Internet DRAFT - draft-westerlund-avtcore-rtp-topologies-update
draft-westerlund-avtcore-rtp-topologies-update
Network Working Group M. Westerlund
Internet-Draft Ericsson
Obsoletes: 5117 (if approved) S. Wenger
Intended status: Informational Vidyo
Expires: August 29, 2013 February 25, 2013
RTP Topologies
draft-westerlund-avtcore-rtp-topologies-update-02
Abstract
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.
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.
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This Internet-Draft will expire on August 29, 2013.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 4
3.2. Point to Point via Middlebox . . . . . . . . . . . . . . . 5
3.2.1. Translators . . . . . . . . . . . . . . . . . . . . . 5
3.2.2. Back to Back RTP sessions . . . . . . . . . . . . . . 8
3.3. Point to Multipoint Using Multicast . . . . . . . . . . . 9
3.3.1. Any Source Multicast (ASM) . . . . . . . . . . . . . . 9
3.3.2. Source Specific Multicast (SSM) . . . . . . . . . . . 11
3.3.3. SSM with Local Unicast Resources . . . . . . . . . . . 12
3.4. Point to Multipoint Using Mesh . . . . . . . . . . . . . . 14
3.5. Point to Multipoint Using the RFC 3550 Translator . . . . 15
3.5.1. Relay - Transport Translator . . . . . . . . . . . . . 15
3.5.2. Media Translator . . . . . . . . . . . . . . . . . . . 16
3.6. Point to Multipoint Using the RFC 3550 Mixer Model . . . . 16
3.6.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 18
3.6.2. Media Switching . . . . . . . . . . . . . . . . . . . 21
3.7. Source Projecting Middlebox . . . . . . . . . . . . . . . 23
3.8. Point to Multipoint Using Video Switching MCUs . . . . . . 26
3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . . 27
3.10. De-composite Endpoint . . . . . . . . . . . . . . . . . . 29
3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 30
3.12. Combining Topologies . . . . . . . . . . . . . . . . . . . 30
4. Comparing Topologies . . . . . . . . . . . . . . . . . . . . . 31
4.1. Topology Properties . . . . . . . . . . . . . . . . . . . 31
4.1.1. All to All Media Transmission . . . . . . . . . . . . 31
4.1.2. Transport or Media Interoperability . . . . . . . . . 32
4.1.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . . 32
4.1.4. Aggregation of Media . . . . . . . . . . . . . . . . . 32
4.1.5. View of All Session Participants . . . . . . . . . . . 32
4.1.6. Loop Detection . . . . . . . . . . . . . . . . . . . . 33
4.2. Comparison of Topologies . . . . . . . . . . . . . . . . . 33
5. Security Considerations . . . . . . . . . . . . . . . . . . . 33
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 36
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 36
8.1. Normative References . . . . . . . . . . . . . . . . . . . 36
8.2. Informative References . . . . . . . . . . . . . . . . . . 36
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 37
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1. Introduction
Real-time Transport Protocol (RTP) [RFC3550] topologies describe
methods for interconnecting RTP entities and their processing
behavior of RTP and RTCP. This document tries to address past and
existing confusion, especially with respect to terms not defined in
RTP but in common use in the conversational communication industry,
such as MCU. In doing so, this memo provides a common information
basis for future discussion and specification work. It attempts to
clarify and explain sections of the 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
3. Topologies
This subsection defines several topologies that are relevant for
codec control but also RTP usage in other contexts. The section
starts with point to point cases, without and with middleboxes. Then
follows a number of different methods for establishing point to
multipoint communication. These are structure around the most
fundamental enabler, i.e. multicast, a mesh of connections,
translators, mixers and source projection middlebox, to finally
discuss MCUs. The section ends by discussing de-composed endpoints,
asymmetric middlebox behaviors and combining topologies.
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 discuss
the handling of RTCP feedback messages as defined in [RFC4585] and
[RFC5104]. Any important differences between the two will be
illuminated in the discussion. At this stage we don't intended to
discuss in detail how each and every feedback messages should be
treated in the various topologies.
3.1. Point to Point
Shortcut name: Topo-Point-to-Point
The Point to Point (PtP) topology (Figure 1) consists of two
endpoints, communicating using unicast. Both RTP and RTCP traffic
are conveyed endpoint-to-endpoint, using unicast traffic only (even
if, in exotic cases, this unicast traffic happens to be conveyed over
an IP-multicast address).
+---+ +---+
| 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
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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. For RTP sessions
which use multiple SSRC per endpoint it can be relevant to implement
support for cross reporting suppression as defined in "Real-Time
Transport Protocol (RTP) Considerations for Endpoints Sending
Multiple Media Streams" [I-D.lennox-avtcore-rtp-multi-stream].
3.2. Point to Point via Middlebox
This section discusses cases where two endpoints communicate but have
one or more middlebox involved in the RTP session.
3.2.1. Translators
Shortcut name: Topo-PtP-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 keeps the SSRC for a stream
across the translation, whereas a Mixer can select a single media
stream, or send out multiple mixed media streams, but always under
its own SSRC, possibly using the CSRC field to indicate the source(s)
of the content. Mixers are more common in point to multipoint cases
than in PtP. The reason is that in PtP use cases the primary focus
is interoperability, such as transcoding to a codec the receiver
supports, which can be done by a media translator.
As specified in Section 7.1 of [RFC3550], the SSRC space is common
for all participants in the RTP session, independent of on which side
of the Translator the session resides. Therefore, it is the
responsibility of the participants to run SSRC collision detection,
and the SSRC is thus a field the Translator cannot change. Any SDES
information associated with a SSRC or CSRC also needs to be forwarded
between the domains for any SSRC/CSRC used in the different domains.
A Translator commonly does not use an SSRC of its own, and is not
visible as an active participant in the session. One reason to have
its own SSRC is when a Translator acts as a quality monitor that
sends RTCP reports and therefore is required to have an SSRC.
Another example is the case when a Translator is prepared to use RTCP
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feedback messages. This may, for example, occur in a translator
configured to detect packet loss of important video packets and wants
to trigger repair by the media sender, by sending feedback messages.
While such feedback could use the SSRC of the target for the
translator, but this in turn would require translation of the targets
RTCP reports to make them consistent. It may be simpler to expose an
additional SSRC in the session, the only concern are endpoints
failing to support the full RTP specification, thus having issues
with multiple SSRCs reporting on the RTP streams sent by that
endpoint.
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.
3.2.1.1. Transport Relay/Anchoring
There exist a number of different types of middleboxes that might be
inserted between two RTP endpoints on the transport level, e.g.
perform changes on the IP/UDP headers, and are, therefore, basic
transport translators. These middleboxes come in many variations
including NAT [RFC3022] traversal by pinning the media path to a
public address domain relay, network topologies where the media flow
is required to pass a particular point for audit by employing
relaying, or preserving privacy by hiding each peers transport
addresses to the other party. Other protocols or functionalities
that provide this behavior are TURN [RFC5766] servers, Session Border
Gateways and Media Processing Nodes with media anchoring
functionalities.
+---+ +---+ +---+
| A |<------>| T |<------->| B |
+---+ +---+ +---+
Figure 2: Point to Point with Translator
What is common for these functions is that they are normally
transparent on RTP level, i.e. they perform no changes on any RTP or
RTCP packet fields, only on the lower layers. However, they may
effect the path the RTP and RTCP packets are routed between the
endpoints in the RTP session, and thereby only indirectly affect the
RTP session. For this reason, one could believe that transport
translator type middleboxes do not need to included in this document.
However, this topology can raise additional requirements the RTP
implementation and its interactions with the signalling solution.
Both in signalling and in certain RTCP field other network addresses
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than those of the relay can occur, due to that B has different
network address than the relay (T). However, implementation not
capable of this will neither not work when endpoints are subject to
NAT.
3.2.1.2. 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 such unicast to
multicast) and the media packetization to allow other transport
protocols to be interconnected to a session (in gateways). Of the
transport Translators, this memo is primarily interested in those
that use RTP on both sides, and this is assumed henceforth.
Translators that bridge between different protocol worlds need to be
concerned about the mapping of the SSRC/CSRC (Contributing Source)
concept to the non-RTP protocol. When designing a Translator to a
non-RTP-based media transport, one crucial factor lies in how to
handle different sources and their identities. This problem space is
not discussed henceforth.
The most basic transport translators that operate below RTP level was
already discussed in Section 3.2.1.1.
3.2.1.3. 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
(down to the sample level or equivalent) utilizing a different media
codec. Media Translators are commonly used to connect entities
without a common interoperability point in the media encoding.
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).
When media translation occurs, the Translator's task regarding
handling of RTCP traffic becomes substantially more complex. In this
case, the Translator needs to rewrite B's RTCP Receiver Report before
forwarding them to A. The rewriting is needed as the stream received
by B is not the same stream as the other participants receive. For
example, the number of packets transmitted to B may be lower than
what A sends, due to the different media format and data rate.
Therefore, if the Receiver Reports were forwarded without changes,
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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.
A variant of translator behaviour worth pointing out is the one
depicted in Figure 3 of an endpoint A sends a media flow to B. On the
path there is a device T that on A's behalf does something with the
media streams, for example adds an RTP session with FEC information
for A's media streams. In this case, T needs to bind the new FEC
streams to A's media stream, for example by using the same CNAME as
A.
+------+ +------+ +------+
| | | | | |
| A |------->| T |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+
Figure 3: When De-composition is a Translator
This type of functionality where T does something with the media
stream on behalf of A is covered under the media translator
definition.
3.2.2. Back to Back RTP sessions
There exist middleboxes that interconnect two endpoints through
themselves not by being part of a common RTP session. Instead they
establish two different RTP sessions, one between A and the middlebox
(MB) and another between the MB and B.
|<--Session A-->| |<--Session B-->|
+------+ +------+ +------+
| A |------->| MB |-------->| B |
+------+ +------+ +------+
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Figure 4: When De-composition is a Translator
The MB acts as a application level gateway and bridges the two RTP
session. This bridging can be as basic as forwarding the RTP
payloads between the sessions, or more complex including media
transcoding. The difference with the single RTP session context is
the handling of the SSRCs and the other session related identifiers,
such as CNAMEs. With two different RTP sessions these can be freely
changed and it becomes the MB task to maintain the right relations.
The signalling or other above-RTP level functionalities referencing
RTP media streams may be what is most impacted by using two RTP
sessions and changing identifiers. The structure with two RTP
sessions also puts a congestion control requirement on the middlebox,
because it becomes fully responsible for the media stream it sources
into each of the sessions.
This can be solved locally or by bridging also statistics from the
receiving endpoint. However, from an implementation point this
requires the implementation to support dealing with a number of
inconsistencies. First, packet loss must be detected for an RTP flow
sent from A to the MB, and that loss must be reported through a
skipped sequence number in the flow from the MB to B. This coupling
and the resulting inconsistencies is conceptually easier to handle
when considering the two flows as belonging to a single RTP session.
3.3. Point to Multipoint Using Multicast
Multicast is a IP layer functionality that is available in some
networks. Two main flavors can be distinguished: Any Source
Multicast (ASM) where any multicast group participant can send to the
group address and expect the packet to reach all group participants;
and Source Specific Multicast (SSM), where only a particular IP host
sends to the multicast group. Both these models are discussed below
in their respective section.
3.3.1. Any Source Multicast (ASM)
Shortcut name: Topo-ASM (was Topo-Multicast)
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+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Figure 5: Point to Multipoint Using Multicast
Point to Multipoint (PtM) is defined here as using a multicast
topology as a transmission model, in which traffic from any
participant reaches all the other participants, except for cases such
as:
o packet loss, or
o when a participant does not wish to receive the traffic for a
specific multicast group and, therefore, has not subscribed to the
IP-multicast group in question. This scenario can occur, for
example, where a multi-media session is distributed using two or
more multicast groups and a participant is subscribed only to a
subset of these sessions.
In the above context, "traffic" encompasses both RTP and RTCP
traffic. The number of participants can vary between one and many,
as RTP and RTCP scale to very large multicast groups (the theoretical
limit of the number of participants in a single RTP session is in the
range of billions). The above can be realized using Any Source
Multicast (ASM).
For feedback usage, it is useful to define a "small multicast group"
as a group where the number of participants is so low (and other
factors such as the connectivity is so good) that it allows the
participants to use early or immediate feedback, as defined in AVPF
[RFC4585]. Even when the environment would allow for the use of a
small multicast group, some applications may still want to use the
more limited options for RTCP feedback available to large multicast
groups, for example when there is a likelyhood that the threshold of
the small multicast group (in terms of participants) may be exceeded
during the lifetime of a session.
RTCP feedback messages in multicast reach, like media, every
subscriber (subject to packet losses and multicast group
subscription). Therefore, the feedback suppression mechanism
discussed in [RFC4585] is typically required. Each individual node
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needs to process every feedback message it receives, not to determine
if it is affected or if the feedback message applies only to some
other participant, but also to derive timing restriction for the
sending of its own feedback messages, if any.
3.3.2. Source Specific Multicast (SSM)
In Any Source Multicast, any of the participants can send to all the
other participants, by sending a packet to the multicast group. In
contrast, Source Specific Multicast [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:
+--------+ +-----+
|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 6: Point to Multipoint using Source Specific Multicast
In the SSM topology (Figure 6) a number of RTP sources (1 to M) are
allowed to send media to the SSM group. These send media to a
dedicated distribution source, which then forwards the media streams
to the multicast group on behalf of the original senders. The media
streams reach the Receivers (R(1) to R(n)). The Receivers' RTCP
cannot be sent to the multicast group, as the SSM multicast group by
definition has only a single source. To support RTCP, an RTP
extension for SSM [RFC5760] was defined. It uses unicast
transmission to send RTCP from each of the receivers to one or more
Feedback Targets (FT). The feedback targets relay the RTCP
unmodified, or provide summary of the participants RTCP reports
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towards the whole group by forwarding the RTCP traffic to the
distribution source. Figure 6 only shows a single feedback target
integrated in the distribution source, but for scalability the FT can
be many and have responsibility for sub-groups of the receivers. For
summary reports, however, there must be a single feedback aggregating
all the summaries to a common message to the whole receiver group.
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.
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 needs to be a member of most or all of these
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.
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 need to have the same RTP and payload type
configuration. Otherwise, A could, for example, be using payload
type 97 to identify the video codec H.264, while B would identify it
as MPEG-2.
Security solutions for this type of group communications are also
challenging. First, the key-management and the security protocol
must support group communication. Source authentication becomes more
difficult and requires special solutions. For more discussion on
this please review Options for Securing RTP Sessions
[I-D.ietf-avtcore-rtp-security-options].
3.3.3. SSM with Local Unicast Resources
[RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions"
results in additional extensions to SSM Topology.
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----------- --------------
| |------------------------------------>| |
| |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| |
| | | |
| Multicast | ---------------- | |
| Source | | Retransmission | | |
| |-------->| Server (RS) | | |
| |.-.-.-.->| | | |
| | | ------------ | | |
----------- | | Feedback | |<.=.=.=.=.| |
| | Target (FT)| |<~~~~~~~~~| RTP Receiver |
PRIMARY MULTICAST | ------------ | | (RTP_Rx) |
RTP SESSION with | | | |
UNICAST FEEDBACK | | | |
| | | |
- - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- -
| | | |
UNICAST BURST | ------------ | | |
(or RETRANSMISSION) | | Burst/ | |<~~~~~~~~>| |
RTP SESSION | | Retrans. | |.........>| |
| |Source (BRS)| |<.=.=.=.=>| |
| ------------ | | |
| | | |
---------------- --------------
-------> Multicast RTP Flow
.-.-.-.> Multicast RTCP Flow
.=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages
.......> Unicast RTP Flow
Figure 7
The Rapid acquisition extension allows an endpoint joining an SSM
multicast session to request media starting with the last sync-point
(from where media can be decoded without prior packets) to be sent at
high speed until such time where, after decoding of these bursted
media packets, the correct media timing is established, i.e. media
packets are received within adequate buffer intervals for this
application. This is accomplished by first establishing an unicast
PtP RTP session between the BRS (Figure 7) and the RTP Receiver.
That session is used to transmit cached packets from the multicast
group at higher then nominal speed so to synchronize the receiver to
the ongoing multicast packet flow. Once the RTP receiver and its
decoder have caught up with the multicast session's current delivery,
the receiver switches over to receiving from the multicast group
directly. The (still existing) PtP RTP session can be used as a
repair channel, i.e. for RTP Retransmission traffic of those packets
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that were not received from the multicast group.
3.4. Point to Multipoint Using Mesh
Shortcut name: Topo-Mesh
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 8: 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.
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.
An alternative structure for establishing the above topology is to
use independent RTP sessions between each pair of peers, i.e. three
different RTP sessions. In some scenarios, the same RTP media stream
is being sent from each sending endpoint. In others, some form of
local adaptation takes place in one or more of the RTP media streams,
rendering them non-identical. From a topologies viewpoint, a
difference exists in the behaviours around RTCP. For example, when a
single RTP session spans all three endpoints and their connecting
flows, a RTCP bandwidth is calculated and used for this single one
joint session. In contrast, when there are multiple independent RTP
sessions, each has its local RTCP bandwidth allocation. Also, when
multiple sessions are used, endpoints not directly involved in these
sessions do not have any awareness of the conditions occurring in
sessions not involving that endpoint. For example, in case of the
three endpoint configuration above, endpoint A has no awareness of
the conditions occurring in the session between endpoints B and C
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(whereas, if a single RTP session were used, it would have such
awareness). Loop detection is also affected. With independent RTP
sessions, the SSRC/CSRC can't be used to determine when a endpoint
receives its own media stream or a mixed media stream including its
own media stream a condition known as a loop. The identification of
loops and, in most cases, its avoidance, has to be achieved by other
means, for example through signaling, or the use of an RTP external
name space binding SSRC/CSRC among any communicating RTP sessions in
the mesh.
3.5. Point to Multipoint Using the RFC 3550 Translator
This section discusses some additional usages related to point to
multipoint of Translators compared to the point to point only cases
in Section 3.2.1.
3.5.1. Relay - Transport Translator
Shortcut name: Topo-PtM-Trn-Translator
This section discusses Transport Translator only usages to enable
multipoint sessions.
+-----+
+---+ / \ +------------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Translator |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+
+-----+
Figure 9: Point to Multipoint Using Multicast
Figure 9 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.
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+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 10: RTP Translator (Relay) with Only Unicast Paths
Another Translator scenario is depicted in Figure 10. 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 set of considerations
those described in the ASM RTP topology. It also puts some
additional signalling requirements onto the session establishment;
for example, a common configuration of RTP payload types is required.
3.5.2. Media Translator
In the context of multipoint communications a Media Translator is not
providing new mechanisms to establish a multipoint session. It is
much more an enabler or facilitator that ensures one or some sub-set
of session participants can participate in the session.
If B in Figure 9 were behind a limited network path, the Translator
may perform media transcoding to allow the traffic received from the
other participants to reach B without overloading the path. This
transcoding can help the other participants in the Multicast part of
the session, by not requiring the quality transmitted by A to be
lowered to the nitrates that B is actually capable of receiving.
3.6. 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 generating a new RTP stream and, in most
cases, by manipulation of the media data. One common application for
a Mixer is to allow a participant to receive a session with a reduced
amount of resources.
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+-----+
+---+ / \ +-----------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Mixer |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +-----------+ +---+
+-----+
Figure 11: 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
the uncompressed domain) and then encodes it for transmission to a
participant. The CSRC Count (CC) and CSRC fields in the RTP header
can be used to indicate the contributors of to the newly generated
stream. The SSRCs of the to-be-mixed streams on the Mixer input
appear as the CSRCs at the Mixer output. That output stream uses a
unique SSRC that identifies the Mixer's stream. The CSRC should be
forwarded between the different conference participants to allow for
loop detection and identification of sources that are part of the
global session. Note that Section 7.1 of RFC 3550 requires the SSRC
space to be shared between domains for these reasons. This also
implies that any SDES information normally needs to be forwarded
across the mixer.
The Mixer is responsible for generating RTCP packets in accordance
with its role. It is a receiver and should therefore send receiver
reports for the media streams it receives. In its role as a media
sender, it should also generate sender reports for those media
streams it sends. As specified in Section 7.3 of RFC 3550, a Mixer
must not forward RTCP unaltered between the two domains.
The Mixer depicted in Figure 11 is involved in three domains that
need to be separated: the any source multicast network (including
participants A and C), participant B, and participant D. Assuming all
four participants in the conference are interested in receiving
content from each other participant, the Mixer produces different
mixed streams for B and D, as the one to B may contain content
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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 3.11 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 by the Mixer may result in the generation
and transmission of RTCP feedback messages by the Mixer to the
participants in the other domain(s). In other cases, a message is
handled by the Mixer itself and therefore not forwarded to any other
domain.
When replacing the multicast network in Figure 11 (to the left of the
Mixer) with individual unicast paths as depicted in Figure 12, the
Mixer model is very similar to the one discussed in Section 3.9
below. Please see the discussion in Section 3.9 about the
differences between these two models.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 12: RTP Mixer with Only Unicast Paths
Lets now discuss in more detail different mixing operations that a
mixer can perform and how they can affect the RTP and RTCP.
3.6.1. Media Mixing
The media mixing mixer is likely the one that most think of when they
hear the term "mixer". Its basic pattern of operation is that it
receives media streams from (typically several) participants. Of
those, it selects (either through static configuration or by dynamic,
content dependent means such as voice activation) the stream(s) to be
included in a media domain mix. Then it creates a single outgoing
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stream from this mix.
The most commonly deployed media mixer is probably the audio mixer,
used in voice conferencing, where the output consists of a mixture of
all the input streams; this needs minimal signalling to be
successfully set up. Audio mixing is relatively straightforward and
commonly possible for a reasonable number of participants. Lets
assume that you want to mix N streams from different participants.
The mixer needs to decode those N streams, typically into the sample
domain. Then it needs to produce N or N+1 mixes, the reasons that
different mixes are needed being that each contributing source get a
mix of all other sources except its own, as this would result in an
echo. When N is lower than the number of all participants one may
produce a Mix of all N streams for the group that are currently not
included in the mix, thus N+1 mixes. These audio streams are then
encoded again, RTP packetized and sent out. In many cases, audio
level normalization is also required before the actual mixing
process.
Video can't really be "mixed" and produce something particularly
useful for the users, however creating an composition out of the
contributed video streams is possible and known as "tiling". For
example the reconstructed, appropriately scaled down videos can be
spatially arranged in a set of tiles, each tile containing the video
from a participant. Tiles can be of different sizes, so that, for
example, a particularly important participant, or the loudest
speaker, is being shown on in larger tile than other participants. A
self-picture can be included in the tiling, which can either be
locally produced or be a feedback from a received and reconstructed
video image (allowing for confidence monitoring, the participant sees
himself/herself just as other participants see him/her). The tiling
normally operates on reconstructed video in the sample domain. The
tiled image is encoded, packetized, and sent by the mixer. It is
possible that a middlebox with media mixing duties contains only a
single mixer of the aforementioned type, in which case all
participants necessarily see the same tiled video, even if it is
being sent over different RTP streams. More common, however, are
mixing arrangement where an individual mixer is available for each
outgoing port of the middlebox, allowing individual compositions for
each participant.
One problem with media mixing is that it consumes both large amount
of media processing (for the actual mixing process in the
uncompressed domain) and encoding resources (for the encoding of the
mixed signal). Another problem is the quality degradation created by
decoding and re-encoding the media that is encapsulated in the RTP
media stream, which is the result of the lossy nature of most, if not
all, commonly used media codecs. A third problem is the latency
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introduced by the media mixing, which can be substantial and
annoyingly noticeable in case of video. The advantage of media
mixing is that it is quite simplistic for the clients to handle the
single media stream (which includes the mixed aggregate of many
sources), as they don't need to handle multiple decodings, local
mixing and composition. In fact, mixers were introduced in pre-RTP
times so that legacy, single stream receiving endpoints can
successfully participate in what a user would recognize as a
multiparty session.
+-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 13: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be very straightforward as
can be seen in Figure 13. The mixer presents one SSRC towards the
receiving client, e.g. MA1 to Peer A; the associated stream of which
is the media mix of the other participants. As, in this example,
each peer receives a different version produced by the mixer, there
is no actual relation between the different RTP sessions in the
actual media or the transport level information. There are, however,
common relationships between RTP1-RTP3 namely SSRC space and identity
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information. When A receives the MA1 stream which is a combination
of BA1 and CA1 streams, the mixer may include CSRC information in the
MA1 stream to identify the contributing source BA1 and CA1, allowing
the receiver to identify the contributing sources even if this were
not possible through the media itself or other signaling means.
The CSRC has, in turn, utility in RTP extensions, like the Mixer to
Client audio levels RTP header extension [RFC6465]. If the SSRC from
endpoint to mixer leg are used as CSRC in another RTP session, then
RTP1, RTP2 and RTP3 become one joint session as they have a common
SSRC space. At this stage, the mixer also need to consider which
RTCP information it needs to expose in the different legs. In the
above scenario, commonly, a mixer would expose nothing more than the
Source Description (SDES) information and RTCP BYE for CSRC leaving
the session. The main goal would be to enable the correct binding
against the application logic and other information sources. This
also enables loop detection in the RTP session.
3.6.2. Media Switching
Media switching mixers are commonly used in such limited
functionality scenarios where no, or only very limited, concurrent
presentation of multiple sources is required by the application. An
RTP Mixer based on media switching avoids the media decoding and
encoding cycle in the mixer, as it conceptually forwards the encoded
media stream as it was being sent to the mixer, but not the
decryption and re-encryption cycle as it rewrites RTP headers.
Forwarding media (in contrast to reconstructing-mixing-encoding
media) reduces the amount of computational resources needed in the
mixer and increases the media quality (both in terms of fidelity and
reduced latency) per transmitted bit.
A media switching mixer maintains a pool of SSRCs representing
conceptual or functional streams the mixer can produce. These
streams are created by selecting media from one of RTP media streams
received by the mixer and forwarded to the peer using the mixer's own
SSRCs. The mixer can switch between available sources if that is
required by the concept for the source, like currently active
speaker. Note that the mixer, in most cases, still need to perform a
certain amount of media processing, as many media formats do not
allow to "tune" into the stream at arbitrary points of their
bitstream.
To achieve a coherent RTP media stream from the mixer's SSRC, the
mixer needs to rewrite the incoming RTP packet's header. First the
SSRC field must be set to the value of the Mixer's SSRC. Secondly,
the sequence number must be the next in the sequence of outgoing
packets it sent. Thirdly the RTP timestamp value needs to be
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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, and each peer is
capable of locally handling two video streams simultaneously.
+-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 14: Media Switching RTP Mixer
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The Media Switching RTP mixer can, similarly to the Media Mixing
Mixer, reduce the bit-rate required for media transmission towards
the different peers by selecting and forwarding only a sub-set of RTP
media streams it receives from the conference participants. In many
practical cases, the link capacities of either direction between
peers and mixer are the same, which effectively limits the subset to
a single media stream.
To ensure that a media receiver can correctly decode the RTP media
stream after a switch, a state saving (frame-based) codec needs to
start its decoding from independent refresh points or similar points
in the bitstream. For some codecs, for example frame based speech
and audio codecs, this is easily achieved by starting the decoding at
RTP packet boundaries (proper packetization on the encoder side
assumed), as each packet boundary provides a refresh point. For
other (mostly video-) codecs, refresh points are less common in the
bitstream or may not be present at all without an explicit request to
the respective encoder. For this purpose there exists the Full Intra
Request [RFC5104] RTCP codec control message.
Also in this type of mixer one could consider to terminate the RTP
sessions fully between the different end-point and mixer legs. The
same arguments and considerations as discussed in Section 3.9 need to
be taken into consideration and apply here.
3.7. 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 middlebox can select which of the potential sources
that are currently actively transmitting media, despite that the
middlebox, in another RTP session, may receive media from that end-
point. This is similar to the media switching Mixer but has some
important differences in RTP details.
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+-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 <----+-+-------| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +---------------------------+
Figure 15: Media Projecting Middlebox
In the six participant conference depicted above in (Figure 15) one
can see that end-point A is aware of five incoming SSRCs, BV1-FV1.
If this middlebox intends to have a similar behaviour as in
Section 3.6.2 where the mixer provides the end-points with the two
latest speaking end-points, then only two out of these five SSRCs
need concurrently transmit media to A. As the middlebox selects the
source in the different RTP sessions that transmit media to the end-
points, each RTP media stream requires some rewriting of RTP header
fields when being projected from one session into another. In
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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.
As the RTP sessions are independent, the SSRC numbers used can also
be handled independently, thereby bypassing the requirement for SSRC
collision detection and avoidance. On the other hand, tools such as
remapping tables between the RTP sessions are required. For example,
the stream that is being sent by endpoint B to the middlebox (BV1)
may use an SSRC value of 12345678. When that media stream is sent to
endpoint F by the middlebox, it can use any SSRC value, e.g.
87654321. As a result, each endpoint may have a different view of
the application usage of a particular SSRC. Any RTP level identity
information, such as SDES items also needs to update the SSRC
referenced, if the included SDES items are intended to be global.
Thus the application must not use SSRC as references to RTP media
streams when communicating with other peers directly. This also
affects loop detection which will fail to work, as there is no common
namespace and identities across the different legs in the
communication session on RTP level. Instead this responsibility
falls onto higher layers.
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).
This version of the middlebox also puts different requirements on the
end-point when it comes to decoder instances and handling of the RTP
media streams providing media. As each projected SSRC can, at any
time, provide media, the end-point either needs to be able to handle
as many decoder instances as the middlebox received, or have
efficient switching of decoder contexts in a more limited set of
actual decoder instances to cope with the switches. The application
also gets more responsibility to update how the media provided is to
be presented to the user.
Note, this could potentially be seen as a media translator which
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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.
3.8. Point to Multipoint Using Video Switching MCUs
Shortcut name: Topo-Video-switch-MCU
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
Figure 16: Point to Multipoint Using a Video Switching MCU
This PtM topology was common before, although the RTCP-terminating
MCUs, as discussed in the next section, where perhaps even more
common. This topology, as well as the following one, was a result of
lack of wide availability of IP multicast technologies, as well as
the simplicity of content switching when compared to content mixing.
The technology is commonly implemented in what is known as "Video
Switching MCUs".
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.
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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.
To avoid the aforementioned issues, the MCU needs to implement two
features. First, it needs to act as a Mixer (see Section 3.6) 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.
3.9. Point to Multipoint Using RTCP-Terminating MCU
Shortcut name: Topo-RTCP-terminating-MCU
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 17: 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
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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
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.
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3.10. 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 its own
network address. A very basic use case for this would be to separate
audio and video processing for a particular endpoint, like a
conference room, into one device handling the audio and another
handling the video, being interconnected by some control functions
allowing them to behave as a single endpoint in all aspects except
for transport Figure 18.
Which decomposition scheme is possible is highly dependent on the RTP
session usage. It is not really feasible to decompose one logical
end-point into two different transport nodes in one RTP session. A
third party monitor would report such an attempt as two entities
being two different end-points with a CNAME collision. As a result,
a fully RTP conformant de-composited endpoint is one where the
different decomposed parts use separate RTP sessions to send and/or
receive media streams intended for them.
+---------------------+
| Endpoint A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+-RTP---\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+-RTP-------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+-SIP---/
| +------------+ |
+---------------------+
Figure 18: De-composite End-Point
In the above usage, let us assume that the different RTP sessions are
used for audio and video. The audio and video parts, however, use a
common CNAME and also have a common clock to ensure that
synchronization and clock drift handling works, despite the
decomposition. Also, the RTCP handling works correctly as long as
only one part of the de-composite is part of each RTP session. That
way any differences in the path between A's audio entity and B and
A's video and B are related to different SSRCs in different RTP
sessions.
The requirement that can be derived from the above usage is that the
transport flows for each RTP session might be under common control,
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but still are addressed to what looks like different endpoints (based
on addresses and ports). This geometry cannot be accomplished using
one RTP session, so in this case, multiple RTP sessions are needed.
3.11. 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 11,
where the Mixer does not need to mix in the direction from B or D
towards the multicast domains with A and C. Instead, the media
streams from B and D are forwarded without changes. Avoiding this
mixing would save media processing resources that perform the mixing
in cases where it isn't needed. However, there would still be a need
to mix B's stream towards D. Only in the direction B -> multicast
domain or D -> multicast domain would it be possible to work as a
Translator. In all other directions, it would function as a Mixer.
The Mixer/Translator would still need to process and change the RTCP
before forwarding it in the directions of B or D to the multicast
domain. One issue is that A and C do not know about the mixed-media
stream the Mixer sends to either B or D. Therefore, any reports
related to these streams must be removed. Also, receiver reports
related to A and C's media stream would be missing. To avoid A and C
thinking that B and D aren't receiving A and C at all, the Mixer
needs to insert locally generated reports reflecting the situation
for the streams from A and C into B and D's Sender Reports. In the
opposite direction, the Receiver Reports from A and C about B's and
D's stream also need to be aggregated into the Mixer's Receiver
Reports sent to B and D. Since B and D only have the Mixer as source
for the stream, all RTCP from A and C must be suppressed by the
Mixer.
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.
3.12. 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 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
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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 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.
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 3.5, 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.
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The transmission of RTCP automatically adapts to any changes in the
number of participants due to the transmission algorithm, defined in
the RTP specification [RFC3550], and the extensions in AVPF [RFC4585]
(when applicable). That way, the resources utilized for RTCP stay
within the bounds configured for the session.
4.1.2. Transport or Media Interoperability
Translators, Mixers, and RTCP-terminating MCU all allow changing the
media encoding or the transport to other properties of the other
domain, thereby providing extended interoperability in cases where
the participants lack a common set of media codecs and/or transport
protocols.
4.1.3. Per Domain Bit-Rate Adaptation
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).
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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 3.8 does not
entirely fulfill this. The one described in Section 3.9 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 3.8, at least as long as it forwards the
RTCP between the participants. However, the MCU in Section 3.9 will
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
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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
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:
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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,
a Translator needs one security association per participant it
translates towards, in the opposite domain. Considering Figure 9,
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.
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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
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4575] 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
[I-D.ietf-avtcore-rtp-security-options]
Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", draft-ietf-avtcore-rtp-security-options-01
(work in progress), October 2012.
[I-D.lennox-avtcore-rtp-multi-stream]
Lennox, J. and M. Westerlund, "Real-Time Transport
Protocol (RTP) Considerations for Endpoints Sending
Multiple Media Streams",
draft-lennox-avtcore-rtp-multi-stream-01 (work in
progress), October 2012.
[RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network
Address Translator (Traditional NAT)", RFC 3022,
January 2001.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for
IP", RFC 4607, August 2006.
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[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.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 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.
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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