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
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
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
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(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.
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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.
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
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.
Multimedia Session: A Multimedia Session is an association among a
group of participants engaged in communication via one or more RTP
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
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.
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.
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
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.
184.108.40.206. 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
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
220.127.116.11. 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 18.104.22.168.
22.214.171.124. 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
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
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.
+------+ +------+ +------+
| | | | | |
| 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.
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
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
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.
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
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
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
The aforementioned situation results in common behavior for RTP
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.
| |------------------------------------>| |
| |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| |
| | | |
| 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