draft-ietf-avtcore-rtp-topologies-update-10.txt   rfc7667.txt 
Network Working Group M. Westerlund Internet Engineering Task Force (IETF) M. Westerlund
Internet-Draft Ericsson Request for Comments: 7667 Ericsson
Obsoletes: 5117 (if approved) S. Wenger Obsoletes: 5117 S. Wenger
Intended status: Informational Vidyo Category: Informational Vidyo
Expires: January 3, 2016 July 2, 2015 ISSN: 2070-1721 November 2015
RTP Topologies RTP Topologies
draft-ietf-avtcore-rtp-topologies-update-10
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 environments based on the Real-time Transport Protocol (RTP).
particular, centralized topologies commonly employed in the video In 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 replaces RFC This document is updated with additional topologies and replaces RFC
5117. 5117.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This document is not an Internet Standards Track specification; it is
provisions of BCP 78 and BCP 79. published for informational purposes.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
This Internet-Draft will expire on January 3, 2016. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7667.
Copyright Notice Copyright Notice
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 4 2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 5
2.2. Definitions related to RTP grouping taxonomy . . . . . . 4 2.2. Definitions Related to RTP Grouping Taxonomy . . . . . . 5
3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . 5 3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 5 3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 6
3.2. Point to Point via Middlebox . . . . . . . . . . . . . . 6 3.2. Point to Point via Middlebox . . . . . . . . . . . . . . 7
3.2.1. Translators . . . . . . . . . . . . . . . . . . . . . 6 3.2.1. Translators . . . . . . . . . . . . . . . . . . . . . 7
3.2.2. Back to Back RTP sessions . . . . . . . . . . . . . . 10 3.2.2. Back-to-Back RTP sessions . . . . . . . . . . . . . . 11
3.3. Point to Multipoint Using Multicast . . . . . . . . . . . 11 3.3. Point to Multipoint Using Multicast . . . . . . . . . . . 12
3.3.1. Any Source Multicast (ASM) . . . . . . . . . . . . . 11 3.3.1. Any-Source Multicast (ASM) . . . . . . . . . . . . . 12
3.3.2. Source Specific Multicast (SSM) . . . . . . . . . . . 13 3.3.2. Source-Specific Multicast (SSM) . . . . . . . . . . . 14
3.3.3. SSM with Local Unicast Resources . . . . . . . . . . 15 3.3.3. SSM with Local Unicast Resources . . . . . . . . . . 15
3.4. Point to Multipoint Using Mesh . . . . . . . . . . . . . 16 3.4. Point to Multipoint Using Mesh . . . . . . . . . . . . . 17
3.5. Point to Multipoint Using the RFC 3550 Translator . . . . 19 3.5. Point to Multipoint Using the RFC 3550 Translator . . . . 20
3.5.1. Relay - Transport Translator . . . . . . . . . . . . 19 3.5.1. Relay - Transport Translator . . . . . . . . . . . . 20
3.5.2. Media Translator . . . . . . . . . . . . . . . . . . 20 3.5.2. Media Translator . . . . . . . . . . . . . . . . . . 21
3.6. Point to Multipoint Using the RFC 3550 Mixer Model . . . 21 3.6. Point to Multipoint Using the RFC 3550 Mixer Model . . . 22
3.6.1. Media Mixing Mixer . . . . . . . . . . . . . . . . . 23 3.6.1. Media-Mixing Mixer . . . . . . . . . . . . . . . . . 24
3.6.2. Media Switching . . . . . . . . . . . . . . . . . . . 26 3.6.2. Media-Switching Mixer . . . . . . . . . . . . . . . . 27
3.7. Selective Forwarding Middlebox . . . . . . . . . . . . . 28 3.7. Selective Forwarding Middlebox . . . . . . . . . . . . . 29
3.8. Point to Multipoint Using Video Switching MCUs . . . . . 32 3.8. Point to Multipoint Using Video-Switching MCUs . . . . . 33
3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 33 3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 34
3.10. Split Component Terminal . . . . . . . . . . . . . . . . 34 3.10. Split Component Terminal . . . . . . . . . . . . . . . . 35
3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 37 3.11. Non-symmetric Mixer/Translators . . . . . . . . . . . . . 38
3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 37 3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 38
4. Topology Properties . . . . . . . . . . . . . . . . . . . . . 38 4. Topology Properties . . . . . . . . . . . . . . . . . . . . . 39
4.1. All to All Media Transmission . . . . . . . . . . . . . . 38 4.1. All-to-All Media Transmission . . . . . . . . . . . . . . 39
4.2. Transport or Media Interoperability . . . . . . . . . . . 39 4.2. Transport or Media Interoperability . . . . . . . . . . . 40
4.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . . . 39 4.3. Per-Domain Bitrate Adaptation . . . . . . . . . . . . . . 40
4.4. Aggregation of Media . . . . . . . . . . . . . . . . . . 40 4.4. Aggregation of Media . . . . . . . . . . . . . . . . . . 41
4.5. View of All Session Participants . . . . . . . . . . . . 40 4.5. View of All Session Participants . . . . . . . . . . . . 41
4.6. Loop Detection . . . . . . . . . . . . . . . . . . . . . 41 4.6. Loop Detection . . . . . . . . . . . . . . . . . . . . . 42
4.7. Consistency between header extensions and RTCP . . . . . 41 4.7. Consistency between Header Extensions and RTCP . . . . . 42
5. Comparison of Topologies . . . . . . . . . . . . . . . . . . 41 5. Comparison of Topologies . . . . . . . . . . . . . . . . . . 42
6. Security Considerations . . . . . . . . . . . . . . . . . . . 42 6. Security Considerations . . . . . . . . . . . . . . . . . . . 43
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 44 7. References . . . . . . . . . . . . . . . . . . . . . . . . . 45
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 44 7.1. Normative References . . . . . . . . . . . . . . . . . . 45
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 44 7.2. Informative References . . . . . . . . . . . . . . . . . 45
9.1. Normative References . . . . . . . . . . . . . . . . . . 44 Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 48
9.2. Informative References . . . . . . . . . . . . . . . . . 45 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 48
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 46
1. Introduction 1. Introduction
Real-time Transport Protocol (RTP) [RFC3550] topologies describe Real-time Transport Protocol (RTP) [RFC3550] topologies describe
methods for interconnecting RTP entities and their processing methods for interconnecting RTP entities and their processing
behavior for RTP and RTCP. This document tries to address past and behavior for RTP and the RTP Control Protocol (RTCP). This document
existing confusion, especially with respect to terms not defined in tries to address past and existing confusion, especially with respect
RTP but in common use in the communication industry, such as the to terms not defined in RTP but in common use in the communication
Multipoint Control Unit or MCU. industry, such as the Multipoint Control Unit or MCU.
When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was
developed the main emphasis lay in the efficient support of point to developed, the main emphasis lay in the efficient support of
point and small multipoint scenarios without centralized multipoint point-to-point and small multipoint scenarios without centralized
control. In practice, however, most multipoint conferences operate multipoint control. In practice, however, most multipoint
utilizing centralized units referred to as MCUs. MCUs may implement conferences operate utilizing centralized units referred to as MCUs.
Mixer or Translator functionality (in RTP [RFC3550] terminology), and MCUs may implement mixer or translator functionality (in RTP
signalling support. They may also contain additional application [RFC3550] terminology) and signaling support. They may also contain
layer functionality. This document focuses on the media transport additional application-layer functionality. This document focuses on
aspects of the MCU that can be realized using RTP, as discussed the media transport aspects of the MCU that can be realized using
below. Further considered are the properties of Mixers and RTP, as discussed below. Further considered are the properties of
Translators, and how some types of deployed MCUs deviate from these mixers and translators, and how some types of deployed MCUs deviate
properties. from these 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 that were not anticipated in RFC 5117;
thus this document replaces [RFC5117]. These architectures use thus, this document replaces [RFC5117]. These architectures use
scalable video coding and simulcasting, and their associated scalable video coding and simulcasting, and their associated
centralized units are referred to as Selective Forwarding Units centralized units are referred to as Selective Forwarding Middleboxes
(SFU). This codification provides a common information basis for (SFMs). This codification provides a common information basis for
future discussion and specification work. future discussion and specification work.
The new topologies are Point to Point via Middlebox (Section 3.2), The new topologies are Point to Point via Middlebox (Section 3.2),
Source Specific Multicast (Section 3.3.2), SSM with Local Unicast Source-Specific Multicast (Section 3.3.2), SSM with Local Unicast
Resources (Section 3.3.3), Point to Multipoint Using Mesh Resources (Section 3.3.3), Point to Multipoint Using Mesh
(Section 3.4), Selective Forwarding Middlebox (Section 3.7), and (Section 3.4), Selective Forwarding Middlebox (Section 3.7), and
Split Component Terminal (Section 3.10). The Point to Multipoint Split Component Terminal (Section 3.10). The Point to Multipoint
Using the RFC 3550 Mixer Model (Section 3.6) has been significantly Using the RFC 3550 Mixer Model (Section 3.6) has been significantly
expanded to cover two different versions namely Media Mixing Mixer expanded to cover two different versions, namely Media-Mixing Mixer
(Section 3.6.1), and Media Switching (Section 3.6.2). (Section 3.6.1) and Media-Switching Mixer (Section 3.6.2).
The document's attempt to clarify and explain sections of the Real- The document's attempt to clarify and explain sections of the RTP
time Transport Protocol (RTP) spec [RFC3550] is informal. It is not spec [RFC3550] is informal. It is not intended to update or change
intended to update or change what is normatively specified within RFC what is normatively specified within RFC 3550.
3550.
2. Definitions 2. Definitions
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 data transfer. end-to-end data transfer.
PtM: Point to Multipoint PtM: Point to Multipoint
PtP: Point to Point PtP: Point to Point
SFU: Selective Forwarding Unit SFM: Selective Forwarding Middlebox
SSM: Source-Specific Multicast SSM: Source-Specific Multicast
SSRC: Synchronization Source SSRC: Synchronization Source
2.2. Definitions related to RTP grouping taxonomy 2.2. Definitions Related to RTP Grouping Taxonomy
[Note to RFC editor: The following definitions have been taken from
draft-ietf-avtext-rtp-grouping-taxonomy-02 (taxonomy draft
henceforth). It is avtcore working group agreement to not delay the
publication of the topologies-update document through a dependency to
the taxonomy draft. If, however, the taxonomy draft and this draft
are in your work queue at the same time and there would be no
significant additional delay (through your schedule, normative
reference citations, or similar) in publishing both documents roughly
in parallel, it would be preferable to replace the definition
language with something like "as in [RFC YYYY]" where YYYY would be
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 [RFC7656].
grouping-taxonomy-02, and are used in capitalized form throughout the
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 two or more Participants communicating with each other via
set of Multimedia Sessions. one or more Multimedia Sessions.
Endpoint: 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 mentity, it is
classified as a single "Endpoint". 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 time
clock synchronized, time progressing, digital media stream, called progressing digital media stream synchronized to a reference
a Source Stream. clock. This stream is called 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 communication via one or more RTP
RTP Sessions. sessions.
3. Topologies 3. Topologies
This subsection defines several topologies that are relevant for This subsection defines several topologies that are relevant for
codec control but also RTP usage in other contexts. The section codec control but also RTP usage in other contexts. The section
starts with point to point cases, with or without middleboxes. Then starts with point-to-point cases, with or without middleboxes. Then
follows a number of different methods for establishing point to it follows a number of different methods for establishing point-to-
multipoint communication. These are structured around the most multipoint communication. These are structured around the most
fundamental enabler, i.e., multicast, a mesh of connections, fundamental enabler, i.e., multicast, a mesh of connections,
translators, mixers and finally MCUs and SFUs. The section ends by translators, mixers, and finally MCUs and SFMs. The section ends by
discussing de-composited terminals, asymmetric middlebox behaviors discussing decomposited terminals, asymmetric middlebox behaviors,
and combining topologies. and combining topologies.
The topologies may be referenced in other documents by a shortcut The topologies may be referenced in other documents by a shortcut
name, indicated by the prefix "Topo-". name, indicated by the prefix "Topo-".
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 The Point-to-Point (PtP) topology (Figure 1) consists of two
endpoints, communicating using unicast. Both RTP and RTCP traffic endpoints, communicating using unicast. Both RTP and RTCP traffic
are conveyed endpoint-to-endpoint, using unicast traffic only (even are conveyed endpoint to endpoint, using unicast traffic only (even
if, in exotic cases, this unicast traffic happens to be conveyed over if, in exotic cases, this unicast traffic happens to be conveyed over
an 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 endpoints and combining the requirements stemming handling multiple endpoints and combining the requirements stemming
from them. Note that an endpoint 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 endpoints. 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 endpoint it can be relevant to implement that use multiple SSRCs 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" [MULTI-STREAM-OPT].
[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 endpoints communicate but have This section discusses cases where two endpoints communicate but 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
mixed media received in multiple RTP streams (in cases like audio media received in multiple RTP streams (in cases like audio mixing or
mixing or video tiling), but always under its own SSRC, possibly video tiling), but always under its own SSRC, possibly using the CSRC
using the CSRC field to indicate the source(s) of the content. field to indicate the source(s) of the content. Mixers are more
Mixers are more common in point to multipoint cases than in PtP. The common in point-to-multipoint cases than in PtP. The reason is that
reason is that in PtP use cases the primary focus of a middlebox is in PtP use cases, the primary focus of a middlebox is enabling
enabling interoperability, between otherwise non-interoperable interoperability, between otherwise non-interoperable endpoints, such
endpoints, such as transcoding to a codec the receiver supports, as transcoding to a codec the receiver supports, which can be done by
which can be done by a media translator. 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 endpoints (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 Source Description (SDES) information
SSRC or CSRC also needs to be forwarded between the domains for any associated with an SSRC or CSRC also needs to be forwarded between
SSRC/CSRC used in the different domains. the domains for any SSRC/CSRC used in the different domains.
A Translator commonly does not use an SSRC of its own, and is not 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 it
to trigger repair by the media sending endpoint, by sending feedback wants to trigger repair by the media sending endpoint, by sending
messages. While such feedback could use the SSRC of the target for feedback messages. While such feedback could use the SSRC of the
the translator (the receiving endpoint), this in turn would require target for the translator (the receiving endpoint), this in turn
translation of the targets RTCP reports to make them consistent. It would require translation of the target RTCP reports to make them
may be simpler to expose an additional SSRC in the session. The only consistent. It may be simpler to expose an additional SSRC in the
concern is endpoints failing to support the full RTP specification session. The only concern is that endpoints failing to support the
may have issues with multiple SSRCs reporting on the RTP streams sent full RTP specification may have issues with multiple SSRCs reporting
by that endpoint, as this use case may be viewed as excotic by on the RTP streams sent by that endpoint, as this use case may be
implementers. viewed as exotic by 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
Shortcut name: Topo-PtP-Relay Shortcut name: Topo-PtP-Relay
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 endpoints 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 and network topologies where the RTP
is required to pass a particular point for audit by employing stream 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 Traversal Using Relays around NAT
Gateways and Media Processing Nodes with media anchoring (TURN) [RFC5766] servers, Session Border Gateways, and Media
functionalities. Processing Nodes with media anchoring functionalities.
+---+ +---+ +---+ +---+ +---+ +---+
| 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 since the RTP and RTCP packets are routed
the endpoints in the RTP session, and thereby indirectly affect the between the endpoints in the RTP session, and thereby they indirectly
RTP session. For this reason, one could believe that transport affect the RTP session. For this reason, one could believe that
translator-type middleboxes do not need to be included in this Transport Translator-type middleboxes do not need to be included in
document. This topology, however, can raise additional requirements this document. This topology, however, can raise additional
in the RTP implementation and its interactions with the signalling requirements in the RTP implementation and its interactions with the
solution. Both in signalling and in certain RTCP fields, network signaling solution. Both in signaling and in certain RTCP fields,
addresses other than those of the relay can occur since B has a network addresses other than those of the relay can occur since B has
different network address than the relay (T). Implementations that a different network address than the relay (T). Implementations that
cannot support this will also not work correctly when endpoints 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 from injecting traffic into a session, which one peer may, in the
fo adequate security in the relay, think it comes from the other absence of adequate security in the relay, think it comes from the
peer. other peer.
3.2.1.2. Transport Translator 3.2.1.2. Transport Translator
Shortcut name: Topo-Trn-Translator Shortcut name: Topo-Trn-Translator
Transport Translators (Topo-Trn-Translator) do not modify the RTP Transport Translators (Topo-Trn-Translator) do not modify the RTP
stream itself, but are concerned with transport parameters. stream itself but are concerned with transport parameters. Transport
Transport parameters, in the sense of this section, comprise the parameters, in the sense of this section, comprise the transport
transport addresses (to bridge different domains such unicast to addresses (to bridge different domains such as unicast to multicast)
multicast) and the media packetization to allow other transport and the media packetization to allow other transport protocols to be
protocols to be interconnected to a session (in gateways). interconnected to a session (in gateways).
Translators that bridge between different protocol worlds need to be Translators that bridge between different protocol worlds need to be
concerned about the mapping of the SSRC/CSRC (Contributing Source) concerned about the mapping of the SSRC/CSRC (Contributing Source)
concept to the non-RTP protocol. When designing a Translator to a concept to the non-RTP protocol. When designing a translator to a
non-RTP-based media transport, an important consideration is how to non-RTP-based media transport, an important consideration is how to
handle different sources and their identities. This problem space is handle different sources and their identities. This problem space is
not discussed henceforth. not discussed henceforth.
Of the transport Translators, this memo is primarily interested in Of the Transport Translators, this memo is primarily interested in
those that use RTP on both sides, and this is assumed henceforth. those that use RTP on both sides, and this is assumed henceforth.
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
Shortcut name: Topo-Media-Translator Shortcut name: Topo-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 endpoints 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 endpoint B's RTCP Receiver case, the translator needs to rewrite endpoint B's RTCP receiver
Report before forwarding them to endpoint A. The rewriting is needed report before forwarding them to endpoint A. The rewriting is needed
as the RTP stream received by B is not the same RTP stream as the as the RTP stream received by B is not the same RTP stream as 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 was substantially behind in reception, while it most
likely it would not be. Therefore, the Translator must translate likely would not be. Therefore, the translator must translate that
that number to a corresponding sequence number for the stream the number to a corresponding sequence number for the stream the
Translator received. Similar requirements exists for most other translator received. Similar requirements exist 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 endpoint 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 endpoint 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 endpoint and itself. In that bandwidth reduction between the sending endpoint and itself. In 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 behavior worth pointing out is the one
depicted in Figure 3 of an endpoint A sending a RTP stream containing depicted in Figure 3 of an endpoint A sending an RTP stream
media (only) to B. On the path there is a device T that on A's containing media (only) to B. On the path, there is a device T that
behalf manipulates the RTP streams. One common example is that T manipulates the RTP streams on A's behalf. One common example is
adds a second RTP stream containing Forward Error Correction (FEC) that T adds a second RTP stream containing Forward Error Correction
information in order to protect A's (non FEC-protected) RTP stream. (FEC) information in order to protect A's (non FEC-protected) RTP
In this case, T needs to semantically bind the new FEC RTP stream to stream. In this case, T needs to semantically bind the new FEC RTP
A's media-carrying RTP stream, for example by using the same CNAME as stream to A's media-carrying RTP stream, for example, by using the
A. same CNAME as 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
RTP stream, while leaving most or all of the original RTP packets RTP stream, while leaving most or all of the original RTP packets
intact (with the exception of certain RTP header fields, such as the intact (with the exception of certain RTP header fields, such as the
sequence number). One example is the injection of meta-data into the sequence number). One example is the injection of metadata into the
RTP stream, carried in their own RTP packets. RTP stream, carried in their own RTP packets.
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
Shortcut name: Topo-Back-To-Back Shortcut name: Topo-Back-To-Back
There exist middleboxes that interconnect two endpoints 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. Instead, they establish 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 |
+------+ +------+ +------+ +------+ +------+ +------+
Figure 4: Back-to-back RTP sessions through Middlebox Figure 4: Back-to-Back RTP Sessions through Middlebox
The middlebox acts as an application-level gateway and bridges the The middlebox acts as an application-level gateway and bridges the
two RTP sessions. This bridging can be as basic as forwarding the two RTP sessions. This bridging can be as basic as forwarding the
RTP payloads between the sessions, or more complex including media RTP payloads between the sessions or more complex including media
transcoding. The difference of this topology relative to the single transcoding. The difference of this topology relative to the single
RTP session context is the handling of the SSRCs and the other RTP session context is the handling of the SSRCs and the other
session-related identifiers, such as CNAMEs. With two different RTP session-related identifiers, such as CNAMEs. With two different RTP
sessions these can be freely changed and it becomes the middlebox's sessions, these can be freely changed and it becomes the middlebox's
respnsibility to maintain the correct relations. responsibility to maintain the correct relations.
The signalling or other above-RTP level functionalities referencing The signaling 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 endpoint two segments or by bridging statistics from the receiving endpoint
through the middlebox to the sending endpoint. 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
Multicast is an IP layer functionality that is available in some Multicast is an IP-layer functionality that is available in some
networks. Two main flavors can be distinguished: Any Source networks. Two main flavors can be distinguished: Any-Source
Multicast (ASM) [RFC1112] where any multicast group participant can Multicast (ASM) [RFC1112] where any multicast group participant can
send to the group address and expect the packet to reach all group send to the group address and expect the packet to reach all group
participants; and Source Specific Multicast (SSM) [RFC3569], where participants and Source-Specific Multicast (SSM) [RFC3569], where
only a particular IP host sends to the multicast group. Each of only a particular IP host sends to the multicast group. Each of
these models are discussed below in their respective sections. these models are discussed below in their respective sections.
3.3.1. Any Source Multicast (ASM) 3.3.1. Any-Source Multicast (ASM)
Shortcut name: Topo-ASM (was Topo-Multicast) Shortcut name: Topo-ASM (was Topo-Multicast)
+-----+ +-----+
+---+ / \ +---+ +---+ / \ +---+
| A |----/ \---| B | | A |----/ \---| B |
+---+ / Multi- \ +---+ +---+ / Multi- \ +---+
+ Cast + + cast +
+---+ \ Network / +---+ +---+ \ Network / +---+
| C |----\ /---| D | | C |----\ /---| D |
+---+ \ / +---+ +---+ \ / +---+
+-----+ +-----+
Figure 5: Point to Multipoint Using Multicast Figure 5: Point to Multipoint Using Multicast
Point to Multipoint (PtM) is defined here as using a multicast Point to Multipoint (PtM) is defined here as using a multicast
topology as a transmission model, in which traffic from any multicast topology as a transmission model, in which traffic from any multicast
group participant reaches all the other multicast group participants, group participant reaches all the other multicast group participants,
except for cases such as: except for cases such as:
o packet loss, or o packet loss, or
o when a multicast group participant does not wish to receive the o when a multicast group participant does not wish to receive the
traffic for a specific multicast group and, therefore, has not traffic for a specific multicast group and, therefore, has not
subscribed to the IP multicast group in question. This scenario subscribed to the IP multicast group in question. This scenario
can occur, for example, where a multimedia session is distributed can occur, for example, where a Multimedia Session is distributed
using two or more multicast groups and a multicast group using two or more multicast groups, and a multicast group
participant is subscribed only to a subset of these sessions. participant is subscribed only to a subset of these sessions.
In the above context, "traffic" encompasses both RTP and RTCP In the above context, "traffic" encompasses both RTP and RTCP
traffic. The number of multicast group participants can vary between traffic. The number of multicast group participants can vary between
one and many, as RTP and RTCP scale to very large multicast groups one and many, as RTP and RTCP scale to very large multicast groups
(the theoretical limit of the number of participants in a single RTP (the theoretical limit of the number of participants in a single RTP
session is in the range of billions). The above can be realized session is in the range of billions). The above can be realized
using Any Source Multicast (ASM). using ASM.
For feedback usage, it is useful to define a "small multicast group" For feedback usage, it is useful to define a "small multicast group"
as a group where the number of multicast group participants is so low as a group where the number of multicast group participants is so low
(and other factors such as the connectivity is so good) that it (and other factors such as the connectivity is so good) that it
allows the participants to use early or immediate feedback, as allows the participants to use early or immediate feedback, as
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 discussed in [RFC4585] is typically required. Each individual
endpoint 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 endpoint, 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)
Shortcut name: Topo-SSM Shortcut name: Topo-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,
creating a topology that looks like the one below: creating a topology that looks like the one below:
+--------+ +-----+ +--------+ +-----+
|Media | | | Source-specific |Media | | | Source-Specific
|Sender 1|<----->| D S | Multicast |Sender 1|<----->| D S | Multicast
+--------+ | I O | +--+----------------> R(1) +--------+ | I O | +--+----------------> R(1)
| S U | | | | | S U | | | |
+--------+ | T R | | +-----------> R(2) | +--------+ | T R | | +-----------> R(2) |
|Media |<----->| R C |->+ | : | | |Media |<----->| R C |->+ | : | |
|Sender 2| | I E | | +------> R(n-1) | | |Sender 2| | I E | | +------> R(n-1) | |
+--------+ | B | | | | | | +--------+ | B | | | | | |
: | U | +--+--> R(n) | | | : | U | +--+--> R(n) | | |
: | T +-| | | | | : | T +-| | | | |
: | I | |<---------+ | | | : | I | |<---------+ | | |
+--------+ | O |F|<---------------+ | | +--------+ | O |F|<---------------+ | |
|Media | | N |T|<--------------------+ | |Media | | N |T|<--------------------+ |
|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 endpoints (RTP In the SSM topology (Figure 6), a number of RTP sending endpoints
sources henceforth) (1 to M) are allowed to send media to the SSM (RTP sources henceforth) (1 to M) are allowed to send media to the
group. These sources send media to a dedicated distribution source, SSM group. These sources send media to a dedicated Distribution
which forwards the RTP streams to the multicast group on behalf of Source, which forwards the RTP streams to the multicast group on
the original RTP sources. The RTP streams reach the receiving behalf of the original RTP sources. The RTP streams reach the
endpoints (Receivers henceforth) (R(1) to R(n)). The Receivers' RTCP receiving endpoints (receivers henceforth) (R(1) to R(n)). The
messages cannot be sent to the multicast group, as the SSM multicast receivers' RTCP messages cannot be sent to the multicast group, as
group by definition has only a single IP sender. To support RTCP, an the SSM multicast group by definition has only a single IP sender.
RTP extension for SSM [RFC5760] was defined. It uses unicast To support RTCP, an RTP extension for SSM [RFC5760] was defined. It
transmission to send RTCP from each of the receivers to one or more uses unicast transmission to send RTCP from each of the receivers to
Feedback Targets (FT). The feedback targets relay the RTCP one or more Feedback Targets (FT). The Feedback Targets relay the
unmodified, or provide a summary of the participants RTCP reports RTCP unmodified, or provide a summary of the participants' RTCP
towards the whole group by forwarding the RTCP traffic to the reports 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
groups of the receivers. For summary reports, however, there subgroups 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, need to be accounted
for. for.
Aforementioned situation results in common behavior for RTP The 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 endpoint 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 signaling 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 signaling 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 signaling requirement: all of
the endpoints need to have the same RTP and payload type the endpoints need to have the same RTP and payload type
configuration. Otherwise, endpoint 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 endpoint 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 unpredictable 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 communication 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
Shortcut name: Topo-SSM-RAMS Shortcut name: Topo-SSM-RAMS
"Unicast-Based Rapid Acquisition of Multicast RTP Sessions" [RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions" [RFC6285]
results in additional extensions to SSM Topology. results in additional extensions to SSM topology.
----------- -------------- ----------- --------------
| |------------------------------------>| | | |------------------------------------>| |
| |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| | | |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| |
| | | | | | | |
| Multicast | ---------------- | | | Multicast | ---------------- | |
| Source | | Retransmission | | | | Source | | Retransmission | | |
| |-------->| Server (RS) | | | | |-------->| Server (RS) | | |
| |.-.-.-.->| | | | | |.-.-.-.->| | | |
| | | ------------ | | | | | | ------------ | | |
----------- | | Feedback | |<.=.=.=.=.| | ----------- | | Feedback | |<.=.=.=.=.| |
| | Target (FT)| |<~~~~~~~~~| RTP Receiver | | | Target (FT)| |<~~~~~~~~~| RTP Receiver |
PRIMARY MULTICAST | ------------ | | (RTP_Rx) | PRIMARY MULTICAST | ------------ | | (RTP_Rx) |
RTP SESSION with | | | | RTP SESSION with | | | |
UNICAST FEEDBACK | | | | UNICAST FEEDBACK | | | |
| | | | | | | |
- - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- - - - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- -
| | | | | | | |
skipping to change at page 15, line 45 skipping to change at page 16, 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 endpoint 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 the decoding of these burst-
media packets, the correct media timing is established, i.e. media delivered media packets, the correct media timing is established,
packets are received within adequate buffer intervals for this i.e., media packets are received within adequate buffer intervals for
application. This is accomplished by first establishing a unicast this application. This is accomplished by first establishing a
PtP RTP session between the Burst/Retransmission Source (BRS, unicast 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
normal speed in order to synchronize the receiver to the ongoing normal speed in order to synchronize the receiver to the ongoing
multicast RTP stream. Once the RTP receiver and its decoder have multicast RTP stream. Once the RTP receiver and its decoder have
caught up with the multicast session's current delivery, the receiver caught up with the multicast session's current delivery, the receiver
switches over to receiving directly from the multicast group. In switches over to receiving directly from the multicast group. In
many deployed application, the (still existing) PtP RTP session is many deployed applications, the (still existing) PtP RTP session is
used as a repair channel, i.e., for RTP Retransmission traffic of used as a repair channel, i.e., for RTP Retransmission traffic of
those packets that were not received from the multicast group. those packets that were not received from the multicast group.
3.4. Point to Multipoint Using Mesh 3.4. Point to Multipoint Using Mesh
Shortcut name: Topo-Mesh Shortcut name: Topo-Mesh
+---+ +---+ +---+ +---+
| A |<---->| B | | A |<---->| B |
+---+ +---+ +---+ +---+
^ ^ ^ ^
\ / \ /
\ / \ /
v v v v
+---+ +---+
| C | | C |
+---+ +---+
Figure 8: Point to Multi-Point using Mesh Figure 8: Point to Multipoint 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 endpoints over multiple joint RTP session involving three or more endpoints over multiple
unicast transport flows, like the joint three endpoint 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 endpoints do the same, everyone will have a joint view As long as all endpoints do the same, everyone will have a joint 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 endpoint 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
skipping to change at page 17, line 29 skipping to change at page 18, line 29
| | +-UDP2------| |-UDP2------+ | | | +-UDP2------| |-UDP2------+ |
| | | +-RTP1----| |-RTP1----+ | | | | | +-RTP1----| |-RTP1----+ | |
| | | | +-Video-| |-Video-+ | | | | | | | +-Video-| |-Video-+ | | |
| +-------+-+-+--->AV1|---------------->| | | | | | +-------+-+-+--->AV1|---------------->| | | | |
| | | | |<----------------|CV1 | | | | | | | | |<----------------|CV1 | | | |
| | | +-------| |-------+ | | | | | | +-------| |-------+ | | |
| | +---------| |---------+ | | | | +---------| |---------+ | |
| +-----------| |-----------+ | | +-----------| |-----------+ |
+----------------------+ +-------------+ +----------------------+ +-------------+
Figure 9: An Multi-unicast Mesh with a joint RTP session Figure 9: A Multi-Unicast Mesh with a Joint RTP Session
Figure 9 depicts endpoints A's view of using a common RTP session Figure 9 depicts endpoint A's view of using a common RTP session when
when establishing the mesh as shown in Figure 8. There is only one establishing the mesh as shown in Figure 8. There is only one RTP
RTP session (RTP1) but two transport flows (UDP1 and UDP2). The session (RTP1) but two transport flows (UDP1 and UDP2). The Media
Media Source (CAM) is encoded and transmitted over the SSRC (AV1) Source (CAM) is encoded and transmitted over the SSRC (AV1) across
across both transport layers. However, as this is a joint RTP both transport layers. However, as this is a joint RTP session, the
session, the two streams must be the same. Thus, an congestion two streams must be the same. Thus, a congestion control adaptation
control adaptation needed for the paths A to B and A to C needs to needed for the paths A to B and A to C needs to use the most
use the most restricting path's properties. 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 endpoint, 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 30 skipping to change at page 19, line 30
| | +-UDP2------| |-UDP2------+ | | | +-UDP2------| |-UDP2------+ |
| | | +-RTP2----| |-RTP2----+ | | | | | +-RTP2----| |-RTP2----+ | |
| | +----+ | | +-Audio-| |-Audio-+ | | | | | +----+ | | +-Audio-| |-Audio-+ | | |
| +->|ENC2|--+-+-+--->AA2|------------->| | | | | | +->|ENC2|--+-+-+--->AA2|------------->| | | | |
| +----+ | | | |<-------------|CA1 | | | | | +----+ | | | |<-------------|CA1 | | | |
| | | +-------| |-------+ | | | | | | +-------| |-------+ | | |
| | +---------| |---------+ | | | | +---------| |---------+ | |
| +-----------| |-----------+ | | +-----------| |-----------+ |
+------------------------+ +-------------+ +------------------------+ +-------------+
Figure 10: An Multi-unicast Mesh with independent RTP session Figure 10: A Multi-Unicast Mesh with an Independent RTP Session
Lets review the topology when independent RTP sessions are used, from Let's review the topology when independent RTP sessions are used from
A's perspective in Figure 10 by considering both how the media is A's perspective in Figure 10 by considering both how the media is
handled and the RTP sessions that are set-up in Figure 10. A's handled and how the RTP sessions are set up in Figure 10. A's
microphone is captured and the audio is fed into two different microphone is captured and the audio is fed into two different
encoder instances, each with a different independent RTP session, encoder instances, each with a different independent RTP session,
i.e. RTP1 and RTP2 respectively. The SSRCs (AA1 and AA2) in each RTP i.e., RTP1 and RTP2, respectively. The SSRCs (AA1 and AA2) in each
session are completely independent and the media bit-rate produced by RTP session are completely independent, and the media bitrate
the encoders can also be tuned differently to address any congestion produced by the encoders can also be tuned differently to address any
control requirements differing for the paths A to B compared to A to congestion control requirements differing for the paths A to B
C. compared to A to 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 endpoints 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, endpoints 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 endpoint those sessions. For example, in the case of the three-endpoint
configuration in Figure 8, endpoint A has no awareness of the configuration in Figure 8, endpoint A has no awareness of the
conditions occurring in the session between endpoints 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 endpoint 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 namespace 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
multipoint of Translators compared to the point to point only cases multipoint of translators compared to the point-to-point cases in
in Section 3.2.1. Section 3.2.1.
3.5.1. Relay - Transport Translator 3.5.1. Relay - Transport Translator
Shortcut name: Topo-PtM-Trn-Translator Shortcut name: Topo-PtM-Trn-Translator
This section discusses Transport Translator only usages to enable This section discusses Transport Translator-only usages to enable
multipoint sessions. multipoint sessions.
+-----+ +-----+
+---+ / \ +------------+ +---+ +---+ / \ +------------+ +---+
| A |<---/ \ | |<---->| B | | A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+ +---+ / \ | | +---+
+ cast +->| Translator | + Multicast +->| 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)
endpoints 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 endpoints 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. both 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 endpoints 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 endpoints. 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 signaling requirements onto the session establishment; for
for example, a common configuration of RTP payload types is required. 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 from using the
using the listening ports on the translator. The translator can, listening ports on the translator to inject traffic. The translator
however, go one step further, and especially if explicit SSRC can, however, go one step further, especially if explicit SSRC
signalling is used, prevent endpoints to send SSRCs other than its signaling is used, to prevent endpoints from sending SSRCs other than
own (that are, for example, used by other participants in the its 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 a 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 packet content can be
operated without the requiring it to have access to the security operated without requiring it to have access to the security contexts
contexts used to protect the RTP/RTCP traffic between the 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
providing new mechanisms to establish a multipoint session. It is not providing new mechanisms to establish a multipoint session. It
more of an enabler, or facilitator, that ensures a given endpoint or is more of an enabler, or facilitator, that ensures a given endpoint
a defined sub-set of endpoints can participate in the session. or a defined subset of endpoints can participate in the session.
If endpoint 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 endpoints to reach B without overloading the received from the other endpoints to reach B without overloading the
path. This transcoding can help the other endpoints in the multicast path. This transcoding can help the other endpoints in the multicast
part of the session, by not requiring the quality transmitted by A to part of the session, by not requiring the quality transmitted by A to
be lowered to the bitrates that B is actually capable of receiving be lowered to the bitrates that B is actually 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.
+-----+ +-----+
+---+ / \ +-----------+ +---+ +---+ / \ +-----------+ +---+
| A |<---/ \ | |<---->| B | | A |<---/ \ | |<---->| B |
+---+ / 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 endpoints 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 endpoints. 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 endpoint, 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 endpoints to allow for CSRC should be forwarded between the different endpoints to allow 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 need to be separated: the Any-Source Multicast network (including
endpoints A and C), endpoint B, and endpoint D. Assuming all four endpoints A and C), endpoint B, and endpoint D. Assuming all four
endpoints in the conference are interested in receiving content from endpoints in the conference are interested in receiving content from
all other endpoints, the Mixer produces different mixed RTP streams all other endpoints, the mixer produces different mixed RTP streams
for B and D, as the one to B may contain content received from D, and 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 and transmission of RTCP feedback messages by the mixer to the
endpoints in the other domain(s). In other cases, a message is endpoints in the other domain(s). In other cases, a message is
handled by the Mixer locally and therefore not forwarded to any other handled by the mixer locally and therefore not forwarded to any other
domain. 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 |
+---+ | | +---+ +---+ | | +---+
| Mixer | | Mixer |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
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 endpoints and selects the stream(s) receives RTP streams from several endpoints and selects the stream(s)
to be included in a media-domain mix. The selection can be through to be included in a media-domain mix. The selection can be through
static configuration or by dynamic, content dependent means such as static configuration or by dynamic, content-dependent means such as
voice activation. The mixer then creates a single outgoing RTP voice activation. The mixer then creates a single outgoing 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-Mixing Mixer is probably the audio
used in voice conferencing, where the output consists of a mixture of mixer, used in voice conferencing, where the output consists of a
all the input audio signals; this needs minimal signalling to be mixture of all the input audio signals; this needs minimal signaling
successfully set up. From a signal processing viewpoint, audio to be 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 endpoints. Assume, for example, that one wants reasonable number of endpoints. Assume, for example, that one wants
to mix N streams from N different endpoints. 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, endpoint gets a mix of all other sources except its own, as this
as this would result in an echo. When N is lower than the number of would result in an echo. When N is lower than the number of all
all endpoints, one may produce a mix of all N streams for the group endpoints, one may produce a mix of all N streams for the group that
that are currently not included in the mix, thus N+1 mixes. These are currently not included in the mix; thus, N+1 mixes. These audio
audio streams are then encoded again, RTP packetized and sent out. streams are then encoded again, RTP packetized, and sent out. In
In many cases, audio level normalization, noise suppression, and many cases, audio level normalization, noise suppression, and similar
similar signal processing steps are also required or desirable before signal processing steps are also required or desirable before the
the actual mixing process commences. 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, with each tile containing the video from
endpoint (typically showing a human participant). Tiles can be of an 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 in a 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 be either locally produced or 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 endpoints. 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 arrangements 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 middlebox, allowing individual compositions for each receiving
endpoint (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 the most commonly used media codecs. A third
is the latency introduced by the media mixing, which can be problem 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-synchronized with high-latency
The advantage of media mixing is that it is straightforward for the video. The advantage of media mixing is that it is straightforward
endpoints to handle the single media stream (which includes the mixed for the endpoints to handle the single media stream (which includes
aggregate of many sources), as they don't need to handle multiple the mixed aggregate of many sources), as they don't need to handle
decodings, local mixing and composition. In fact, mixers were multiple decodings, local mixing, and composition. In fact, mixers
introduced in pre-RTP times so that legacy, single stream receiving were introduced in pre-RTP times so that legacy, single stream
endpoints (that, in some protocol environments, actually didn't need receiving endpoints (that, in some protocol environments, actually
to be aware of the multipoint nature of the conference) could didn't need to be aware of the multipoint nature of the conference)
successfully participate in what a user would recognize as a could 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 34 skipping to change at page 26, line 34
+-C---------+ | | I | | +-C---------+ | | I | |
| +-RTP3----| |-RTP3------+ | X | | | +-RTP3----| |-RTP3------+ | X | |
| | +-Audio-| |-Audio---+ | +---+ | E | | | | +-Audio-| |-Audio---+ | +---+ | E | |
| | | CA1|--------->|---------+-+-|DEC|->| R | | | | | CA1|--------->|---------+-+-|DEC|->| R | |
| | | |<---------|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,
can be seen in Figure 15. The mixer presents one SSRC towards the as can be seen in Figure 15. The mixer presents one SSRC towards the
receiving endpoint, e.g., MA1 to Peer A, where the associated stream receiving endpoint, e.g., MA1 to Peer A, where the associated stream
is the media mix of the other endpoints. As each peer, in this 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 Sources
and CA1, allowing the receiver to identify the contributing sources BA1 and CA1, allowing the receiver to identify the Contributing
even if this were not possible through the media itself or through Sources even if this were not possible through the media itself or
other signaling means. through 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 RTP header
Client audio levels RTP header extension [RFC6465]. If the SSRCs extension for Mixer-to-Client Audio Level Indication [RFC6465]. If
from the endpoint to mixer paths are used as CSRCs in another RTP the SSRCs from the endpoint to mixer paths are used as CSRCs in
session, then RTP1, RTP2 and RTP3 become one joint session as they another RTP session, then RTP1, RTP2, and RTP3 become one joint
have a common SSRC space. At this stage, the mixer also needs to session as they have a common SSRC space. At this stage, the mixer
consider which RTCP information it needs to expose in the different also needs to consider which RTCP information it needs to expose in
paths. In the above scenario, a mixer would normally expose nothing the different paths. In the above scenario, a mixer would normally
more than the Source Description (SDES) information and RTCP BYE for expose nothing more than the SDES information and RTCP BYE for a CSRC
a CSRC leaving the session. The main goal would be to enable the leaving the session. The main goal would be to enable the correct
correct binding against the application logic and other information binding against the application logic and other information sources.
sources. This also enables loop detection in the RTP session. This also enables loop detection in the RTP session.
3.6.2. Media Switching 3.6.2. Media-Switching Mixer
Media switching mixers are used in limited functionality scenarios Media-Switching Mixers are used in limited functionality scenarios
where no, or only very limited, concurrent presentation of multiple where no, or only very limited, concurrent presentation of multiple
sources is required by the application, to more complex multi-stream sources is required by the application and also in more complex
usages with receiver mixing or tiling, including combined with multi-stream usages with receiver mixing or tiling, including
simulcast and/or scalability between source and mixer. An RTP Mixer combined with simulcast and/or scalability between source and mixer.
based on media switching avoids the media decoding and encoding An RTP mixer based on media switching avoids the media decoding and
operations in the mixer, as it conceptually forwards the encoded encoding operations in the mixer, as it conceptually forwards the
media stream as it was being sent to the mixer. It does not avoid, encoded media stream as it was being sent to the mixer. It does not
however, the decryption and re-encryption cycle as it rewrites RTP avoid, however, the decryption and re-encryption cycle as it rewrites
headers. Forwarding media (in contrast to reconstructing-mixing- RTP headers. Forwarding media (in contrast to reconstructing-mixing-
encoding media) reduces the amount of computational resources needed encoding media) reduces the amount of computational resources needed
in the mixer and increases the media quality (both in terms of in the mixer and increases the media quality (both in terms of
fidelity and reduced latency). fidelity and reduced latency).
A media switching mixer maintains a pool of SSRCs representing A Media-Switching Mixer maintains a pool of SSRCs representing
conceptual or functional RTP streams that the mixer can produce. conceptual or functional RTP streams that the mixer can produce.
These RTP streams are created by selecting media from one of the RTP These RTP streams are created by selecting media from one of the RTP
streams received by the mixer and forwarded to the peer using the streams received by the mixer and forwarded to the peer using the
mixer's own SSRCs. The mixer can switch between available sources if mixer's own SSRCs. The mixer can switch between available sources if
that is required by the concept for the source, like the currently that is required by the concept for the source, like the currently
active speaker. Note that the mixer, in most cases, still needs to active speaker. Note that the mixer, in most cases, still needs to
perform a certain amount of media processing, as many media formats perform a certain amount of media processing, as many media formats
do not allow to "tune into" the stream at arbitrary points in their do not allow to "tune into" the stream at arbitrary points in their
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 the Media Source.
depending on the negotiation of the RTP payload type, the value Finally, depending on the negotiation of the RTP payload type, the
representing this particular RTP payload configuration may have to be value representing this particular RTP payload configuration may have
changed if the different endpoint-to-mixer paths have not arrived on to be changed if the different endpoint-to-mixer paths have not
the same numbering for a given configuration. This also requires arrived on the same numbering for a given configuration. This also
that the different endpoints support a common set of codecs, requires 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 endpoints (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 endpoint. The mixer has thus two SSRCs sending video each receiving endpoint. Thus, the mixer has 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 45 skipping to change at page 28, line 46
+-F---------+ | | | | +-F---------+ | | | |
| +-RTP6----| |-RTP6------+ | | | | +-RTP6----| |-RTP6------+ | | |
| | +-Video-| |-Video---+ | | | | | | +-Video-| |-Video---+ | | | |
| | | FV1|------------>|---------+-+------->| | | | | | FV1|------------>|---------+-+------->| | |
| | | |<------------|MV11 <---+-+-AV1----| | | | | | |<------------|MV11 <---+-+-AV1----| | |
| | | |<------------|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 Mixer can, similarly to the Media-Mixing Mixer,
Mixer, reduce the bit-rate required for media transmission towards reduce the bitrate required for media transmission towards the
the different peers by selecting and forwarding only a sub-set of RTP different peers by selecting and forwarding only a subset of RTP
streams it receives from the sending endpoints. In cases the mixer streams it receives from the sending endpoints. In case 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 endpoint, both based on subsets of streams to forward to a receiving endpoint, both based on
transport or endpoint restrictions as well as application logic. transport or endpoint restrictions as well as application logic.
To ensure that a media receiver in an endpoint 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 fully terminating the RTP
sessions between the different endpoint 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- or make available, all potential RTP sources (SSRCs) into a per-
endpoint, 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 endpoints. 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 | |
| | | |<------------|FV1 <----+-+-------| C | | | | | |<------------|FV1 <----+-+-------| C | |
skipping to change at page 29, line 44 skipping to change at page 30, line 44
| | | FV1|------------>|---------+-+------>| | | | | | FV1|------------>|---------+-+------>| | |
| | | |<------------|AV1 <----+-+-------| | | | | | |<------------|AV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | | | | | | : : : |: : : : : : : : :| | |
| | | |<------------|EV1 <----+-+-------| | | | | | |<------------|EV1 <----+-+-------| | |
| | +-------| |---------+ | | | | | | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ | | +---------| |-----------+ +-----+ |
+-----------+ +---------------------------+ +-----------+ +---------------------------+
Figure 17: Selective Forwarding Middlebox Figure 17: Selective Forwarding Middlebox
In the six endpoint conference depicted above in (Figure 17) one can In the six endpoint conference depicted above (in Figure 17), one can
see that endpoint A is aware of five incoming SSRCs, BV1-FV1. If 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 endpoints with the two latest speaking where the mixer provides the endpoints with the two latest speaking
endpoints, 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 endpoints, each RTP different RTP sessions that transmit media to the endpoints, each RTP
stream requires rewriting of certain RTP header fields when being stream requires the 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. an RTP session. The timestamp (possibly offset) stays the same.
The RTP sessions can be considered independent, resulting in that the The RTP sessions can be considered independent, resulting in that the
SSRC numbers used can also be handled independently. This simplifies SSRC numbers used can also be handled independently. This simplifies
the SSRC collision detection and avoidance, but require tools such as the SSRC collision detection and avoidance but requires tools such as
remapping tables between the RTP sessions. Using independent RTP remapping tables between the RTP sessions. Using independent RTP
sessions are not required, as the switching behavior is possible to sessions is not required, as it is possible for the switching
perform also with a common SSRC space. However, in this case behavior to also perform with a common SSRC space. However, in this
collision detection and handling becomes a different problem. It is case, collision detection and handling becomes a different problem.
up to the implementation to use a single common SSRC space or It is up to the implementation to use a single common SSRC space or
separate ones. separate ones.
Using separate SSRC spaces has some implications. For example, the Using separate SSRC spaces has some implications. For example, the
RTP stream that is being sent by endpoint B to the middlebox (BV1) RTP stream that is being sent by endpoint B to the middlebox (BV1)
may use an SSRC value of 12345678. When that RTP stream is sent to may use an SSRC value of 12345678. When that RTP stream is sent to
endpoint F by the middlebox, it can use any SSRC value, e.g. endpoint F by the middlebox, it can use any SSRC value, e.g.,
87654321. As a result, each endpoint may have a different view of 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 the RTP level. Instead, this responsibility falls onto higher
layers.
The middlebox is also responsible for receiving any RTCP codec The middlebox is also responsible for receiving any RTCP codec
control requests coming from an endpoint, and decide if it can act on control requests coming from an endpoint and deciding if it can act
the request locally or needs to translate the request into the RTP on the request locally or needs to translate the request into the RTP
session/transport leg that contains the media source. Both endpoints session/transport leg that contains the Media Source. Both endpoints
and the middlebox need to implement conference related codec control and the 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 that switching points
points between the sources, and Temporary Maximum Media Bit-rate be provided between the sources and Temporary Maximum Media Bitrate
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 bitrate (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 endpoints. 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 bitrate, desired
layout, etc. Contrary to transcoding MCUs, these "Selective layout, etc. Contrary to transcoding MCUs, SFMs have extremely low
Forwarding Units" (SFUs) have extremely low delay, and provide delay and provide features that are typically associated with high-
features that are typically associated with high-end systems end systems (personalized layout, error localization) without any
(personalized layout, error localization) without any signal signal processing at the middlebox. They are also capable of scaling
processing at the middlebox. They are also capable of scaling to a to a large number of concurrent users, and--due to their very low
large number of concurrent users, and--due to their very low delay-- 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
endpoint 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 endpoint either needs to be able to handle as many provide media, the endpoint either needs to be able to handle as 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 that includes an on/off logic as part of its media
translation. The topology has the property that all SSRCs present in translation. The topology has the property that all SSRCs present in
the session are visible to an endpoint. It also has mixer aspects, the session are visible to an endpoint. It also has mixer aspects,
as the streams it provides are not basically translated version, but as the streams it provides are not basically translated versions, but
instead they have conceptual property assigned to them and can be instead they have conceptual property assigned to them and can be
both turned on/off as well as being fully or partially delivered. both turned on/off as well as fully or partially delivered. Thus,
Thus this topology appears to be some hybrid between the translator this topology appears to be some hybrid between the translator and
and mixer model. mixer model.
The differences between selective forwarding middlebox and a The differences between a Selective Forwarding Middlebox and a
switching mixer (Section 3.6.2) are minor, and they share most Switching-Media Mixer (Section 3.6.2) are minor, and they share most
properties. The above requirement on having a large number of properties. The above requirement on having a large number of
decoding instances or requiring efficient switching of decoder decoding instances or requiring efficient switching of decoder
contexts, are one point of difference. The other is how the contexts, are one point of difference. The other is how the
identification is performed, where the Mixer uses CSRC to provide identification is performed, where the mixer uses CSRC to provide
information on what is included in a particular RTP stream that information on what is included in a particular RTP stream that
represent a particular concept. Selective forwarding gets the source represents a particular concept. Selective forwarding gets the
information through the SSRC, and instead uses other mechanisms to source information through the SSRC and instead uses other mechanisms
indicate the streams intended usage, if needed. to indicate the streams intended usage, if needed.
3.8. Point to Multipoint Using Video Switching MCUs 3.8. Point to Multipoint Using Video-Switching MCUs
Shortcut name: Topo-Video-switch-MCU Shortcut name: Topo-Video-switch-MCU
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |------| Multipoint |------| B | | A |------| Multipoint |------| B |
+---+ | Control | +---+ +---+ | Control | +---+
| Unit | | Unit |
+---+ | (MCU) | +---+ +---+ | (MCU) | +---+
| C |------| |------| D | | C |------| |------| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 18: Point to Multipoint Using a Video Switching MCU Figure 18: Point to Multipoint Using a Video-Switching MCU
This PtM topology was popular in early implementations of multipoint This PtM topology was popular in early implementations of multipoint
videoconferencing systems due to its simplicity, and the videoconferencing systems due to its simplicity, and the
corresponding middlebox design has been known as a "video switching corresponding middlebox design has been known as a "video-switching
MCU". The more complex RTCP-terminating MCUs, discussed in the next MCU". The more complex RTCP-terminating MCUs, discussed in the next
section, became the norm, however, when technology allowed section, became the norm, however, when technology allowed
implementations at acceptable costs. implementations at acceptable costs.
A video switching MCU forwards to a participant a single media A video-switching MCU forwards to a participant a single media
stream, selected from the available streams. The criteria for stream, selected from the available streams. The criteria for
selection are often based on voice activity in the audio-visual selection are often based on voice activity in the audio-visual
conference, but other conference management mechanisms (like conference, but other conference management mechanisms (like
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 bitrate, 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 endpoints. 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 endpoints 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
that one implement a centralized video switching conference using a that one implement a centralized video-switching conference using a
Mixer according to RFC 3550, instead of the shortcut implementation mixer according to RFC 3550, instead of the shortcut implementation
described here. described here.
3.9. Point to Multipoint Using RTCP-Terminating MCU 3.9. Point to Multipoint Using RTCP-Terminating MCU
Shortcut name: Topo-RTCP-terminating-MCU Shortcut name: Topo-RTCP-terminating-MCU
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B | | A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+ +---+ | Control | +---+
| Unit | | Unit |
skipping to change at page 33, line 35 skipping to change at page 34, line 35
| 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 endpoint 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 endpoints, 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 bitrate,
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 and the RTP stream the MCU sends. In other words,
outgoing RTP streams typically use a different SSRC, and may well use the outgoing RTP streams typically use a different SSRC, and may well
a different payload type (PT), even if this different PT happens to use a different payload type (PT), even if this different PT happens
be mapped to the same media type. This is a result of the to be mapped to the same media type. This is a result of the
individually negotiated RTP session for each endpoint. 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 endpoint. 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 signaled 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 endpoints implement this feature. Instead, the identification of the endpoints
whose content is included in the Mixer's output is not indicated whose content is included in the mixer's output is not indicated
through any explicit RTP mechanism. That is, most deployed MCUs set through any explicit RTP mechanism. That is, most deployed MCUs set
the CSRC Count (CC) field in the RTP header to zero, thereby the CC field in the RTP header to zero, thereby indicating no
indicating no available CSRC information, even if they could identify available CSRC information, even if they could identify the original
the original sending endpoints as suggested in RTP. 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
carelessly connecting two misconfigured MCUs, a loop could be carelessly connecting two misconfigured MCUs, a loop could be
generated. generated.
2. There is no information about active media senders available in 2. There is no information about active media senders available in
the RTP packet. As this information is missing, receivers cannot the RTP packet. As this information is missing, receivers cannot
use it. It also deprives the client of information related to use it. It also deprives the client of information related to
currently active senders in a machine-usable way, thus preventing currently active senders in a machine-usable way, thus preventing
clients from indicating currently active speakers in user clients from indicating currently active speakers in user
interfaces, etc. interfaces, etc.
Note that many/most deployed MCUs (and video conferencing endpoints) Note that many/most deployed MCUs (and video conferencing endpoints)
rely on signalling layer mechanisms for the identification of the rely on signaling-layer mechanisms for the identification of the
contributing sources, for example, a SIP conferencing package Contributing Sources, for example, a SIP conferencing package
[RFC4575]. This alleviates, to some extent, the aforementioned [RFC4575]. This alleviates, to some extent, the aforementioned
issues resulting from ignoring RTP's CSRC mechanism. issues resulting from ignoring RTP's CSRC mechanism.
3.10. Split Component Terminal 3.10. Split Component Terminal
Shortcut name: Topo-Split-Terminal Shortcut name: Topo-Split-Terminal
In some applications, for example in some telepresence systems, In some applications, for example, in some telepresence systems,
terminals may be not integrated into a single functional unit, but terminals may not be integrated into a single functional unit but
composed of more than one subunits. For example, a telepresence room composed of more than one subunits. For example, a telepresence room
terminal employing multiple cameras and monitors may consist of terminal employing multiple cameras and monitors may consist of
multiple video conferencing subunits, each capable of handling a multiple video conferencing subunits, each capable of handling a
single camera and monitor. Another example would be a video single camera and monitor. Another example would be a video
conferencing terminal in which audio is handled by one subunit, and conferencing terminal in which audio is handled by one subunit, and
video by another. Each of these subunits uses its own physical video by another. Each of these subunits uses its own physical
network interface (for example: Ethernet jack) and network address. network interface (for example: Ethernet jack) and network address.
The various (media processing) subunits need (logically and The various (media processing) subunits need (logically and
physically) to be interconnected by control functionality, but their physically) to be interconnected by control functionality, but their
media plane functionality may be split. This type of terminals is media plane functionality may be split. These types of terminals are
referred to as split component terminals. Historically, the earliest referred to as split component terminals. Historically, the earliest
split component terminals were perhaps the independent audio and split component terminals were perhaps the independent audio and
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 endpoint 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 |
| +------------+ | | +------------+ |
skipping to change at page 35, line 47 skipping to change at page 36, line 47
| +------------+ | | +------------+ |
+---------------------+ +---------------------+
Figure 20: Split Component Terminal Figure 20: Split Component Terminal
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 subunits share a common clock, and can be synchronized, the
the RTP streams generated from those Source Streams need to include RTP streams generated from those Source Streams need to include the
the same CNAME in their RTCP SDES packets. The use of a common CNAME same CNAME in their RTCP SDES packets. The use of a common CNAME for
for RTP flows carried in different transport-layer flows is entirely RTP flows carried in different transport-layer flows is entirely
normal for RTP and RTCP senders, and fully compliant RTP endpoints, normal for RTP and RTCP senders, and fully compliant RTP endpoints,
middle-boxes, and other tools should have no problem with this. middleboxes, 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 endpoint scenario, which is not further perhaps a multihomed 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 gracefully endpoints that use a common
across multiple transport layer flows gracefully: they report an CNAME across multiple transport-layer flows: they report an error
error condition that two separate endpoints are using the same CNAME. condition in which two separate endpoints are using the same CNAME.
Depending on the sophistication of the support staff, such erroneous Depending on the sophistication of the support staff, such erroneous
reports can lead to support issues. reports can lead to support issues.
Aforementioned support issue can sometimes be avoided if each of the The aforementioned support issue can sometimes be avoided if each of
subunits of a split component terminal is configured to use a the 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
video subunit with a common CNAME are close to unavoidable. video subunit with a common CNAME are close to unavoidable.
The different RTP streams comprising a split terminal system can form The different RTP streams comprising a split terminal system can form
a single RTP session or they can form multiple RTP sessions, a single RTP session or they can form multiple RTP sessions,
depending on the visibility of their SSRC values in RTCP reports. If depending on the visibility of their SSRC values in RTCP reports. If
the receiver of the RTP streams sent by the split terminal sends the receiver of the RTP streams sent by the split terminal sends
reports relating to all of the RTP flows (i.e., to each SSRC) in each reports relating to all of the RTP flows (i.e., to each SSRC) in each
RTCP report then a single RTP session is formed. Alternatively, if RTCP report, then a single RTP session is formed. Alternatively, if
the receiver of the RTP streams sent by the split terminal does not the receiver of the RTP streams sent by the split terminal does not
send cross-reports in RTCP, then the audio and video form separate send cross-reports in RTCP, then the audio and video form separate
RTP sessions. RTP sessions.
For example, in the Figure 20, B will send RTCP reports to each of For example, in Figure 20, B will send RTCP reports to each of the
the sub-units of A. If the RTCP packets that B sends to the audio subunits of A. If the RTCP packets that B sends to the audio subunit
sub-unit of A include reports on the reception quality of the video of A include reports on the reception quality of the video as well as
as well as the audio, and similarly if the RTCP packets that B sends the audio, and similarly if the RTCP packets that B sends to the
to the video sub-unit of A include reports on the reception quality video subunit of A include reports on the reception quality of the
of the audio as well as video, then a single RTP session is formed. audio as well as video, then a single RTP session is formed.
However, if the RTCP packets B sends to the audio sub-unit of A only However, if the RTCP packets B sends to the audio subunit of A only
report on the received audio, and the RTCP packet B sends to the report on the received audio, and the RTCP packets B sends to the
video sub-unit of A only report on the received video, then there are video subunit of A only report on the received video, then there are
two separate RTP sessions. two separate RTP sessions.
Forming a single RTP session across the RTP streams sent by the Forming a single RTP session across the RTP streams sent by the
different sub-units of a split terminal gives each sub-unit different subunits of a split terminal gives each subunit visibility
visibility into reception quality of RTP streams sent by the other into reception quality of RTP streams sent by the other subunits.
sub-units. This information can help diagnose reception quality
problems, but at the cost of increased RTCP bandwidth use.
RTP streams sent by the sub-units of a split terminal need to use the This information can help diagnose reception quality problems, but at
the cost of increased RTCP bandwidth use.
RTP streams sent by the subunits of a split terminal need to use the
same CNAME in their RTCP packets if they are to be synchronized, same CNAME in their RTCP packets if they are to be synchronized,
irrespective of whether a single RTP session is formed or not. irrespective of whether a single RTP session is formed or not.
3.11. Non-Symmetric Mixer/Translators 3.11. Non-symmetric Mixer/Translators
Shortcut name: Topo-Asymmetric Shortcut name: Topo-Asymmetric
It is theoretically possible to construct an MCU that is a Mixer in It is theoretically possible to construct an MCU that is a mixer in
one direction and a Translator in another. The main reason to one direction and a translator in another. The main reason to
consider this would be to allow topologies similar to Figure 13, consider this would be to allow topologies similar to Figure 13,
where the Mixer does not need to mix in the direction from B or D where the mixer does not need to mix in the direction from B or D
towards the multicast domains with A and C. Instead, the RTP streams towards the multicast domains with A and C. Instead, the RTP streams
from B and D are forwarded without changes. Avoiding this mixing from B and D are forwarded without changes. Avoiding this mixing
would save media processing resources that perform the mixing in would save media processing resources that perform the mixing in
cases where it isn't needed. However, there would still be a need to cases where it isn't needed. However, there would still be a need to
mix B's media towards D. Only in the direction B -> multicast domain mix B's media towards D. Only in the direction B -> multicast domain
or D -> multicast domain would it be possible to work as a or D -> multicast domain would it be possible to work as a
Translator. In all other directions, it would function as a Mixer. translator. In all other directions, it would function as a mixer.
The Mixer/Translator would still need to process and change the RTCP The mixer/translator would still need to process and change the RTCP
before forwarding it in the directions of B or D to the multicast before forwarding it in the directions of B or D to the multicast
domain. One issue is that A and C do not know about the mixed-media domain. One issue is that A and C do not know about the mixed-media
stream the Mixer sends to either B or D. Therefore, any reports stream the mixer sends to either B or D. Therefore, any reports
related to these streams must be removed. Also, receiver reports related to these streams must be removed. Also, receiver reports
related to A and C's RTP streams would be missing. To avoid A and C related to A's and C's RTP streams would be missing. To avoid A and
thinking that B and D aren't receiving A and C at all, the Mixer C thinking that B and D aren't receiving A and C at all, the mixer
needs to insert locally generated reports reflecting the situation needs to insert locally generated reports reflecting the situation
for the streams from A and C into B and D's Sender Reports. In the for the streams from A and C into B's and D's sender reports. In the
opposite direction, the Receiver Reports from A and C about B's and opposite direction, the receiver reports from A and C about B's and
D's stream also need to be aggregated into the Mixer's Receiver D's streams also need to be aggregated into the mixer's receiver
Reports sent to B and D. Since B and D only have the Mixer as source reports sent to B and D. Since B and D only have the mixer as source
for the stream, all RTCP from A and C must be suppressed by the for the stream, all RTCP from A and C must be suppressed by the
Mixer. mixer.
This topology is so problematic and it is so easy to get the RTCP This topology is so problematic, and it is so easy to get the RTCP
processing wrong, that it is not recommended for implementation. processing wrong, that it is not recommended for implementation.
3.12. Combining Topologies 3.12. Combining Topologies
Topologies can be combined and linked to each other using Mixers or Topologies can be combined and linked to each other using mixers or
Translators. However, care must be taken in handling the SSRC/CSRC translators. However, care must be taken in handling the SSRC/CSRC
space. A Mixer does not forward RTCP from sources in other domains, space. A mixer does not forward RTCP from sources in other domains,
but instead generates its own RTCP packets for each domain it mixes but instead generates its own RTCP packets for each domain it mixes
into, including the necessary Source Description (SDES) information into, including the necessary SDES information for both the CSRCs and
for both the CSRCs and the SSRCs. Thus, in a mixed domain, the only the SSRCs. Thus, in a mixed domain, the only SSRCs seen will be the
SSRCs seen will be the ones present in the domain, while there can be ones present in the domain, while there can be CSRCs from all the
CSRCs from all the domains connected together with a combination of domains connected together with a combination of mixers and
Mixers and Translators. The combined SSRC and CSRC space is common translators. The combined SSRC and CSRC space is common over any
over any Translator or Mixer. It is important to facilitate loop translator or mixer. It is important to facilitate loop detection,
detection, something that is likely to be even more important in something that is likely to be even more important in combined
combined topologies due to the mixed behavior between the domains. topologies due to the mixed behavior between the domains. Any
Any hybrid, like the Topo-Video-switch-MCU or Topo-Asymmetric, hybrid, like the Topo-Video-switch-MCU or Topo-Asymmetric, requires
requires considerable thought on how RTCP is dealt with. considerable thought on how RTCP is dealt with.
4. Topology Properties 4. Topology Properties
The topologies discussed in Section 3 have different properties. The topologies discussed in Section 3 have different properties.
This section describes these properties. Note that, even if a This section describes these properties. Note that, even if a
certain property is supported within a particular topology concept, certain property is supported within a particular topology concept,
the necessary functionality may be optional to implement. the necessary functionality may be optional to implement.
4.1. All to All Media Transmission 4.1. All-to-All Media Transmission
To recapitulate, multicast, and in particular Any Source Multicast To recapitulate, multicast, and in particular ASM, provides the
(ASM), provides the functionality that everyone may send to, or functionality that everyone may send to, or receive from, everyone
receive from, everyone else within the session. Source-specific else within the session. SSM can provide a similar functionality by
Multicast (SSM) can provide a similar functionality by having anyone having anyone intending to participate as a sender to send its media
intending to participate as sender to send its media to the SSM to the SSM Distribution Source. The SSM Distribution Source forwards
distribution source. The SSM distribution source forwards the media the media to all receivers subscribed to the multicast group. Mesh,
to all receivers subscribed to the multicast group. Mesh, MCUs, MCUs, mixers, Selective Forwarding Middleboxes (SFMs), and
Mixers, SFMs and Translators may all provide that functionality at translators may all provide that functionality at least on some basic
least on some basic level. However, there are some differences in level. However, there are some differences in which type of
which type of reachability they provide. reachability they provide.
The topologies that comes closest to emulating Any Source IP The topologies that come closest to emulating Any-Source IP
Multicast, with all-to-all transmission capabilities, are the Multicast, with all-to-all transmission capabilities, are the
transport Translator function called "relay" in Section 3.5, as well Transport Translator function called "relay" in Section 3.5, as well
as the Mesh with joint RTP sessions (Section 3.4). Media as the Mesh with joint RTP sessions (Section 3.4). Media
Translators, Mesh with independent RTP Sessions, Mixers, SFUs and the Translators, Mesh with independent RTP Sessions, mixers, SFUs, and
MCU variants do not provide a fully meshed forwarding on the 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 endpoint considers the path to the least capable transmitting endpoint considers the path to the least-capable
receiving endpoint. Otherwise, the media transmissions may overload receiving endpoint. Otherwise, the media transmissions may overload
that path. Therefore, a sending endpoint needs to monitor the path that path. Therefore, a sending endpoint needs to monitor the path
from itself to any of the receiving endpoints, 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 endpoints 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 perform
performing this monitoring, at least on a medium time scale. 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 on the topology. Both ASM and SSM have the benefit
benefit that only one copy of each packet traverses a particular that only one copy of each packet traverses a particular link. Using
link. Using a relay causes the transmission of one copy of a packet a relay causes the transmission of one copy of a packet per
per endpoint-to-relay path and packet transmitted. However, in most 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 endpoints (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 topologies causes N-1 streams of transmitted packets to traverse the
link from the endpoint, in an N endpoint mesh. How long the first-hop link from the endpoint, in a mesh with N endpoints. How
different paths are common, is highly situation dependent. long the 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, RTCP-terminating MCUs, 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 endpoints 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 endpoint'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 endpoint's encoding if the encoded streams do not match the receiving endpoint's
capabilities. capabilities.
4.3. Per Domain Bit-Rate Adaptation 4.3. Per-Domain Bitrate Adaptation
Endpoints 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. In the ASM, SSM, Mesh with common RTP session, and
Transport Relay scenario, each individual sending endpoint has to Transport Relay scenarios, each individual sending endpoint has to
adapt to the receiving endpoint behind the least capable path, adapt to the receiving endpoint behind the least-capable path,
yielding suboptimal quality for the endpoints behind the more capable yielding suboptimal quality for the endpoints behind the more capable
paths. This is no longer an issue when Media Translators, Mixers, paths. This is no longer an issue when Media Translators, mixers,
SFM or MCUs are involved, as each endpoint only needs to adapt to the SFMs, or MCUs are involved, as each endpoint only needs to adapt to
slowest path within its own domain. The Translator, Mixer, SFM, or the slowest path within its own domain. The translator, mixer, SFM,
MCU topologies all require their respective outgoing RTP streams to or MCU topologies all require their respective outgoing RTP streams
adjust the bit-rate, packet-rate, etc., to adapt to the least capable to adjust the bitrate, packet rate, etc., to adapt to the least-
path in each of the other domains. That way one can avoid lowering capable path in each of the other domains. That way one can avoid
the quality to the least-capable endpoint in all the domains at the lowering the quality to the least-capable endpoint in all the domains
cost (complexity, delay, equipment) of the Mixer, SFM or Translator, at the cost (complexity, delay, equipment) of the mixer, SFM, or
and potentially media sender (multicast/layered encoding and sending translator, and potentially the media sender (multicast/layered
the different representations). encoding and sending 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 endpoints with transmissions share the available bitrate. 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 use a mixer, or MCU, to aggregate the
aggregate the multiple RTP streams into a single one, where the multiple RTP streams into a single one, where the single RTP stream
single RTP stream takes up less resources in terms of bit-rate. This takes up less resources in terms of bitrate. This aggregation can be
aggregation can be performed according to different methods. Mixing performed according to different methods. Mixing or selection are
or selection are two common methods. Selection is almost always two common methods. Selection is almost always possible and easy to
possible and easy to implement. Mixing requires resources in the implement. Mixing requires resources in the mixer and may be
mixer, and may be relatively easy and not impairing the quality too relatively easy and not impair the quality too badly (audio) or quite
badly (audio) or quite difficult (video tiling, which is not only difficult (video tiling, which is not only computationally complex
computationally complex but also reduces the pixel count per stream, but also reduces the pixel count per stream, with corresponding loss
with corresponding loss in perceptual quality). 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 SDES. In topologies that
In topologies that provide a full all-to-all functionality, i.e. ASM, provide a full all-to-all functionality, i.e., ASM, Mesh with common
Mesh with common RTP session, Relay a compliant RTP implementation RTP session, and relay, a compliant RTP implementation offers the
offers the functionality directly as specified in RTP. In topologies functionality directly as specified in RTP. In topologies that do
that do not offer all-to-all communication, it is necessary that RTCP not offer all-to-all communication, it is necessary that RTCP is
is handled correctly in domain bridging function. RTP includes handled correctly in domain bridging functions. RTP includes
explicit specification text for Translators and Mixers, and for SFMs explicit specification text for translators and mixers, and for SFMs
the required functionality can be derived from that text. However, the required functionality can be derived from that text. However,
the MCU described in Section 3.8 cannot offer the full functionality the MCU described in Section 3.8 cannot offer the full functionality
for session participant identification through RTP means. The for session participant identification through RTP means. The
topologies that create independent RTP sessions per endpoint or pair topologies that create independent RTP sessions per endpoint or pair
of endpoints, like Back-to-Back RTP session, MESH with independent of endpoints, like a Back-to-Back RTP session, MESH with independent
RTP sessions, and the RTCP terminating MCU RTCP terminating MCU RTP sessions, and the RTCP terminating MCU (Section 3.9), with an
(Section 3.9), with an exception of SFM, do not support RTP based exception of SFM, do not support RTP-based identification of session
identification of session participants. In all those cases, other participants. In all those cases, other non-RTP-based mechanisms
non-RTP based mechanisms need to be implemented if such knowledge is need to be implemented if such knowledge is required or desirable.
required or desirable. When it comes to SFM the SSRC name space is When it comes to SFM, the SSRC namespace is not necessarily joint.
not necessarily joint, instead identification will require knowledge Instead, identification will require knowledge of SSRC/CSRC mappings
of SSRC/CSRC mappings that the SFM performed, see Section 3.7. 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 Section 3.8, at least as long as it forwards the RTCP between the
endpoints. 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, and SFMs 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 "A General generate them. Modern header extensions generally follow "A General
Mechanism for RTP Header Extensions" [RFC5285], which allows for all Mechanism for RTP Header Extensions" [RFC5285], which allows for all
of the above. Examples for such header extensions include the mid of the above. Examples for such header extensions include the Media
(media ID) in [I-D.ietf-mmusic-sdp-bundle-negotiation]. At the time ID (MID) in [SDP-BUNDLE]. At the time of writing, there was also a
of writing there was also a proposal for how to include any SDES into proposal for how to include some SDES into an RTP header extension
an RTP header extension [I-D.westerlund-avtext-sdes-hdr-ext]. [RTCP-SDES].
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 the 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 MID sent in the RTP
the RTP header extension and in the RTCP SDES message could lead to 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.
Middleboxes unaware of the nature of a header extension, as specified Middleboxes unaware of the nature of a header extension, as specified
in [RFC5285], are free to forward or discard header extensions. in [RFC5285], are free to forward or discard header extensions.
5. Comparison of Topologies 5. Comparison of Topologies
The table below attempts to summarize the properties of the different The table below attempts to summarize the properties of the different
topologies. The legend to the topology abbreviations are: Topo- topologies. The legend to the topology abbreviations are:
Point-to-Point (PtP), Topo-ASM (ASM), Topo-SSM (SSM), Topo-Trns- Topo-Point-to-Point (PtP), Topo-ASM (ASM), Topo-SSM (SSM), Topo-Trn-
Translator (TT), Topo-Media-Translator (including Transport Translator (TT), Topo-Media-Translator (including Transport
Translator) (MT), Topo-Mesh with joint session (MJS), Topo-Mesh with Translator) (MT), Topo-Mesh with joint session (MJS), Topo-Mesh with
individual sessions (MIS), Topo-Mixer (Mix), Topo-Asymmetric (ASY), individual sessions (MIS), Topo-Mixer (Mix), Topo-Asymmetric (ASY),
Topo-Video-switch-MCU (VSM), and Topo-RTCP-terminating-MCU (RTM), Topo-Video-switch-MCU (VSM), Topo-RTCP-terminating-MCU (RTM), and
Selective Forwarding Middlebox (SFM). In the table below, Y Selective Forwarding Middlebox (SFM). In the table below, Y
indicates Yes or full support, N indicates No support, (Y) indicates indicates Yes or full support, N indicates No support, (Y) indicates
partial support, and N/A indicates not applicable. partial support, and N/A indicates not applicable.
Property PtP ASM SSM TT MT MJS MIS Mix ASY VSM RTM SFM Property PtP ASM SSM TT MT MJS MIS Mix ASY VSM RTM SFM
--------------------------------------------------------------------- ---------------------------------------------------------------------
All to All media N Y (Y) Y Y Y (Y) (Y) (Y) (Y) (Y) (Y) All-to-All Media N Y (Y) Y Y Y (Y) (Y) (Y) (Y) (Y) (Y)
Interoperability N/A N N Y Y Y Y Y Y N Y Y Interoperability N/A N N Y Y Y Y Y Y N Y Y
Per Domain Adaptation N/A N N N Y N Y Y Y N Y Y Per-Domain Adaptation N/A N N N Y N Y Y Y N Y Y
Aggregation of media N N N N N N N Y (Y) Y Y N Aggregation of Media N N N N N N N Y (Y) Y Y N
Full Session View Y Y Y Y Y Y N Y Y (Y) N Y Full Session View Y Y Y Y Y Y N Y Y (Y) N Y
Loop Detection Y Y Y Y Y Y N Y Y (Y) N N Loop Detection Y Y Y Y Y Y N Y Y (Y) N N
Please note that the Media Translator also includes the transport Please note that the Media Translator also includes the Transport
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 mixers, SFMs,
SFMs and Translators modify packets, thus preventing the use of and translators modify packets, thus preventing the use of integrity
integrity and source authentication, unless they are trusted devices and source authentication, unless they are trusted devices that take
that take part in the security context, e.g., the device can send part in the security context, e.g., the device can send Secure Real-
Secure Realtime Transport Protocol (SRTP) and Secure Realtime time Transport Protocol (SRTP) and Secure Real-time Transport Control
Transport Control Protocol (SRTCP) [RFC3711] packets to endpoints in Protocol (SRTCP) [RFC3711] packets to endpoints in the Communication
the Communication Session. If encryption is employed, the media Session. If encryption is employed, the Media Translator, SFM, and
Translator, SFM and Mixer need to be able to decrypt the media to mixer need to be able to decrypt the media to perform its function.
perform its function. A transport Translator may be used without A Transport Translator may be used without access to the encrypted
access to the encrypted payload in cases where it translates parts payload in cases where it translates parts that are not included in
that are not included in the encryption and integrity protection, for the encryption and integrity protection, for example, IP address and
example, IP address and UDP port numbers in a media stream using SRTP UDP port numbers in a media stream using SRTP [RFC3711]. However, in
[RFC3711]. However, in general, the Translator, SFM or Mixer needs general, the translator, SFM, or mixer needs to be part of the
to be part of the signalling context and get the necessary security signaling context and get the necessary security associations (e.g.,
associations (e.g., SRTP crypto contexts) established with its RTP SRTP crypto contexts) established with its RTP session participants.
session participants.
Including the Mixer, SFM and Translator in the security context Including the mixer, SFM, and translator in the security context
allows the entity, if subverted or misbehaving, to perform a number allows the entity, if subverted or misbehaving, to perform a number
of very serious attacks as it has full access. It can perform all of very serious attacks as it has full access. It can perform all
the attacks possible (see RFC 3550 and any applicable profiles) as if the attacks possible (see RFC 3550 and any applicable profiles) as if
the media session were not protected at all, while giving the the media session were not protected at all, while giving the
impression to the human session participants that they are protected. impression to the human session participants that they are protected.
Transport Translators have no interactions with cryptography that Transport Translators have no interactions with cryptography that
works above the transport layer, such as SRTP, since that sort of work above the transport layer, such as SRTP, since that sort of
Translator leaves the RTP header and payload unaltered. Media translator leaves the RTP header and payload unaltered. Media
Translators, on the other hand, have strong interactions with Translators, on the other hand, have strong interactions with
cryptography, since they alter the RTP payload. A media Translator cryptography, since they alter the RTP payload. A Media Translator
in a session that uses cryptographic protection needs to perform in a session that uses cryptographic protection needs to perform
cryptographic processing to both inbound and outbound packets. cryptographic processing to both inbound and outbound packets.
A media Translator may need to use different cryptographic keys for A Media Translator may need to use different cryptographic keys for
the inbound and outbound processing. For SRTP, different keys are the inbound and outbound processing. For SRTP, different keys are
required, because an RFC 3550 media Translator leaves the SSRC required, because an RFC 3550 Media Translator leaves the SSRC
unchanged during its packet processing, and SRTP key sharing is only unchanged during its packet processing, and SRTP key sharing is only
allowed when distinct SSRCs can be used to protect distinct packet allowed when distinct SSRCs can be used to protect distinct packet
streams. streams.
When the media Translator uses different keys to process inbound and When the Media Translator uses different keys to process inbound and
outbound packets, each session participant needs to be provided with outbound packets, each session participant needs to be provided with
the appropriate key, depending on whether they are listening to the the appropriate key, depending on whether they are listening to the
Translator or the original source. (Note that there is an translator or the original source. (Note that there is an
architectural difference between RTP media translation, in which architectural difference between RTP media translation, in which
participants can rely on the RTP Payload Type field of a packet to participants can rely on the RTP payload type field of a packet to
determine appropriate processing, and cryptographically protected determine appropriate processing, and cryptographically protected
media translation, in which participants must use information that is media translation, in which participants must use information that is
not carried in the packet.) not carried in the packet.)
When using security mechanisms with Translators, SFMs and Mixers, it When using security mechanisms with translators, SFMs, and mixers, it
is possible that the Translator, SFM or Mixer could create different is possible that the translator, SFM, or mixer could create different
security associations for the different domains they are working in. security associations for the different domains they are working in.
Doing so has some implications: Doing so has some implications:
First, it might weaken security if the Mixer/Translator accepts a First, it might weaken security if the mixer/translator accepts a
weaker algorithm or key in one domain than in another. Therefore, weaker algorithm or key in one domain rather than in another.
care should be taken that appropriately strong security parameters Therefore, care should be taken that appropriately strong security
are negotiated in all domains. In many cases, "appropriate" parameters are negotiated in all domains. In many cases,
translates to "similar" strength. If a key management system does "appropriate" translates to "similar" strength. If a key-management
allow the negotiation of security parameters resulting in a different system does allow the negotiation of security parameters resulting in
strength of the security, then this system should notify the a different strength of the security, then this system should notify
participants in the other domains about this. the participants in the other domains about this.
Second, the number of crypto contexts (keys and security related Second, the number of crypto contexts (keys and security-related
state) needed (for example, in SRTP [RFC3711]) may vary between state) needed (for example, in SRTP [RFC3711]) may vary between
Mixers, SFMs and Translators. A Mixer normally needs to represent mixers, SFMs, and translators. A mixer normally needs to represent
only a single SSRCs per domain and therefore needs to create only one only a single SSRC per domain and therefore needs to create only one
security association (SRTP crypto context) per domain. In contrast, security association (SRTP crypto context) per domain. In contrast,
a Translator needs one security association per participant it a translator needs one security association per participant it
translates towards, in the opposite domain. Considering Figure 11, translates towards, in the opposite domain. Considering Figure 11,
the Translator needs two security associations towards the multicast the translator needs two security associations towards the multicast
domain, one for B and one for D. It may be forced to maintain a set 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 ASM, SSM, Relay, and Mesh (with common RTP session) topologies
session are all topologies with multiple endpoints that require each have multiple endpoints that require shared knowledge about the
shared knowledge about the different crypto contexts for the different crypto contexts for the endpoints. These multiparty
endpoints. These multi-party topologies have special requirements on topologies have special requirements on the key management as well as
the key-management as well as the security functions. Specifically the security functions. Specifically, source authentication in these
source-authentication in these environments has special requirements. 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
document. Please review RTP Security Options [RFC7201] to get a document. Please review RTP Security Options [RFC7201] to get a
better understanding of most of the available options. better understanding of most of the available options.
7. IANA Considerations 7. References
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
8. Acknowledgements
The authors would like to thank Mark Baugher, Bo Burman, Ben
Campbell, Umesh Chandra, Alex Eleftheriadis, Roni Even, Ladan Gharai,
Geoff Hunt, Suresh Krishnan, Keith Lantz, Jonathan Lennox, Scarlet
Liuyan, Suhas Nandakumar, Colin Perkins, and Dan Wing for their help
in reviewing and improving this document.
9. References
9.1. Normative References 7.1. 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, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [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,
2006. DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
9.2. Informative References [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Grouping Semantics and
Mechanisms for Real-Time Transport Protocol (RTP)
Sources", RFC 7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] 7.2. Informative References
[MULTI-STREAM-OPT]
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-05 (work Work in Progress, draft-ietf-avtcore-rtp-multi-stream-
in progress), February 2015. optimisation-08, October 2015.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-22 (work in progress), June 2015.
[I-D.westerlund-avtext-sdes-hdr-ext]
Westerlund, M., Even, R., and M. Zanaty, "RTP Header
Extension for RTCP Source Description Items", draft-
westerlund-avtext-sdes-hdr-ext-03 (work in progress),
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, DOI 10.17487/RFC1112, August 1989,
<http://www.rfc-editor.org/info/rfc1112>.
[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,
2001. DOI 10.17487/RFC3022, January 2001,
<http://www.rfc-editor.org/info/rfc3022>.
[RFC3569] Bhattacharyya, S., "An Overview of Source-Specific [RFC3569] Bhattacharyya, S., Ed., "An Overview of Source-Specific
Multicast (SSM)", RFC 3569, July 2003. Multicast (SSM)", RFC 3569, DOI 10.17487/RFC3569, July
2003, <http://www.rfc-editor.org/info/rfc3569>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004. RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Initiation Protocol (SIP) Event Package for Conference Session Initiation Protocol (SIP) Event Package for
State", RFC 4575, August 2006. Conference State", RFC 4575, DOI 10.17487/RFC4575, August
2006, <http://www.rfc-editor.org/info/rfc4575>.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for [RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for
IP", RFC 4607, August 2006. IP", RFC 4607, DOI 10.17487/RFC4607, August 2006,
<http://www.rfc-editor.org/info/rfc4607>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, [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, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008. DOI 10.17487/RFC5117, January 2008,
<http://www.rfc-editor.org/info/rfc5117>.
[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, DOI 10.17487/RFC5285, July
2008, <http://www.rfc-editor.org/info/rfc5285>.
[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,
DOI 10.17487/RFC5760, February 2010,
<http://www.rfc-editor.org/info/rfc5760>.
[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,
DOI 10.17487/RFC5766, April 2010,
<http://www.rfc-editor.org/info/rfc5766>.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax, [RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP "Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011. Sessions", RFC 6285, DOI 10.17487/RFC6285, June 2011,
<http://www.rfc-editor.org/info/rfc6285>.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time [RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
Transport Protocol (RTP) Header Extension for Mixer-to- time Transport Protocol (RTP) Header Extension for Mixer-
Client Audio Level Indication", RFC 6465, December 2011. to-Client Audio Level Indication", RFC 6465,
DOI 10.17487/RFC6465, December 2011,
<http://www.rfc-editor.org/info/rfc6465>.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, April 2014. Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
<http://www.rfc-editor.org/info/rfc7201>.
[RTCP-SDES]
Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
Header Extension for RTCP Source Description Items", Work
in Progress, draft-ietf-avtext-sdes-hdr-ext-02, July 2015.
[SDP-BUNDLE]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", Work in Progress,
draft-ietf-mmusic-sdp-bundle-negotiation-23, July 2015.
Acknowledgements
The authors would like to thank Mark Baugher, Bo Burman, Ben
Campbell, Umesh Chandra, Alex Eleftheriadis, Roni Even, Ladan Gharai,
Geoff Hunt, Suresh Krishnan, Keith Lantz, Jonathan Lennox, Scarlet
Liuyan, Suhas Nandakumar, Colin Perkins, and Dan Wing for their help
in reviewing and improving this document.
Authors' Addresses Authors' Addresses
Magnus Westerlund Magnus Westerlund
Ericsson Ericsson
Farogatan 6 Farogatan 2
SE-164 80 Kista SE-164 80 Kista
Sweden Sweden
Phone: +46 10 714 82 87 Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com Email: magnus.westerlund@ericsson.com
Stephan Wenger Stephan Wenger
Vidyo Vidyo
433 Hackensack Ave 433 Hackensack Ave
Hackensack, NJ 07601 Hackensack, NJ 07601
USA United States
Email: stewe@stewe.org Email: stewe@stewe.org
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