representations).
</t>
</section>
<section title="Aggregation of Media">
<t> In the all-to-all media property mentioned above and provided by ASM,
SSM, Mesh with common RTP session, and relay, all simultaneous media
transmissions share the available bit-rate. For End Points with
limited reception capabilities, this may result in a situation where
even a minimal acceptable media quality cannot be accomplished, because multiple
RTP streams need to share the same
resources. One solution to this problem is to provide for
a Mixer, or MCU to aggregate the multiple RTP streams into a single one,
where the single RTP stream takes up less resources in terms of bit-rate.
This aggregation can be performed according to different methods.
Mixing or selection are two common methods. Selection is almost always possible
and easy to implement. Mixing requires resources in the mixer, and may be
relatively easy and not impairing the quality too badly (audio) or quite
difficult (video tiling, which is not only computationally complex but also
reduces the pixel count per stream, with corresponding loss in perceptual quality).
</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).
In topologies that provide a full all-to-all
functionality, i.e. ASM, Mesh with common RTP session, Relay a compliant RTP
implementation offers the functionality directly as specified in RTP. In topologies
that do not offer all-to-all communication, it is necessary that RTCP is handled
correctly in domain bridging function. RTP includes explicit specification text for
Translators and Mixers, and for SFMs the required functionality can be derived from
that text. However, the MCU described in
<xref target="sec-ptm-switch-mcu"/> cannot offer the full functionality for
session participant identification through RTP means.
The topologies that create independent
RTP sessions per End Point or pair of End Points, like Back-to-Back RTP
session, MESH with independent RTP sessions, and the RTCP terminating
MCU <xref target="sec-ptm-mcu">RTCP
terminating MCU</xref> do not support RTP based identification of session
participants. In all those cases, other non-RTP based mechanisms need to be
implemented if such knowledge is required or desirable.</t>
</section>
<section title="Loop Detection">
<t>In complex topologies with multiple interconnected domains, it is
possible to unintentionally form media loops. RTP and RTCP support detecting such
loops, as long as the SSRC and CSRC identities are maintained and
correctly set in forwarded packets. Loop detection will work in ASM,
SSM, Mesh with joint RTP session, and Relay. It is likely that loop
detection works for the video switching MCU <xref
target="sec-ptm-switch-mcu"/>, at least as long as it forwards the
RTCP between the End Points. However, the Back-to-Back RTP
sessions, Mesh with independent RTP sessions, SFM, will definitely
break the loop detection mechanism.</t>
<!--MW: Considering adding security with several aspects, source authentication,
confidentiality, need to trust middlebox. Or is security consideration
complete in this regards, but should be included below?-->
</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-ASM (ASM), Topo-SSM (SSM),
Topo-Trns-Translator (TT), Topo-Media-Translator (including Transport
Translator) (MT), Topo-Mesh with joint session (MJS), Topo-Mesh with
individual sessions (MIS), Topo-Mixer (Mix), Topo-Asymmetric (ASY),
Topo-Video-switch-MCU (VSM), and Topo-RTCP-terminating-MCU (RTM),
Selective Forwarding Middlebox (SFM). 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 ASM SSM TT MT MJS MIS Mix ASY VSM RTM SFM
---------------------------------------------------------------------
All to All media N Y (Y) Y Y Y (Y) (Y) (Y) (Y) (Y) (Y)
Interoperability N/A N N Y Y Y Y Y Y N Y Y
Per Domain Adaptation N/A N N N Y N Y Y Y N Y Y
Aggregation of media N N N N N N N Y (Y) Y Y N
Full Session View Y Y Y Y Y Y N Y Y (Y) N Y
Loop Detection Y Y Y Y Y Y N Y Y (Y) N 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, SFMs and Translators has impact on security and
the security functions used. The primary issue is that both Mixers, SFMs
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
End Points in the Communication Session. If encryption is employed, the media Translator, SFM 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, SFM 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, SFM 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 human 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, SFMs and Mixers, it
is possible that the Translator, SFM 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, SFMs and Translators. A Mixer normally needs to
represent only a single SSRCs 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>The multicast based (ASM and SSM), Relay and Mesh with common RTP
session are all topologies with multiple End Points that require shared
knowledge about the different crypto contexts for the End Points. These
multi-party topologies have special requirements on the key-management as
well as the security functions. Specifically source-authentication in
these environments has special requirements.</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="RFC7201">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 Mark Baugher, Bo Burman, Umesh Chandra, Alex
Eleftheriadis, Roni Even, Ladan Gharai, Geoff Hunt, Keith Lantz, and Colin Perkins
for their help in reviewing and improving 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.1112'?>
<?rfc include='reference.RFC.3022'?>
<?rfc include='reference.RFC.3569'?>
<?rfc include='reference.RFC.4607'?>
<?rfc include="reference.RFC.5104"?>
<?rfc include="reference.RFC.5760"?>
<?rfc include='reference.RFC.5766'?>
<!--
<?rfc include='reference.RFC.6285'?>
-->
<?rfc include='reference.RFC.6465'?>
<?rfc include='reference.RFC.7201'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-multi-stream-optimisation'?>
</references>
</back>
</rfc>
| PAFTECH AB 2003-2026 | 2026-04-23 20:42:37 |