draft-ietf-avtcore-rtp-topologies-update-05.txt   draft-ietf-avtcore-rtp-topologies-update-06.txt 
Network Working Group M. Westerlund Network Working Group M. Westerlund
Internet-Draft Ericsson Internet-Draft Ericsson
Obsoletes: 5117 (if approved) S. Wenger Obsoletes: 5117 (if approved) S. Wenger
Intended status: Informational Vidyo Intended status: Informational Vidyo
Expires: May 16, 2015 November 12, 2014 Expires: September 3, 2015 March 2, 2015
RTP Topologies RTP Topologies
draft-ietf-avtcore-rtp-topologies-update-05 draft-ietf-avtcore-rtp-topologies-update-06
Abstract Abstract
This document discusses point to point and multi-endpoint topologies This document discusses point to point and multi-endpoint topologies
used in Real-time Transport Protocol (RTP)-based environments. In used in Real-time Transport Protocol (RTP)-based environments. In
particular, centralized topologies commonly employed in the video particular, centralized topologies commonly employed in the video
conferencing industry are mapped to the RTP terminology. conferencing industry are mapped to the RTP terminology.
This document is updated with additional topologies and is intended This document is updated with additional topologies and replaces RFC
to replace RFC 5117. 5117.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on May 16, 2015. This Internet-Draft will expire on September 3, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
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the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
skipping to change at page 2, line 42 skipping to change at page 2, line 42
3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 33 3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 33
3.10. Split Component Terminal . . . . . . . . . . . . . . . . 34 3.10. Split Component Terminal . . . . . . . . . . . . . . . . 34
3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 37 3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 37
3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 37 3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 37
4. Topology Properties . . . . . . . . . . . . . . . . . . . . . 38 4. Topology Properties . . . . . . . . . . . . . . . . . . . . . 38
4.1. All to All Media Transmission . . . . . . . . . . . . . . 38 4.1. All to All Media Transmission . . . . . . . . . . . . . . 38
4.2. Transport or Media Interoperability . . . . . . . . . . . 39 4.2. Transport or Media Interoperability . . . . . . . . . . . 39
4.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . . . 39 4.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . . . 39
4.4. Aggregation of Media . . . . . . . . . . . . . . . . . . 40 4.4. Aggregation of Media . . . . . . . . . . . . . . . . . . 40
4.5. View of All Session Participants . . . . . . . . . . . . 40 4.5. View of All Session Participants . . . . . . . . . . . . 40
4.6. Loop Detection . . . . . . . . . . . . . . . . . . . . . 40 4.6. Loop Detection . . . . . . . . . . . . . . . . . . . . . 41
4.7. Consistency between header extensions and RTCP . . . . . 41 4.7. Consistency between header extensions and RTCP . . . . . 41
5. Comparison of Topologies . . . . . . . . . . . . . . . . . . 41 5. Comparison of Topologies . . . . . . . . . . . . . . . . . . 41
6. Security Considerations . . . . . . . . . . . . . . . . . . . 42 6. Security Considerations . . . . . . . . . . . . . . . . . . . 42
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 44 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 44
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 44 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 44
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 44 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 44
9.1. Normative References . . . . . . . . . . . . . . . . . . 44 9.1. Normative References . . . . . . . . . . . . . . . . . . 44
9.2. Informative References . . . . . . . . . . . . . . . . . 45 9.2. Informative References . . . . . . . . . . . . . . . . . 45
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 46 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 46
skipping to change at page 3, line 28 skipping to change at page 3, line 28
utilizing centralized units referred to as MCUs. MCUs may implement utilizing centralized units referred to as MCUs. MCUs may implement
Mixer or Translator functionality (in RTP [RFC3550] terminology), and Mixer or Translator functionality (in RTP [RFC3550] terminology), and
signalling support. They may also contain additional application signalling support. They may also contain additional application
layer functionality. This document focuses on the media transport layer functionality. This document focuses on the media transport
aspects of the MCU that can be realized using RTP, as discussed aspects of the MCU that can be realized using RTP, as discussed
below. Further considered are the properties of Mixers and below. Further considered are the properties of Mixers and
Translators, and how some types of deployed MCUs deviate from these Translators, and how some types of deployed MCUs deviate from these
properties. properties.
This document also codifies new multipoint architectures that have This document also codifies new multipoint architectures that have
recently been introduced and which were not anticipated in RFC 5117. recently been introduced and which were not anticipated in RFC 5117,
These architectures use scalable video coding and simulcasting, and thus this document replaces [RFC5117]. These architectures use
their associated centralized units are referred to as Selective scalable video coding and simulcasting, and their associated
Forwarding Units (SFU). This codification provides a common centralized units are referred to as Selective Forwarding Units
information basis for future discussion and specification work. (SFU). This codification provides a common information basis for
future discussion and specification work.
The document's attempt to clarify and explain sections of the Real- The document's attempt to clarify and explain sections of the Real-
time Transport Protocol (RTP) spec [RFC3550] is informal. It is not time Transport Protocol (RTP) spec [RFC3550] is informal. It is not
intended to update or change what is normatively specified within RFC intended to update or change what is normatively specified within RFC
3550. 3550.
2. Definitions 2. Definitions
2.1. Glossary 2.1. Glossary
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2.1. Glossary 2.1. Glossary
ASM: Any Source Multicast ASM: Any Source Multicast
AVPF: The Extended RTP Profile for RTCP-based Feedback AVPF: The Extended RTP Profile for RTCP-based Feedback
CSRC: Contributing Source CSRC: Contributing Source
Link: The data transport to the next IP hop Link: The data transport to the next IP hop
Middlebox: A device that is on the Path that media travel between Middlebox: A device that is on the Path that media travel between
two Endpoints two endpoints
MCU: Multipoint Control Unit MCU: Multipoint Control Unit
Path: The concatenation of multiple links, resulting in an end-to- Path: The concatenation of multiple links, resulting in an end-to-
end data transfer. end data transfer.
PtM: Point to Multipoint PtM: Point to Multipoint
PtP: Point to Point PtP: Point to Point
skipping to change at page 4, line 42 skipping to change at page 4, line 44
the RFC number of the published taxonomy draft.] the RFC number of the published taxonomy draft.]
The following definitions have been taken from draft-ietf-avtext-rtp- The following definitions have been taken from draft-ietf-avtext-rtp-
grouping-taxonomy-02, and are used in capitalized form throughout the grouping-taxonomy-02, and are used in capitalized form throughout the
document. document.
Communication Session: A Communication Session is an association Communication Session: A Communication Session is an association
among group of participants communicating with each other via a among group of participants communicating with each other via a
set of Multimedia Sessions. set of Multimedia Sessions.
End Point: A single addressable entity sending or receiving RTP Endpoint: A single addressable entity sending or receiving RTP
packets. It may be decomposed into several functional blocks, but packets. It may be decomposed into several functional blocks, but
as long as it behaves as a single RTP stack entity it is as long as it behaves as a single RTP stack entity it is
classified as a single "End Point". classified as a single "Endpoint".
Media Source: A Media Source is the logical source of a reference Media Source: A Media Source is the logical source of a reference
clock synchronized, time progressing, digital media stream, called clock synchronized, time progressing, digital media stream, called
a Source Stream. a Source Stream.
Multimedia Session: A multimedia session is an association among a Multimedia Session: A multimedia session is an association among a
group of participants engaged in the communication via one or more group of participants engaged in the communication via one or more
RTP Sessions. RTP Sessions.
3. Topologies 3. Topologies
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For each of the RTP-defined topologies, we discuss how RTP, RTCP, and 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 carried media are handled. With respect to RTCP, we also discuss
the handling of RTCP feedback messages as defined in [RFC4585] and the handling of RTCP feedback messages as defined in [RFC4585] and
[RFC5104]. [RFC5104].
3.1. Point to Point 3.1. Point to Point
Shortcut name: Topo-Point-to-Point Shortcut name: Topo-Point-to-Point
The Point to Point (PtP) topology (Figure 1) consists of two End The Point to Point (PtP) topology (Figure 1) consists of two
Points, communicating using unicast. Both RTP and RTCP traffic are endpoints, communicating using unicast. Both RTP and RTCP traffic
conveyed endpoint-to-endpoint, using unicast traffic only (even if, are conveyed endpoint-to-endpoint, using unicast traffic only (even
in exotic cases, this unicast traffic happens to be conveyed over an if, in exotic cases, this unicast traffic happens to be conveyed over
IP-multicast address). an IP-multicast address).
+---+ +---+ +---+ +---+
| A |<------->| B | | A |<------->| B |
+---+ +---+ +---+ +---+
Figure 1: Point to Point Figure 1: Point to Point
The main property of this topology is that A sends to B, and only B, 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 while B sends to A, and only A. This avoids all complexities of
handling multiple End Points and combining the requirements stemming handling multiple endpoints and combining the requirements stemming
from them. Note that an End Point can still use multiple RTP from them. Note that an endpoint can still use multiple RTP
Synchronization Sources (SSRCs) in an RTP session. The number of 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 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. only to system level limitations like the number range of ports.
RTCP feedback messages for the indicated SSRCs are communicated RTCP feedback messages for the indicated SSRCs are communicated
directly between the End Points. Therefore, this topology poses directly between the endpoints. Therefore, this topology poses
minimal (if any) issues for any feedback messages. For RTP sessions minimal (if any) issues for any feedback messages. For RTP sessions
which use multiple SSRC per End Point it can be relevant to implement which use multiple SSRC per endpoint it can be relevant to implement
support for cross-reporting suppression as defined in "Sending support for cross-reporting suppression as defined in "Sending
Multiple Media Streams in a Single RTP Session" Multiple Media Streams in a Single RTP Session"
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]. [I-D.ietf-avtcore-rtp-multi-stream-optimisation].
3.2. Point to Point via Middlebox 3.2. Point to Point via Middlebox
This section discusses cases where two End Points communicate but This section discusses cases where two endpoints communicate but have
have one or more middleboxes involved in the RTP session. one or more middleboxes involved in the RTP session.
3.2.1. Translators 3.2.1. Translators
Shortcut name: Topo-PtP-Translator Shortcut name: Topo-PtP-Translator
Two main categories of Translators can be distinguished; Transport Two main categories of Translators can be distinguished; Transport
Translators and Media translators. Both Translator types share Translators and Media translators. Both Translator types share
common attributes that separate them from Mixers. For each RTP common attributes that separate them from Mixers. For each RTP
stream that the Translator receives, it generates an individual 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 in the other domain. A translator keeps the SSRC for an RTP
stream across the translation, whereas a Mixer can select a single stream across the translation, whereas a Mixer can select a single
RTP stream from multiple received RTP streams (in cases like audio/ RTP stream from multiple received RTP streams (in cases like audio/
video switching), or send out an RTP stream composed of multiple video switching), or send out an RTP stream composed of multiple
mixed media received in multiple RTP streams (in cases like audio mixed media received in multiple RTP streams (in cases like audio
mixing or video tiling), but always under its own SSRC, possibly mixing or video tiling), but always under its own SSRC, possibly
using the CSRC field to indicate the source(s) of the content. 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 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 reason is that in PtP use cases the primary focus of a middlebox is
enabling interoperability, between otherwise non-interoperable End enabling interoperability, between otherwise non-interoperable
Points, such as transcoding to a codec the receiver supports, which endpoints, such as transcoding to a codec the receiver supports,
can be done by a media translator. which can be done by a media translator.
As specified in Section 7.1 of [RFC3550], the SSRC space is common 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 for all participants in the RTP session, independent of on which side
of the Translator the session resides. Therefore, it is the of the Translator the session resides. Therefore, it is the
responsibility of the End Points (as the RTP session participants) to responsibility of the endpoints (as the RTP session participants) to
run SSRC collision detection, and the SSRC is thus a field the run SSRC collision detection, and the SSRC is thus a field the
Translator cannot change. Any SDES information associated with a Translator cannot change. Any SDES information associated with a
SSRC or CSRC also needs to be forwarded between the domains for any SSRC or CSRC also needs to be forwarded between the domains for any
SSRC/CSRC used in the different domains. SSRC/CSRC used in the different domains.
A Translator commonly does not use an SSRC of its own, and is not 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 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 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. sends RTCP reports and therefore is required to have an SSRC.
Another example is the case when a Translator is prepared to use RTCP Another example is the case when a Translator is prepared to use RTCP
feedback messages. This may, for example, occur in a translator feedback messages. This may, for example, occur in a translator
configured to detect packet loss of important video packets and wants configured to detect packet loss of important video packets and wants
to trigger repair by the media sending End Point, by sending feedback to trigger repair by the media sending endpoint, by sending feedback
messages. While such feedback could use the SSRC of the target for messages. While such feedback could use the SSRC of the target for
the translator (the receiving End Point), this in turn would require the translator (the receiving endpoint), this in turn would require
translation of the targets RTCP reports to make them consistent. It translation of the targets RTCP reports to make them consistent. It
may be simpler to expose an additional SSRC in the session. The only may be simpler to expose an additional SSRC in the session. The only
concern is End Points failing to support the full RTP specification concern is endpoints failing to support the full RTP specification
may have issues with multiple SSRCs reporting on the RTP streams sent may have issues with multiple SSRCs reporting on the RTP streams sent
by that End Point, as this use case may be viewed as excotic by by that endpoint, as this use case may be viewed as excotic by
implementers. implementers.
In general, a Translator implementation should consider which RTCP In general, a Translator implementation should consider which RTCP
feedback messages or codec-control messages it needs to understand in feedback messages or codec-control messages it needs to understand in
relation to the functionality of the Translator itself. This is relation to the functionality of the Translator itself. This is
completely in line with the requirement to also translate RTCP completely in line with the requirement to also translate RTCP
messages between the domains. messages between the domains.
3.2.1.1. Transport Relay/Anchoring 3.2.1.1. Transport Relay/Anchoring
There exist a number of different types of middleboxes that might be There exist a number of different types of middleboxes that might be
inserted between two End Points on the transport level, e.g., to inserted between two endpoints on the transport level, e.g., to
perform changes on the IP/UDP headers, and are, therefore, basic perform changes on the IP/UDP headers, and are, therefore, basic
transport translators. These middleboxes come in many variations transport translators. These middleboxes come in many variations
including NAT [RFC3022] traversal by pinning the media path to a including NAT [RFC3022] traversal by pinning the media path to a
public address domain relay, network topologies where the RTP stream public address domain relay, network topologies where the RTP stream
is required to pass a particular point for audit by employing is required to pass a particular point for audit by employing
relaying, or preserving privacy by hiding each peer's transport relaying, or preserving privacy by hiding each peer's transport
addresses to the other party. Other protocols or functionalities addresses to the other party. Other protocols or functionalities
that provide this behavior are TURN [RFC5766] servers, Session Border that provide this behavior are TURN [RFC5766] servers, Session Border
Gateways and Media Processing Nodes with media anchoring Gateways and Media Processing Nodes with media anchoring
functionalities. functionalities.
skipping to change at page 7, line 47 skipping to change at page 7, line 47
+---+ +---+ +---+ +---+ +---+ +---+
| A |<------>| T |<------->| B | | A |<------>| T |<------->| B |
+---+ +---+ +---+ +---+ +---+ +---+
Figure 2: Point to Point with Translator Figure 2: Point to Point with Translator
A common element in these functions is that they are normally A common element in these functions is that they are normally
transparent at the RTP level, i.e., they perform no changes on any 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 RTP or RTCP packet fields and only affect the lower layers. They may
affect, however, the path the RTP and RTCP packets are routed between affect, however, the path the RTP and RTCP packets are routed between
the End Points in the RTP session, and thereby indirectly affect the the endpoints in the RTP session, and thereby indirectly affect the
RTP session. For this reason, one could believe that transport RTP session. For this reason, one could believe that transport
translator-type middleboxes do not need to be included in this translator-type middleboxes do not need to be included in this
document. This topology, however, can raise additional requirements document. This topology, however, can raise additional requirements
in the RTP implementation and its interactions with the signalling in the RTP implementation and its interactions with the signalling
solution. Both in signalling and in certain RTCP fields, network solution. Both in signalling and in certain RTCP fields, network
addresses other than those of the relay can occur since B has a addresses other than those of the relay can occur since B has a
different network address than the relay (T). Implementations that different network address than the relay (T). Implementations that
cannot support this will also not work correctly when End Points are cannot support this will also not work correctly when endpoints are
subject to NAT. subject to NAT.
The transport relay implementations also have to take into account The transport relay implementations also have to take into account
security considerations. In particular, source address filtering of security considerations. In particular, source address filtering of
incoming packets is usually important in relays, to prevent attackers incoming packets is usually important in relays, to prevent attackers
to inject traffic into a session, which one peer may, in the absence to inject traffic into a session, which one peer may, in the absence
fo adequate security in the relay, think it comes from the other fo adequate security in the relay, think it comes from the other
peer. peer.
3.2.1.2. Transport Translator 3.2.1.2. Transport Translator
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The most basic transport translators that operate below the RTP level The most basic transport translators that operate below the RTP level
were already discussed in Section 3.2.1.1. were already discussed in Section 3.2.1.1.
3.2.1.3. Media Translator 3.2.1.3. Media Translator
Media Translators (Topo-Media-Translator) modify the media inside the Media Translators (Topo-Media-Translator) modify the media inside the
RTP stream. This process is commonly known as transcoding. The RTP stream. This process is commonly known as transcoding. The
modification of the media can be as small as removing parts of 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 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 (down to the sample level or equivalent) utilizing a different media
codec. Media Translators are commonly used to connect End Points codec. Media Translators are commonly used to connect endpoints
without a common interoperability point in the media encoding. without a common interoperability point in the media encoding.
Stand-alone Media Translators are rare. Most commonly, a combination Stand-alone Media Translators are rare. Most commonly, a combination
of Transport and Media Translator is used to translate both the media of Transport and Media Translator is used to translate both the media
and the transport aspects of the RTP stream carrying the media and the transport aspects of the RTP stream carrying the media
between two transport domains. between two transport domains.
When media translation occurs, the Translator's task regarding When media translation occurs, the Translator's task regarding
handling of RTCP traffic becomes substantially more complex. In this handling of RTCP traffic becomes substantially more complex. In this
case, the Translator needs to rewrite End Point B's RTCP Receiver case, the Translator needs to rewrite endpoint B's RTCP Receiver
Report before forwarding them to End Point A. The rewriting is Report before forwarding them to endpoint A. The rewriting is needed
needed as the RTP stream received by B is not the same RTP stream as as the RTP stream received by B is not the same RTP stream as the
the other participants receive. For example, the number of packets other participants receive. For example, the number of packets
transmitted to B may be lower than what A sends, due to the different 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 media format and data rate. Therefore, if the Receiver Reports were
forwarded without changes, the extended highest sequence number would forwarded without changes, the extended highest sequence number would
indicate that B were substantially behind in reception, while most indicate that B were substantially behind in reception, while most
likely it would not be. Therefore, the Translator must translate likely it would not be. Therefore, the Translator must translate
that number to a corresponding sequence number for the stream the that number to a corresponding sequence number for the stream the
Translator received. Similar requirements exists for most other Translator received. Similar requirements exists for most other
fields in the RTCP Receiver Reports. fields in the RTCP Receiver Reports.
A media Translator may in some cases act on behalf of the "real" A media Translator may in some cases act on behalf of the "real"
source (the End Point originally sending the media to the Translator) source (the endpoint originally sending the media to the Translator)
and respond to RTCP feedback messages. This may occur, for example, and respond to RTCP feedback messages. This may occur, for example,
when a receiving End Point requests a bandwidth reduction, and the when a receiving endpoint requests a bandwidth reduction, and the
media Translator has not detected any congestion or other reasons for media Translator has not detected any congestion or other reasons for
bandwidth reduction between the sending End Point and itself. In bandwidth reduction between the sending endpoint and itself. In that
that case, it is sensible that the media Translator reacts to codec case, it is sensible that the media Translator reacts to codec
control messages itself, for example, by transcoding to a lower media control messages itself, for example, by transcoding to a lower media
rate. rate.
A variant of translator behaviour worth pointing out is the one A variant of translator behaviour worth pointing out is the one
depicted in Figure 3 of an End Point A sending a RTP stream depicted in Figure 3 of an endpoint A sending a RTP stream containing
containing media (only) to B. On the path there is a device T that media (only) to B. On the path there is a device T that on A's
on A's behalf manipulates the RTP streams. One common example is behalf manipulates the RTP streams. One common example is that T
that T adds a second RTP stream containing Forward Error Correction adds a second RTP stream containing Forward Error Correction (FEC)
(FEC) information in order to protect A's (non FEC-protected) RTP information in order to protect A's (non FEC-protected) RTP stream.
stream. In this case, T needs to semantically bind the new FEC RTP In this case, T needs to semantically bind the new FEC RTP stream to
stream to A's media-carrying RTP stream, for example by using the A's media-carrying RTP stream, for example by using the same CNAME as
same CNAME as A. A.
+------+ +------+ +------+ +------+ +------+ +------+
| | | | | | | | | | | |
| A |------->| T |-------->| B | | A |------->| T |-------->| B |
| | | |---FEC-->| | | | | |---FEC-->| |
+------+ +------+ +------+ +------+ +------+ +------+
Figure 3: Media Translator adding FEC Figure 3: Media Translator adding FEC
there may also be cases where information is added into the original there may also be cases where information is added into the original
skipping to change at page 10, line 18 skipping to change at page 10, line 18
Similarly, a Media Translator can sometimes remove information from Similarly, a Media Translator can sometimes remove information from
the RTP stream, while otherwise leaving the remaining RTP packets the RTP stream, while otherwise leaving the remaining RTP packets
unchanged (again with the exception of certain RTP header fields). unchanged (again with the exception of certain RTP header fields).
Either type of functionality where T manipulates the RTP stream, or Either type of functionality where T manipulates the RTP stream, or
adds an accompanying RTP stream, on behalf of A is also covered under adds an accompanying RTP stream, on behalf of A is also covered under
the media translator definition. the media translator definition.
3.2.2. Back to Back RTP sessions 3.2.2. Back to Back RTP sessions
There exist middleboxes that interconnect two End Points A and B There exist middleboxes that interconnect two endpoints A and B
through themselves (MB), but not by being part of a common RTP through themselves (MB), but not by being part of a common RTP
session. They establish instead two different RTP sessions, one session. They establish instead two different RTP sessions, one
between A and the middlebox and another between the middlebox and B. between A and the middlebox and another between the middlebox and B.
This topology is called Topo-Back-To-Back This topology is called Topo-Back-To-Back
|<--Session A-->| |<--Session B-->| |<--Session A-->| |<--Session B-->|
+------+ +------+ +------+ +------+ +------+ +------+
| A |------->| MB |-------->| B | | A |------->| MB |-------->| B |
+------+ +------+ +------+ +------+ +------+ +------+
skipping to change at page 10, line 48 skipping to change at page 10, line 48
respnsibility to maintain the correct relations. respnsibility to maintain the correct relations.
The signalling or other above-RTP level functionalities referencing The signalling or other above-RTP level functionalities referencing
RTP streams may be what is most impacted by using two RTP sessions RTP streams may be what is most impacted by using two RTP sessions
and changing identifiers. The structure with two RTP sessions also and changing identifiers. The structure with two RTP sessions also
puts a congestion control requirement on the middlebox, because it puts a congestion control requirement on the middlebox, because it
becomes fully responsible for the media stream it sources into each becomes fully responsible for the media stream it sources into each
of the sessions. of the sessions.
Adherence to congestion control can be solved locally on each of the Adherence to congestion control can be solved locally on each of the
two segments, or by bridging statistics from the receiving End Point two segments, or by bridging statistics from the receiving endpoint
through the middlebox to the sending End Point. From an through the middlebox to the sending endpoint. From an
implementation point, however, the latter requires dealing with a implementation point, however, the latter requires dealing with a
number of inconsistencies. First, packet loss must be detected for number of inconsistencies. First, packet loss must be detected for
an RTP stream sent from A to the middlebox, and that loss must be 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 reported through a skipped sequence number in the RTP stream from the
middlebox to B. This coupling and the resulting inconsistencies are middlebox to B. This coupling and the resulting inconsistencies are
conceptually easier to handle when considering the two RTP streams as conceptually easier to handle when considering the two RTP streams as
belonging to a single RTP session. belonging to a single RTP session.
3.3. Point to Multipoint Using Multicast 3.3. Point to Multipoint Using Multicast
skipping to change at page 12, line 23 skipping to change at page 12, line 23
defined in AVPF [RFC4585]. Even when the environment would allow for defined in AVPF [RFC4585]. Even when the environment would allow for
the use of a small multicast group, some applications may still want the use of a small multicast group, some applications may still want
to use the more limited options for RTCP feedback available to large to use the more limited options for RTCP feedback available to large
multicast groups, for example when there is a likelihood that the multicast groups, for example when there is a likelihood that the
threshold of the small multicast group (in terms of multicast group threshold of the small multicast group (in terms of multicast group
participants) may be exceeded during the lifetime of a session. participants) may be exceeded during the lifetime of a session.
RTCP feedback messages in multicast reach, like media data, every RTCP feedback messages in multicast reach, like media data, every
subscriber (subject to packet losses and multicast group subscriber (subject to packet losses and multicast group
subscription). Therefore, the feedback suppression mechanism subscription). Therefore, the feedback suppression mechanism
discussed in [RFC4585] is typically required. Each individual End discussed in [RFC4585] is typically required. Each individual
Point that is a multicast group participant needs to process every endpoint that is a multicast group participant needs to process every
feedback message it receives, not only to determine if it is affected feedback message it receives, not only to determine if it is affected
or if the feedback message applies only to some other End Point, but or if the feedback message applies only to some other endpoint, but
also to derive timing restrictions for the sending of its own also to derive timing restrictions for the sending of its own
feedback messages, if any. feedback messages, if any.
3.3.2. Source Specific Multicast (SSM) 3.3.2. Source Specific Multicast (SSM)
In Any Source Multicast, any of the multicast group participants can In Any Source Multicast, any of the multicast group participants can
send to all the other multicast group participants, by sending a send to all the other multicast group participants, by sending a
packet to the multicast group. In contrast, Source Specific packet to the multicast group. In contrast, Source Specific
Multicast [RFC3569][RFC4607] refers to scenarios where only a single Multicast [RFC3569][RFC4607] refers to scenarios where only a single
source (Distribution Source) can send to the multicast group, source (Distribution Source) can send to the multicast group,
skipping to change at page 13, line 29 skipping to change at page 13, line 29
|Sender M|<----->| | |<-------------------------+ |Sender M|<----->| | |<-------------------------+
+--------+ +-----+ RTCP Unicast +--------+ +-----+ RTCP Unicast
FT = Feedback Target FT = Feedback Target
Transport from the Feedback Target to the Distribution Transport from the Feedback Target to the Distribution
Source is via unicast or multicast RTCP if they are not Source is via unicast or multicast RTCP if they are not
co-located. co-located.
Figure 6: Point to Multipoint using Source Specific Multicast Figure 6: Point to Multipoint using Source Specific Multicast
In the SSM topology (Figure 6) a number of RTP sending End Points In the SSM topology (Figure 6) a number of RTP sending endpoints (RTP
(RTP sources henceforth) (1 to M) are allowed to send media to the sources henceforth) (1 to M) are allowed to send media to the SSM
SSM group. These sources send media to a dedicated distribution group. These sources send media to a dedicated distribution source,
source, which forwards the RTP streams to the multicast group on which forwards the RTP streams to the multicast group on behalf of
behalf of the original RTP sources. The RTP streams reach the the original RTP sources. The RTP streams reach the receiving
receiving End Points (Receivers henceforth) (R(1) to R(n)). The endpoints (Receivers henceforth) (R(1) to R(n)). The Receivers' RTCP
Receivers' RTCP messages cannot be sent to the multicast group, as messages cannot be sent to the multicast group, as the SSM multicast
the SSM multicast group by definition has only a single IP sender. group by definition has only a single IP sender. To support RTCP, an
To support RTCP, an RTP extension for SSM [RFC5760] was defined. It RTP extension for SSM [RFC5760] was defined. It uses unicast
uses unicast transmission to send RTCP from each of the receivers to transmission to send RTCP from each of the receivers to one or more
one or more Feedback Targets (FT). The feedback targets relay the Feedback Targets (FT). The feedback targets relay the RTCP
RTCP unmodified, or provide a summary of the participants RTCP unmodified, or provide a summary of the participants RTCP reports
reports towards the whole group by forwarding the RTCP traffic to the towards the whole group by forwarding the RTCP traffic to the
distribution source. Figure 6 only shows a single feedback target distribution source. Figure 6 only shows a single feedback target
integrated in the distribution source, but for scalability the FT can integrated in the distribution source, but for scalability the FT can
be distributed and each instance can have responsibility for sub- be distributed and each instance can have responsibility for sub-
groups of the receivers. For summary reports, however, there groups of the receivers. For summary reports, however, there
typically must be a single feedback target aggregating all the typically must be a single feedback target aggregating all the
summaries to a common message to the whole receiver group. summaries to a common message to the whole receiver group.
The RTP extension for SSM specifies how feedback (both reception The RTP extension for SSM specifies how feedback (both reception
information and specific feedback events) are handled. The more information and specific feedback events) are handled. The more
general problems associated with the use of multicast, where everyone general problems associated with the use of multicast, where everyone
receives what the distribution source sends needs to be accounted receives what the distribution source sends needs to be accounted
for. for.
Aforementioned situation results in common behavior for RTP Aforementioned situation results in common behavior for RTP
multicast: multicast:
1. Multicast applications often use a group of RTP sessions, not 1. Multicast applications often use a group of RTP sessions, not
one. Each End Point needs to be a member of most or all of these one. Each endpoint needs to be a member of most or all of these
RTP sessions in order to perform well. RTP sessions in order to perform well.
2. Within each RTP session, the number of media sinks is likely to 2. Within each RTP session, the number of media sinks is likely to
be much larger than the number of RTP sources. be much larger than the number of RTP sources.
3. Multicast applications need signalling functions to identify the 3. Multicast applications need signalling functions to identify the
relationships between RTP sessions. relationships between RTP sessions.
4. Multicast applications need signalling functions to identify the 4. Multicast applications need signalling functions to identify the
relationships between SSRCs in different RTP sessions. relationships between SSRCs in different RTP sessions.
All multicast configurations share a signalling requirement: all of All multicast configurations share a signalling requirement: all of
the End Points need to have the same RTP and payload type the endpoints need to have the same RTP and payload type
configuration. Otherwise, End Point A could, for example, be using configuration. Otherwise, endpoint A could, for example, be using
payload type 97 to identify the video codec H.264, while End Point B payload type 97 to identify the video codec H.264, while endpoint B
would identify it as MPEG-2, with unpredicatble but almost certainly would identify it as MPEG-2, with unpredicatble but almost certainly
not visually pleasing results. not visually pleasing results.
Security solutions for this type of group communications are also Security solutions for this type of group communications are also
challenging. First, the key-management and the security protocol challenging. First, the key-management and the security protocol
must support group communication. Source authentication becomes more must support group communication. Source authentication becomes more
difficult and requires specialized solutions. For more discussion on difficult and requires specialized solutions. For more discussion on
this please review Options for Securing RTP Sessions [RFC7201]. this please review Options for Securing RTP Sessions [RFC7201].
3.3.3. SSM with Local Unicast Resources 3.3.3. SSM with Local Unicast Resources
skipping to change at page 15, line 38 skipping to change at page 15, line 38
---------------- -------------- ---------------- --------------
-------> Multicast RTP Stream -------> Multicast RTP Stream
.-.-.-.> Multicast RTCP Stream .-.-.-.> Multicast RTCP Stream
.=.=.=.> Unicast RTCP Reports .=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages ~~~~~~~> Unicast RTCP Feedback Messages
.......> Unicast RTP Stream .......> Unicast RTP Stream
Figure 7: SSM with Local Unicast Resources (RAMS) Figure 7: SSM with Local Unicast Resources (RAMS)
The Rapid acquisition extension allows an End Point joining an SSM The Rapid acquisition extension allows an endpoint joining an SSM
multicast session to request media starting with the last sync-point multicast session to request media starting with the last sync-point
(from where media can be decoded without requiring context (from where media can be decoded without requiring context
established by the decoding of prior packets) to be sent at high established by the decoding of prior packets) to be sent at high
speed until such time where, after decoding of these burst-delivered speed until such time where, after decoding of these burst-delivered
media packets, the correct media timing is established, i.e. media media packets, the correct media timing is established, i.e. media
packets are received within adequate buffer intervals for this packets are received within adequate buffer intervals for this
application. This is accomplished by first establishing a unicast application. This is accomplished by first establishing a unicast
PtP RTP session between the Burst/Retransmission Source (BRS, PtP RTP session between the Burst/Retransmission Source (BRS,
Figure 7) and the RTP Receiver. The unicast session is used to Figure 7) and the RTP Receiver. The unicast session is used to
transmit cached packets from the multicast group at higher then transmit cached packets from the multicast group at higher then
skipping to change at page 16, line 26 skipping to change at page 16, line 26
\ / \ /
\ / \ /
v v v v
+---+ +---+
| C | | C |
+---+ +---+
Figure 8: Point to Multi-Point using Mesh Figure 8: Point to Multi-Point using Mesh
Based on the RTP session definition, it is clearly possible to have a Based on the RTP session definition, it is clearly possible to have a
joint RTP session involving three or more End Points over multiple joint RTP session involving three or more endpoints over multiple
unicast transport flows, like the joint three End point session unicast transport flows, like the joint three endpoint session
depicted above. In this case, A needs to send its RTP streams and 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. RTCP packets to both B and C over their respective transport flows.
As long as all End Points do the same, everyone will have a joint As long as all endpoints do the same, everyone will have a joint view
view of the RTP session. of the RTP session.
This topology does not create any additional requirements beyond the This topology does not create any additional requirements beyond the
need to have multiple transport flows associated with a single RTP need to have multiple transport flows associated with a single RTP
session. Note that an End Point may use a single local port to session. Note that an endpoint may use a single local port to
receive all these transport flows (in which case the sending port, IP receive all these transport flows (in which case the sending port, IP
address, or SSRC can be used to demultiplex), or it might have address, or SSRC can be used to demultiplex), or it might have
separate local reception ports for each of the End Points. separate local reception ports for each of the endpoints.
+-A--------------------+ +-A--------------------+
|+---+ | |+---+ |
||CAM| | +-B-----------+ ||CAM| | +-B-----------+
|+---+ +-UDP1------| |-UDP1------+ | |+---+ +-UDP1------| |-UDP1------+ |
| | | +-RTP1----| |-RTP1----+ | | | | | +-RTP1----| |-RTP1----+ | |
| V | | +-Video-| |-Video-+ | | | | V | | +-Video-| |-Video-+ | | |
|+----+ | | | |<----------------|BV1 | | | | |+----+ | | | |<----------------|BV1 | | | |
||ENC |----+-+-+--->AV1|---------------->| | | | | ||ENC |----+-+-+--->AV1|---------------->| | | | |
|+----+ | | +-------| |-------+ | | | |+----+ | | +-------| |-------+ | | |
skipping to change at page 17, line 31 skipping to change at page 17, line 31
| | | | +-Video-| |-Video-+ | | | | | | | +-Video-| |-Video-+ | | |
| +-------+-+-+--->AV1|---------------->| | | | | | +-------+-+-+--->AV1|---------------->| | | | |
| | | | |<----------------|CV1 | | | | | | | | |<----------------|CV1 | | | |
| | | +-------| |-------+ | | | | | | +-------| |-------+ | | |
| | +---------| |---------+ | | | | +---------| |---------+ | |
| +-----------| |-----------+ | | +-----------| |-----------+ |
+----------------------+ +-------------+ +----------------------+ +-------------+
Figure 9: An Multi-unicast Mesh with a joint RTP session Figure 9: An Multi-unicast Mesh with a joint RTP session
A joint RTP session from End Point A's perspective for the Mesh A joint RTP session from endpoint A's perspective for the Mesh
depicted in Figure 8 with a joint RTP session have multiple transport depicted in Figure 8 with a joint RTP session have multiple transport
flows, here enumerated as UDP1 and UDP2. However, there is only one flows, here enumerated as UDP1 and UDP2. However, there is only one
RTP session (RTP1). The Media Source (CAM) is encoded and RTP session (RTP1). The Media Source (CAM) is encoded and
transmitted over the SSRC (AV1) across both transport layers. transmitted over the SSRC (AV1) across both transport layers.
However, as this is a joint RTP session, the two streams must be the However, as this is a joint RTP session, the two streams must be the
same. Thus, an congestion control adaptation needed for the paths A same. Thus, an congestion control adaptation needed for the paths A
to B and A to C needs to use the most restricting path's properties. 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 An alternative structure for establishing the above topology is to
use independent RTP sessions between each pair of peers, i.e., three use independent RTP sessions between each pair of peers, i.e., three
different RTP sessions. In some scenarios, the same RTP stream may different RTP sessions. In some scenarios, the same RTP stream may
be sent from the transmitting End Point, however it also supports be sent from the transmitting endpoint, however it also supports
local adaptation taking place in one or more of the RTP streams, local adaptation taking place in one or more of the RTP streams,
rendering them non-identical. rendering them non-identical.
+-A----------------------+ +-B-----------+ +-A----------------------+ +-B-----------+
|+---+ | | | |+---+ | | |
||MIC| +-UDP1------| |-UDP1------+ | ||MIC| +-UDP1------| |-UDP1------+ |
|+---+ | +-RTP1----| |-RTP1----+ | | |+---+ | +-RTP1----| |-RTP1----+ | |
| | +----+ | | +-Audio-| |-Audio-+ | | | | | +----+ | | +-Audio-| |-Audio-+ | | |
| +->|ENC1|--+-+-+--->AA1|------------->| | | | | | +->|ENC1|--+-+-+--->AA1|------------->| | | | |
| | +----+ | | | |<-------------|BA1 | | | | | | +----+ | | | |<-------------|BA1 | | | |
skipping to change at page 18, line 45 skipping to change at page 18, line 45
microphone is captured and the audio is fed into two different microphone is captured and the audio is fed into two different
encoder instances, each being associated with two independent RTP encoder instances, each being associated with two independent RTP
sessions (RTP1 and RTP2). The SSRCs (AA1 and AA2) in each RTP sessions (RTP1 and RTP2). The SSRCs (AA1 and AA2) in each RTP
session are completely independent and the media bit-rate produced by session are completely independent and the media bit-rate produced by
the encoders can also be tuned differently to address any congestion the encoders can also be tuned differently to address any congestion
control requirements differing for the paths A to B compared to A to control requirements differing for the paths A to B compared to A to
C. C.
From a topologies viewpoint, an important difference exists in the From a topologies viewpoint, an important difference exists in the
behavior around RTCP. First, when a single RTP session spans all behavior around RTCP. First, when a single RTP session spans all
three End Points A, B, and C, and their connecting RTP streams, a three endpoints A, B, and C, and their connecting RTP streams, a
common RTCP bandwidth is calculated and used for this single joint common RTCP bandwidth is calculated and used for this single joint
session. In contrast, when there are multiple independent RTP session. In contrast, when there are multiple independent RTP
sessions, each RTP session has its local RTCP bandwidth allocation. sessions, each RTP session has its local RTCP bandwidth allocation.
Further, when multiple sessions are used, End Points not directly Further, when multiple sessions are used, endpoints not directly
involved in a session do not have any awareness of the conditions in involved in a session do not have any awareness of the conditions in
those sessions. For example, in the case of the three End Point those sessions. For example, in the case of the three endpoint
configuration in Figure 8, End Point A has no awareness of the configuration in Figure 8, endpoint A has no awareness of the
conditions occurring in the session between End Points B and C conditions occurring in the session between endpoints B and C
(whereas, if a single RTP session were used, it would have such (whereas, if a single RTP session were used, it would have such
awareness). awareness).
Loop detection is also affected. With independent RTP sessions, the Loop detection is also affected. With independent RTP sessions, the
SSRC/CSRC cannot be used to determine when an End Point receives its SSRC/CSRC cannot be used to determine when an endpoint receives its
own media stream, or a mixed media stream including its own media own media stream, or a mixed media stream including its own media
stream (a condition known as a loop). The identification of loops stream (a condition known as a loop). The identification of loops
and, in most cases, their avoidance, has to be achieved by other and, in most cases, their avoidance, has to be achieved by other
means, for example through signaling or the use of an RTP external means, for example through signaling or the use of an RTP external
name space binding SSRC/CSRC among any communicating RTP sessions in name space binding SSRC/CSRC among any communicating RTP sessions in
the mesh. the mesh.
3.5. Point to Multipoint Using the RFC 3550 Translator 3.5. Point to Multipoint Using the RFC 3550 Translator
This section discusses some additional usages related to point to This section discusses some additional usages related to point to
skipping to change at page 19, line 45 skipping to change at page 19, line 45
+ cast +->| Translator | + cast +->| Translator |
+---+ \ Network / | | +---+ +---+ \ Network / | | +---+
| C |<---\ / | |<---->| D | | C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+ +---+ \ / +------------+ +---+
+-----+ +-----+
Figure 11: Point to Multipoint Using Multicast Figure 11: Point to Multipoint Using Multicast
Figure 11 depicts an example of a Transport Translator performing at Figure 11 depicts an example of a Transport Translator performing at
least IP address translation. It allows the (non-multicast-capable) least IP address translation. It allows the (non-multicast-capable)
End Points B and D to take part in an any source multicast session endpoints B and D to take part in an any source multicast session
involving End Points A and C, by having the Translator forward their involving endpoints A and C, by having the Translator forward their
unicast traffic to the multicast addresses in use, and vice versa. 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 It must also forward B's traffic to D, and vice versa, to provide
each of B and D with a complete view of the session. each of B and D with a complete view of the session.
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Translator | | Translator |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 12: RTP Translator (Relay) with Only Unicast Paths Figure 12: RTP Translator (Relay) with Only Unicast Paths
Another Translator scenario is depicted in Figure 12. The Translator Another Translator scenario is depicted in Figure 12. The Translator
in this case connects multiple End Points through unicast. This can in this case connects multiple endpoints through unicast. This can
be implemented using a very simple transport Translator which, in be implemented using a very simple transport Translator which, in
this document, is called a relay. The relay forwards all traffic it this document, is called a relay. The relay forwards all traffic it
receives, both RTP and RTCP, to all other End Points. In doing so, a receives, both RTP and RTCP, to all other endpoints. In doing so, a
multicast network is emulated without relying on a multicast-capable multicast network is emulated without relying on a multicast-capable
network infrastructure. network infrastructure.
For RTCP feedback this results in a similar set of considerations to For RTCP feedback this results in a similar set of considerations to
those described in the ASM RTP topology. It also puts some those described in the ASM RTP topology. It also puts some
additional signalling requirements onto the session establishment; additional signalling requirements onto the session establishment;
for example, a common configuration of RTP payload types is required. for example, a common configuration of RTP payload types is required.
Transport translators and relays should always consider implementing Transport translators and relays should always consider implementing
source address filtering, to prevent attackers to inject traffic source address filtering, to prevent attackers to inject traffic
using the listening ports on the translator. The translator can, using the listening ports on the translator. The translator can,
however, go one step further, and especially if explicit SSRC however, go one step further, and especially if explicit SSRC
signalling is used, prevent End points to send SSRCs other than its signalling is used, prevent endpoints to send SSRCs other than its
own (that are, for example, used by other participants in the own (that are, for example, used by other participants in the
session). This can improve the security properties of the session, session). This can improve the security properties of the session,
despite the use of group keys that on cryptographic level allows despite the use of group keys that on cryptographic level allows
anyone to impersonate another in the same RTP session. anyone to impersonate another in the same RTP session.
A Translator that doesn't change the RTP/RTCP packets content can be A Translator that doesn't change the RTP/RTCP packets content can be
operated without the requiring it to have access to the security operated without the requiring it to have access to the security
contexts used to protect the RTP/RTCP traffic between the contexts used to protect the RTP/RTCP traffic between the
participants. participants.
3.5.2. Media Translator 3.5.2. Media Translator
In the context of multipoint communications a Media Translator is not In the context of multipoint communications a Media Translator is not
providing new mechanisms to establish a multipoint session. It is providing new mechanisms to establish a multipoint session. It is
more of an enabler, or facilitator, that ensures a given End Point or more of an enabler, or facilitator, that ensures a given endpoint or
a defined sub-set of End Points can participate in the session. a defined sub-set of endpoints can participate in the session.
If End Point B in Figure 11 were behind a limited network path, the If endpoint B in Figure 11 were behind a limited network path, the
Translator may perform media transcoding to allow the traffic Translator may perform media transcoding to allow the traffic
received from the other End Points to reach B without overloading the received from the other endpoints to reach B without overloading the
path. This transcoding can help the other End Points in the path. This transcoding can help the other endpoints in the multicast
multicast part of the session, by not requiring the quality part of the session, by not requiring the quality transmitted by A to
transmitted by A to be lowered to the bitrates that B is actually be lowered to the bitrates that B is actually capable of receiving
capable of receiving (and vice versa). (and vice versa).
3.6. Point to Multipoint Using the RFC 3550 Mixer Model 3.6. Point to Multipoint Using the RFC 3550 Mixer Model
Shortcut name: Topo-Mixer Shortcut name: Topo-Mixer
A Mixer is a middlebox that aggregates multiple RTP streams that are 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 part of a session by generating one or more new RTP streams and, in
most cases, by manipulating the media data. One common application most cases, by manipulating the media data. One common application
for a Mixer is to allow a participant to receive a session with a for a Mixer is to allow a participant to receive a session with a
reduced amount of resources. reduced amount of resources.
skipping to change at page 21, line 32 skipping to change at page 21, line 32
+---+ / Multi- \ | | +---+ +---+ / Multi- \ | | +---+
+ cast +->| Mixer | + cast +->| Mixer |
+---+ \ Network / | | +---+ +---+ \ Network / | | +---+
| C |<---\ / | |<---->| D | | C |<---\ / | |<---->| D |
+---+ \ / +-----------+ +---+ +---+ \ / +-----------+ +---+
+-----+ +-----+
Figure 13: Point to Multipoint Using the RFC 3550 Mixer Model Figure 13: Point to Multipoint Using the RFC 3550 Mixer Model
A Mixer can be viewed as a device terminating the RTP streams A Mixer can be viewed as a device terminating the RTP streams
received from other End Points in the same RTP session. Using the received from other endpoints in the same RTP session. Using the
media data carried in the received RTP streams, a Mixer generates media data carried in the received RTP streams, a Mixer generates
derived RTP streams that are sent to the receiving End Points. derived RTP streams that are sent to the receiving endpoints.
The content that the Mixer provides is the mixed aggregate of what 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 the Mixer receives over the PtP or PtM paths, which are part of the
same Communication Session. same Communication Session.
The Mixer creates the Media Source and the source RTP stream just The Mixer creates the Media Source and the source RTP stream just
like an End Point, as it mixes the content (often in the uncompressed like an endpoint, as it mixes the content (often in the uncompressed
domain) and then encodes and packetizes it for transmission to a domain) and then encodes and packetizes it for transmission to a
receiving endpoint. The CSRC Count (CC) and CSRC fields in the RTP receiving endpoint. The CSRC Count (CC) and CSRC fields in the RTP
header can be used to indicate the contributors to the newly header can be used to indicate the contributors to the newly
generated RTP stream. The SSRCs of the to-be-mixed streams on the 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 Mixer input appear as the CSRCs at the Mixer output. That output
stream uses a unique SSRC that identifies the Mixer's stream. The stream uses a unique SSRC that identifies the Mixer's stream. The
CSRC should be forwarded between the different End Points to allow CSRC should be forwarded between the different endpoints to allow for
for loop detection and identification of sources that are part of the loop detection and identification of sources that are part of the
Communication Session. Note that Section 7.1 of RFC 3550 requires Communication Session. Note that Section 7.1 of RFC 3550 requires
the SSRC space to be shared between domains for these reasons. This the SSRC space to be shared between domains for these reasons. This
also implies that any SDES information normally needs to be forwarded also implies that any SDES information normally needs to be forwarded
across the mixer. across the mixer.
The Mixer is responsible for generating RTCP packets in accordance The Mixer is responsible for generating RTCP packets in accordance
with its role. It is an RTP receiver and should therefore send RTCP with its role. It is an RTP receiver and should therefore send RTCP
receiver reports for the RTP streams it receives and terminates. In receiver reports for the RTP streams it receives and terminates. In
its role as an RTP sender, it should also generate RTCP sender 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 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 of RFC 3550, a Mixer must not forward RTCP unaltered between the two
domains. domains.
The Mixer depicted in Figure 13 is involved in three domains that The Mixer depicted in Figure 13 is involved in three domains that
need to be separated: the any source multicast network (including End need to be separated: the any source multicast network (including
Points A and C), End Point B, and End Point D. Assuming all four End endpoints A and C), endpoint B, and endpoint D. Assuming all four
Points in the conference are interested in receiving content from all endpoints in the conference are interested in receiving content from
other End Points, the Mixer produces different mixed RTP streams for all other endpoints, the Mixer produces different mixed RTP streams
B and D, as the one to B may contain content received from D, and 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 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 in each domain where it is the receiving entity and transmitter of
mixed content. mixed content.
In the multicast domain, a Mixer still needs to provide a mixed view 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 of the other domains. This makes the Mixer simpler to implement and
avoids any issues with advanced RTCP handling or loop detection, avoids any issues with advanced RTCP handling or loop detection,
which would be problematic if the Mixer were providing non-symmetric which would be problematic if the Mixer were providing non-symmetric
behavior. Please see Section 3.11 for more discussion on this topic. behavior. Please see Section 3.11 for more discussion on this topic.
The mixing operation, however, in each domain could potentially be The mixing operation, however, in each domain could potentially be
different. different.
A Mixer is responsible for receiving RTCP feedback messages and A Mixer is responsible for receiving RTCP feedback messages and
handling them appropriately. The definition of "appropriate" depends handling them appropriately. The definition of "appropriate" depends
on the message itself and the context. In some cases, the reception 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 of a codec-control message by the Mixer may result in the generation
and transmission of RTCP feedback messages by the Mixer to the End and transmission of RTCP feedback messages by the Mixer to the
Points in the other domain(s). In other cases, a message is handled endpoints in the other domain(s). In other cases, a message is
by the Mixer locally and therefore not forwarded to any other domain. 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 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) with individual unicast paths as depicted in Figure 14, the
Mixer model is very similar to the one discussed in Section 3.9 Mixer model is very similar to the one discussed in Section 3.9
below. Please see the discussion in Section 3.9 about the below. Please see the discussion in Section 3.9 about the
differences between these two models. differences between these two models.
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
skipping to change at page 23, line 22 skipping to change at page 23, line 22
Figure 14: RTP Mixer with Only Unicast Paths Figure 14: RTP Mixer with Only Unicast Paths
We now discuss in more detail the different mixing operations that a We now discuss in more detail the different mixing operations that a
mixer can perform and how they can affect RTP and RTCP behavior. mixer can perform and how they can affect RTP and RTCP behavior.
3.6.1. Media Mixing Mixer 3.6.1. Media Mixing Mixer
The media mixing mixer is likely the one that most think of when they 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 hear the term "mixer". Its basic mode of operation is that it
receives RTP streams from several End Points and selects the receives RTP streams from several endpoints and selects the stream(s)
stream(s) to be included in a media-domain mix. The selection can be to be included in a media-domain mix. The selection can be through
through static configuration or by dynamic, content dependent means static configuration or by dynamic, content dependent means such as
such as voice activation. The mixer then creates a single outgoing voice activation. The mixer then creates a single outgoing RTP
RTP stream from this mix. stream from this mix.
The most commonly deployed media mixer is probably the audio mixer, The most commonly deployed media mixer is probably the audio mixer,
used in voice conferencing, where the output consists of a mixture of used in voice conferencing, where the output consists of a mixture of
all the input audio signals; this needs minimal signalling to be all the input audio signals; this needs minimal signalling to be
successfully set up. From a signal processing viewpoint, audio successfully set up. From a signal processing viewpoint, audio
mixing is relatively straightforward and commonly possible for a mixing is relatively straightforward and commonly possible for a
reasonable number of End Points. Assume, for example, that one wants reasonable number of endpoints. Assume, for example, that one wants
to mix N streams from N different End Points. The mixer needs to to mix N streams from N different endpoints. The mixer needs to
decode those N streams, typically into the sample domain, and then decode those N streams, typically into the sample domain, and then
produce N or N+1 mixes. Different mixes are needed so that each produce N or N+1 mixes. Different mixes are needed so that each
contributing source gets a mix of all other sources except its own, contributing source gets a mix of all other sources except its own,
as this would result in an echo. When N is lower than the number of as this would result in an echo. When N is lower than the number of
all End points, one may produce a mix of all N streams for the group 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 that are currently not included in the mix, thus N+1 mixes. These
audio streams are then encoded again, RTP packetized and sent out. audio streams are then encoded again, RTP packetized and sent out.
In many cases, audio level normalization, noise suppression, and In many cases, audio level normalization, noise suppression, and
similar signal processing steps are also required or desirable before similar signal processing steps are also required or desirable before
the actual mixing process commences. the actual mixing process commences.
In video, the term "mixing" has a different interpretation than In video, the term "mixing" has a different interpretation than
audio. It is commonly used to refer to the process of spatially audio. It is commonly used to refer to the process of spatially
combining contributed video streams, which is also known as "tiling". combining contributed video streams, which is also known as "tiling".
The reconstructed, appropriately scaled down videos can be spatially The reconstructed, appropriately scaled down videos can be spatially
arranged in a set of tiles, each tile containing the video from an arranged in a set of tiles, each tile containing the video from an
End Point (typically showing a human participant). Tiles can be of endpoint (typically showing a human participant). Tiles can be of
different sizes, so that, for example, a particularly important different sizes, so that, for example, a particularly important
participant, or the loudest speaker, is being shown on in larger tile participant, or the loudest speaker, is being shown on in larger tile
than other participants. A self-view picture can be included in the than other participants. A self-view picture can be included in the
tiling, which can either be locally produced or be a feedback from a tiling, which can either be locally produced or be a feedback from a
mixer-received and reconstructed video image. Such remote loopback mixer-received and reconstructed video image. Such remote loopback
allows for confidence monitoring, i.e., it enables the participant to allows for confidence monitoring, i.e., it enables the participant to
see himself/herself in the same quality as other participants see see himself/herself in the same quality as other participants see
him/her. The tiling normally operates on reconstructed video in the him/her. The tiling normally operates on reconstructed video in the
sample domain. The tiled image is encoded, packetized, and sent by sample domain. The tiled image is encoded, packetized, and sent by
the mixer to the receiving End Points. It is possible that a the mixer to the receiving endpoints. It is possible that a
middlebox with media mixing duties contains only a single mixer of middlebox with media mixing duties contains only a single mixer of
the aforementioned type, in which case all participants necessarily the aforementioned type, in which case all participants necessarily
see the same tiled video, even if it is being sent over different RTP see the same tiled video, even if it is being sent over different RTP
streams. More common, however, are mixing arrangement where an streams. More common, however, are mixing arrangement where an
individual mixer is available for each outgoing port of the individual mixer is available for each outgoing port of the
middlebox, allowing individual compositions for each receiving End middlebox, allowing individual compositions for each receiving
Point (a feature commonly referred to as personalized layout). endpoint (a feature commonly referred to as personalized layout).
One problem with media mixing is that it consumes both large amounts One problem with media mixing is that it consumes both large amounts
of media processing resources (for the decoding and mixing process in of media processing resources (for the decoding and mixing process in
the uncompressed domain) and encoding resources (for the encoding of the uncompressed domain) and encoding resources (for the encoding of
the mixed signal). Another problem is the quality degradation the mixed signal). Another problem is the quality degradation
created by decoding and re-encoding the media, which is the result of created by decoding and re-encoding the media, which is the result of
the lossy nature of most commonly used media codecs. A third problem the lossy nature of most commonly used media codecs. A third problem
is the latency introduced by the media mixing, which can be is the latency introduced by the media mixing, which can be
substantial and annoyingly noticeable in case of video, or in case of substantial and annoyingly noticeable in case of video, or in case of
audio if that mixed audio is lip-sychronized with high latency video. audio if that mixed audio is lip-sychronized with high latency video.
The advantage of media mixing is that it is straightforward for the The advantage of media mixing is that it is straightforward for the
End Points to handle the single media stream (which includes the endpoints to handle the single media stream (which includes the mixed
mixed aggregate of many sources), as they don't need to handle aggregate of many sources), as they don't need to handle multiple
multiple decodings, local mixing and composition. In fact, mixers decodings, local mixing and composition. In fact, mixers were
were introduced in pre-RTP times so that legacy, single stream introduced in pre-RTP times so that legacy, single stream receiving
receiving endpoints (that, in some protocol environments, actually endpoints (that, in some protocol environments, actually didn't need
didn't need to be aware of the multipoint nature of the conference) to be aware of the multipoint nature of the conference) could
could successfully participate in what a user would recognize as a successfully participate in what a user would recognize as a
multiparty video conference. multiparty video conference.
+-A---------+ +-MIXER----------------------+ +-A---------+ +-MIXER----------------------+
| +-RTP1----| |-RTP1------+ +-----+ | | +-RTP1----| |-RTP1------+ +-----+ |
| | +-Audio-| |-Audio---+ | +---+ | | | | | +-Audio-| |-Audio---+ | +---+ | | |
| | | AA1|--------->|---------+-+-|DEC|->| | | | | | AA1|--------->|---------+-+-|DEC|->| | |
| | | |<---------|MA1 <----+ | +---+ | | | | | | |<---------|MA1 <----+ | +---+ | | |
| | | | |(BA1+CA1)|\| +---+ | | | | | | | |(BA1+CA1)|\| +---+ | | |
| | +-------| |---------+ +-|ENC|<-| B+C | | | | +-------| |---------+ +-|ENC|<-| B+C | |
| +---------| |-----------+ +---+ | | | | +---------| |-----------+ +---+ | | |
skipping to change at page 25, line 38 skipping to change at page 25, line 38
| | | |<---------|MA3 <----+ | +---+ | | | | | | |<---------|MA3 <----+ | +---+ | | |
| | +-------| |(AA1+BA1)|\| +---+ | | | | | +-------| |(AA1+BA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+B | | | +---------| |---------+ +-|ENC|<-| A+B | |
+-----------+ |-----------+ +---+ +-----+ | +-----------+ |-----------+ +---+ +-----+ |
+----------------------------+ +----------------------------+
Figure 15: Session and SSRC details for Media Mixer Figure 15: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be a very simple process, as 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 can be seen in Figure 15. The mixer presents one SSRC towards the
receiving End Point, e.g., MA1 to Peer A, where the associated stream receiving endpoint, e.g., MA1 to Peer A, where the associated stream
is the media mix of the other End Points. As each peer, in this 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 example, receives a different version of a mix from the mixer, there
is no actual relation between the different RTP sessions in terms of is no actual relation between the different RTP sessions in terms of
actual media or transport level information. There are, however, actual media or transport level information. There are, however,
common relationships between RTP1-RTP3, namely SSRC space and common relationships between RTP1-RTP3, namely SSRC space and
identity information. When A receives the MA1 stream which is a identity information. When A receives the MA1 stream which is a
combination of BA1 and CA1 streams, the mixer may include CSRC combination of BA1 and CA1 streams, the mixer may include CSRC
information in the MA1 stream to identify the contributing source BA1 information in the MA1 stream to identify the contributing source BA1
and CA1, allowing the receiver to identify the contributing sources and CA1, allowing the receiver to identify the contributing sources
even if this were not possible through the media itself or through even if this were not possible through the media itself or through
other signaling means. other signaling means.
The CSRC has, in turn, utility in RTP extensions, like the Mixer to The CSRC has, in turn, utility in RTP extensions, like the Mixer to
Client audio levels RTP header extension [RFC6465]. If the SSRCs Client audio levels RTP header extension [RFC6465]. If the SSRCs
from the End Point to mixer paths are used as CSRCs in another RTP 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 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 have a common SSRC space. At this stage, the mixer also needs to
consider which RTCP information it needs to expose in the different consider which RTCP information it needs to expose in the different
paths. In the above scenario, a mixer would normally expose nothing paths. In the above scenario, a mixer would normally expose nothing
more than the Source Description (SDES) information and RTCP BYE for more than the Source Description (SDES) information and RTCP BYE for
a CSRC leaving the session. The main goal would be to enable the a CSRC leaving the session. The main goal would be to enable the
correct binding against the application logic and other information correct binding against the application logic and other information
sources. This also enables loop detection in the RTP session. sources. This also enables loop detection in the RTP session.
3.6.2. Media Switching 3.6.2. Media Switching
skipping to change at page 26, line 49 skipping to change at page 26, line 49
bitstream. bitstream.
To achieve a coherent RTP stream from the mixer's SSRC, the mixer 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 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 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 sequence number must be the next in the sequence of outgoing packets
it sent. Third, the RTP timestamp value needs to be adjusted using it sent. Third, the RTP timestamp value needs to be adjusted using
an offset that changes each time one switches media source. Finally, an offset that changes each time one switches media source. Finally,
depending on the negotiation of the RTP payload type, the value depending on the negotiation of the RTP payload type, the value
representing this particular RTP payload configuration may have to be representing this particular RTP payload configuration may have to be
changed if the different End Point-to-mixer paths have not arrived on changed if the different endpoint-to-mixer paths have not arrived on
the same numbering for a given configuration. This also requires the same numbering for a given configuration. This also requires
that the different End Points support a common set of codecs, that the different endpoints support a common set of codecs,
otherwise media transcoding for codec compatibility would still be otherwise media transcoding for codec compatibility would still be
required. required.
We now consider the operation of a media switching mixer that We now consider the operation of a media switching mixer that
supports a video conference with six participating End Points (A-F) supports a video conference with six participating endpoints (A-F)
where the two most recent speakers in the conference are shown to where the two most recent speakers in the conference are shown to
each receiving End Point. The mixer has thus two SSRCs sending video each receiving endpoint. The mixer has thus two SSRCs sending video
to each peer, and each peer is capable of locally handling two video to each peer, and each peer is capable of locally handling two video
streams simultaneously. streams simultaneously.
+-A---------+ +-MIXER----------------------+ +-A---------+ +-MIXER----------------------+
| +-RTP1----| |-RTP1------+ +-----+ | | +-RTP1----| |-RTP1------+ +-----+ |
| | +-Video-| |-Video---+ | | | | | | +-Video-| |-Video---+ | | | |
| | | AV1|------------>|---------+-+------->| S | | | | | AV1|------------>|---------+-+------->| S | |
| | | |<------------|MV1 <----+-+-BV1----| W | | | | | |<------------|MV1 <----+-+-BV1----| W | |
| | | |<------------|MV2 <----+-+-EV1----| I | | | | | |<------------|MV2 <----+-+-EV1----| I | |
| | +-------| |---------+ | | T | | | | +-------| |---------+ | | T | |
skipping to change at page 27, line 50 skipping to change at page 27, line 50
| | | |<------------|MV12 <---+-+-EV1----| | | | | | |<------------|MV12 <---+-+-EV1----| | |
| | +-------| |---------+ | | | | | | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ | | +---------| |-----------+ +-----+ |
+-----------+ +----------------------------+ +-----------+ +----------------------------+
Figure 16: Media Switching RTP Mixer Figure 16: Media Switching RTP Mixer
The Media Switching RTP mixer can, similarly to the Media Mixing The Media Switching RTP mixer can, similarly to the Media Mixing
Mixer, reduce the bit-rate required for media transmission towards Mixer, reduce the bit-rate required for media transmission towards
the different peers by selecting and forwarding only a sub-set of RTP the different peers by selecting and forwarding only a sub-set of RTP
streams it receives from the sending End Points. In cases the mixer streams it receives from the sending endpoints. In cases the mixer
receives simulcast transmissions or a scalable encoding of the media receives simulcast transmissions or a scalable encoding of the media
source, the mixer has more degrees of freedom to select streams or source, the mixer has more degrees of freedom to select streams or
sub-sets of stream to forward to a receiving End Point, both based on sub-sets of stream to forward to a receiving endpoint, both based on
transport or End Point restrictions as well as application logic. transport or endpoint restrictions as well as application logic.
To ensure that a media receiver in an End Point can correctly decode 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 the media in the RTP stream after a switch, a codec that uses
temporal prediction needs to start its decoding from independent temporal prediction needs to start its decoding from independent
refresh points, or points in the bitstream offering similar refresh points, or points in the bitstream offering similar
functionality (like "dirty refresh points"). For some codecs, for functionality (like "dirty refresh points"). For some codecs, for
example frame based speech and audio codecs, this is easily achieved example frame based speech and audio codecs, this is easily achieved
by starting the decoding at RTP packet boundaries, as each packet by starting the decoding at RTP packet boundaries, as each packet
boundary provides a refresh point (assuming proper packetization on boundary provides a refresh point (assuming proper packetization on
the encoder side). For other codecs, particularly in video, refresh the encoder side). For other codecs, particularly in video, refresh
points are less common in the bitstream or may not be present at all points are less common in the bitstream or may not be present at all
without an explicit request to the respective encoder. The Full without an explicit request to the respective encoder. The Full
Intra Request [RFC5104] RTCP codec control message has been defined Intra Request [RFC5104] RTCP codec control message has been defined
for this purpose. for this purpose.
In this type of mixer one could consider to fully terminate the RTP In this type of mixer one could consider to fully terminate the RTP
sessions between the different End Point and mixer paths. The same sessions between the different endpoint and mixer paths. The same
arguments and considerations as discussed in Section 3.9 need to be arguments and considerations as discussed in Section 3.9 need to be
taken into consideration and apply here. taken into consideration and apply here.
3.7. Selective Forwarding Middlebox 3.7. Selective Forwarding Middlebox
Another method for handling media in the RTP mixer is to "project", Another method for handling media in the RTP mixer is to "project",
or make available, all potential RTP sources (SSRCs) into a per-End or make available, all potential RTP sources (SSRCs) into a per-
Point, independent RTP session. The middlebox can select which of endpoint, independent RTP session. The middlebox can select which of
the potential sources that are currently actively transmitting media the potential sources that are currently actively transmitting media
will be sent to each of the End Points. This is similar to the 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. switching Mixer but has some important differences in RTP details.
+-A---------+ +-Middlebox-----------------+ +-A---------+ +-Middlebox-----------------+
| +-RTP1----| |-RTP1------+ +-----+ | | +-RTP1----| |-RTP1------+ +-----+ |
| | +-Video-| |-Video---+ | | | | | | +-Video-| |-Video---+ | | | |
| | | AV1|------------>|---------+-+------>| | | | | | AV1|------------>|---------+-+------>| | |
| | | |<------------|BV1 <----+-+-------| S | | | | | |<------------|BV1 <----+-+-------| S | |
| | | |<------------|CV1 <----+-+-------| W | | | | | |<------------|CV1 <----+-+-------| W | |
| | | |<------------|DV1 <----+-+-------| I | | | | | |<------------|DV1 <----+-+-------| I | |
| | | |<------------|EV1 <----+-+-------| T | | | | | |<------------|EV1 <----+-+-------| T | |
skipping to change at page 29, line 44 skipping to change at page 29, line 44
| | | FV1|------------>|---------+-+------>| | | | | | FV1|------------>|---------+-+------>| | |
| | | |<------------|AV1 <----+-+-------| | | | | | |<------------|AV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | | | | | | : : : |: : : : : : : : :| | |
| | | |<------------|EV1 <----+-+-------| | | | | | |<------------|EV1 <----+-+-------| | |
| | +-------| |---------+ | | | | | | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ | | +---------| |-----------+ +-----+ |
+-----------+ +---------------------------+ +-----------+ +---------------------------+
Figure 17: Selective Forwarding Middlebox Figure 17: Selective Forwarding Middlebox
In the six End Point conference depicted above in (Figure 17) one can In the six endpoint conference depicted above in (Figure 17) one can
see that End Point A is aware of five incoming SSRCs, BV1-FV1. If 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 this middlebox intends to have a similar behavior as in Section 3.6.2
where the mixer provides the End Points with the two latest speaking where the mixer provides the endpoints with the two latest speaking
End Points, then only two out of these five SSRCs need concurrently endpoints, then only two out of these five SSRCs need concurrently
transmit media to A. As the middlebox selects the source in the transmit media to A. As the middlebox selects the source in the
different RTP sessions that transmit media to the End points, each different RTP sessions that transmit media to the endpoints, each RTP
RTP stream requires rewriting of certain RTP header fields when being stream requires rewriting of certain RTP header fields when being
projected from one session into another. In particular, the sequence projected from one session into another. In particular, the sequence
number needs to be consecutively incremented based on the packet number needs to be consecutively incremented based on the packet
actually being transmitted in each RTP session. Therefore, the RTP actually being transmitted in each RTP session. Therefore, the RTP
sequence number offset will change each time a source is turned on in sequence number offset will change each time a source is turned on in
a RTP session. The timestamp (possibly offset) stays the same. a RTP session. The timestamp (possibly offset) stays the same.
As the RTP sessions are independent, the SSRC numbers used can also As the RTP sessions are independent, the SSRC numbers used can also
be handled independently, thereby bypassing the requirement for SSRC be handled independently, thereby bypassing the requirement for SSRC
collision detection and avoidance. On the other hand, tools such as collision detection and avoidance. On the other hand, tools such as
remapping tables between the RTP sessions are required. For example, remapping tables between the RTP sessions are required. For example,
the RTP stream that is being sent by End Point B to the middlebox 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 (BV1) may use an SSRC value of 12345678. When that RTP stream is
sent to End Point F by the middlebox, it can use any SSRC value, e.g. sent to endpoint F by the middlebox, it can use any SSRC value, e.g.
87654321. As a result, each End Point may have a different view of 87654321. As a result, each endpoint may have a different view of
the application usage of a particular SSRC. Any RTP level identity the application usage of a particular SSRC. Any RTP level identity
information, such as SDES items also needs to update the SSRC information, such as SDES items also needs to update the SSRC
referenced, if the included SDES items are intended to be global. referenced, if the included SDES items are intended to be global.
Thus the application must not use SSRC as references to RTP streams Thus the application must not use SSRC as references to RTP streams
when communicating with other peers directly. This also affects loop when communicating with other peers directly. This also affects loop
detection which will fail to work, as there is no common namespace detection which will fail to work, as there is no common namespace
and identities across the different legs in the communication session and identities across the different legs in the communication session
on RTP level. Instead this responsibility falls onto higher layers. on RTP level. Instead this responsibility falls onto higher layers.
The middlebox is also responsible to receive any RTCP codec control The middlebox is also responsible to receive any RTCP codec control
requests coming from an End Point, and decide if it can act on the requests coming from an endpoint, and decide if it can act on the
request locally or needs to translate the request into the RTP request locally or needs to translate the request into the RTP
session that contains the media source. Both End Points and the session that contains the media source. Both endpoints and the
middlebox need to implement conference related codec control middlebox need to implement conference related codec control
functionalities to provide a good experience. Commonly used are Full functionalities to provide a good experience. Commonly used are Full
Intra Request to request from the media source to provide switching Intra Request to request from the media source to provide switching
points between the sources, and Temporary Maximum Media Bit-rate points between the sources, and Temporary Maximum Media Bit-rate
Request (TMMBR) to enable the middlebox to aggregate congestion Request (TMMBR) to enable the middlebox to aggregate congestion
control responses towards the media source so to enable it to adjust control responses towards the media source so to enable it to adjust
its bit-rate (obviously only in case the limitation is not in the its bit-rate (obviously only in case the limitation is not in the
source to middlebox link). source to middlebox link).
The selective forwarding middlebox has been introduced in recently The selective forwarding middlebox has been introduced in recently
developed videoconferencing systems in conjunction with, and to developed videoconferencing systems in conjunction with, and to
capitalize on, scalable video coding as well as simulcasting. An capitalize on, scalable video coding as well as simulcasting. An
example of scalable video coding is Annex G of H.264, but other example of scalable video coding is Annex G of H.264, but other
codecs, including H.264 AVC and VP8 also exhibit scalability, albeit codecs, including H.264 AVC and VP8 also exhibit scalability, albeit
only in the temporal dimension. In both scalable coding and only in the temporal dimension. In both scalable coding and
simulcast cases the video signal is represented by a set of two or simulcast cases the video signal is represented by a set of two or
more bitstreams, providing a corresponding number of distinct more bitstreams, providing a corresponding number of distinct
fidelity points. The middlebox selects which parts of a scalable fidelity points. The middlebox selects which parts of a scalable
bitstream (or which bitstream, in the case of simulcasting) to bitstream (or which bitstream, in the case of simulcasting) to
forward to each of the receiving End Points. The decision may be forward to each of the receiving endpoints. The decision may be
driven by a number of factors, such as available bit rate, desired driven by a number of factors, such as available bit rate, desired
layout, etc. Contrary to transcoding MCUs, these "Selective layout, etc. Contrary to transcoding MCUs, these "Selective
Forwarding Units" (SFUs) have extremely low delay, and provide Forwarding Units" (SFUs) have extremely low delay, and provide
features that are typically associated with high-end systems features that are typically associated with high-end systems
(personalized layout, error localization) without any signal (personalized layout, error localization) without any signal
processing at the middlebox. They are also capable of scaling to a processing at the middlebox. They are also capable of scaling to a
large number of concurrent users, and--due to their very low delay-- large number of concurrent users, and--due to their very low delay--
can also be cascaded. can also be cascaded.
This version of the middlebox also puts different requirements on the This version of the middlebox also puts different requirements on the
End Point when it comes to decoder instances and handling of the RTP endpoint when it comes to decoder instances and handling of the RTP
streams providing media. As each projected SSRC can, at any time, streams providing media. As each projected SSRC can, at any time,
provide media, the End Point either needs to be able to handle as provide media, the endpoint either needs to be able to handle as many
many decoder instances as the middlebox received, or have efficient decoder instances as the middlebox received, or have efficient
switching of decoder contexts in a more limited set of actual decoder switching of decoder contexts in a more limited set of actual decoder
instances to cope with the switches. The application also gets more instances to cope with the switches. The application also gets more
responsibility to update how the media provided is to be presented to responsibility to update how the media provided is to be presented to
the user. the user.
Note that this topology could potentially be seen as a media Note that this topology could potentially be seen as a media
translator which include an on/off logic as part of its media translator which include an on/off logic as part of its media
translation. The main difference would be a common global SSRC space translation. The main difference would be a common global SSRC space
in the case of the Media Translator and the mapped one used in the in the case of the Media Translator and the mapped one used in the
above. It also has mixer aspects, as the streams it provides are not above. It also has mixer aspects, as the streams it provides are not
skipping to change at page 32, line 35 skipping to change at page 32, line 35
presentation mode or explicit floor control) are known to exist as presentation mode or explicit floor control) are known to exist as
well. well.
The video switching MCU may also perform media translation to modify The video switching MCU may also perform media translation to modify
the content in bit-rate, encoding, or resolution. However, it still the content in bit-rate, encoding, or resolution. However, it still
may indicate the original sender of the content through the SSRC. In may indicate the original sender of the content through the SSRC. In
this case, the values of the CC and CSRC fields are retained. this case, the values of the CC and CSRC fields are retained.
If not terminating RTP, the RTCP Sender Reports are forwarded for the If not terminating RTP, the RTCP Sender Reports are forwarded for the
currently selected sender. All RTCP Receiver Reports are freely currently selected sender. All RTCP Receiver Reports are freely
forwarded between the End points. In addition, the MCU may also forwarded between the endpoints. In addition, the MCU may also
originate RTCP control traffic in order to control the session and/or originate RTCP control traffic in order to control the session and/or
report on status from its viewpoint. report on status from its viewpoint.
The video switching MCU has most of the attributes of a Translator. The video switching MCU has most of the attributes of a Translator.
However, its stream selection is a mixing behavior. This behavior However, its stream selection is a mixing behavior. This behavior
has some RTP and RTCP issues associated with it. The suppression of has some RTP and RTCP issues associated with it. The suppression of
all but one RTP stream results in most participants seeing only a 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 subset of the sent RTP streams at any given time, often a single RTP
stream per conference. Therefore, RTCP Receiver Reports only report stream per conference. Therefore, RTCP Receiver Reports only report
on these RTP streams. Consequently, the End Points emitting RTP on these RTP streams. Consequently, the endpoints emitting RTP
streams that are not currently forwarded receive a view of the streams that are not currently forwarded receive a view of the
session that indicates their RTP streams disappear somewhere en session that indicates their RTP streams disappear somewhere en
route. This makes the use of RTCP for congestion control, or any route. This makes the use of RTCP for congestion control, or any
type of quality reporting, very problematic. type of quality reporting, very problematic.
To avoid the aforementioned issues, the MCU needs to implement two 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 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 forward the selected RTP stream under its own SSRC and with the
appropriate CSRC values. Second, the MCU needs to modify the RTCP appropriate CSRC values. Second, the MCU needs to modify the RTCP
RRs it forwards between the domains. As a result, it is recommended RRs it forwards between the domains. As a result, it is recommended
skipping to change at page 33, line 28 skipping to change at page 33, line 28
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B | | A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+ +---+ | Control | +---+
| Unit | | Unit |
+---+ | (MCU) | +---+ +---+ | (MCU) | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 19: Point to Multipoint Using Content Modifying MCUs Figure 19: Point to Multipoint Using Content Modifying MCUs
In this PtM scenario, each End Point runs an RTP point-to-point In this PtM scenario, each endpoint runs an RTP point-to-point
session between itself and the MCU. This is a very commonly deployed session between itself and the MCU. This is a very commonly deployed
topology in multipoint video conferencing. The content that the MCU topology in multipoint video conferencing. The content that the MCU
provides to each participant is either: provides to each participant is either:
a. a selection of the content received from the other End Points, or 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 b. the mixed aggregate of what the MCU receives from the other PtP
paths, which are part of the same Communication Session. paths, which are part of the same Communication Session.
In case (a), the MCU may modify the content in terms of bit-rate, In case (a), the MCU may modify the content in terms of bit-rate,
encoding format, or resolution. No explicit RTP mechanism is used to encoding format, or resolution. No explicit RTP mechanism is used to
establish the relationship between the original RTP stream of the establish the relationship between the original RTP stream of the
media being sent RTP stream the MCU sends. In other words, the media being sent RTP stream the MCU sends. In other words, the
outgoing RTP streams typically use a different SSRC, and may well use outgoing RTP streams typically use a different SSRC, and may well use
a different payload type (PT), even if this different PT happens to 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 be mapped to the same media type. This is a result of the
individually negotiated RTP session for each End Point. individually negotiated RTP session for each endpoint.
In case (b), the MCU is the Media Source and generates the Source RTP 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 Stream as it mixes the received content and then encodes and
packetizes it for transmission to an End Point. According to RTP packetizes it for transmission to an endpoint. According to RTP
[RFC3550], the SSRC of the contributors are to be signalled using the [RFC3550], the SSRC of the contributors are to be signalled using the
CSRC/CC mechanism. In practice, today, most deployed MCUs do not CSRC/CC mechanism. In practice, today, most deployed MCUs do not
implement this feature. Instead, the identification of the End implement this feature. Instead, the identification of the endpoints
Points whose content is included in the Mixer's output is not whose content is included in the Mixer's output is not indicated
indicated through any explicit RTP mechanism. That is, most deployed through any explicit RTP mechanism. That is, most deployed MCUs set
MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby the CSRC Count (CC) field in the RTP header to zero, thereby
indicating no available CSRC information, even if they could identify indicating no available CSRC information, even if they could identify
the original sending End Points as suggested in RTP. the original sending endpoints as suggested in RTP.
The main feature that sets this topology apart from what RFC 3550 The main feature that sets this topology apart from what RFC 3550
describes is the breaking of the common RTP session across the describes is the breaking of the common RTP session across the
centralized device, such as the MCU. This results in the loss of centralized device, such as the MCU. This results in the loss of
explicit RTP-level indication of all participants. If one were using explicit RTP-level indication of all participants. If one were using
the mechanisms available in RTP and RTCP to signal this explicitly, the mechanisms available in RTP and RTCP to signal this explicitly,
the topology would follow the approach of an RTP Mixer. The lack of the topology would follow the approach of an RTP Mixer. The lack of
explicit indication has at least the following potential problems: explicit indication has at least the following potential problems:
1. Loop detection cannot be performed on the RTP level. When 1. Loop detection cannot be performed on the RTP level. When
skipping to change at page 35, line 16 skipping to change at page 35, line 16
video conference software tools used over the MBONE in the late video conference software tools used over the MBONE in the late
1990s. 1990s.
An example for such a split component terminal is depicted in An example for such a split component terminal is depicted in
Figure 20. Within split component terminal A, at least audio and Figure 20. Within split component terminal A, at least audio and
video subunits are addressed by their own network addresses. In some video subunits are addressed by their own network addresses. In some
of these systems, the control stack subunit may also have its own of these systems, the control stack subunit may also have its own
network address. network address.
From an RTP viewpoint, each of the subunits terminates RTP, and acts From an RTP viewpoint, each of the subunits terminates RTP, and acts
as an End Point in the sense that each subunit includes its own, as an endpoint in the sense that each subunit includes its own,
independent RTP stack. However, as the subunits are semantically independent RTP stack. However, as the subunits are semantically
part of the same terminal, it is appropriate that this semantic part of the same terminal, it is appropriate that this semantic
relationship is expressed in RTCP protocol elements, namely in the relationship is expressed in RTCP protocol elements, namely in the
CNAME. CNAME.
+---------------------+ +---------------------+
| Endpoint A | | Endpoint A |
| Local Area Network | | Local Area Network |
| +------------+ | | +------------+ |
| +->| Audio |<+-RTP---\ | +->| Audio |<+-RTP---\
skipping to change at page 35, line 47 skipping to change at page 35, line 47
It is further sensible that the subunits share a common clock from It is further sensible that the subunits share a common clock from
which RTP and RTCP clocks are derived, to facilitate synchronization which RTP and RTCP clocks are derived, to facilitate synchronization
and avoid clock drift. and avoid clock drift.
To indicate that audio and video Source Streams generated by To indicate that audio and video Source Streams generated by
different sub-units share a common clock, and can be synchronized, different sub-units share a common clock, and can be synchronized,
the RTP streams generated from those Source Streams need to include 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 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 for RTP flows carried in different transport-layer flows is entirely
normal for RTP and RTCP senders, and fully compliant RTP End Points, normal for RTP and RTCP senders, and fully compliant RTP endpoints,
middle-boxes, and other tools should have no problem with this. middle-boxes, and other tools should have no problem with this.
However, outside of the split component terminal scenario (and However, outside of the split component terminal scenario (and
perhaps a multi-homed End Point scenario, which is not further perhaps a multi-homed endpoint scenario, which is not further
discussed herein), the use of a common CNAME in RTP streams sent from 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 separate endpoints (as opposed to a common CNAME for RTP streams sent
on different transport layer flows between two endpoints) is rare. on different transport layer flows between two endpoints) is rare.
It has been reported that at least some third party tools like some It has been reported that at least some third party tools like some
network monitors do not handle endpoints that use of a common CNAME network monitors do not handle endpoints that use of a common CNAME
across multiple transport layer flows gracefully: they report an across multiple transport layer flows gracefully: they report an
error condition that two separate End Points are using the same error condition that two separate endpoints are using the same CNAME.
CNAME. Depending on the sophistication of the support staff, such Depending on the sophistication of the support staff, such erroneous
erroneous reports can lead to support issues. reports can lead to support issues.
Aforementioned support issue can sometimes be avoided if each of the Aforementioned support issue can sometimes be avoided if each of the
subunits of a split component terminal is configured to use a subunits of a split component terminal is configured to use a
different CNAME, with the synchronization between the RTP streams different CNAME, with the synchronization between the RTP streams
being indicated by some non-RTP signaling channel rather than using a being indicated by some non-RTP signaling channel rather than using a
common CNAME sent in RTCP. This complicates the signaling, common CNAME sent in RTCP. This complicates the signaling,
especially in cases where there are multiple SSRCs in use with especially in cases where there are multiple SSRCs in use with
complex synchronization requirements, as is the same in many current complex synchronization requirements, as is the same in many current
telepresence systems. Unless one uses RTCP terminating topologies telepresence systems. Unless one uses RTCP terminating topologies
such as Topo-RTCP-terminating-MCU, sessions involving more than one such as Topo-RTCP-terminating-MCU, sessions involving more than one
skipping to change at page 38, line 40 skipping to change at page 38, line 40
Closest to true IP-multicast-based, all-to-all transmission comes Closest to true IP-multicast-based, all-to-all transmission comes
perhaps the transport Translator function called "relay" in perhaps the transport Translator function called "relay" in
Section 3.5, as well as the Mesh with joint RTP sessions. Media Section 3.5, as well as the Mesh with joint RTP sessions. Media
Translators, Mesh with independent RTP Sessions, Mixers, SFUs and the Translators, Mesh with independent RTP Sessions, Mixers, SFUs and the
MCU variants do not provide a fully meshed forwarding on the MCU variants do not provide a fully meshed forwarding on the
transport level; instead, they only allow limited forwarding of transport level; instead, they only allow limited forwarding of
content from the other session participants. content from the other session participants.
The "all to all media transmission" requires that any media The "all to all media transmission" requires that any media
transmitting End Point considers the path to the least capable transmitting endpoint considers the path to the least capable
receiving End Point. Otherwise, the media transmissions may overload receiving endpoint. Otherwise, the media transmissions may overload
that path. Therefore, a sending End Point needs to monitor the path that path. Therefore, a sending endpoint needs to monitor the path
from itself to any of the receiving End Points, to detect the from itself to any of the receiving endpoints, to detect the
currently least capable receiver, and adapt its sending rate currently least capable receiver, and adapt its sending rate
accordingly. As multiple End Points may send simultaneously, the accordingly. As multiple endpoints may send simultaneously, the
available resources may vary. RTCP's Receiver Reports help available resources may vary. RTCP's Receiver Reports help
performing this monitoring, at least on a medium time scale. performing this monitoring, at least on a medium time scale.
The resource consumption for performing all to all transmission The resource consumption for performing all to all transmission
varies depending with the topology. Both ASM and SSM have the varies depending with the topology. Both ASM and SSM have the
benefit that only one copy of each packet traverses a particular benefit that only one copy of each packet traverses a particular
link. Using a relay causes the transmission of one copy of a packet link. Using a relay causes the transmission of one copy of a packet
per End Point-to-relay path and packet transmitted. However, in most per endpoint-to-relay path and packet transmitted. However, in most
cases the links carrying the multiple copies will be the ones close 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 to the relay (which can be assumed to be part of the network
infrastructure with good connectivity to the backbone), rather than infrastructure with good connectivity to the backbone), rather than
the End Points (which may be behind slower access links). The Mesh the endpoints (which may be behind slower access links). The Mesh
causes N-1 streams of transmitted packets to traverse the first hop causes N-1 streams of transmitted packets to traverse the first hop
link from the End Point, in an N End Point mesh. How long the link from the endpoint, in an N endpoint mesh. How long the
different paths are common, is highly situation dependent. different paths are common, is highly situation dependent.
The transmission of RTCP by design adapts to any changes in the The transmission of RTCP by design adapts to any changes in the
number of participants due to the transmission algorithm, defined in number of participants due to the transmission algorithm, defined in
the RTP specification [RFC3550], and the extensions in AVPF [RFC4585] the RTP specification [RFC3550], and the extensions in AVPF [RFC4585]
(when applicable). That way, the resources utilized for RTCP stay (when applicable). That way, the resources utilized for RTCP stay
within the bounds configured for the session. within the bounds configured for the session.
4.2. Transport or Media Interoperability 4.2. Transport or Media Interoperability
All Translators, Mixers, and RTCP-terminating MCU, and Mesh with All Translators, Mixers, and RTCP-terminating MCU, and Mesh with
individual RTP sessions, allow changing the media encoding or the individual RTP sessions, allow changing the media encoding or the
transport to other properties of the other domain, thereby providing transport to other properties of the other domain, thereby providing
extended interoperability in cases where the End Points lack a common extended interoperability in cases where the endpoints lack a common
set of media codecs and/or transport protocols. Selective Forwarding set of media codecs and/or transport protocols. Selective Forwarding
Middleboxes can adopt the transport, and (at least) selectively Middleboxes can adopt the transport, and (at least) selectively
forward the encoded streams that match a receiving End Point's forward the encoded streams that match a receiving endpoint's
capability. It requires an additional translator to change the media capability. It requires an additional translator to change the media
encoding if the encoded streams do not match the receiving End encoding if the encoded streams do not match the receiving endpoint's
Point's capabilities. capabilities.
4.3. Per Domain Bit-Rate Adaptation 4.3. Per Domain Bit-Rate Adaptation
End Points are often connected to each other with a heterogeneous set Endpoints are often connected to each other with a heterogeneous set
of paths. This makes congestion control in a Point to Multipoint set of paths. This makes congestion control in a Point to Multipoint set
problematic. For the ASM, SSM, Mesh with common RTP session, and problematic. For the ASM, SSM, Mesh with common RTP session, and
Transport Relay scenario, each individual sending End Point has to Transport Relay scenario, each individual sending endpoint has to
adapt to the receiving End Point behind the least capable path, adapt to the receiving endpoint behind the least capable path,
yielding suboptimal quality for the End Points behind the more yielding suboptimal quality for the endpoints behind the more capable
capable paths. This is no longer an issue when Media Translators, paths. This is no longer an issue when Media Translators, Mixers,
Mixers, SFM or MCUs are involved, as each End Point only needs to SFM or MCUs are involved, as each endpoint only needs to adapt to the
adapt to the slowest path within its own domain. The Translator, slowest path within its own domain. The Translator, Mixer, SFM, or
Mixer, SFM, or MCU topologies all require their respective outgoing MCU topologies all require their respective outgoing RTP streams to
RTP streams to adjust the bit-rate, packet-rate, etc., to adapt to adjust the bit-rate, packet-rate, etc., to adapt to the least capable
the least capable path in each of the other domains. That way one path in each of the other domains. That way one can avoid lowering
can avoid lowering the quality to the least-capable End Point in all the quality to the least-capable endpoint in all the domains at the
the domains at the cost (complexity, delay, equipment) of the Mixer, cost (complexity, delay, equipment) of the Mixer, SFM or Translator,
SFM or Translator, and potentially media sender (multicast/layered and potentially media sender (multicast/layered encoding and sending
encoding and sending the different representations). the different representations).
4.4. Aggregation of Media 4.4. Aggregation of Media
In the all-to-all media property mentioned above and provided by ASM, In the all-to-all media property mentioned above and provided by ASM,
SSM, Mesh with common RTP session, and relay, all simultaneous media SSM, Mesh with common RTP session, and relay, all simultaneous media
transmissions share the available bit-rate. For End Points with transmissions share the available bit-rate. For endpoints with
limited reception capabilities, this may result in a situation where limited reception capabilities, this may result in a situation where
even a minimal acceptable media quality cannot be accomplished, even a minimal acceptable media quality cannot be accomplished,
because multiple RTP streams need to share the same resources. One because multiple RTP streams need to share the same resources. One
solution to this problem is to provide for a Mixer, or MCU to solution to this problem is to provide for a Mixer, or MCU to
aggregate the multiple RTP streams into a single one, where the aggregate the multiple RTP streams into a single one, where the
single RTP stream takes up less resources in terms of bit-rate. This single RTP stream takes up less resources in terms of bit-rate. This
aggregation can be performed according to different methods. Mixing aggregation can be performed according to different methods. Mixing
or selection are two common methods. Selection is almost always or selection are two common methods. Selection is almost always
possible and easy to implement. Mixing requires resources in the possible and easy to implement. Mixing requires resources in the
mixer, and may be relatively easy and not impairing the quality too mixer, and may be relatively easy and not impairing the quality too
badly (audio) or quite difficult (video tiling, which is not only badly (audio) or quite difficult (video tiling, which is not only
computationally complex but also reduces the pixel count per stream, computationally complex but also reduces the pixel count per stream,
with corresponding loss in perceptual quality). with corresponding loss in perceptual quality).
4.5. View of All Session Participants 4.5. View of All Session Participants
The RTP protocol includes functionality to identify the session The RTP protocol includes functionality to identify the session
participants through the use of the SSRC and CSRC fields. In participants through the use of the SSRC and CSRC fields. In
addition, it is capable of carrying some further identity information addition, it is capable of carrying some further identity information
about these participants using the RTCP Source Descriptors (SDES). about these participants using the RTCP Source Descriptors (SDES).
In topologies that provide a full all-to-all functionality, i.e. In topologies that provide a full all-to-all functionality, i.e. ASM,
ASM, Mesh with common RTP session, Relay a compliant RTP Mesh with common RTP session, Relay a compliant RTP implementation
implementation offers the functionality directly as specified in RTP. offers the functionality directly as specified in RTP. In topologies
In topologies that do not offer all-to-all communication, it is that do not offer all-to-all communication, it is necessary that RTCP
necessary that RTCP is handled correctly in domain bridging function. is handled correctly in domain bridging function. RTP includes
RTP includes explicit specification text for Translators and Mixers, explicit specification text for Translators and Mixers, and for SFMs
and for SFMs the required functionality can be derived from that the required functionality can be derived from that text. However,
text. However, the MCU described in Section 3.8 cannot offer the the MCU described in Section 3.8 cannot offer the full functionality
full functionality for session participant identification through RTP for session participant identification through RTP means. The
means. The topologies that create independent RTP sessions per End topologies that create independent RTP sessions per endpoint or pair
Point or pair of End Points, like Back-to-Back RTP session, MESH with of endpoints, like Back-to-Back RTP session, MESH with independent
independent RTP sessions, and the RTCP terminating MCU RTCP RTP sessions, and the RTCP terminating MCU RTCP terminating MCU
terminating MCU (Section 3.9) do not support RTP based identification (Section 3.9), with an exception of SFM, do not support RTP based
of session participants. In all those cases, other non-RTP based identification of session participants. In all those cases, other
mechanisms need to be implemented if such knowledge is required or non-RTP based mechanisms need to be implemented if such knowledge is
desirable. required or desirable. When it comes to SFM the SSRC name space is
not necessarily joint, instead identification will require knowledge
of SSRC/CSRC mappings that the SFM performed, see Section 3.7.
4.6. Loop Detection 4.6. Loop Detection
In complex topologies with multiple interconnected domains, it is In complex topologies with multiple interconnected domains, it is
possible to unintentionally form media loops. RTP and RTCP support possible to unintentionally form media loops. RTP and RTCP support
detecting such loops, as long as the SSRC and CSRC identities are detecting such loops, as long as the SSRC and CSRC identities are
maintained and correctly set in forwarded packets. Loop detection maintained and correctly set in forwarded packets. Loop detection
will work in ASM, SSM, Mesh with joint RTP session, and Relay. It is will work in ASM, SSM, Mesh with joint RTP session, and Relay. It is
likely that loop detection works for the video switching MCU likely that loop detection works for the video switching MCU
Section 3.8, at least as long as it forwards the RTCP between the End Section 3.8, at least as long as it forwards the RTCP between the
Points. However, the Back-to-Back RTP sessions, Mesh with endpoints. However, the Back-to-Back RTP sessions, Mesh with
independent RTP sessions, SFM, will definitely break the loop independent RTP sessions, SFM, will definitely break the loop
detection mechanism. detection mechanism.
4.7. Consistency between header extensions and RTCP 4.7. Consistency between header extensions and RTCP
Some RTP header extensions have relevance not only end-to-end, but 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 also hop-to-hop, meaning at least some of the middleboxes in the path
are aware of their potential presence through signaling, intercept are aware of their potential presence through signaling, intercept
and interpret such header extensions and potentially also rewrite or and interpret such header extensions and potentially also rewrite or
generate them. Modern header extensions generally follow RFC 5285 generate them. Modern header extensions generally follow RFC 5285
[RFC5285], which allows for all of the above. Examples for such [RFC5285], which allows for all of the above. Examples for such
header extensions include the mid (media ID) in [draft-ietf-mmusic- header extensions include the mid (media ID) in [draft-ietf-mmusic-
sdp-bundle-negotiation-12] [I-D.ietf-mmusic-sdp-bundle-negotiation]. sdp-bundle-negotiation-12] [I-D.ietf-mmusic-sdp-bundle-negotiation].
There is also a generalization of mapping RTCP SDES into an RTP At the time of writing there was also a proposal for how to include
header extension [draft-westerlund-avtext-dses-hdr-ext] any SDES into an RTP header extension [draft-westerlund-avtext-dses-
[I-D.westerlund-avtext-sdes-hdr-ext]. hdr-ext] [I-D.westerlund-avtext-sdes-hdr-ext].
When such header extensions are in use, any middlebox that When such header extensions are in use, any middlebox that
understands it must ensure consistency between the extensions it sees understands it must ensure consistency between the extensions it sees
and/or generates, and the RTCP it receives and generates. For and/or generates, and the RTCP it receives and generates. For
example, the mid of bundle is sent in an RTP header extension and example, the mid of bundle is sent in an RTP header extension and
also in an RTCP SDES message. This apparent redundancy was also in an RTCP SDES message. This apparent redundancy was
introduced as unaware middleboxes may choose to discard RTP header introduced as unaware middleboxes may choose to discard RTP header
extensions. Obviously, inconsistency between the media ID sent in extensions. Obviously, inconsistency between the media ID sent in
the RTP header extension and in the RTCP SDES message could lead to the RTP header extension and in the RTCP SDES message could lead to
undesirable results, and, therefore, consistency is needed. undesirable results, and, therefore, consistency is needed.
skipping to change at page 42, line 25 skipping to change at page 42, line 29
Translator functionality. Translator functionality.
6. Security Considerations 6. Security Considerations
The use of Mixers, SFMs and Translators has impact on security and The use of Mixers, SFMs and Translators has impact on security and
the security functions used. The primary issue is that both Mixers, the security functions used. The primary issue is that both Mixers,
SFMs and Translators modify packets, thus preventing the use of SFMs and Translators modify packets, thus preventing the use of
integrity and source authentication, unless they are trusted devices integrity and source authentication, unless they are trusted devices
that take part in the security context, e.g., the device can send that take part in the security context, e.g., the device can send
Secure Realtime Transport Protocol (SRTP) and Secure Realtime Secure Realtime Transport Protocol (SRTP) and Secure Realtime
Transport Control Protocol (SRTCP) [RFC3711] packets to End Points in Transport Control Protocol (SRTCP) [RFC3711] packets to endpoints in
the Communication Session. If encryption is employed, the media the Communication Session. If encryption is employed, the media
Translator, SFM and Mixer need to be able to decrypt the media to Translator, SFM and Mixer need to be able to decrypt the media to
perform its function. A transport Translator may be used without perform its function. A transport Translator may be used without
access to the encrypted payload in cases where it translates parts access to the encrypted payload in cases where it translates parts
that are not included in the encryption and integrity protection, for that are not included in the encryption and integrity protection, for
example, IP address and UDP port numbers in a media stream using SRTP example, IP address and UDP port numbers in a media stream using SRTP
[RFC3711]. However, in general, the Translator, SFM or Mixer needs [RFC3711]. However, in general, the Translator, SFM or Mixer needs
to be part of the signalling context and get the necessary security to be part of the signalling context and get the necessary security
associations (e.g., SRTP crypto contexts) established with its RTP associations (e.g., SRTP crypto contexts) established with its RTP
session participants. session participants.
skipping to change at page 44, line 6 skipping to change at page 44, line 9
domain, one for B and one for D. It may be forced to maintain a set 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 of totally independent security associations between itself and B and
D respectively, so as to avoid two-time pad occurrences. These D respectively, so as to avoid two-time pad occurrences. These
contexts must also be capable of handling all the sources present in contexts must also be capable of handling all the sources present in
the other domains. Hence, using completely independent security the other domains. Hence, using completely independent security
associations (for certain keying mechanisms) may force a Translator associations (for certain keying mechanisms) may force a Translator
to handle N*DM keys and related state; where N is the total number of 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. SSRCs used over all domains and DM is the total number of domains.
The multicast based (ASM and SSM), Relay and Mesh with common RTP The multicast based (ASM and SSM), Relay and Mesh with common RTP
session are all topologies with multiple End Points that require session are all topologies with multiple endpoints that require
shared knowledge about the different crypto contexts for the End shared knowledge about the different crypto contexts for the
Points. These multi-party topologies have special requirements on endpoints. These multi-party topologies have special requirements on
the key-management as well as the security functions. Specifically the key-management as well as the security functions. Specifically
source-authentication in these environments has special requirements. source-authentication in these environments has special requirements.
There exist a number of different mechanisms to provide keys to the There exist a number of different mechanisms to provide keys to the
different participants. One example is the choice between group keys different participants. One example is the choice between group keys
and unique keys per SSRC. The appropriate keying model is impacted and unique keys per SSRC. The appropriate keying model is impacted
by the topologies one intends to use. The final security properties by the topologies one intends to use. The final security properties
are dependent on both the topologies in use and the keying are dependent on both the topologies in use and the keying
mechanisms' properties, and need to be considered by the application. mechanisms' properties, and need to be considered by the application.
Exactly which mechanisms are used is outside of the scope of this Exactly which mechanisms are used is outside of the scope of this
skipping to change at page 44, line 33 skipping to change at page 44, line 36
This document makes no request of IANA. This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. RFC.
8. Acknowledgements 8. Acknowledgements
The authors would like to thank Mark Baugher, Bo Burman, Umesh The authors would like to thank Mark Baugher, Bo Burman, Umesh
Chandra, Alex Eleftheriadis, Roni Even, Ladan Gharai, Geoff Hunt, Chandra, Alex Eleftheriadis, Roni Even, Ladan Gharai, Geoff Hunt,
Keith Lantz, Jonathan Lennox, Colin Perkins, and Suhas Nandakumar for Keith Lantz, Jonathan Lennox, Scarlet Liuyan, Suhas Nandakumar, and
their help in reviewing and improving this document. Colin Perkins for their help in reviewing and improving this
document.
9. References 9. References
9.1. Normative References 9.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006. 2006.
9.2. Informative References 9.2. Informative References
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session: "Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback", Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-04 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-05 (work
in progress), August 2014. in progress), February 2015.
[I-D.ietf-mmusic-sdp-bundle-negotiation] [I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings, Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session "Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-12 (work in progress), October 2014. negotiation-16 (work in progress), January 2015.
[I-D.westerlund-avtext-sdes-hdr-ext] [I-D.westerlund-avtext-sdes-hdr-ext]
Westerlund, M., Even, R., and M. Zanaty, "RTP Header Westerlund, M., Even, R., and M. Zanaty, "RTP Header
Extension for RTCP Source Description Items", draft- Extension for RTCP Source Description Items", draft-
westerlund-avtext-sdes-hdr-ext-03 (work in progress), westerlund-avtext-sdes-hdr-ext-03 (work in progress),
November 2014. November 2014.
[RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5, [RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5,
RFC 1112, August 1989. RFC 1112, August 1989.
[RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network [RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network
Address Translator (Traditional NAT)", RFC 3022, January Address Translator (Traditional NAT)", RFC 3022, January
2001. 2001.
[RFC3569] Bhattacharyya, S., "An Overview of Source-Specific [RFC3569] Bhattacharyya, S., "An Overview of Source-Specific
Multicast (SSM)", RFC 3569, July 2003. Multicast (SSM)", RFC 3569, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for [RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for
IP", RFC 4607, August 2006. IP", RFC 4607, August 2006.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile "Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008. with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008. Header Extensions", RFC 5285, July 2008.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control [RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760, February 2010. Sessions with Unicast Feedback", RFC 5760, February 2010.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
 End of changes. 125 change blocks. 
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