RTP Topologies
draft-westerlund-avtcore-rtp-topologies-update-01
The information below is for an old version of the document.
| Document | Type |
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|---|---|---|---|
| Authors | Magnus Westerlund , Stephan Wenger | ||
| Last updated | 2012-10-15 | ||
| Replaced by | draft-ietf-avtcore-rtp-topologies-update, RFC 7667 | ||
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| Consensus boilerplate | Unknown | ||
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draft-westerlund-avtcore-rtp-topologies-update-01
Network Working Group M. Westerlund
Internet-Draft Ericsson
Obsoletes: 5117 (if approved) S. Wenger
Intended status: Informational Vidyo
Expires: April 18, 2013 October 15, 2012
RTP Topologies
draft-westerlund-avtcore-rtp-topologies-update-01
Abstract
This document discusses multi-endpoint topologies used in Real-time
Transport Protocol (RTP)-based environments. In particular,
centralized topologies commonly employed in the video conferencing
industry are mapped to the RTP terminology.
This document is updated with additional topologies and are intended
to replace RFC 5117.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on April 18, 2013.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
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include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.2. Indicating Requirement Levels . . . . . . . . . . . . . . 4
3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 4
3.2. Point to Multipoint Using Multicast . . . . . . . . . . . 5
3.2.1. Any Source Multicast (ASM) . . . . . . . . . . . . . . 5
3.2.2. Source Specific Multicast (SSM) . . . . . . . . . . . 6
3.2.3. SSM with Local Unicast Resources . . . . . . . . . . . 8
3.3. Point to Multipoint Using Mesh . . . . . . . . . . . . . . 8
3.4. Point to Multipoint Using the RFC 3550 Translator . . . . 9
3.4.1. Relay - Transport Translator . . . . . . . . . . . . . 10
3.4.2. Media Translator . . . . . . . . . . . . . . . . . . . 11
3.5. Point to Multipoint Using the RFC 3550 Mixer Model . . . . 13
3.5.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 15
3.5.2. Media Switching . . . . . . . . . . . . . . . . . . . 17
3.6. Source Projecting Middlebox . . . . . . . . . . . . . . . 19
3.7. Point to Multipoint Using Video Switching MCUs . . . . . . 21
3.8. Point to Multipoint Using RTCP-Terminating MCU . . . . . . 23
3.9. De-composite Endpoint . . . . . . . . . . . . . . . . . . 24
3.10. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 25
3.11. Combining Topologies . . . . . . . . . . . . . . . . . . . 26
4. Comparing Topologies . . . . . . . . . . . . . . . . . . . . . 26
4.1. Topology Properties . . . . . . . . . . . . . . . . . . . 27
4.1.1. All to All Media Transmission . . . . . . . . . . . . 27
4.1.2. Transport or Media Interoperability . . . . . . . . . 27
4.1.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . . 28
4.1.4. Aggregation of Media . . . . . . . . . . . . . . . . . 28
4.1.5. View of All Session Participants . . . . . . . . . . . 28
4.1.6. Loop Detection . . . . . . . . . . . . . . . . . . . . 28
4.2. Comparison of Topologies . . . . . . . . . . . . . . . . . 29
5. Security Considerations . . . . . . . . . . . . . . . . . . . 29
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 31
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 31
8.1. Normative References . . . . . . . . . . . . . . . . . . . 31
8.2. Informative References . . . . . . . . . . . . . . . . . . 32
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 32
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1. Introduction
When working on the Codec Control Messages [RFC5104], considerable
confusion was noticed in the community with respect to terms such as
Multipoint Control Unit (MCU), Mixer, and Translator, and their usage
in various topologies. This document tries to address this confusion
by providing a common information basis for future discussion and
specification work. It attempts to clarify and explain sections of
the Real-time Transport Protocol (RTP) spec [RFC3550] in an informal
way. It is not intended to update or change what is normatively
specified within RFC 3550.
When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was
developed the main emphasis lay in the efficient support of point to
point and small multipoint scenarios without centralized multipoint
control. However, in practice, many small multipoint conferences
operate utilizing devices known as Multipoint Control Units (MCUs).
MCUs may implement Mixer or Translator (in RTP [RFC3550] terminology)
functionality and signalling support. They may also contain
additional application functionality. This document focuses on the
media transport aspects of the MCU that can be realized using RTP, as
discussed below. Further considered are the properties of Mixers and
Translators, and how some types of deployed MCUs deviate from these
properties.
2. Definitions
2.1. Glossary
ASM: Any Source Multicast
AVPF: The Extended RTP Profile for RTCP-based Feedback
CSRC: Contributing Source
Link: The data transport to the next IP hop
MCU: Multipoint Control Unit
Path: The concatenation of multiple links, resulting in an end-to-
end data transfer.
PtM: Point to Multipoint
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PtP: Point to Point
SSM: Source-Specific Multicast
SSRC: Synchronization Source
2.2. Indicating Requirement Levels
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
The RFC 2119 language is used in this document to highlight those
important requirements and/or resulting solutions that are necessary
to address the issues raised in this document.
3. Topologies
This subsection defines several topologies that are relevant for
codec control but also RTP usage in other contexts. The first relate
to the RTP system model utilizing multicast and/or unicast, as
envisioned in RFC 3550. Two later topologies (MCU and RTCP
terminating), in contrast, describe the deployed system models as
used in many H.323 [H323] video conferences, where both the media
streams and the RTP Control Protocol (RTCP) control traffic terminate
at the MCU. In these two cases, the media sender does not receive
the (unmodified or Translator-modified) Receiver Reports from all
sources (which it needs to interpret based on Synchronization Source
(SSRC) values) and therefore has no full information about all the
endpoint's situation as reported in RTCP Receiver Reports (RRs).
More topologies can be constructed by combining any of the models;
see Section 3.11.
The topologies may be referenced in other documents by a shortcut
name, indicated by the prefix "Topo-".
For each of the RTP-defined topologies, we discuss how RTP, RTCP, and
the carried media are handled. With respect to RTCP, we also
introduce the handling of RTCP feedback messages as defined in
[RFC4585] and [RFC5104]. Any important differences between the two
will be illuminated in the discussion.
3.1. Point to Point
Shortcut name: Topo-Point-to-Point
The Point to Point (PtP) topology (Figure 1) consists of two
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endpoints, communicating using unicast. Both RTP and RTCP traffic
are conveyed endpoint-to-endpoint, using unicast traffic only (even
if, in exotic cases, this unicast traffic happens to be conveyed over
an IP-multicast address).
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 1: Point to Point
The main property of this topology is that A sends to B, and only B,
while B sends to A, and only A. This avoids all complexities of
handling multiple endpoints and combining the requirements from them.
Note that an endpoint can still use multiple RTP Synchronization
Sources (SSRCs) in an RTP session. The number of RTP sessions in use
between A and B can also be of any number.
RTCP feedback messages for the indicated SSRCs are communicated
directly between the endpoints. Therefore, this topology poses
minimal (if any) issues for any feedback messages.
3.2. Point to Multipoint Using Multicast
Multicast is a IP layer functionality that is available in some
networks. It comes in two main flavors, Any Source Multicast (ASM)
where any multicast group participant can send to the group address
and expect the packet to reach all group participants. The other
model is Source Specific Multicast (SSM) where only a particular IP
host is allowed to send to the multicast group. Both these models
are discussed below in their respective section.
3.2.1. Any Source Multicast (ASM)
Shortcut name: Topo-ASM (was Topo-Multicast)
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Figure 2: Point to Multipoint Using Multicast
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Point to Multipoint (PtM) is defined here as using a multicast
topology as a transmission model, in which traffic from any
participant reaches all the other participants, except for cases such
as:
o packet loss, or
o when a participant does not wish to receive the traffic for a
specific multicast group and therefore has not subscribed to the
IP-multicast group in question. This is for the cases where a
multi-media session is distributed using two or more multicast
groups.
In the above context, "traffic" encompasses both RTP and RTCP
traffic. The number of participants can vary between one and many,
as RTP and RTCP scale to very large multicast groups (the theoretical
limit of the number of participants in a single RTP session is
approximately two billion). The above can be realized using Any
Source Multicast (ASM).
For feedback usage it is relevant to make distinction of that subset
of multicast sessions wherein the number of participants in the
multicast group is so low that it allows the participants to use
early or immediate feedback, as defined in AVPF [RFC4585]. This
document refers to those groups as "small multicast groups". Some
applications may still want to use larger multicast groups where the
RTCP feedback possibilities are more limited.
RTCP feedback messages in multicast will, like media, reach everyone
(subject to packet losses and multicast group subscription).
Therefore, the feedback suppression mechanism discussed in [RFC4585]
is required. Each individual node needs to process every feedback
message it receives to determine if it is affected or if the feedback
message applies only to some other participant.
3.2.2. Source Specific Multicast (SSM)
In Any Source Multicast, any of the participants can send to all the
other participants, simply by sending a packet to the multicast
group. That is not possible in Source Specific Multicast [RFC4607]
where only a single source (Distribution Source) can send to the
multicast group, creating a topology that looks like the one below:
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+--------+ +-----+
|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.
Figure 3: Point to Multipoint using Source Specific Multicast
In the SSM topology (Figure 3) a number of RTP sources (1 to M) are
allowed to send media to the SSM group. These send media to the
distribution source which then forwards the media streams to the
multicast group. The media streams reach the Receivers (R(1) to
R(n)). The Receivers' RTCP cannot be sent to the multicast group.
To support RTCP, an RTP extension for SSM [RFC5760] was defined to
use unicast transmission to send RTCP from the receivers to one or
more Feedback Targets (FT).
The RTP extension for SSM deals with how feedback both general
reception information and specific feedback events are generally
handled. The general problems of multicast that everyone will
receive what the distribution source sends needs to be accounted for.
The result of this is some common behaviours for RTP multicast:
1. Multicast applications often use a group of RTP sessions, not
one. Each endpoint will need to be a member of a number of RTP
sessions in order to perform well.
2. Within each RTP session, the number of media sinks is likely to
be much larger than the number of RTP sources.
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3. Multicast applications need signalling functions to identify the
relationships between RTP sessions.
4. Multicast applications need signalling functions to identify the
relationships between SSRCs in different RTP sessions.
All multicast configurations share a signalling requirement; all of
the participants will need to have the same RTP and payload type
configuration. Otherwise, A could for example be using payload type
97 as the video codec H.264 while B thinks it is MPEG-2.
Security solutions for this type of group communications are also
challenging. First of all the key-management and the security
protocol must support group communication. Source authentication
becomes more difficult and requires special solutions. For more
discussion on this please review Options for Securing RTP Sessions
[I-D.ietf-avtcore-rtp-security-options].
3.2.3. SSM with Local Unicast Resources
[RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions"
results in additional extensions to SSM Topology. Should be
described.
3.3. Point to Multipoint Using Mesh
Shortcut name: Topo-Mesh
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 4: Point to Multi-Point using Mesh
Based on the RTP session definition, it is clearly possible to have a
joint RTP session over multiple unicast transport flows like the
above three endpoint joint session. In this case, A needs to send
its' media streams and RTCP packets to both B and C over their
respective transport flows. As long as all participants do the same,
everyone will have a joint view of the RTP session.
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This doesn't create any additional requirements beyond the need to
have multiple transport flows associated with a single RTP session.
Note that an endpoint may use a single local port to receive all
these transport flows, or it might have separate local reception
ports for each of the endpoints.
There exists an alternative structure for establishing the above
topology which uses independent RTP sessions between each pair of
peers, i.e. three different RTP sessions. Unless independently
adapted the same RTP media stream could be sent in both of the RTP
sessions an endpoint has. The difference exists in the behaviours
around RTCP, for example common RTCP bandwidth for one joint session,
rather than three independent pools, and the awareness based on RTCP
reports between the peers of how that third leg is doing.
3.4. Point to Multipoint Using the RFC 3550 Translator
Shortcut name: Topo-Translator
Two main categories of Translators can be distinguished; Transport
Translators and Media translators. Both Translator types share
common attributes that separate them from Mixers. For each media
stream that the Translator receives, it generates an individual
stream in the other domain. A Translator always keeps the SSRC for a
stream across the translation, where a Mixer can select a media
stream, or send them out mixed, always under its own SSRC, using the
CSRC field to indicate the source(s) of the content.
As specified in Section 7.1 of [RFC3550], the SSRC space is common
for all participants in the session, independent of on which side
they are of the Translator. Therefore, it is the responsibility of
the participants to run SSRC collision detection, and the SSRC is a
field the Translator should not change.
A Translator commonly does not use an SSRC of its own, and is not
visible as an active participant in the session. One reason is when
a Translator acts as a quality monitor that sends RTCP reports and
therefore is required to have an SSRC. Another example is the case
when a Translator is prepared to use RTCP feedback messages. This
may, for example, occur when it suffers packet loss of important
video packets and wants to trigger repair by the media sender, by
sending feedback messages. This can be done using the SSRC of the
target for the translator, but this requires translation of the
targets RTCP reports to make them consistent, so it is likely simpler
to expose an additional SSRC in the session.
In general, a Translator implementation should consider which RTCP
feedback messages or codec-control messages it needs to understand in
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relation to the functionality of the Translator itself. This is
completely in line with the requirement to also translate RTCP
messages between the domains.
The RTCP translation process can be trivial, for example, when
Transport Translators just need to adjust IP addresses and transport
protocol ports, or they can be quite complex as in the case of media
Translators. See Section 7.2 of [RFC3550].
3.4.1. Relay - Transport Translator
Transport Translators (Topo-Trn-Translator) do not modify the media
stream itself, but are concerned with transport parameters.
Transport parameters, in the sense of this section, comprise the
transport addresses (to bridge different domains) and the media
packetization to allow other transport protocols to be interconnected
to a session (in gateways). Of the transport Translators, this memo
is primarily interested in those that use RTP on both sides, and this
is assumed henceforth. Translators that bridge between different
protocol worlds need to be concerned about the mapping of the SSRC/
CSRC (Contributing Source) concept to the non-RTP protocol. When
designing a Translator to a non-RTP-based media transport, one
crucial factor lies in how to handle different sources and their
identities. This problem space is not discussed henceforth.
+-----+
+---+ / \ +------------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Translator |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+
+-----+
Figure 5: Point to Multipoint Using Multicast
Figure 5 depicts an example of a Transport Translator performing at
least IP address translation. It allows the (non-multicast-capable)
participants B and D to take part in an any source multicast session
by having the Translator forward their unicast traffic to the
multicast addresses in use, and vice versa. It must also forward B's
traffic to D, and vice versa, to provide each of B and D with a
complete view of the session.
Also a point to point communication can end up in a situation when
the peer it is communicating with needs basic transport translators
functions. This include NAT traversal by pinning the media path to a
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public address domain relay, network topologies where the media flow
is required to pass a particular point by employing relaying or
preserving privacy by hiding each peers transport addresses to the
other party.
+---+ +---+ +---+
| A |<------>| T |<------->| B |
+---+ +---+ +---+
Point to Point with Translator
This type of very basic relay service should in most case need to
have no RTP functionality. Thus one can believe that they do not
need to included in this document. However, due to that the network
level addressing and the RTP identifier view of the RTP session and
who the peer is doesn't match as in the PtP unicast scenario depicted
above this topology can raise additional requirements.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 6: RTP Translator (Relay) with Only Unicast Paths
Another Translator scenario is depicted in Figure 6. Herein, the
Translator connects multiple users of a conference through unicast.
This can be implemented using a very simple transport Translator,
which in this document is called a relay. The relay forwards all
traffic it receives, both RTP and RTCP, to all other participants.
In doing so, a multicast network is emulated without relying on a
multicast-capable network infrastructure.
For RTCP feedback this results in a similar considerations that arise
for the ASM RTP topology. It also puts some special signalling
requirements where common configuration of RTP payload types for
example are required.
3.4.2. Media Translator
Media Translators (Topo-Media-Translator), in contrast, modify the
media stream itself. This process is commonly known as transcoding.
The modification of the media stream can be as small as removing
parts of the stream, and it can go all the way to a full transcoding
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(down to the sample level or equivalent) utilizing a different media
codec. Media Translators are commonly used to connect entities
without a common interoperability point.
Stand-alone Media Translators are rare. Most commonly, a combination
of Transport and Media Translators are used to translate both the
media stream and the transport aspects of a stream between two
transport domains (or clouds).
If B in Figure 5 were behind a limited network path, the Translator
may perform media transcoding to allow the traffic received from the
other participants to reach B without overloading the path.
When, in the example depicted in Figure 5, the Translator acts only
as a Transport Translator, then the RTCP traffic can simply be
forwarded, similar to the media traffic. However, when media
translation occurs, the Translator's task becomes substantially more
complex, even with respect to the RTCP traffic. In this case, the
Translator needs to rewrite B's RTCP Receiver Report before
forwarding them to D and the multicast network. The rewriting is
needed as the stream received by B is not the same stream as the
other participants receive. For example, the number of packets
transmitted to B may be lower than what D receives, due to the
different media format and data rate. Therefore, if the Receiver
Reports were forwarded without changes, the extended highest sequence
number would indicate that B were substantially behind in reception,
while it most likely it would not be. Therefore, the Translator must
translate that number to a corresponding sequence number for the
stream the Translator received. Similar arguments can be made for
most other fields in the RTCP Receiver Reports.
A media Translator may in some cases act on behalf of the "real"
source and respond to RTCP feedback messages. This may occur, for
example, when a receiver requests a bandwidth reduction, and the
media Translator has not detected any congestion or other reasons for
bandwidth reduction between the media source and itself. In that
case, it is sensible that the media Translator reacts to the codec
control messages itself, for example, by transcoding to a lower media
rate. If it were not reacting, the media quality in the media
sender's domain may suffer, as a result of the media sender adjusting
its media rate (and quality) according to the needs of the slow past-
Translator endpoint, at the expense of the rate and quality of all
other session participants.
A variant of translator behaviour worth pointing out is the one
depicted in Figure 7 of an endpoint A sends a media flow to B. On the
path there is a device T that on A's behalf does something with the
media streams, for example adds an RTP session with FEC information
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for A's media streams. T will in this case need to bind the new FEC
streams to A's media stream, for example by using the same CNAME as
A.
+------+ +------+ +------+
| | | | | |
| A |------->| T |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+
Figure 7: When De-composition is a Translator
This type of functionality where T does something with the media
stream on behalf of A is clearly covered under the media translator
definition.
3.5. Point to Multipoint Using the RFC 3550 Mixer Model
Shortcut name: Topo-Mixer
A Mixer is a middlebox that aggregates multiple RTP streams, which
are part of a session, by manipulation of the media data and
generating a new RTP stream. One common application for a Mixer is
to allow a participant to receive a session with a reduced amount of
resources.
+-----+
+---+ / \ +-----------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Mixer |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +-----------+ +---+
+-----+
Figure 8: Point to Multipoint Using the RFC 3550 Mixer Model
A Mixer can be viewed as a device terminating the media streams
received from other session participants. Using the media data from
the received media streams, a Mixer generates a media stream that is
sent to the session participant.
The content that the Mixer provides is the mixed aggregate of what
the Mixer receives over the PtP or PtM paths, which are part of the
same conference session.
The Mixer is the content source, as it mixes the content (often in
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the uncompressed domain) and then encodes it for transmission to a
participant. The CSRC Count (CC) and CSRC fields in the RTP header
are used to indicate the contributors of to the newly generated
stream. The SSRCs of the to-be-mixed streams on the Mixer input
appear as the CSRCs at the Mixer output. That output stream uses a
unique SSRC that identifies the Mixer's stream. The CSRC should be
forwarded between the two domains to allow for loop detection and
identification of sources that are part of the global session. Note
that Section 7.1 of RFC 3550 requires the SSRC space to be shared
between domains for these reasons.
The Mixer is responsible for generating RTCP packets in accordance
with its role. It is a receiver and should therefore send reception
reports for the media streams it receives. In its role as a media
sender, it should also generate Sender Reports for those media
streams sent. As specified in Section 7.3 of RFC 3550, a Mixer must
not forward RTCP unaltered between the two domains.
The Mixer depicted in Figure 8 is involved in three domains that need
to be separated: the any source multicast network, participant B, and
participant D. The Mixer produces different mixed streams to B and D,
as the one to B may contain content received from D, and vice versa.
However, the Mixer may only need one SSRC per media type in each
domain that is the receiving entity and transmitter of mixed content.
In the multicast domain, a Mixer still needs to provide a mixed view
of the other domains. This makes the Mixer simpler to implement and
avoids any issues with advanced RTCP handling or loop detection,
which would be problematic if the Mixer were providing non-symmetric
behavior. Please see Section 3.10 for more discussion on this topic.
However, the mixing operation in each domain could potentially be
different.
A Mixer is responsible for receiving RTCP feedback messages and
handling them appropriately. The definition of "appropriate" depends
on the message itself and the context. In some cases, the reception
of a codec-control message may result in the generation and
transmission of RTCP feedback messages by the Mixer to the
participants in the other domain. In other cases, a message is
handled by the Mixer itself and therefore not forwarded to any other
domain.
When replacing the multicast network in Figure 8 (to the left of the
Mixer) with individual unicast paths as depicted in Figure 9, the
Mixer model is very similar to the one discussed in Section 3.8
below. Please see the discussion in Section 3.8 about the
differences between these two models.
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+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 9: RTP Mixer with Only Unicast Paths
Lets now discuss in more detail different mixing operations that a
mixer can perform and how that can affect the RTP and RTCP.
3.5.1. Media Mixing
The media mixing mixer is likely the one that most thinks of when
they hear the term mixer. Its basic pattern of operation is that it
will receive the different participants RTP media stream. Select
which that are to be included in a media domain mix of the incoming
RTP media streams. Then create a single outgoing stream from this
mix.
The most commonly deployed media mixer is probably the audio mixer,
used in voice conferencing, where the output consists of some mixture
of all the input streams; this needs minimal signalling to be
successful. Audio mixing is straight forward and commonly possible
to do for a number of participants. Lets assume that you want to mix
N number of streams from different participants. Then the mixer need
to perform N decodings. Then it needs to produce N or N+1 mixes, the
reasons that different mixes are needed are so that each contributing
source get a mix which don't contain themselves, as this would result
in an echo. When N is lower than the number of all participants one
may produce a Mix of all N streams for the group that are currently
not included in the mix, thus N+1 mixes. These audio streams are
then encoded again, RTP packetised and sent out.
Video can't really be "mixed" and produce something particular useful
for the users, however creating an composition out of the contributed
video streams can be done. In fact it can be done in a number of
ways, tiling the different streams creating a chessboard, selecting
someone as more important and showing them large and a number of
other sources as smaller overlays is another. Also here one commonly
need to produce a number of different compositions so that the
contributing part doesn't need to see themselves. Then the mixer re-
encodes the created video stream, RTP packetise it and send it out
The problem with media mixing is that it both consume large amount of
media processing and encoding resources. The second is the quality
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degradation created by decoding and re-encoding the RTP media stream.
Its advantage is that it is quite simplistic for the clients to
handle as they don't need to handle local mixing and composition.
+-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 | |
+-----------+ |-----------+ +---+ +-----+ |
+----------------------------+
Figure 10: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be very straight forward as
can be seen in Figure 10. The mixer present one SSRC towards the
peer client, e.g. MA1 to Peer A, which is the media mix of the other
participants. As each peer receives a different version produced by
the mixer there are no actual relation between the different RTP
sessions in the actual media or the transport level information.
There is however one connection between RTP1-RTP3 in this figure. It
has to do with the SSRC space and the identity information. When A
receives the MA1 stream which is a combination of BA1 and CA1
streams, the mixer may include CSRC information in the MA1 stream to
identify the contributing source BA1 and CA1.
The CSRC has in its turn utility in RTP extensions, like the in Mixer
to Client audio levels RTP header extension [RFC6465]. If the SSRC
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from endpoint to mixer leg are used as CSRC in another RTP session
then RTP1, RTP2 and RTP3 becomes one joint session as they have a
common SSRC space. At this stage the mixer also need to consider
which RTCP information it need to expose in the different legs. For
the above situation commonly nothing more than the Source Description
(SDES) information and RTCP BYE for CSRC need to be exposed. The
main goal would be to enable the correct binding against the
application logic and other information sources. This also enables
loop detection in the RTP session.
3.5.2. Media Switching
An RTP Mixer based on media switching avoids the media decoding and
encoding cycle in the mixer, but not the decryption and re-encryption
cycle as it rewrites RTP headers. This both reduces the amount of
computational resources needed in the mixer and increases the media
quality per transmitted bit. This is achieve by letting the mixer
have a number of SSRCs that represents conceptual or functional
streams the mixer produces. These streams are created by selecting
media from one of the by the mixer received RTP media streams and
forward the media using the mixers own SSRCs. The mixer can then
switch between available sources if that is required by the concept
for the source, like currently active speaker.
To achieve a coherent RTP media stream from the mixer's SSRC the
mixer is forced to rewrite the incoming RTP packet's header. First
the SSRC field must be set to the value of the Mixer's SSRC.
Secondly, the sequence number must be the next in the sequence of
outgoing packets it sent. Thirdly the RTP timestamp value needs to
be adjusted using an offset that changes each time one switch media
source. Finally depending on the negotiation the RTP payload type
value representing this particular RTP payload configuration may have
to be changed if the different endpoint mixer legs have not arrived
on the same numbering for a given configuration. This also requires
that the different end-points do support a common set of codecs,
otherwise media transcoding for codec compatibility is still
required.
Lets consider the operation of media switching mixer that supports a
video conference with six participants (A-F) where the two latest
speakers in the conference are shown to each participants. Thus the
mixer has two SSRCs sending video to each peer.
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+-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----| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +----------------------------+
Figure 11: Media Switching RTP Mixer
The Media Switching RTP mixer can similar to the Media Mixing one
reduce the bit-rate needed towards the different peers by selecting
and switching in a sub-set of RTP media streams out of the ones it
receives from the conference participants.
To ensure that a media receiver can correctly decode the RTP media
stream after a switch, it becomes necessary to ensure for state
saving codecs that they start from default state at the point of
switching. Thus one common tool for video is to request that the
encoding creates an intra picture, something that isn't dependent on
earlier state. This can be done using Full Intra Request [RFC5104]
RTCP codec control message.
Also in this type of mixer one could consider to terminate the RTP
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sessions fully between the different end-point and mixer legs. The
same arguments and considerations as discussed in Section 3.8 applies
here.
3.6. Source Projecting Middlebox
Another method for handling media in the RTP mixer is to project all
potential RTP sources (SSRCs) into a per end-point independent RTP
session. The mixer can then select which of the potential sources
that are currently actively transmitting media, despite that the
mixer in another RTP session receives media from that end-point.
This is similar to the media switching Mixer but have some important
differences in RTP details.
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+-A---------+ +-MIXER---------------------+
| +-RTP1----| |-RTP1------+ +-----+ |
| | +-Video-| |-Video---+ | | | |
| | | AV1|------------>|---------+-+------>| | |
| | | |<------------|BV1 <----+-+-------| S | |
| | | |<------------|CV1 <----+-+-------| W | |
| | | |<------------|DV1 <----+-+-------| I | |
| | | |<------------|EV1 <----+-+-------| T | |
| | | |<------------|FV1 <----+-+-------| C | |
| | +-------| |---------+ | | H | |
| +---------| |-----------+ | | |
+-----------+ | | M | |
| | A | |
+-B---------+ | | T | |
| +-RTP2----| |-RTP2------+ | R | |
| | +-Video-| |-Video---+ | | I | |
| | | BV1|------------>|---------+-+------>| X | |
| | | |<------------|AV1 <----+-+-------| | |
| | | |<------------|CV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | |
| | | |<------------|FV1 <----+-+-------| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ | | |
+-----------+ | | | |
: : : :
: : : :
+-F---------+ | | | |
| +-RTP6----| |-RTP6------+ | | |
| | +-Video-| |-Video---+ | | | |
| | | CV1|------------>|---------+-+------>| | |
| | | |<------------|AV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | |
| | | |<------------|EV1 <----+-+-------| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +---------------------------+
Figure 12: Media Projecting Mixer
So in this six participant conference depicted above in (Figure 12)
one can see that end-point A will in this case be aware of 5 incoming
SSRCs, BV1-FV1. If this mixer intend to have the same behaviour as
in Section 3.5.2 where the mixer provides the end-points with the two
latest speaking end-points, then only two out of these five SSRCs
will concurrently transmit media to A. As the mixer selects which
source in the different RTP sessions that transmit media to the end-
points each RTP media stream will require some rewriting when being
projected from one session into another. The main thing is that the
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sequence number will need to be consecutively incremented based on
the packet actually being transmitted in each RTP session. Thus the
RTP sequence number offset will change each time a source is turned
on in a RTP session.
As the RTP sessions are independent the SSRC numbers used can be
handled independently also thus working around any SSRC collisions by
having remapping tables between the RTP sessions. This will result
that each endpoint may have a different view of the application usage
of a particular SSRC. Thus the application must not use SSRC as
references to RTP media streams when communicating with other peers
directly.
The mixer will also be responsible to act on any RTCP codec control
requests coming from an end-point and decide if it can act on it
locally or needs to translate the request into the RTP session that
contains the media source. Both end-points and the mixer will need
to implement conference related codec control functionalities to
provide a good experience. Full Intra Request to request from the
media source to provide switching points between the sources,
Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
to aggregate congestion control response towards the media source and
have it adjust its bit-rate in case the limitation is not in the
source to mixer link.
This version of the mixer also puts different requirements on the
end-point when it comes to decoder instances and handling of the RTP
media streams providing media. As each projected SSRC can at any
time provide media the end-point either needs to handle having thus
many allocated decoder instances or have efficient switching of
decoder contexts in a more limited set of actual decoder instances to
cope with the switches. The WebRTC application also gets more
responsibility to update how the media provides is to be presented to
the user.
Note, this could potentially be seen as a media translator which
include an on/off logic as part of its media translation. The main
difference would be a common global SSRC space in the case of the
Media Translator and the mapped one used in the above.
3.7. Point to Multipoint Using Video Switching MCUs
Shortcut name: Topo-Video-switch-MCU
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+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
Figure 13: Point to Multipoint Using a Video Switching MCU
This PtM topology is still deployed today, although the RTCP-
terminating MCUs, as discussed in the next section, are perhaps more
common. This topology, as well as the following one, reflect today's
lack of wide availability of IP multicast technologies, as well as
the simplicity of content switching when compared to content mixing.
The technology is commonly implemented in what is known as "Video
Switching MCUs".
A video switching MCU forwards to a participant a single media
stream, selected from the available streams. The criteria for
selection are often based on voice activity in the audio-visual
conference, but other conference management mechanisms (like
presentation mode or explicit floor control) are known to exist as
well.
The video switching MCU may also perform media translation to modify
the content in bit-rate, encoding, or resolution. However, it still
may indicate the original sender of the content through the SSRC. In
this case, the values of the CC and CSRC fields are retained.
If not terminating RTP, the RTCP Sender Reports are forwarded for the
currently selected sender. All RTCP Receiver Reports are freely
forwarded between the participants. In addition, the MCU may also
originate RTCP control traffic in order to control the session and/or
report on status from its viewpoint.
The video switching MCU has most of the attributes of a Translator.
However, its stream selection is a mixing behavior. This behavior
has some RTP and RTCP issues associated with it. The suppression of
all but one media stream results in most participants seeing only a
subset of the sent media streams at any given time, often a single
stream per conference. Therefore, RTCP Receiver Reports only report
on these streams. Consequently, the media senders that are not
currently forwarded receive a view of the session that indicates
their media streams disappear somewhere en route. This makes the use
of RTCP for congestion control, or any type of quality reporting,
very problematic.
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To avoid the aforementioned issues, the MCU needs to implement two
features. First, it needs to act as a Mixer (see Section 3.5) and
forward the selected media stream under its own SSRC and with the
appropriate CSRC values. Second, the MCU needs to modify the RTCP
RRs it forwards between the domains. As a result, it is RECOMMENDED
that one implement a centralized video switching conference using a
Mixer according to RFC 3550, instead of the shortcut implementation
described here.
3.8. Point to Multipoint Using RTCP-Terminating MCU
Shortcut name: Topo-RTCP-terminating-MCU
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 14: Point to Multipoint Using Content Modifying MCUs
In this PtM scenario, each participant runs an RTP point-to-point
session between itself and the MCU. This is a very commonly deployed
topology in multipoint video conferencing. The content that the MCU
provides to each participant is either:
a. a selection of the content received from the other participants,
or
b. the mixed aggregate of what the MCU receives from the other PtP
paths, which are part of the same conference session.
In case a), the MCU may modify the content in bit-rate, encoding, or
resolution. No explicit RTP mechanism is used to establish the
relationship between the original media sender and the version the
MCU sends. In other words, the outgoing sessions typically use a
different SSRC, and may well use a different payload type (PT), even
if this different PT happens to be mapped to the same media type.
This is a result of the individually negotiated session for each
participant.
In case b), the MCU is the content source as it mixes the content and
then encodes it for transmission to a participant. According to RTP
[RFC3550], the SSRC of the contributors are to be signalled using the
CSRC/CC mechanism. In practice, today, most deployed MCUs do not
implement this feature. Instead, the identification of the
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participants whose content is included in the Mixer's output is not
indicated through any explicit RTP mechanism. That is, most deployed
MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby
indicating no available CSRC information, even if they could identify
the content sources as suggested in RTP.
The main feature that sets this topology apart from what RFC 3550
describes is the breaking of the common RTP session across the
centralized device, such as the MCU. This results in the loss of
explicit RTP-level indication of all participants. If one were using
the mechanisms available in RTP and RTCP to signal this explicitly,
the topology would follow the approach of an RTP Mixer. The lack of
explicit indication has at least the following potential problems:
1. Loop detection cannot be performed on the RTP level. When
carelessly connecting two misconfigured MCUs, a loop could be
generated.
2. There is no information about active media senders available in
the RTP packet. As this information is missing, receivers cannot
use it. It also deprives the client of information related to
currently active senders in a machine-usable way, thus preventing
clients from indicating currently active speakers in user
interfaces, etc.
Note that deployed MCUs (and endpoints) rely on signalling layer
mechanisms for the identification of the contributing sources, for
example, a SIP conferencing package [RFC4575]. This alleviates, to
some extent, the aforementioned issues resulting from ignoring RTP's
CSRC mechanism.
As a result of the shortcomings of this topology, it is RECOMMENDED
to instead implement the Mixer concept as specified by RFC 3550.
3.9. De-composite Endpoint
The implementation of an application may desire to send a subset of
the application's data to each of multiple devices, each with their
own network address. A very basic use case for this would be to
separate audio and video processing for a particular endpoint, like a
conference room, into one device handling the audio and another
handling the video, being interconnected by some control functions
allowing them to behave as a single endpoint in all aspects except
for transport Figure 15.
Which decomposition that is possible is highly dependent on the RTP
session usage. It is not really feasible to decomposed one logical
end-point into two different transport node in one RTP session. From
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a third party monitor of such an attempt the two entities would look
like two different end-points with a CNAME collision. This put a
requirement on that the only type of de-composited endpoint that RTP
really supports is one where the different parts have separate RTP
sessions to send and/or receive media streams intended for them.
+---------------------+
| Endpoint A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+-RTP---\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+-RTP-------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+-SIP---/
| +------------+ |
+---------------------+
Figure 15: De-composite End-Point
In the above usage, let us assume that the RTP sessions are different
for audio and video. The audio and video parts will use a common
CNAME and also have a common clock to ensure that synchronisation and
clock drift handling works despite the decomposition. Also the RTCP
handling works correctly as long as only one part of the de-composite
is part of each RTP session. That way any differences in the path
between A's audio entity and B and A's video and B are related to
different SSRCs in different RTP sessions.
The requirements that can derived from the above usage is that the
transport flows for each RTP session might be under common control
but still go to what looks like different endpoints based on
addresses and ports. This geometry cannot be accomplished using one
RTP session, so in this case, multiple RTP sessions are needed.
3.10. Non-Symmetric Mixer/Translators
Shortcut name: Topo-Asymmetric
It is theoretically possible to construct an MCU that is a Mixer in
one direction and a Translator in another. The main reason to
consider this would be to allow topologies similar to Figure 8, where
the Mixer does not need to mix in the direction from B or D towards
the multicast domains with A and C. Instead, the media streams from B
and D are forwarded without changes. Avoiding this mixing would save
media processing resources that perform the mixing in cases where it
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isn't needed. However, there would still be a need to mix B's stream
towards D. Only in the direction B -> multicast domain or D ->
multicast domain would it be possible to work as a Translator. In
all other directions, it would function as a Mixer.
The Mixer/Translator would still need to process and change the RTCP
before forwarding it in the directions of B or D to the multicast
domain. One issue is that A and C do not know about the mixed-media
stream the Mixer sends to either B or D. Thus, any reports related to
these streams must be removed. Also, receiver reports related to A
and C's media stream would be missing. To avoid A and C thinking
that B and D aren't receiving A and C at all, the Mixer needs to
insert its Receiver Reports for the streams from A and C into B and
D's Sender Reports. In the opposite direction, the Receiver Reports
from A and C about B's and D's stream also need to be aggregated into
the Mixer's Receiver Reports sent to B and D. Since B and D only have
the Mixer as source for the stream, all RTCP from A and C must be
suppressed by the Mixer.
This topology is so problematic and it is so easy to get the RTCP
processing wrong, that it is NOT RECOMMENDED to implement this
topology.
3.11. Combining Topologies
Topologies can be combined and linked to each other using Mixers or
Translators. However, care must be taken in handling the SSRC/CSRC
space. A Mixer will not forward RTCP from sources in other domains,
but will instead generate its own RTCP packets for each domain it
mixes into, including the necessary Source Description (SDES)
information for both the CSRCs and the SSRCs. Thus, in a mixed
domain, the only SSRCs seen will be the ones present in the domain,
while there can be CSRCs from all the domains connected together with
a combination of Mixers and Translators. The combined SSRC and CSRC
space is common over any Translator or Mixer. This is important to
facilitate loop detection, something that is likely to be even more
important in combined topologies due to the mixed behavior between
the domains. Any hybrid, like the Topo-Video-switch-MCU or Topo-
Asymmetric, requires considerable thought on how RTCP is dealt with.
4. Comparing Topologies
The topologies discussed in Section 3 have different properties.
This section first lists these properties and then maps the different
topologies to them. Please note that even if a certain property is
supported within a particular topology concept, the necessary
functionality may, in many cases, be optional to implement.
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Note: This section has not yet been updated with the new additions of
topologies.
4.1. Topology Properties
4.1.1. All to All Media Transmission
Multicast, at least Any Source Multicast (ASM), provides the
functionality that everyone may send to, or receive from, everyone
else within the session. MCUs, Mixers, and Translators may all
provide that functionality at least on some basic level. However,
there are some differences in which type of reachability they
provide.
The transport Translator function called "relay", in Section 3.4, is
the one that provides the emulation of ASM that is closest to true
IP-multicast-based, all to all transmission. Media Translators,
Mixers, and the MCU variants do not provide a fully meshed forwarding
on the transport level; instead, they only allow limited forwarding
of content from the other session participants.
The "all to all media transmission" requires that any media
transmitting entity considers the path to the least capable receiver.
Otherwise, the media transmissions may overload that path.
Therefore, a media sender needs to monitor the path from itself to
any of the participants, to detect the currently least capable
receiver, and adapt its sending rate accordingly. As multiple
participants may send simultaneously, the available resources may
vary. RTCP's Receiver Reports help performing this monitoring, at
least on a medium time scale.
The transmission of RTCP automatically adapts to any changes in the
number of participants due to the transmission algorithm, defined in
the RTP specification [RFC3550], and the extensions in AVPF [RFC4585]
(when applicable). That way, the resources utilized for RTCP stay
within the bounds configured for the session.
4.1.2. Transport or Media Interoperability
Translators, Mixers, and RTCP-terminating MCU all allow changing the
media encoding or the transport to other properties of the other
domain, thereby providing extended interoperability in cases where
the participants lack a common set of media codecs and/or transport
protocols.
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4.1.3. Per Domain Bit-Rate Adaptation
Participants are most likely to be connected to each other with a
heterogeneous set of paths. This makes congestion control in a Point
to Multipoint set problematic. For the ASM and "relay" scenario,
each individual sender has to adapt to the receiver with the least
capable path. This is no longer necessary when Media Translators,
Mixers, or MCUs are involved, as each participant only needs to adapt
to the slowest path within its own domain. The Translator, Mixer, or
MCU topologies all require their respective outgoing streams to
adjust the bit-rate, packet-rate, etc., to adapt to the least capable
path in each of the other domains. That way one can avoid lowering
the quality to the least-capable participant in all the domains at
the cost (complexity, delay, equipment) of the Mixer or Translator.
4.1.4. Aggregation of Media
In the all to all media property mentioned above and provided by ASM,
all simultaneous media transmissions share the available bit-rate.
For participants with limited reception capabilities, this may result
in a situation where even a minimal acceptable media quality cannot
be accomplished. This is the result of multiple media streams
needing to share the available resources. The solution to this
problem is to provide for a Mixer or MCU to aggregate the multiple
streams into a single one. This aggregation can be performed
according to different methods. Mixing or selection are two common
methods.
4.1.5. View of All Session Participants
The RTP protocol includes functionality to identify the session
participants through the use of the SSRC and CSRC fields. In
addition, it is capable of carrying some further identity information
about these participants using the RTCP Source Descriptors (SDES).
To maintain this functionality, it is necessary that RTCP is handled
correctly in domain bridging function. This is specified for
Translators and Mixers. The MCU described in Section 3.7 does not
entirely fulfill this. The one described in Section 3.8 does not
support this at all.
4.1.6. Loop Detection
In complex topologies with multiple interconnected domains, it is
possible to form media loops. RTP and RTCP support detecting such
loops, as long as the SSRC and CSRC identities are correctly set in
forwarded packets. It is likely that loop detection works for the
MCU, described in Section 3.7, at least as long as it forwards the
RTCP between the participants. However, the MCU in Section 3.8 will
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definitely break the loop detection mechanism.
4.2. Comparison of Topologies
The table below attempts to summarize the properties of the different
topologies. The legend to the topology abbreviations are: Topo-
Point-to-Point (PtP), Topo-Multicast (Multic), Topo-Trns-Translator
(TTrn), Topo-Media-Translator (including Transport Translator)
(MTrn), Topo-Mixer (Mixer), Topo-Asymmetric (ASY), Topo-Video-switch-
MCU (MCUs), and Topo-RTCP-terminating-MCU (MCUt). In the table
below, Y indicates Yes or full support, N indicates No support, (Y)
indicates partial support, and N/A indicates not applicable.
Property PtP Multic TTrn MTrn Mixer ASY MCUs MCUt
------------------------------------------------------------------
All to All media N Y Y Y (Y) (Y) (Y) (Y)
Interoperability N/A N Y Y Y Y N Y
Per Domain Adaptation N/A N N Y Y Y N Y
Aggregation of media N N N N Y (Y) Y Y
Full Session View Y Y Y Y Y Y (Y) N
Loop Detection Y Y Y Y Y Y (Y) N
Please note that the Media Translator also includes the transport
Translator functionality.
5. Security Considerations
The use of Mixers and Translators has impact on security and the
security functions used. The primary issue is that both Mixers and
Translators modify packets, thus preventing the use of integrity and
source authentication, unless they are trusted devices that take part
in the security context, e.g., the device can send Secure Realtime
Transport Protocol (SRTP) and Secure Realtime Transport Control
Protocol (SRTCP) [RFC3711] packets to session endpoints. If
encryption is employed, the media Translator and Mixer need to be
able to decrypt the media to perform its function. A transport
Translator may be used without access to the encrypted payload in
cases where it translates parts that are not included in the
encryption and integrity protection, for example, IP address and UDP
port numbers in a media stream using SRTP [RFC3711]. However, in
general, the Translator or Mixer needs to be part of the signalling
context and get the necessary security associations (e.g., SRTP
crypto contexts) established with its RTP session participants.
Including the Mixer and Translator in the security context allows the
entity, if subverted or misbehaving, to perform a number of very
serious attacks as it has full access. It can perform all the
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attacks possible (see RFC 3550 and any applicable profiles) as if the
media session were not protected at all, while giving the impression
to the session participants that they are protected.
Transport Translators have no interactions with cryptography that
works above the transport layer, such as SRTP, since that sort of
Translator leaves the RTP header and payload unaltered. Media
Translators, on the other hand, have strong interactions with
cryptography, since they alter the RTP payload. A media Translator
in a session that uses cryptographic protection needs to perform
cryptographic processing to both inbound and outbound packets.
A media Translator may need to use different cryptographic keys for
the inbound and outbound processing. For SRTP, different keys are
required, because an RFC 3550 media Translator leaves the SSRC
unchanged during its packet processing, and SRTP key sharing is only
allowed when distinct SSRCs can be used to protect distinct packet
streams.
When the media Translator uses different keys to process inbound and
outbound packets, each session participant needs to be provided with
the appropriate key, depending on whether they are listening to the
Translator or the original source. (Note that there is an
architectural difference between RTP media translation, in which
participants can rely on the RTP Payload Type field of a packet to
determine appropriate processing, and cryptographically protected
media translation, in which participants must use information that is
not carried in the packet.)
When using security mechanisms with Translators and Mixers, it is
possible that the Translator or Mixer could create different security
associations for the different domains they are working in. Doing so
has some implications:
First, it might weaken security if the Mixer/Translator accepts a
weaker algorithm or key in one domain than in another. Therefore,
care should be taken that appropriately strong security parameters
are negotiated in all domains. In many cases, "appropriate"
translates to "similar" strength. If a key management system does
allow the negotiation of security parameters resulting in a different
strength of the security, then this system SHOULD notify the
participants in the other domains about this.
Second, the number of crypto contexts (keys and security related
state) needed (for example, in SRTP [RFC3711]) may vary between
Mixers and Translators. A Mixer normally needs to represent only a
single SSRC per domain and therefore needs to create only one
security association (SRTP crypto context) per domain. In contrast,
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a Translator needs one security association per participant it
translates towards, in the opposite domain. Considering Figure 5,
the Translator needs two security associations towards the multicast
domain, one for B and one for D. It may be forced to maintain a set
of totally independent security associations between itself and B and
D respectively, so as to avoid two-time pad occurrences. These
contexts must also be capable of handling all the sources present in
the other domains. Hence, using completely independent security
associations (for certain keying mechanisms) may force a Translator
to handle N*DM keys and related state; where N is the total number of
SSRCs used over all domains and DM is the total number of domains.
There exist a number of different mechanisms to provide keys to the
different participants. One example is the choice between group keys
and unique keys per SSRC. The appropriate keying model is impacted
by the topologies one intends to use. The final security properties
are dependent on both the topologies in use and the keying
mechanisms' properties, and need to be considered by the application.
Exactly which mechanisms are used is outside of the scope of this
document. Please review RTP Security Options
[I-D.ietf-avtcore-rtp-security-options] to get a better understanding
of most of the available options.
6. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
7. Acknowledgements
The authors would like to thank Bo Burman, Umesh Chandra, Roni Even,
Keith Lantz, Ladan Gharai, Geoff Hunt, and Mark Baugher for their
help in reviewing this document.
8. References
8.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
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Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
8.2. Informative References
[H323] ITU-T Recommendation H.323, "Packet-based multimedia
communications systems", June 2006.
[I-D.ietf-avtcore-rtp-security-options]
Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", draft-ietf-avtcore-rtp-security-options-00
(work in progress), July 2012.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for
IP", RFC 4607, August 2006.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760, February 2010.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 2011.
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Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Stephan Wenger
Vidyo
433 Hackensack Ave
Hackensack, NJ 07601
USA
Email: stewe@stewe.org
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