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INFORMATIONAL
Internet Engineering Task Force (IETF) M. Westerlund
Request for Comments: 7667 Ericsson
Obsoletes: 5117 S. Wenger
Category: Informational Vidyo
ISSN: 2070-1721 November 2015
RTP Topologies
Abstract
This document discusses point-to-point and multi-endpoint topologies
used in environments based on the Real-time Transport Protocol (RTP).
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 replaces RFC
5117.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7667.
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Copyright Notice
Copyright (c) 2015 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
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.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 5
2.2. Definitions Related to RTP Grouping Taxonomy . . . . . . 5
3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 6
3.2. Point to Point via Middlebox . . . . . . . . . . . . . . 7
3.2.1. Translators . . . . . . . . . . . . . . . . . . . . . 7
3.2.2. Back-to-Back RTP sessions . . . . . . . . . . . . . . 11
3.3. Point to Multipoint Using Multicast . . . . . . . . . . . 12
3.3.1. Any-Source Multicast (ASM) . . . . . . . . . . . . . 12
3.3.2. Source-Specific Multicast (SSM) . . . . . . . . . . . 14
3.3.3. SSM with Local Unicast Resources . . . . . . . . . . 15
3.4. Point to Multipoint Using Mesh . . . . . . . . . . . . . 17
3.5. Point to Multipoint Using the RFC 3550 Translator . . . . 20
3.5.1. Relay - Transport Translator . . . . . . . . . . . . 20
3.5.2. Media Translator . . . . . . . . . . . . . . . . . . 21
3.6. Point to Multipoint Using the RFC 3550 Mixer Model . . . 22
3.6.1. Media-Mixing Mixer . . . . . . . . . . . . . . . . . 24
3.6.2. Media-Switching Mixer . . . . . . . . . . . . . . . . 27
3.7. Selective Forwarding Middlebox . . . . . . . . . . . . . 29
3.8. Point to Multipoint Using Video-Switching MCUs . . . . . 33
3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 34
3.10. Split Component Terminal . . . . . . . . . . . . . . . . 35
3.11. Non-symmetric Mixer/Translators . . . . . . . . . . . . . 38
3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 38
4. Topology Properties . . . . . . . . . . . . . . . . . . . . . 39
4.1. All-to-All Media Transmission . . . . . . . . . . . . . . 39
4.2. Transport or Media Interoperability . . . . . . . . . . . 40
4.3. Per-Domain Bitrate Adaptation . . . . . . . . . . . . . . 40
4.4. Aggregation of Media . . . . . . . . . . . . . . . . . . 41
4.5. View of All Session Participants . . . . . . . . . . . . 41
4.6. Loop Detection . . . . . . . . . . . . . . . . . . . . . 42
4.7. Consistency between Header Extensions and RTCP . . . . . 42
5. Comparison of Topologies . . . . . . . . . . . . . . . . . . 42
6. Security Considerations . . . . . . . . . . . . . . . . . . . 43
7. References . . . . . . . . . . . . . . . . . . . . . . . . . 45
7.1. Normative References . . . . . . . . . . . . . . . . . . 45
7.2. Informative References . . . . . . . . . . . . . . . . . 45
Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 48
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 48
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1. Introduction
Real-time Transport Protocol (RTP) [RFC3550] topologies describe
methods for interconnecting RTP entities and their processing
behavior for RTP and the RTP Control Protocol (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 communication
industry, such as the Multipoint Control Unit or MCU.
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. In practice, however, most multipoint
conferences operate utilizing centralized units referred to as MCUs.
MCUs may implement mixer or translator functionality (in RTP
[RFC3550] terminology) and signaling support. They may also contain
additional application-layer functionality. This document focuses on
the media transport aspects of the MCU that can be realized using
RTP, as discussed below. Further considered are the properties of
mixers and translators, and how some types of deployed MCUs deviate
from these properties.
This document also codifies new multipoint architectures that have
recently been introduced and that were not anticipated in RFC 5117;
thus, this document replaces [RFC5117]. These architectures use
scalable video coding and simulcasting, and their associated
centralized units are referred to as Selective Forwarding Middleboxes
(SFMs). This codification provides a common information basis for
future discussion and specification work.
The new topologies are Point to Point via Middlebox (Section 3.2),
Source-Specific Multicast (Section 3.3.2), SSM with Local Unicast
Resources (Section 3.3.3), Point to Multipoint Using Mesh
(Section 3.4), Selective Forwarding Middlebox (Section 3.7), and
Split Component Terminal (Section 3.10). The Point to Multipoint
Using the RFC 3550 Mixer Model (Section 3.6) has been significantly
expanded to cover two different versions, namely Media-Mixing Mixer
(Section 3.6.1) and Media-Switching Mixer (Section 3.6.2).
The document's attempt to clarify and explain sections of the RTP
spec [RFC3550] is informal. It is not intended to update or change
what is normatively specified within RFC 3550.
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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
Middlebox: A device that is on the Path that media travel between
two endpoints
MCU: Multipoint Control Unit
Path: The concatenation of multiple links, resulting in an
end-to-end data transfer.
PtM: Point to Multipoint
PtP: Point to Point
SFM: Selective Forwarding Middlebox
SSM: Source-Specific Multicast
SSRC: Synchronization Source
2.2. Definitions Related to RTP Grouping Taxonomy
The following definitions have been taken from [RFC7656].
Communication Session: A Communication Session is an association
among two or more Participants communicating with each other via
one or more Multimedia Sessions.
Endpoint: A single addressable entity sending or receiving RTP
packets. It may be decomposed into several functional blocks, but
as long as it behaves as a single RTP stack mentity, it is
classified as a single "endpoint".
Media Source: A Media Source is the logical source of a time
progressing digital media stream synchronized to a reference
clock. This stream is called a Source Stream.
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Multimedia Session: A Multimedia Session is an association among a
group of participants engaged in communication via one or more RTP
sessions.
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, with or without middleboxes. Then
it follows a number of different methods for establishing point-to-
multipoint communication. These are structured around the most
fundamental enabler, i.e., multicast, a mesh of connections,
translators, mixers, and finally MCUs and SFMs. The section ends by
discussing decomposited terminals, 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].
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
handling multiple endpoints and combining the requirements stemming
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, subject
only to system-level limitations like the number range of ports.
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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
that use multiple SSRCs per endpoint, it can be relevant to implement
support for cross-reporting suppression as defined in "Sending
Multiple Media Streams in a Single RTP Session" [MULTI-STREAM-OPT].
3.2. Point to Point via Middlebox
This section discusses cases where two endpoints communicate but have
one or more middleboxes 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 RTP
stream that the translator receives, it generates an individual RTP
stream in the other domain. A translator keeps the SSRC for an RTP
stream across the translation, whereas a mixer can select a single
RTP stream from multiple received RTP streams (in cases like audio/
video switching) or send out an RTP stream composed of multiple mixed
media received in multiple RTP streams (in cases like audio mixing or
video tiling), but always under its own SSRC, possibly using the CSRC
field to indicate the source(s) of the content. Mixers are more
common in point-to-multipoint cases than in PtP. The reason is that
in PtP use cases, the primary focus of a middlebox is enabling
interoperability, between otherwise non-interoperable endpoints, 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 endpoints (as the RTP session participants) to
run SSRC collision detection, and the SSRC is thus a field the
translator cannot change. Any Source Description (SDES) information
associated with an 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 RTP session. One reason to
have its own SSRC is when a translator acts as a quality monitor that
sends RTCP reports and therefore is required to have an SSRC.
Another example is the case when a translator is prepared to use RTCP
feedback messages. This may, for example, occur in a translator
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configured to detect packet loss of important video packets, and it
wants to trigger repair by the media sending endpoint, by sending
feedback messages. While such feedback could use the SSRC of the
target for the translator (the receiving endpoint), this in turn
would require translation of the target RTCP reports to make them
consistent. It may be simpler to expose an additional SSRC in the
session. The only concern is that endpoints failing to support the
full RTP specification may have issues with multiple SSRCs reporting
on the RTP streams sent by that endpoint, as this use case may be
viewed as exotic by implementers.
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
Shortcut name: Topo-PtP-Relay
There exist a number of different types of middleboxes that might be
inserted between two endpoints on the transport level, e.g., to
perform changes on the IP/UDP headers, and are, therefore, basic
Transport Translators. These middleboxes come in many variations
including NAT [RFC3022] traversal by pinning the media path to a
public address domain relay and network topologies where the RTP
stream is required to pass a particular point for audit by employing
relaying, or preserving privacy by hiding each peer's transport
addresses to the other party. Other protocols or functionalities
that provide this behavior are Traversal Using Relays around NAT
(TURN) [RFC5766] servers, Session Border Gateways, and Media
Processing Nodes with media anchoring functionalities.
+---+ +---+ +---+
| A |<------>| T |<------->| B |
+---+ +---+ +---+
Figure 2: Point to Point with Translator
A common element in these functions is that they are normally
transparent at the RTP level, i.e., they perform no changes on any
RTP or RTCP packet fields and only affect the lower layers. They may
affect, however, the path since the RTP and RTCP packets are routed
between the endpoints in the RTP session, and thereby they indirectly
affect the RTP session. For this reason, one could believe that
Transport Translator-type middleboxes do not need to be included in
this document. This topology, however, can raise additional
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requirements in the RTP implementation and its interactions with the
signaling solution. Both in signaling and in certain RTCP fields,
network addresses other than those of the relay can occur since B has
a different network address than the relay (T). Implementations that
cannot support this will also not work correctly when endpoints are
subject to NAT.
The Transport Relay implementations also have to take into account
security considerations. In particular, source address filtering of
incoming packets is usually important in relays, to prevent attackers
from injecting traffic into a session, which one peer may, in the
absence of adequate security in the relay, think it comes from the
other peer.
3.2.1.2. Transport Translator
Shortcut name: Topo-Trn-Translator
Transport Translators (Topo-Trn-Translator) do not modify the RTP
stream itself but are concerned with transport parameters. Transport
parameters, in the sense of this section, comprise the transport
addresses (to bridge different domains such as unicast to multicast)
and the media packetization to allow other transport protocols to be
interconnected to a session (in gateways).
Translators that bridge between different protocol worlds need to be
concerned about the mapping of the SSRC/CSRC (Contributing Source)
concept to the non-RTP protocol. When designing a translator to a
non-RTP-based media transport, an important consideration is how to
handle different sources and their identities. This problem space is
not discussed henceforth.
Of the Transport Translators, this memo is primarily interested in
those that use RTP on both sides, and this is assumed henceforth.
The most basic Transport Translators that operate below the RTP level
were already discussed in Section 3.2.1.1.
3.2.1.3. Media Translator
Shortcut name: Topo-Media-Translator
Media Translators (Topo-Media-Translator) modify the media inside the
RTP stream. This process is commonly known as transcoding. The
modification of the media can be as small as removing parts of the
stream, and it can go all the way to a full decoding and re-encoding
(down to the sample level or equivalent) utilizing a different media
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codec. Media Translators are commonly used to connect endpoints
without a common interoperability point in the media encoding.
Stand-alone Media Translators are rare. Most commonly, a combination
of Transport and Media Translator is used to translate both the media
and the transport aspects of the RTP stream carrying the media
between two transport domains.
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 endpoint B's RTCP receiver
report before forwarding them to endpoint A. The rewriting is needed
as the RTP stream received by B is not the same RTP stream as the
other participants receive. For example, the number of packets
transmitted to B may be lower than what A sends, due to the different
media format and data rate. Therefore, if the receiver reports were
forwarded without changes, the extended highest sequence number would
indicate that B was substantially behind in reception, while it most
likely would not be. Therefore, the translator must translate that
number to a corresponding sequence number for the stream the
translator received. Similar requirements exist for most other
fields in the RTCP receiver reports.
A Media Translator may in some cases act on behalf of the "real"
source (the endpoint originally sending the media to the translator)
and respond to RTCP feedback messages. This may occur, for example,
when a receiving endpoint requests a bandwidth reduction, and the
Media Translator has not detected any congestion or other reasons for
bandwidth reduction between the sending endpoint and itself. In that
case, it is sensible that the Media Translator reacts to codec
control messages itself, for example, by transcoding to a lower media
rate.
A variant of translator behavior worth pointing out is the one
depicted in Figure 3 of an endpoint A sending an RTP stream
containing media (only) to B. On the path, there is a device T that
manipulates the RTP streams on A's behalf. One common example is
that T adds a second RTP stream containing Forward Error Correction
(FEC) information in order to protect A's (non FEC-protected) RTP
stream. In this case, T needs to semantically bind the new FEC RTP
stream to A's media-carrying RTP stream, for example, by using the
same CNAME as A.
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+------+ +------+ +------+
| | | | | |
| A |------->| T |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+
Figure 3: Media Translator Adding FEC
There may also be cases where information is added into the original
RTP stream, while leaving most or all of the original RTP packets
intact (with the exception of certain RTP header fields, such as the
sequence number). One example is the injection of metadata into the
RTP stream, carried in their own RTP packets.
Similarly, a Media Translator can sometimes remove information from
the RTP stream, while otherwise leaving the remaining RTP packets
unchanged (again with the exception of certain RTP header fields).
Either type of functionality where T manipulates the RTP stream, or
adds an accompanying RTP stream, on behalf of A is also covered under
the Media Translator definition.
3.2.2. Back-to-Back RTP sessions
Shortcut name: Topo-Back-To-Back
There exist middleboxes that interconnect two endpoints (A and B)
through themselves (MB), but not by being part of a common RTP
session. Instead, they establish two different RTP sessions: one
between A and the middlebox and another between the middlebox and B.
This topology is called Topo-Back-To-Back.
|<--Session A-->| |<--Session B-->|
+------+ +------+ +------+
| A |------->| MB |-------->| B |
+------+ +------+ +------+
Figure 4: Back-to-Back RTP Sessions through Middlebox
The middlebox acts as an application-level gateway and bridges the
two RTP sessions. This bridging can be as basic as forwarding the
RTP payloads between the sessions or more complex including media
transcoding. The difference of this topology relative to the single
RTP session context is the handling of the SSRCs and the other
session-related identifiers, such as CNAMEs. With two different RTP
sessions, these can be freely changed and it becomes the middlebox's
responsibility to maintain the correct relations.
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The signaling or other above RTP-level functionalities referencing
RTP streams may be what is most impacted by using two RTP sessions
and changing identifiers. The structure with two RTP sessions also
puts a congestion control requirement on the middlebox, because it
becomes fully responsible for the media stream it sources into each
of the sessions.
Adherence to congestion control can be solved locally on each of the
two segments or by bridging statistics from the receiving endpoint
through the middlebox to the sending endpoint. From an
implementation point, however, the latter requires dealing with a
number of inconsistencies. First, packet loss must be detected for
an RTP stream sent from A to the middlebox, and that loss must be
reported through a skipped sequence number in the RTP stream from the
middlebox to B. This coupling and the resulting inconsistencies are
conceptually easier to handle when considering the two RTP streams as
belonging to a single RTP session.
3.3. Point to Multipoint Using Multicast
Multicast is an IP-layer functionality that is available in some
networks. Two main flavors can be distinguished: Any-Source
Multicast (ASM) [RFC1112] 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) [RFC3569], where
only a particular IP host sends to the multicast group. Each of
these models are discussed below in their respective sections.
3.3.1. Any-Source Multicast (ASM)
Shortcut name: Topo-ASM (was Topo-Multicast)
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Figure 5: 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 multicast
group participant reaches all the other multicast group participants,
except for cases such as:
o packet loss, or
o when a multicast group participant does not wish to receive the
traffic for a specific multicast group and, therefore, has not
subscribed to the IP multicast group in question. This scenario
can occur, for example, where a Multimedia Session is distributed
using two or more multicast groups, and a multicast group
participant is subscribed only to a subset of these sessions.
In the above context, "traffic" encompasses both RTP and RTCP
traffic. The number of multicast group participants can vary between
one and many, as RTP and RTCP scale to very large multicast groups
(the theoretical limit of the number of participants in a single RTP
session is in the range of billions). The above can be realized
using ASM.
For feedback usage, it is useful to define a "small multicast group"
as a group where the number of multicast group participants is so low
(and other factors such as the connectivity is so good) that it
allows the participants to use early or immediate feedback, as
defined in 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 likelihood that the
threshold of the small multicast group (in terms of multicast group
participants) may be exceeded during the lifetime of a session.
RTCP feedback messages in multicast reach, like media data, every
subscriber (subject to packet losses and multicast group
subscription). Therefore, the feedback suppression mechanism
discussed in [RFC4585] is typically required. Each individual
endpoint that is a multicast group participant needs to process every
feedback message it receives, not only to determine if it is affected
or if the feedback message applies only to some other endpoint but
also to derive timing restrictions for the sending of its own
feedback messages, if any.
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3.3.2. Source-Specific Multicast (SSM)
Shortcut name: Topo-SSM
In Any-Source Multicast, any of the multicast group participants can
send to all the other multicast group participants, by sending a
packet to the multicast group. In contrast, Source-Specific
Multicast [RFC3569][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 sending endpoints
(RTP sources henceforth) (1 to M) are allowed to send media to the
SSM group. These sources send media to a dedicated Distribution
Source, which forwards the RTP streams to the multicast group on
behalf of the original RTP sources. The RTP streams reach the
receiving endpoints (receivers henceforth) (R(1) to R(n)). The
receivers' RTCP messages cannot be sent to the multicast group, as
the SSM multicast group by definition has only a single IP sender.
To support RTCP, an 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 a summary of the participants' RTCP
reports towards the whole group by forwarding the RTCP traffic to the
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Distribution Source. Figure 6 only shows a single Feedback Target
integrated in the Distribution Source, but for scalability the FT can
be distributed and each instance can have responsibility for
subgroups of the receivers. For summary reports, however, there
typically must be a single Feedback Target aggregating all the
summaries to a common message to the whole receiver group.
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, need to be accounted
for.
The aforementioned situation results in common behavior 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 signaling functions to identify the
relationships between RTP sessions.
4. Multicast applications need signaling functions to identify the
relationships between SSRCs in different RTP sessions.
All multicast configurations share a signaling requirement: all of
the endpoints need to have the same RTP and payload type
configuration. Otherwise, endpoint A could, for example, be using
payload type 97 to identify the video codec H.264, while endpoint B
would identify it as MPEG-2, with unpredictable but almost certainly
not visually pleasing results.
Security solutions for this type of group communication are also
challenging. First, the key management and the security protocol
must support group communication. Source authentication becomes more
difficult and requires specialized solutions. For more discussion on
this, please review "Options for Securing RTP Sessions" [RFC7201].
3.3.3. SSM with Local Unicast Resources
Shortcut name: Topo-SSM-RAMS
"Unicast-Based Rapid Acquisition of Multicast RTP Sessions" [RFC6285]
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 Stream
.-.-.-.> Multicast RTCP Stream
.=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages
.......> Unicast RTP Stream
Figure 7: SSM with Local Unicast Resources (RAMS)
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 requiring context
established by the decoding of prior packets) to be sent at high
speed until such time where, after the decoding of these burst-
delivered media packets, the correct media timing is established,
i.e., media packets are received within adequate buffer intervals for
this application. This is accomplished by first establishing a
unicast PtP RTP session between the Burst/Retransmission Source (BRS)
(Figure 7) and the RTP Receiver. The unicast session is used to
transmit cached packets from the multicast group at higher then
normal speed in order to synchronize the receiver to the ongoing
multicast RTP stream. Once the RTP receiver and its decoder have
caught up with the multicast session's current delivery, the receiver
switches over to receiving directly from the multicast group. In
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many deployed applications, the (still existing) PtP RTP session is
used as a repair channel, i.e., for RTP Retransmission traffic of
those packets 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 Multipoint Using Mesh
Based on the RTP session definition, it is clearly possible to have a
joint RTP session involving three or more endpoints over multiple
unicast transport flows, like the joint three-endpoint session
depicted above. In this case, A needs to send its RTP streams and
RTCP packets to both B and C over their respective transport flows.
As long as all endpoints do the same, everyone will have a joint view
of the RTP session.
This topology does not create any additional requirements beyond the
need to have multiple transport flows associated with a single RTP
session. Note that an endpoint may use a single local port to
receive all these transport flows (in which case the sending port, IP
address, or SSRC can be used to demultiplex), or it might have
separate local reception ports for each of the endpoints.
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+-A--------------------+
|+---+ |
||CAM| | +-B-----------+
|+---+ +-UDP1------| |-UDP1------+ |
| | | +-RTP1----| |-RTP1----+ | |
| V | | +-Video-| |-Video-+ | | |
|+----+ | | | |<----------------|BV1 | | | |
||ENC |----+-+-+--->AV1|---------------->| | | | |
|+----+ | | +-------| |-------+ | | |
| | | +---------| |---------+ | |
| | +-----------| |-----------+ |
| | | +-------------+
| | |
| | | +-C-----------+
| | +-UDP2------| |-UDP2------+ |
| | | +-RTP1----| |-RTP1----+ | |
| | | | +-Video-| |-Video-+ | | |
| +-------+-+-+--->AV1|---------------->| | | | |
| | | | |<----------------|CV1 | | | |
| | | +-------| |-------+ | | |
| | +---------| |---------+ | |
| +-----------| |-----------+ |
+----------------------+ +-------------+
Figure 9: A Multi-Unicast Mesh with a Joint RTP Session
Figure 9 depicts endpoint A's view of using a common RTP session when
establishing the mesh as shown in Figure 8. There is only one RTP
session (RTP1) but two transport flows (UDP1 and UDP2). The Media
Source (CAM) is encoded and transmitted over the SSRC (AV1) across
both transport layers. However, as this is a joint RTP session, the
two streams must be the same. Thus, a congestion control adaptation
needed for the paths A to B and A to C needs to use the most
restricting path's properties.
An alternative structure for establishing the above topology is to
use independent RTP sessions between each pair of peers, i.e., three
different RTP sessions. In some scenarios, the same RTP stream may
be sent from the transmitting endpoint; however, it also supports
local adaptation taking place in one or more of the RTP streams,
rendering them non-identical.
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+-A----------------------+ +-B-----------+
|+---+ | | |
||MIC| +-UDP1------| |-UDP1------+ |
|+---+ | +-RTP1----| |-RTP1----+ | |
| | +----+ | | +-Audio-| |-Audio-+ | | |
| +->|ENC1|--+-+-+--->AA1|------------->| | | | |
| | +----+ | | | |<-------------|BA1 | | | |
| | | | +-------| |-------+ | | |
| | | +---------| |---------+ | |
| | +-----------| |-----------+ |
| | ------------| |-------------|
| | | |-------------+
| | |
| | | +-C-----------+
| | | | |
| | +-UDP2------| |-UDP2------+ |
| | | +-RTP2----| |-RTP2----+ | |
| | +----+ | | +-Audio-| |-Audio-+ | | |
| +->|ENC2|--+-+-+--->AA2|------------->| | | | |
| +----+ | | | |<-------------|CA1 | | | |
| | | +-------| |-------+ | | |
| | +---------| |---------+ | |
| +-----------| |-----------+ |
+------------------------+ +-------------+
Figure 10: A Multi-Unicast Mesh with an Independent RTP Session
Let's review the topology when independent RTP sessions are used from
A's perspective in Figure 10 by considering both how the media is
handled and how the RTP sessions are set up in Figure 10. A's
microphone is captured and the audio is fed into two different
encoder instances, each with a different independent RTP session,
i.e., RTP1 and RTP2, respectively. The SSRCs (AA1 and AA2) in each
RTP session are completely independent, and the media bitrate
produced by the encoders can also be tuned differently to address any
congestion control requirements differing for the paths A to B
compared to A to C.
From a topologies viewpoint, an important difference exists in the
behavior around RTCP. First, when a single RTP session spans all
three endpoints A, B, and C, and their connecting RTP streams, a
common RTCP bandwidth is calculated and used for this single joint
session. In contrast, when there are multiple independent RTP
sessions, each RTP session has its local RTCP bandwidth allocation.
Further, when multiple sessions are used, endpoints not directly
involved in a session do not have any awareness of the conditions in
those sessions. For example, in the case of the three-endpoint
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configuration in Figure 8, endpoint A has no awareness of the
conditions occurring in the session between endpoints B and C
(whereas if a single RTP session were used, it would have such
awareness).
Loop detection is also affected. With independent RTP sessions, the
SSRC/CSRC cannot be used to determine when an 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, their avoidance, has to be achieved by other
means, for example, through signaling or the use of an RTP external
namespace 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 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 |
+---+ / \ | | +---+
+ Multicast +->| Translator |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+
+-----+
Figure 11: Point to Multipoint Using Multicast
Figure 11 depicts an example of a Transport Translator performing at
least IP address translation. It allows the (non-multicast-capable)
endpoints B and D to take part in an Any-Source Multicast session
involving endpoints A and C, by having the translator forward their
unicast traffic to the multicast addresses in use, and vice versa.
It must also forward B's traffic to D, and vice versa, to provide
both B and D with a complete view of the session.
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+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 12: RTP Translator (Relay) with Only Unicast Paths
Another translator scenario is depicted in Figure 12. The translator
in this case connects multiple endpoints 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 endpoints. 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 to
those described in the ASM RTP topology. It also puts some
additional signaling requirements onto the session establishment; for
example, a common configuration of RTP payload types is required.
Transport Translators and relays should always consider implementing
source address filtering, to prevent attackers from using the
listening ports on the translator to inject traffic. The translator
can, however, go one step further, especially if explicit SSRC
signaling is used, to prevent endpoints from sending SSRCs other than
its own (that are, for example, used by other participants in the
session). This can improve the security properties of the session,
despite the use of group keys that on a cryptographic level allows
anyone to impersonate another in the same RTP session.
A translator that doesn't change the RTP/RTCP packet content can be
operated without requiring it to have access to the security contexts
used to protect the RTP/RTCP traffic between the participants.
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 more of an enabler, or facilitator, that ensures a given endpoint
or a defined subset of endpoints can participate in the session.
If endpoint B in Figure 11 were behind a limited network path, the
translator may perform media transcoding to allow the traffic
received from the other endpoints to reach B without overloading the
path. This transcoding can help the other endpoints in the multicast
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part of the session, by not requiring the quality transmitted by A to
be lowered to the bitrates that B is actually capable of receiving
(and vice versa).
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 that are
part of a session by generating one or more new RTP streams and, in
most cases, by manipulating the media data. One common application
for a mixer is to allow a participant to receive a session with a
reduced amount of resources.
+-----+
+---+ / \ +-----------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ cast +->| Mixer |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +-----------+ +---+
+-----+
Figure 13: Point to Multipoint Using the RFC 3550 Mixer Model
A mixer can be viewed as a device terminating the RTP streams
received from other endpoints in the same RTP session. Using the
media data carried in the received RTP streams, a mixer generates
derived RTP streams that are sent to the receiving endpoints.
The content that the mixer provides is the mixed aggregate of what
the mixer receives over the PtP or PtM paths, which are part of the
same Communication Session.
The mixer creates the Media Source and the source RTP stream just
like an endpoint, as it mixes the content (often in the uncompressed
domain) and then encodes and packetizes it for transmission to a
receiving endpoint. The CSRC Count (CC) and CSRC fields in the RTP
header can be used to indicate the contributors to the newly
generated RTP stream. The SSRCs of the to-be-mixed streams on the
mixer input appear as the CSRCs at the mixer output. That output
stream uses a unique SSRC that identifies the mixer's stream. The
CSRC should be forwarded between the different endpoints to allow for
loop detection and identification of sources that are part of the
Communication Session. Note that Section 7.1 of RFC 3550 requires
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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 an RTP receiver and should therefore send RTCP
receiver reports for the RTP streams it receives and terminates. In
its role as an RTP sender, it should also generate RTCP sender
reports for those RTP streams it sends. As specified in Section 7.3
of RFC 3550, a mixer must not forward RTCP unaltered between the two
domains.
The mixer depicted in Figure 13 is involved in three domains that
need to be separated: the Any-Source Multicast network (including
endpoints A and C), endpoint B, and endpoint D. Assuming all four
endpoints in the conference are interested in receiving content from
all other endpoints, the mixer produces different mixed RTP streams
for B and D, as the one to B may contain content received from D, and
vice versa. However, the mixer may only need one SSRC per media type
in each domain where it is the receiving entity and transmitter of
mixed content.
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.
The mixing operation, however, 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
endpoints in the other domain(s). In other cases, a message is
handled by the mixer locally and therefore not forwarded to any other
domain.
When replacing the multicast network in Figure 13 (to the left of the
mixer) with individual unicast paths as depicted in Figure 14, 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.
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+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 14: RTP Mixer with Only Unicast Paths
We now discuss in more detail the different mixing operations that a
mixer can perform and how they can affect RTP and RTCP behavior.
3.6.1. Media-Mixing Mixer
The Media-Mixing Mixer is likely the one that most think of when they
hear the term "mixer". Its basic mode of operation is that it
receives RTP streams from several endpoints and selects the stream(s)
to be included in a media-domain mix. The selection can be through
static configuration or by dynamic, content-dependent means such as
voice activation. The mixer then creates a single outgoing RTP
stream from this mix.
The most commonly deployed Media-Mixing Mixer is probably the audio
mixer, used in voice conferencing, where the output consists of a
mixture of all the input audio signals; this needs minimal signaling
to be successfully set up. From a signal processing viewpoint, audio
mixing is relatively straightforward and commonly possible for a
reasonable number of endpoints. Assume, for example, that one wants
to mix N streams from N different endpoints. The mixer needs to
decode those N streams, typically into the sample domain, and then
produce N or N+1 mixes. Different mixes are needed so that each
endpoint gets a mix of all other sources except its own, as this
would result in an echo. When N is lower than the number of all
endpoints, one may produce a mix of all N streams for the group that
are currently not included in the mix; thus, N+1 mixes. These audio
streams are then encoded again, RTP packetized, and sent out. In
many cases, audio level normalization, noise suppression, and similar
signal processing steps are also required or desirable before the
actual mixing process commences.
In video, the term "mixing" has a different interpretation than
audio. It is commonly used to refer to the process of spatially
combining contributed video streams, which is also known as "tiling".
The reconstructed, appropriately scaled down videos can be spatially
arranged in a set of tiles, with each tile containing the video from
an endpoint (typically showing a human participant). Tiles can be of
different sizes so that, for example, a particularly important
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participant, or the loudest speaker, is being shown in a larger tile
than other participants. A self-view picture can be included in the
tiling, which can be either locally produced or feedback from a
mixer-received and reconstructed video image. Such remote loopback
allows for confidence monitoring, i.e., it enables the participant to
see himself/herself in the same quality as other participants see
him/her. The tiling normally operates on reconstructed video in the
sample domain. The tiled image is encoded, packetized, and sent by
the mixer to the receiving endpoints. 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 arrangements where an
individual mixer is available for each outgoing port of the
middlebox, allowing individual compositions for each receiving
endpoint (a feature commonly referred to as personalized layout).
One problem with media mixing is that it consumes both large amounts
of media processing resources (for the decoding and mixing process in
the uncompressed domain) and encoding resources (for the encoding of
the mixed signal). Another problem is the quality degradation
created by decoding and re-encoding the media, which is the result of
the lossy nature of the most commonly used media codecs. A third
problem is the latency introduced by the media mixing, which can be
substantial and annoyingly noticeable in case of video, or in case of
audio if that mixed audio is lip-synchronized with high-latency
video. The advantage of media mixing is that it is straightforward
for the endpoints to handle the single media stream (which includes
the mixed aggregate of many sources), as they don't need to handle
multiple decodings, local mixing, and composition. In fact, mixers
were introduced in pre-RTP times so that legacy, single stream
receiving endpoints (that, in some protocol environments, actually
didn't need to be aware of the multipoint nature of the conference)
could successfully participate in what a user would recognize as a
multiparty video conference.
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+-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 <----+ | +---+ | | |
| | +-------| |(AA1+CA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+C | |
+-----------+ |-----------+ +---+ | | |
| | M | |
+-C---------+ | | I | |
| +-RTP3----| |-RTP3------+ | X | |
| | +-Audio-| |-Audio---+ | +---+ | E | |
| | | CA1|--------->|---------+-+-|DEC|->| R | |
| | | |<---------|MA3 <----+ | +---+ | | |
| | +-------| |(AA1+BA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+B | |
+-----------+ |-----------+ +---+ +-----+ |
+----------------------------+
Figure 15: Session and SSRC Details for Media Mixer
From an RTP perspective, media mixing can be a very simple process,
as can be seen in Figure 15. The mixer presents one SSRC towards the
receiving endpoint, e.g., MA1 to Peer A, where the associated stream
is the media mix of the other endpoints. As each peer, in this
example, receives a different version of a mix from the mixer, there
is no actual relation between the different RTP sessions in terms of
actual media or transport-level information. There are, however,
common relationships between RTP1-RTP3, namely SSRC space and
identity information. When A receives the MA1 stream, which is a
combination of BA1 and CA1 streams, the mixer may include CSRC
information in the MA1 stream to identify the Contributing Sources
BA1 and CA1, allowing the receiver to identify the Contributing
Sources even if this were not possible through the media itself or
through other signaling means.
The CSRC has, in turn, utility in RTP extensions, like the RTP header
extension for Mixer-to-Client Audio Level Indication [RFC6465]. If
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the SSRCs from the endpoint to mixer paths are used as CSRCs in
another RTP session, then RTP1, RTP2, and RTP3 become one joint
session as they have a common SSRC space. At this stage, the mixer
also needs to consider which RTCP information it needs to expose in
the different paths. In the above scenario, a mixer would normally
expose nothing more than the SDES information and RTCP BYE for a CSRC
leaving the session. The main goal would be to enable the correct
binding against the application logic and other information sources.
This also enables loop detection in the RTP session.
3.6.2. Media-Switching Mixer
Media-Switching Mixers are used in limited functionality scenarios
where no, or only very limited, concurrent presentation of multiple
sources is required by the application and also in more complex
multi-stream usages with receiver mixing or tiling, including
combined with simulcast and/or scalability between source and mixer.
An RTP mixer based on media switching avoids the media decoding and
encoding operations in the mixer, as it conceptually forwards the
encoded media stream as it was being sent to the mixer. It does not
avoid, however, the decryption and re-encryption cycle as it rewrites
RTP headers. Forwarding media (in contrast to reconstructing-mixing-
encoding media) reduces the amount of computational resources needed
in the mixer and increases the media quality (both in terms of
fidelity and reduced latency).
A Media-Switching Mixer maintains a pool of SSRCs representing
conceptual or functional RTP streams that the mixer can produce.
These RTP streams are created by selecting media from one of the RTP
streams received by the mixer and forwarded to the peer using the
mixer's own SSRCs. The mixer can switch between available sources if
that is required by the concept for the source, like the currently
active speaker. Note that the mixer, in most cases, still needs to
perform a certain amount of media processing, as many media formats
do not allow to "tune into" the stream at arbitrary points in their
bitstream.
To achieve a coherent RTP stream from the mixer's SSRC, the mixer
needs to rewrite the incoming RTP packet's header. First, the SSRC
field must be set to the value of the mixer's SSRC. Second, the
sequence number must be the next in the sequence of outgoing packets
it sent. Third, the RTP timestamp value needs to be adjusted using
an offset that changes each time one switches the Media Source.
Finally, depending on the negotiation of the RTP payload type, the
value representing this particular RTP payload configuration may have
to be changed if the different endpoint-to-mixer paths have not
arrived on the same numbering for a given configuration. This also
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requires that the different endpoints support a common set of codecs,
otherwise media transcoding for codec compatibility would still be
required.
We now consider the operation of a Media-Switching Mixer that
supports a video conference with six participating endpoints (A-F)
where the two most recent speakers in the conference are shown to
each receiving endpoint. 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---+ | | | |
| | | FV1|------------>|---------+-+------->| | |
| | | |<------------|MV11 <---+-+-AV1----| | |
| | | |<------------|MV12 <---+-+-EV1----| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +----------------------------+
Figure 16: Media-Switching RTP Mixer
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The Media-Switching Mixer can, similarly to the Media-Mixing Mixer,
reduce the bitrate required for media transmission towards the
different peers by selecting and forwarding only a subset of RTP
streams it receives from the sending endpoints. In case the mixer
receives simulcast transmissions or a scalable encoding of the Media
Source, the mixer has more degrees of freedom to select streams or
subsets of streams to forward to a receiving endpoint, both based on
transport or endpoint restrictions as well as application logic.
To ensure that a media receiver in an endpoint can correctly decode
the media in the RTP stream after a switch, a codec that uses
temporal prediction needs to start its decoding from independent
refresh points, or points in the bitstream offering similar
functionality (like "dirty refresh points"). For some codecs, for
example, frame-based speech and audio codecs, this is easily achieved
by starting the decoding at RTP packet boundaries, as each packet
boundary provides a refresh point (assuming proper packetization on
the encoder side). For other codecs, particularly in video, refresh
points are less common in the bitstream or may not be present at all
without an explicit request to the respective encoder. The Full
Intra Request [RFC5104] RTCP codec control message has been defined
for this purpose.
In this type of mixer, one could consider fully terminating the RTP
sessions between the different endpoint and mixer paths. The same
arguments and considerations as discussed in Section 3.9 need to be
taken into consideration and apply here.
3.7. Selective Forwarding Middlebox
Another method for handling media in the RTP mixer is to "project",
or make available, all potential RTP sources (SSRCs) into a per-
endpoint, independent RTP session. The middlebox can select which of
the potential sources that are currently actively transmitting media
will be sent to each of the endpoints. 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 17: Selective Forwarding Middlebox
In the six endpoint conference depicted above (in Figure 17), one can
see that endpoint A is aware of five incoming SSRCs, BV1-FV1. If
this middlebox intends to have a similar behavior as in Section 3.6.2
where the mixer provides the endpoints with the two latest speaking
endpoints, 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 endpoints, each RTP
stream requires the rewriting of certain RTP header fields when being
projected from one session into another. In particular, the sequence
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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
an RTP session. The timestamp (possibly offset) stays the same.
The RTP sessions can be considered independent, resulting in that the
SSRC numbers used can also be handled independently. This simplifies
the SSRC collision detection and avoidance but requires tools such as
remapping tables between the RTP sessions. Using independent RTP
sessions is not required, as it is possible for the switching
behavior to also perform with a common SSRC space. However, in this
case, collision detection and handling becomes a different problem.
It is up to the implementation to use a single common SSRC space or
separate ones.
Using separate SSRC spaces has some implications. For example, the
RTP stream that is being sent by endpoint B to the middlebox (BV1)
may use an SSRC value of 12345678. When that RTP 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 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 the RTP level. Instead, this responsibility falls onto higher
layers.
The middlebox is also responsible for receiving any RTCP codec
control requests coming from an endpoint and deciding if it can act
on the request locally or needs to translate the request into the RTP
session/transport leg that contains the Media Source. Both endpoints
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 that switching points
be provided between the sources and Temporary Maximum Media Bitrate
Request (TMMBR) to enable the middlebox to aggregate congestion
control responses towards the Media Source so to enable it to adjust
its bitrate (obviously, only in case the limitation is not in the
source to middlebox link).
The Selective Forwarding Middlebox has been introduced in recently
developed videoconferencing systems in conjunction with, and to
capitalize on, scalable video coding as well as simulcasting. An
example of scalable video coding is Annex G of H.264, but other
codecs, including H.264 AVC and VP8, also exhibit scalability, albeit
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only in the temporal dimension. In both scalable coding and
simulcast cases, the video signal is represented by a set of two or
more bitstreams, providing a corresponding number of distinct
fidelity points. The middlebox selects which parts of a scalable
bitstream (or which bitstream, in the case of simulcasting) to
forward to each of the receiving endpoints. The decision may be
driven by a number of factors, such as available bitrate, desired
layout, etc. Contrary to transcoding MCUs, SFMs have extremely low
delay and provide features that are typically associated with high-
end systems (personalized layout, error localization) without any
signal processing at the middlebox. They are also capable of scaling
to a large number of concurrent users, and--due to their very low
delay--can also be cascaded.
This version of the middlebox also puts different requirements on the
endpoint when it comes to decoder instances and handling of the RTP
streams providing media. As each projected SSRC can, at any time,
provide media, the endpoint 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 that this topology could potentially be seen as a Media
Translator that includes an on/off logic as part of its media
translation. The topology has the property that all SSRCs present in
the session are visible to an endpoint. It also has mixer aspects,
as the streams it provides are not basically translated versions, but
instead they have conceptual property assigned to them and can be
both turned on/off as well as fully or partially delivered. Thus,
this topology appears to be some hybrid between the translator and
mixer model.
The differences between a Selective Forwarding Middlebox and a
Switching-Media Mixer (Section 3.6.2) are minor, and they share most
properties. The above requirement on having a large number of
decoding instances or requiring efficient switching of decoder
contexts, are one point of difference. The other is how the
identification is performed, where the mixer uses CSRC to provide
information on what is included in a particular RTP stream that
represents a particular concept. Selective forwarding gets the
source information through the SSRC and instead uses other mechanisms
to indicate the streams intended usage, if needed.
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3.8. Point to Multipoint Using Video-Switching MCUs
Shortcut name: Topo-Video-switch-MCU
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
Figure 18: Point to Multipoint Using a Video-Switching MCU
This PtM topology was popular in early implementations of multipoint
videoconferencing systems due to its simplicity, and the
corresponding middlebox design has been known as a "video-switching
MCU". The more complex RTCP-terminating MCUs, discussed in the next
section, became the norm, however, when technology allowed
implementations at acceptable costs.
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 bitrate, 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 endpoints. 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 RTP stream results in most participants seeing only a
subset of the sent RTP streams at any given time, often a single RTP
stream per conference. Therefore, RTCP receiver reports only report
on these RTP streams. Consequently, the endpoints emitting RTP
streams that are not currently forwarded receive a view of the
session that indicates their RTP streams disappear somewhere en
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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 RTP stream under its own SSRC and with the
appropriate CSRC values. Second, the MCU needs to modify the RTCP
RRs it forwards between the domains. As a result, it is recommended
that one implement a centralized video-switching conference using a
mixer according to RFC 3550, instead of the shortcut implementation
described here.
3.9. Point to Multipoint Using RTCP-Terminating MCU
Shortcut name: Topo-RTCP-terminating-MCU
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 19: Point to Multipoint Using Content Modifying MCUs
In this PtM scenario, each endpoint 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 endpoints or
b. the mixed aggregate of what the MCU receives from the other PtP
paths, which are part of the same Communication Session.
In case (a), the MCU may modify the content in terms of bitrate,
encoding format, or resolution. No explicit RTP mechanism is used to
establish the relationship between the original RTP stream of the
media being sent and the RTP stream the MCU sends. In other words,
the outgoing RTP streams typically use a different SSRC, and may well
use a different payload type (PT), even if this different PT happens
to be mapped to the same media type. This is a result of the
individually negotiated RTP session for each endpoint.
In case (b), the MCU is the Media Source and generates the Source RTP
Stream as it mixes the received content and then encodes and
packetizes it for transmission to an endpoint. According to RTP
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[RFC3550], the SSRC of the contributors are to be signaled using the
CSRC/CC mechanism. In practice, today, most deployed MCUs do not
implement this feature. Instead, the identification of the endpoints
whose content is included in the mixer's output is not indicated
through any explicit RTP mechanism. That is, most deployed MCUs set
the CC field in the RTP header to zero, thereby indicating no
available CSRC information, even if they could identify the original
sending endpoints 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 many/most deployed MCUs (and video conferencing endpoints)
rely on signaling-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.
3.10. Split Component Terminal
Shortcut name: Topo-Split-Terminal
In some applications, for example, in some telepresence systems,
terminals may not be integrated into a single functional unit but
composed of more than one subunits. For example, a telepresence room
terminal employing multiple cameras and monitors may consist of
multiple video conferencing subunits, each capable of handling a
single camera and monitor. Another example would be a video
conferencing terminal in which audio is handled by one subunit, and
video by another. Each of these subunits uses its own physical
network interface (for example: Ethernet jack) and network address.
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The various (media processing) subunits need (logically and
physically) to be interconnected by control functionality, but their
media plane functionality may be split. These types of terminals are
referred to as split component terminals. Historically, the earliest
split component terminals were perhaps the independent audio and
video conference software tools used over the MBONE in the late
1990s.
An example for such a split component terminal is depicted in
Figure 20. Within split component terminal A, at least audio and
video subunits are addressed by their own network addresses. In some
of these systems, the control stack subunit may also have its own
network address.
From an RTP viewpoint, each of the subunits terminates RTP and acts
as an endpoint in the sense that each subunit includes its own,
independent RTP stack. However, as the subunits are semantically
part of the same terminal, it is appropriate that this semantic
relationship is expressed in RTCP protocol elements, namely in the
CNAME.
+---------------------+
| Endpoint A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+-RTP---\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+-RTP-------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+-SIP---/
| +------------+ |
+---------------------+
Figure 20: Split Component Terminal
It is further sensible that the subunits share a common clock from
which RTP and RTCP clocks are derived, to facilitate synchronization
and avoid clock drift.
To indicate that audio and video Source Streams generated by
different subunits share a common clock, and can be synchronized, the
RTP streams generated from those Source Streams need to include the
same CNAME in their RTCP SDES packets. The use of a common CNAME for
RTP flows carried in different transport-layer flows is entirely
normal for RTP and RTCP senders, and fully compliant RTP endpoints,
middleboxes, and other tools should have no problem with this.
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However, outside of the split component terminal scenario (and
perhaps a multihomed endpoint scenario, which is not further
discussed herein), the use of a common CNAME in RTP streams sent from
separate endpoints (as opposed to a common CNAME for RTP streams sent
on different transport-layer flows between two endpoints) is rare.
It has been reported that at least some third-party tools like some
network monitors do not handle gracefully endpoints that use a common
CNAME across multiple transport-layer flows: they report an error
condition in which two separate endpoints are using the same CNAME.
Depending on the sophistication of the support staff, such erroneous
reports can lead to support issues.
The aforementioned support issue can sometimes be avoided if each of
the subunits of a split component terminal is configured to use a
different CNAME, with the synchronization between the RTP streams
being indicated by some non-RTP signaling channel rather than using a
common CNAME sent in RTCP. This complicates the signaling,
especially in cases where there are multiple SSRCs in use with
complex synchronization requirements, as is the same in many current
telepresence systems. Unless one uses RTCP terminating topologies
such as Topo-RTCP-terminating-MCU, sessions involving more than one
video subunit with a common CNAME are close to unavoidable.
The different RTP streams comprising a split terminal system can form
a single RTP session or they can form multiple RTP sessions,
depending on the visibility of their SSRC values in RTCP reports. If
the receiver of the RTP streams sent by the split terminal sends
reports relating to all of the RTP flows (i.e., to each SSRC) in each
RTCP report, then a single RTP session is formed. Alternatively, if
the receiver of the RTP streams sent by the split terminal does not
send cross-reports in RTCP, then the audio and video form separate
RTP sessions.
For example, in Figure 20, B will send RTCP reports to each of the
subunits of A. If the RTCP packets that B sends to the audio subunit
of A include reports on the reception quality of the video as well as
the audio, and similarly if the RTCP packets that B sends to the
video subunit of A include reports on the reception quality of the
audio as well as video, then a single RTP session is formed.
However, if the RTCP packets B sends to the audio subunit of A only
report on the received audio, and the RTCP packets B sends to the
video subunit of A only report on the received video, then there are
two separate RTP sessions.
Forming a single RTP session across the RTP streams sent by the
different subunits of a split terminal gives each subunit visibility
into reception quality of RTP streams sent by the other subunits.
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This information can help diagnose reception quality problems, but at
the cost of increased RTCP bandwidth use.
RTP streams sent by the subunits of a split terminal need to use the
same CNAME in their RTCP packets if they are to be synchronized,
irrespective of whether a single RTP session is formed or not.
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 13,
where the mixer does not need to mix in the direction from B or D
towards the multicast domains with A and C. Instead, the RTP streams
from B and D are forwarded without changes. Avoiding this mixing
would save media processing resources that perform the mixing in
cases where it isn't needed. However, there would still be a need to
mix B's media towards D. Only in the direction B -> multicast domain
or D -> multicast domain would it be possible to work as a
translator. In all other directions, it would function as a mixer.
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's and C's RTP streams would be missing. To avoid A and
C thinking that B and D aren't receiving A and C at all, the mixer
needs to insert locally generated reports reflecting the situation
for the streams from A and C into B's and D's sender reports. In the
opposite direction, the receiver reports from A and C about B's and
D's streams 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 for implementation.
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 SDES information for both the CSRCs and
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the SSRCs. Thus, in a mixed domain, the only SSRCs seen will be the
ones present in the domain, while there can be CSRCs from all the
domains connected together with a combination of mixers and
translators. The combined SSRC and CSRC space is common over any
translator or mixer. It is important to facilitate loop detection,
something that is likely to be even more important in combined
topologies due to the mixed behavior between the domains. Any
hybrid, like the Topo-Video-switch-MCU or Topo-Asymmetric, requires
considerable thought on how RTCP is dealt with.
4. Topology Properties
The topologies discussed in Section 3 have different properties.
This section describes these properties. Note that, even if a
certain property is supported within a particular topology concept,
the necessary functionality may be optional to implement.
4.1. All-to-All Media Transmission
To recapitulate, multicast, and in particular ASM, provides the
functionality that everyone may send to, or receive from, everyone
else within the session. SSM can provide a similar functionality by
having anyone intending to participate as a sender to send its media
to the SSM Distribution Source. The SSM Distribution Source forwards
the media to all receivers subscribed to the multicast group. Mesh,
MCUs, mixers, Selective Forwarding Middleboxes (SFMs), and
translators may all provide that functionality at least on some basic
level. However, there are some differences in which type of
reachability they provide.
The topologies that come closest to emulating Any-Source IP
Multicast, with all-to-all transmission capabilities, are the
Transport Translator function called "relay" in Section 3.5, as well
as the Mesh with joint RTP sessions (Section 3.4). Media
Translators, Mesh with independent RTP Sessions, mixers, SFUs, and
the MCU variants do not provide a fully meshed forwarding on the
transport level; instead, they only allow limited forwarding of
content from the other session participants.
The "all-to-all media transmission" requires that any media
transmitting endpoint considers the path to the least-capable
receiving endpoint. Otherwise, the media transmissions may overload
that path. Therefore, a sending endpoint needs to monitor the path
from itself to any of the receiving endpoints, to detect the
currently least-capable receiver and adapt its sending rate
accordingly. As multiple endpoints may send simultaneously, the
available resources may vary. RTCP's receiver reports help perform
this monitoring, at least on a medium time scale.
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The resource consumption for performing all-to-all transmission
varies depending on the topology. Both ASM and SSM have the benefit
that only one copy of each packet traverses a particular link. Using
a relay causes the transmission of one copy of a packet per
endpoint-to-relay path and packet transmitted. However, in most
cases, the links carrying the multiple copies will be the ones close
to the relay (which can be assumed to be part of the network
infrastructure with good connectivity to the backbone) rather than
the endpoints (which may be behind slower access links). The Mesh
topologies causes N-1 streams of transmitted packets to traverse the
first-hop link from the endpoint, in a mesh with N endpoints. How
long the different paths are common is highly situation dependent.
The transmission of RTCP by design 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.
4.2. Transport or Media Interoperability
All translators, mixers, RTCP-terminating MCUs, and Mesh with
individual RTP sessions allow changing the media encoding or the
transport to other properties of the other domain, thereby providing
extended interoperability in cases where the endpoints lack a common
set of media codecs and/or transport protocols. Selective Forwarding
Middleboxes can adopt the transport and (at least) selectively
forward the encoded streams that match a receiving endpoint's
capability. It requires an additional translator to change the media
encoding if the encoded streams do not match the receiving endpoint's
capabilities.
4.3. Per-Domain Bitrate Adaptation
Endpoints are often connected to each other with a heterogeneous set
of paths. This makes congestion control in a Point-to-Multipoint set
problematic. In the ASM, SSM, Mesh with common RTP session, and
Transport Relay scenarios, each individual sending endpoint has to
adapt to the receiving endpoint behind the least-capable path,
yielding suboptimal quality for the endpoints behind the more capable
paths. This is no longer an issue when Media Translators, mixers,
SFMs, or MCUs are involved, as each endpoint only needs to adapt to
the slowest path within its own domain. The translator, mixer, SFM,
or MCU topologies all require their respective outgoing RTP streams
to adjust the bitrate, 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 endpoint in all the domains
at the cost (complexity, delay, equipment) of the mixer, SFM, or
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translator, and potentially the media sender (multicast/layered
encoding and sending the different representations).
4.4. Aggregation of Media
In the all-to-all media property mentioned above and provided by ASM,
SSM, Mesh with common RTP session, and relay, all simultaneous media
transmissions share the available bitrate. For endpoints with
limited reception capabilities, this may result in a situation where
even a minimal, acceptable media quality cannot be accomplished,
because multiple RTP streams need to share the same resources. One
solution to this problem is to use a mixer, or MCU, to aggregate the
multiple RTP streams into a single one, where the single RTP stream
takes up less resources in terms of bitrate. This aggregation can be
performed according to different methods. Mixing or selection are
two common methods. Selection is almost always possible and easy to
implement. Mixing requires resources in the mixer and may be
relatively easy and not impair the quality too badly (audio) or quite
difficult (video tiling, which is not only computationally complex
but also reduces the pixel count per stream, with corresponding loss
in perceptual quality).
4.5. 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 SDES. In topologies that
provide a full all-to-all functionality, i.e., ASM, Mesh with common
RTP session, and relay, a compliant RTP implementation offers the
functionality directly as specified in RTP. In topologies that do
not offer all-to-all communication, it is necessary that RTCP is
handled correctly in domain bridging functions. RTP includes
explicit specification text for translators and mixers, and for SFMs
the required functionality can be derived from that text. However,
the MCU described in Section 3.8 cannot offer the full functionality
for session participant identification through RTP means. The
topologies that create independent RTP sessions per endpoint or pair
of endpoints, like a Back-to-Back RTP session, MESH with independent
RTP sessions, and the RTCP terminating MCU (Section 3.9), with an
exception of SFM, do not support RTP-based identification of session
participants. In all those cases, other non-RTP-based mechanisms
need to be implemented if such knowledge is required or desirable.
When it comes to SFM, the SSRC namespace is not necessarily joint.
Instead, identification will require knowledge of SSRC/CSRC mappings
that the SFM performed; see Section 3.7.
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4.6. Loop Detection
In complex topologies with multiple interconnected domains, it is
possible to unintentionally form media loops. RTP and RTCP support
detecting such loops, as long as the SSRC and CSRC identities are
maintained and correctly set in forwarded packets. Loop detection
will work in ASM, SSM, Mesh with joint RTP session, and relay. It is
likely that loop detection works for the video-switching MCU,
Section 3.8, at least as long as it forwards the RTCP between the
endpoints. However, the Back-to-Back RTP sessions, Mesh with
independent RTP sessions, and SFMs will definitely break the loop
detection mechanism.
4.7. Consistency between Header Extensions and RTCP
Some RTP header extensions have relevance not only end to end but
also hop to hop, meaning at least some of the middleboxes in the path
are aware of their potential presence through signaling, intercept
and interpret such header extensions, and potentially also rewrite or
generate them. Modern header extensions generally follow "A General
Mechanism for RTP Header Extensions" [RFC5285], which allows for all
of the above. Examples for such header extensions include the Media
ID (MID) in [SDP-BUNDLE]. At the time of writing, there was also a
proposal for how to include some SDES into an RTP header extension
[RTCP-SDES].
When such header extensions are in use, any middlebox that
understands it must ensure consistency between the extensions it sees
and/or generates and the RTCP it receives and generates. For
example, the MID of the bundle is sent in an RTP header extension and
also in an RTCP SDES message. This apparent redundancy was
introduced as unaware middleboxes may choose to discard RTP header
extensions. Obviously, inconsistency between the MID sent in the RTP
header extension and in the RTCP SDES message could lead to
undesirable results, and, therefore, consistency is needed.
Middleboxes unaware of the nature of a header extension, as specified
in [RFC5285], are free to forward or discard header extensions.
5. 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-ASM (ASM), Topo-SSM (SSM), Topo-Trn-
Translator (TT), Topo-Media-Translator (including Transport
Translator) (MT), Topo-Mesh with joint session (MJS), Topo-Mesh with
individual sessions (MIS), Topo-Mixer (Mix), Topo-Asymmetric (ASY),
Topo-Video-switch-MCU (VSM), Topo-RTCP-terminating-MCU (RTM), and
Selective Forwarding Middlebox (SFM). In the table below, Y
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indicates Yes or full support, N indicates No support, (Y) indicates
partial support, and N/A indicates not applicable.
Property PtP ASM SSM TT MT MJS MIS Mix ASY VSM RTM SFM
---------------------------------------------------------------------
All-to-All Media N Y (Y) Y Y Y (Y) (Y) (Y) (Y) (Y) (Y)
Interoperability N/A N N Y Y Y Y Y Y N Y Y
Per-Domain Adaptation N/A N N N Y N Y Y Y N Y Y
Aggregation of Media N N N N N N N Y (Y) Y Y N
Full Session View Y Y Y Y Y Y N Y Y (Y) N Y
Loop Detection Y Y Y Y Y Y N Y Y (Y) N N
Please note that the Media Translator also includes the Transport
Translator functionality.
6. Security Considerations
The use of mixers, SFMs, and translators has impact on security and
the security functions used. The primary issue is that mixers, SFMs,
and translators modify packets, thus preventing the use of integrity
and source authentication, unless they are trusted devices that take
part in the security context, e.g., the device can send Secure Real-
time Transport Protocol (SRTP) and Secure Real-time Transport Control
Protocol (SRTCP) [RFC3711] packets to endpoints in the Communication
Session. If encryption is employed, the Media Translator, SFM, and
mixer need to be able to decrypt the media to perform its function.
A Transport Translator may be used without access to the encrypted
payload in cases where it translates parts that are not included in
the encryption and integrity protection, for example, IP address and
UDP port numbers in a media stream using SRTP [RFC3711]. However, in
general, the translator, SFM, or mixer needs to be part of the
signaling context and get the necessary security associations (e.g.,
SRTP crypto contexts) established with its RTP session participants.
Including the mixer, SFM, and translator in the security context
allows the entity, if subverted or misbehaving, to perform a number
of very serious attacks as it has full access. It can perform all
the attacks possible (see RFC 3550 and any applicable profiles) as if
the media session were not protected at all, while giving the
impression to the human session participants that they are protected.
Transport Translators have no interactions with cryptography that
work 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.
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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, SFMs, and mixers, it
is possible that the translator, SFM, or mixer could create different
security associations for the different domains they are working in.
Doing so has some implications:
First, it might weaken security if the mixer/translator accepts a
weaker algorithm or key in one domain rather 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, SFMs, 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 11,
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.
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The ASM, SSM, Relay, and Mesh (with common RTP session) topologies
each have multiple endpoints that require shared knowledge about the
different crypto contexts for the endpoints. These multiparty
topologies have special requirements on the key management as well as
the security functions. Specifically, source authentication in these
environments has special requirements.
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 [RFC7201] to get a
better understanding of most of the available options.
7. References
7.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[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,
DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Grouping Semantics and
Mechanisms for Real-Time Transport Protocol (RTP)
Sources", RFC 7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>.
7.2. Informative References
[MULTI-STREAM-OPT]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback",
Work in Progress, draft-ietf-avtcore-rtp-multi-stream-
optimisation-08, October 2015.
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RFC 7667 RTP Topologies November 2015
[RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5,
RFC 1112, DOI 10.17487/RFC1112, August 1989,
<http://www.rfc-editor.org/info/rfc1112>.
[RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network
Address Translator (Traditional NAT)", RFC 3022,
DOI 10.17487/RFC3022, January 2001,
<http://www.rfc-editor.org/info/rfc3022>.
[RFC3569] Bhattacharyya, S., Ed., "An Overview of Source-Specific
Multicast (SSM)", RFC 3569, DOI 10.17487/RFC3569, July
2003, <http://www.rfc-editor.org/info/rfc3569>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, DOI 10.17487/RFC4575, August
2006, <http://www.rfc-editor.org/info/rfc4575>.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for
IP", RFC 4607, DOI 10.17487/RFC4607, August 2006,
<http://www.rfc-editor.org/info/rfc4607>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
DOI 10.17487/RFC5117, January 2008,
<http://www.rfc-editor.org/info/rfc5117>.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
2008, <http://www.rfc-editor.org/info/rfc5285>.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760,
DOI 10.17487/RFC5760, February 2010,
<http://www.rfc-editor.org/info/rfc5760>.
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RFC 7667 RTP Topologies November 2015
[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,
DOI 10.17487/RFC5766, April 2010,
<http://www.rfc-editor.org/info/rfc5766>.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, DOI 10.17487/RFC6285, June 2011,
<http://www.rfc-editor.org/info/rfc6285>.
[RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
time Transport Protocol (RTP) Header Extension for Mixer-
to-Client Audio Level Indication", RFC 6465,
DOI 10.17487/RFC6465, December 2011,
<http://www.rfc-editor.org/info/rfc6465>.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
<http://www.rfc-editor.org/info/rfc7201>.
[RTCP-SDES]
Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
Header Extension for RTCP Source Description Items", Work
in Progress, draft-ietf-avtext-sdes-hdr-ext-02, July 2015.
[SDP-BUNDLE]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", Work in Progress,
draft-ietf-mmusic-sdp-bundle-negotiation-23, July 2015.
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Acknowledgements
The authors would like to thank Mark Baugher, Bo Burman, Ben
Campbell, Umesh Chandra, Alex Eleftheriadis, Roni Even, Ladan Gharai,
Geoff Hunt, Suresh Krishnan, Keith Lantz, Jonathan Lennox, Scarlet
Liuyan, Suhas Nandakumar, Colin Perkins, and Dan Wing for their help
in reviewing and improving this document.
Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 2
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
United States
Email: stewe@stewe.org
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