draft-ietf-avtcore-rtp-topologies-update-02.txt   draft-ietf-avtcore-rtp-topologies-update-03.txt 
Network Working Group M. Westerlund Network Working Group M. Westerlund
Internet-Draft Ericsson Internet-Draft Ericsson
Obsoletes: 5117 (if approved) S. Wenger Obsoletes: 5117 (if approved) S. Wenger
Intended status: Informational Vidyo Intended status: Informational Vidyo
Expires: November 28, 2014 May 27, 2014 Expires: February 8, 2015 August 7, 2014
RTP Topologies RTP Topologies
draft-ietf-avtcore-rtp-topologies-update-02 draft-ietf-avtcore-rtp-topologies-update-03
Abstract Abstract
This document discusses point to point and multi-endpoint topologies This document discusses point to point and multi-endpoint topologies
used in Real-time Transport Protocol (RTP)-based environments. In used in Real-time Transport Protocol (RTP)-based environments. In
particular, centralized topologies commonly employed in the video particular, centralized topologies commonly employed in the video
conferencing industry are mapped to the RTP terminology. conferencing industry are mapped to the RTP terminology.
This document is updated with additional topologies and is intended This document is updated with additional topologies and is intended
to replace RFC 5117. to replace RFC 5117.
skipping to change at page 1, line 37 skipping to change at page 1, line 37
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on November 28, 2014. This Internet-Draft will expire on February 8, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
skipping to change at page 2, line 13 skipping to change at page 2, line 13
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 3 2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . 4 2.2. Definitions related to RTP grouping taxonomy . . . . . . 4
3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 4 3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.2. Point to Point via Middlebox . . . . . . . . . . . . . . 5 3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 5
3.2.1. Translators . . . . . . . . . . . . . . . . . . . . . 5 3.2. Point to Point via Middlebox . . . . . . . . . . . . . . 6
3.2.2. Back to Back RTP sessions . . . . . . . . . . . . . . 9 3.2.1. Translators . . . . . . . . . . . . . . . . . . . . . 6
3.3. Point to Multipoint Using Multicast . . . . . . . . . . . 10 3.2.2. Back to Back RTP sessions . . . . . . . . . . . . . . 10
3.3.1. Any Source Multicast (ASM) . . . . . . . . . . . . . 10 3.3. Point to Multipoint Using Multicast . . . . . . . . . . . 11
3.3.2. Source Specific Multicast (SSM) . . . . . . . . . . . 11 3.3.1. Any Source Multicast (ASM) . . . . . . . . . . . . . 11
3.3.3. SSM with Local Unicast Resources . . . . . . . . . . 13 3.3.2. Source Specific Multicast (SSM) . . . . . . . . . . . 12
3.4. Point to Multipoint Using Mesh . . . . . . . . . . . . . 15 3.3.3. SSM with Local Unicast Resources . . . . . . . . . . 14
3.5. Point to Multipoint Using the RFC 3550 Translator . . . . 18 3.4. Point to Multipoint Using Mesh . . . . . . . . . . . . . 16
3.5.1. Relay - Transport Translator . . . . . . . . . . . . 18 3.5. Point to Multipoint Using the RFC 3550 Translator . . . . 19
3.5.2. Media Translator . . . . . . . . . . . . . . . . . . 19 3.5.1. Relay - Transport Translator . . . . . . . . . . . . 19
3.6. Point to Multipoint Using the RFC 3550 Mixer Model . . . 20 3.5.2. Media Translator . . . . . . . . . . . . . . . . . . 20
3.6.1. Media Mixing . . . . . . . . . . . . . . . . . . . . 22 3.6. Point to Multipoint Using the RFC 3550 Mixer Model . . . 21
3.6.2. Media Switching . . . . . . . . . . . . . . . . . . . 25 3.6.1. Media Mixing Mixer . . . . . . . . . . . . . . . . . 23
3.7. Selective Forwarding Middlebox . . . . . . . . . . . . . 27 3.6.2. Media Switching . . . . . . . . . . . . . . . . . . . 26
3.8. Point to Multipoint Using Video Switching MCUs . . . . . 30 3.7. Selective Forwarding Middlebox . . . . . . . . . . . . . 28
3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 32 3.8. Point to Multipoint Using Video Switching MCUs . . . . . 31
3.10. Split Component Endpoint . . . . . . . . . . . . . . . . 33 3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 33
3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 34 3.10. Split Component Terminal . . . . . . . . . . . . . . . . 34
3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 35 3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 36
4. Comparing Topologies . . . . . . . . . . . . . . . . . . . . 35 3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 36
4.1. Topology Properties . . . . . . . . . . . . . . . . . . . 36 4. Comparing Topologies . . . . . . . . . . . . . . . . . . . . 37
4.1.1. All to All Media Transmission . . . . . . . . . . . . 36 4.1. Topology Properties . . . . . . . . . . . . . . . . . . . 37
4.1.2. Transport or Media Interoperability . . . . . . . . . 37 4.1.1. All to All Media Transmission . . . . . . . . . . . . 37
4.1.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . 37 4.1.2. Transport or Media Interoperability . . . . . . . . . 38
4.1.4. Aggregation of Media . . . . . . . . . . . . . . . . 37 4.1.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . 38
4.1.5. View of All Session Participants . . . . . . . . . . 38 4.1.4. Aggregation of Media . . . . . . . . . . . . . . . . 39
4.1.6. Loop Detection . . . . . . . . . . . . . . . . . . . 38 4.1.5. View of All Session Participants . . . . . . . . . . 39
4.2. Comparison of Topologies . . . . . . . . . . . . . . . . 39 4.1.6. Loop Detection . . . . . . . . . . . . . . . . . . . 39
5. Security Considerations . . . . . . . . . . . . . . . . . . . 39 4.2. Comparison of Topologies . . . . . . . . . . . . . . . . 40
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 41 5. Security Considerations . . . . . . . . . . . . . . . . . . . 40
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 41 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 42
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 41 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 43
8.1. Normative References . . . . . . . . . . . . . . . . . . 41 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 43
8.2. Informative References . . . . . . . . . . . . . . . . . 42 8.1. Normative References . . . . . . . . . . . . . . . . . . 43
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 43 8.2. Informative References . . . . . . . . . . . . . . . . . 43
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44
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 of RTP and RTCP. This document tries to address past and behavior of RTP and RTCP. This document tries to address past and
existing confusion, especially with respect to terms not defined in existing confusion, especially with respect to terms not defined in
RTP but in common use in the conversational communication industry, RTP but in common use in the conversational communication industry,
such as the Multipoint Control Unit or MCU. such as the Multipoint Control Unit or MCU.
skipping to change at page 4, line 20 skipping to change at page 4, line 20
PtM: Point to Multipoint PtM: Point to Multipoint
PtP: Point to Point PtP: Point to Point
SFU: Selective Forwarding Unit SFU: Selective Forwarding Unit
SSM: Source-Specific Multicast SSM: Source-Specific Multicast
SSRC: Synchronization Source SSRC: Synchronization Source
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-
grouping-taxonomy-02, and are used in capitalized form throughout the
document.
Communication Session: A Communication Session is an association
among group of participants communicating with each other via a
set of Multimedia Sessions.
End Point: A single addressable entity sending or receiving RTP
packets. It may be decomposed into several functional blocks, but
as long as it behaves as a single RTP stack entity it is
classified as a single "End Point".
Media Source: A Media Source is the logical source of a reference
clock synchronized, time progressing, digital media stream, called
a Source Stream.
Multimedia Session: A multimedia session is an association among a
group of participants engaged in the communication via one or more
RTP 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 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 SFUs. The section ends by
discussing de-composited endpoints, asymmetric middlebox behaviors discussing de-composited 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 End
endpoints, communicating using unicast. Both RTP and RTCP traffic Points, communicating using unicast. Both RTP and RTCP traffic are
are conveyed endpoint-to-endpoint, using unicast traffic only (even conveyed endpoint-to-endpoint, using unicast traffic only (even if,
if, in exotic cases, this unicast traffic happens to be conveyed over in exotic cases, this unicast traffic happens to be conveyed over an
an IP-multicast address). 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 End Points and combining the requirements stemming
from them. Note that an endpoint can still use multiple RTP from them. Note that an End Point 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 End Points. 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 which use multiple SSRC per End Point it can be relevant to implement
support for cross-reporting suppression as defined in "Sending support for cross-reporting suppression as defined in "Sending
Multiple Media Streams in a Single RTP Session" Multiple Media Streams in a Single RTP Session"
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]. [I-D.ietf-avtcore-rtp-multi-stream-optimisation].
3.2. Point to Point via Middlebox 3.2. Point to Point via Middlebox
This section discusses cases where two endpoints communicate but have This section discusses cases where two End Points communicate but
one or more middleboxes involved in the RTP session. have 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 media common attributes that separate them from Mixers. For each RTP
stream that the Translator receives, it generates an individual stream that the Translator receives, it generates an individual RTP
stream in the other domain. A translator keeps the SSRC for a stream stream in the other domain. A translator keeps the SSRC for an RTP
across the translation, whereas a Mixer can select a single media stream across the translation, whereas a Mixer can select a single
stream, or send out multiple mixed media streams, but always under RTP stream from multiple received RTP streams (in cases like audio/
its own SSRC, possibly using the CSRC field to indicate the source(s) video switching), or send out an RTP stream composed of multiple
of the content. Mixers are more common in point to multipoint cases mixed media received in multiple RTP streams (in cases like audio
than in PtP. The reason is that in PtP use cases the primary focus mixing or video tiling), but always under its own SSRC, possibly
is interoperability, such as transcoding to a codec the receiver using the CSRC field to indicate the source(s) of the content.
supports, which can be done by a media translator. Mixers are more common in point to multipoint cases than in PtP. The
reason is that in PtP use cases the primary focus of a middlebox is
enabling interoperability, between otherwise non-interoperable End
Points, such as transcoding to a codec the receiver supports, which
can be done by a media translator.
As specified in Section 7.1 of [RFC3550], the SSRC space is common 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 participants to run SSRC collision detection, responsibility of the End Points (as the RTP session participants) to
and the SSRC is thus a field the Translator cannot change. Any SDES run SSRC collision detection, and the SSRC is thus a field the
information associated with a SSRC or CSRC also needs to be forwarded Translator cannot change. Any SDES information associated with a
between the domains for any SSRC/CSRC used in the different domains. SSRC or CSRC also needs to be forwarded between the domains for any
SSRC/CSRC used in the different domains.
A Translator commonly does not use an SSRC of its own, and is not A Translator commonly does not use an SSRC of its own, and is not
visible as an active participant in the session. One reason to have visible as an active participant in the RTP session. One reason to
its own SSRC is when a Translator acts as a quality monitor that have its own SSRC is when a Translator acts as a quality monitor that
sends RTCP reports and therefore is required to have an SSRC. sends RTCP reports and therefore is required to have an SSRC.
Another example is the case when a Translator is prepared to use RTCP Another example is the case when a Translator is prepared to use RTCP
feedback messages. This may, for example, occur in a translator feedback messages. This may, for example, occur in a translator
configured to detect packet loss of important video packets and wants configured to detect packet loss of important video packets and wants
to trigger repair by the media sender, by sending feedback messages. to trigger repair by the media sending End Point, by sending feedback
While such feedback could use the SSRC of the target for the messages. While such feedback could use the SSRC of the target for
translator, this in turn would require translation of the targets the translator (the receiving End Point), this in turn would require
RTCP reports to make them consistent. It may be simpler to expose an translation of the targets RTCP reports to make them consistent. It
additional SSRC in the session. The only concern is endpoints may be simpler to expose an additional SSRC in the session. The only
failing to support the full RTP specification, thus having issues concern is End Points failing to support the full RTP specification
with multiple SSRCs reporting on the RTP streams sent by that may have issues with multiple SSRCs reporting on the RTP streams sent
endpoint. by that End Point, as this use case may be viewed as excotic 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
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 RTP endpoints on the transport level, e.g., to inserted between two End Points 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 media flow public address domain relay, network topologies where the RTP stream
is required to pass a particular point for audit by employing is required to pass a particular point for audit by employing
relaying, or preserving privacy by hiding each peer's transport relaying, or preserving privacy by hiding each peer's transport
addresses to the other party. Other protocols or functionalities addresses to the other party. Other protocols or functionalities
that provide this behavior are TURN [RFC5766] servers, Session Border that provide this behavior are TURN [RFC5766] servers, Session Border
Gateways and Media Processing Nodes with media anchoring Gateways and Media Processing Nodes with media anchoring
functionalities. functionalities.
+---+ +---+ +---+ +---+ +---+ +---+
| A |<------>| T |<------->| B | | A |<------>| T |<------->| B |
+---+ +---+ +---+ +---+ +---+ +---+
Figure 2: Point to Point with Translator Figure 2: Point to Point with Translator
A common element in these functions is that they are normally A common element in these functions is that they are normally
transparent at the RTP level, i.e., they perform no changes on any transparent at the RTP level, i.e., they perform no changes on any
RTP or RTCP packet fields and only affect the lower layers. They may RTP or RTCP packet fields and only affect the lower layers. They may
affect, however, the path the RTP and RTCP packets are routed between affect, however, the path the RTP and RTCP packets are routed between
the endpoints in the RTP session, and thereby only indirectly affect the End Points in the RTP session, and thereby indirectly affect the
the RTP session. For this reason, one could believe that transport RTP session. For this reason, one could believe that transport
translator-type middleboxes do not need to be included in this translator-type middleboxes do not need to be included in this
document. This topology, however, can raise additional requirements document. This topology, however, can raise additional requirements
in the RTP implementation and its interactions with the signalling in the RTP implementation and its interactions with the signalling
solution. Both in signalling and in certain RTCP fields, network solution. Both in signalling and in certain RTCP fields, network
addresses other than those of the relay can occur since B has a addresses other than those of the relay can occur since B has a
different network address than the relay (T). Implementations that different network address than the relay (T). Implementations that
can not support this will also not work correctly when endpoints are cannot support this will also not work correctly when End Points are
subject to NAT. subject to NAT.
The transport relay implementation also have some considerations, The transport relay implementations also have to take into account
where security considerations are an important aspect. Source security considerations. In particular, source address filtering of
address filtering of incoming packets are usually important in incoming packets is usually important in relays, to prevent attackers
relays, to prevent attackers to inject traffic into a session, which to inject traffic into a session, which one peer may, in the absence
one peer will think comes from the other peer. fo adequate security in the relay, think it comes from the other
peer.
3.2.1.2. Transport Translator 3.2.1.2. Transport Translator
Transport Translators (Topo-Trn-Translator) do not modify the media 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 parameters, in the sense of this section, comprise the Transport parameters, in the sense of this section, comprise the
transport addresses (to bridge different domains such unicast to transport addresses (to bridge different domains such unicast to
multicast) and the media packetization to allow other transport multicast) and the media packetization to allow other transport
protocols to be interconnected to a session (in gateways). Of the protocols to be interconnected to a session (in gateways).
transport Translators, this memo is primarily interested in those
that use RTP on both sides, and this is assumed henceforth.
Translators that bridge between different protocol worlds need to be 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
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
Media Translators (Topo-Media-Translator) modify the media stream Media Translators (Topo-Media-Translator) modify the media inside the
itself. This process is commonly known as transcoding. The RTP stream. This process is commonly known as transcoding. The
modification of the media stream can be as small as removing parts of modification of the media can be as small as removing parts of the
the stream, and it can go all the way to a full decoding and re- stream, and it can go all the way to a full decoding and re-encoding
encoding (down to the sample level or equivalent) utilizing a (down to the sample level or equivalent) utilizing a different media
different media codec. Media Translators are commonly used to codec. Media Translators are commonly used to connect End Points
connect entities without a common interoperability point in the media without a common interoperability point in the media encoding.
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
stream and the transport aspects of a stream between two transport and the transport aspects of the RTP stream carrying the media
domains (or clouds). 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 B's RTCP Receiver Report before case, the Translator needs to rewrite End Point B's RTCP Receiver
forwarding them to A. The rewriting is needed as the stream received Report before forwarding them to End Point A. The rewriting is
by B is not the same stream as the other participants receive. For needed as the RTP stream received by B is not the same RTP stream as
example, the number of packets transmitted to B may be lower than the other participants receive. For example, the number of packets
what A sends, due to the different media format and data rate. transmitted to B may be lower than what A sends, due to the different
Therefore, if the Receiver Reports were forwarded without changes, media format and data rate. Therefore, if the Receiver Reports were
the extended highest sequence number would indicate that B were forwarded without changes, the extended highest sequence number would
substantially behind in reception, while most likely it would not be. indicate that B were substantially behind in reception, while most
Therefore, the Translator must translate that number to a likely it would not be. Therefore, the Translator must translate
corresponding sequence number for the stream the Translator received. that number to a corresponding sequence number for the stream the
Similar arguments can be made for most other fields in the RTCP Translator received. Similar requirements exists for most other
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 and respond to RTCP feedback messages. This may occur, for source (the End Point originally sending the media to the Translator)
example, when a receiver requests a bandwidth reduction, and the and respond to RTCP feedback messages. This may occur, for example,
when a receiving End Point 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 media source and itself. In that bandwidth reduction between the sending End Point and itself. In
case, it is sensible that the media Translator reacts to the codec that case, it is sensible that the media Translator reacts to codec
control messages itself, for example, by transcoding to a lower media control messages itself, for example, by transcoding to a lower media
rate. rate.
A variant of translator behaviour worth pointing out is the one A variant of translator behaviour worth pointing out is the one
depicted in Figure 3 of an endpoint A sends a media flow to B. On depicted in Figure 3 of an End Point A sending a RTP stream
the path there is a device T that on A's behalf does something with containing media (only) to B. On the path there is a device T that
the media streams, for example adds an RTP session with FEC on A's behalf manipulates the RTP streams. One common example is
information for A's media streams. In this case, T needs to bind the that T adds a second RTP stream containing Forward Error Correction
new FEC streams to A's media stream, for example by using the same (FEC) information in order to protect A's (non FEC-protected) RTP
CNAME as A. stream. In this case, T needs to semantically bind the new FEC RTP
stream to A's media-carrying RTP stream, for example by using the
same CNAME as A.
+------+ +------+ +------+ +------+ +------+ +------+
| | | | | | | | | | | |
| A |------->| T |-------->| B | | A |------->| T |-------->| B |
| | | |---FEC-->| | | | | |---FEC-->| |
+------+ +------+ +------+ +------+ +------+ +------+
Figure 3: When De-composition is a Translator Figure 3: Media Translator adding FEC
This type of functionality where T does something with the media there may also be cases where information is added into the original
stream on behalf of A is covered under the media translator RTP stream, while leaving most or all of the original RTP packets
definition. intact (with the exception of certain RTP header fields, such as the
sequence number). One example is the injection of meta-data into the
RTP stream, carried in their own RTP packets.
Similarly, a Media Translator can sometimes remove information from
the RTP stream, while otherwise leaving teh remaining RTP packets
unchanged (again with the exception of certain RTP header fields).
Either type of functionality where T manipulates the RTP stream, or
adds an accompanying RTP stream, on behalf of A is also covered under
the media translator definition.
3.2.2. Back to Back RTP sessions 3.2.2. Back to Back RTP sessions
There exist middleboxes that interconnect two endpoints through There exist middleboxes that interconnect two End Points A and B
themselves, but not by being part of a common RTP session. They through themselves (MB), but not by being part of a common RTP
establish instead two different RTP sessions, one between A and the session. They establish instead two different RTP sessions, one
middlebox and another between the middlebox and B. This topology is between A and the middlebox and another between the middlebox and B.
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: When De-composition is a Translator 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 with the single RTP session context is transcoding. The difference of this topology relative to the single
the handling of the SSRCs and the other session-related identifiers, RTP session context is the handling of the SSRCs and the other
such as CNAMEs. With two different RTP sessions these can be freely session-related identifiers, such as CNAMEs. With two different RTP
changed and it becomes the middlebox's task to maintain the correct sessions these can be freely changed and it becomes the middlebox's
relations. respnsibility to maintain the correct relations.
The signalling or other above-RTP level functionalities referencing The signalling or other above-RTP level functionalities referencing
RTP media streams may be what is most impacted by using two RTP RTP streams may be what is most impacted by using two RTP sessions
sessions and changing identifiers. The structure with two RTP and changing identifiers. The structure with two RTP sessions also
sessions also puts a congestion control requirement on the middlebox, puts a congestion control requirement on the middlebox, because it
because it becomes fully responsible for the media stream it sources becomes fully responsible for the media stream it sources into each
into each of the sessions. of the sessions.
Adherence to congestion control can be solved locally or by bridging Adherence to congestion control can be solved locally on each of the
also statistics from the receiving endpoint. From an implementation two segments, or by bridging statistics from the receiving End Point
point, however, this requires dealing with a number of through the middlebox to the sending End Point. From an
inconsistencies. First, packet loss must be detected for an RTP flow implementation point, however, the latter requires dealing with a
sent from A to the middlebox, and that loss must be reported through number of inconsistencies. First, packet loss must be detected for
a skipped sequence number in the flow from the middlebox to B. This an RTP stream sent from A to the middlebox, and that loss must be
coupling and the resulting inconsistencies is conceptually easier to reported through a skipped sequence number in the RTP stream from the
handle when considering the two flows as belonging to a single RTP middlebox to B. This coupling and the resulting inconsistencies are
session. conceptually easier to handle when considering the two RTP streams as
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. Both these only a particular IP host sends to the multicast group. Both these
models are discussed below in their respective sections. models are discussed below in their respective sections.
skipping to change at page 10, line 36 skipping to change at page 11, line 37
+---+ / 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 topology as a transmission model, in which traffic from any multicast
participant reaches all the other participants, except for cases such group participant reaches all the other multicast group participants,
as: except for cases such as:
o packet loss, or o packet loss, or
o when a participant does not wish to receive the traffic for a o when a multicast group participant does not wish to receive the
specific multicast group and, therefore, has not subscribed to the traffic for a specific multicast group and, therefore, has not
IP multicast group in question. This scenario can occur, for subscribed to the IP multicast group in question. This scenario
example, where a multimedia session is distributed using two or can occur, for example, where a multimedia session is distributed
more multicast groups and a participant is subscribed only to a using two or more multicast groups and a multicast group
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 participants can vary between one and many, traffic. The number of multicast group participants can vary between
as RTP and RTCP scale to very large multicast groups (the theoretical one and many, as RTP and RTCP scale to very large multicast groups
limit of the number of participants in a single RTP session is in the (the theoretical limit of the number of participants in a single RTP
range of billions). The above can be realized using Any Source session is in the range of billions). The above can be realized
Multicast (ASM). using Any Source Multicast (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 participants is so low (and other as a group where the number of multicast group participants is so low
factors such as the connectivity is so good) that it allows the (and other factors such as the connectivity is so good) that it
participants to use early or immediate feedback, as defined in AVPF allows the participants to use early or immediate feedback, as
[RFC4585]. Even when the environment would allow for the use of a defined in AVPF [RFC4585]. Even when the environment would allow for
small multicast group, some applications may still want to use the the use of a small multicast group, some applications may still want
more limited options for RTCP feedback available to large multicast to use the more limited options for RTCP feedback available to large
groups, for example when there is a likelihood that the threshold of multicast groups, for example when there is a likelihood that the
the small multicast group (in terms of participants) may be exceeded threshold of the small multicast group (in terms of multicast group
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 node discussed in [RFC4585] is typically required. Each individual End
needs to process every feedback message it receives, not only to Point that is a multicast group participant needs to process every
determine if it is affected or if the feedback message applies only feedback message it receives, not only to determine if it is affected
to some other participant, but also to derive timing restrictions for or if the feedback message applies only to some other End Point, but
the sending of its own feedback messages, if any. also to derive timing restrictions for the sending of its own
feedback messages, if any.
3.3.2. Source Specific Multicast (SSM) 3.3.2. Source Specific Multicast (SSM)
In Any Source Multicast, any of the participants can send to all the In Any Source Multicast, any of the multicast group participants can
other participants, by sending a packet to the multicast group. In send to all the other multicast group participants, by sending a
contrast, Source Specific Multicast [RFC3569][RFC4607] refers to packet to the multicast group. In contrast, Source Specific
scenarios where only a single source (Distribution Source) can send Multicast [RFC3569][RFC4607] refers to scenarios where only a single
to the multicast group, creating a topology that looks like the one source (Distribution Source) can send to the multicast group,
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 | | | | | |
skipping to change at page 12, line 29 skipping to change at page 13, line 29
|Sender M|<----->| | |<-------------------------+ |Sender M|<----->| | |<-------------------------+
+--------+ +-----+ RTCP Unicast +--------+ +-----+ RTCP Unicast
FT = Feedback Target FT = Feedback Target
Transport from the Feedback Target to the Distribution Transport from the Feedback Target to the Distribution
Source is via unicast or multicast RTCP if they are not Source is via unicast or multicast RTCP if they are not
co-located. co-located.
Figure 6: Point to Multipoint using Source Specific Multicast Figure 6: Point to Multipoint using Source Specific Multicast
In the SSM topology (Figure 6) a number of RTP sources (1 to M) are In the SSM topology (Figure 6) a number of RTP sending End Points
allowed to send media to the SSM group. These sources send media to (RTP sources henceforth) (1 to M) are allowed to send media to the
a dedicated distribution source, which forwards the media streams to SSM group. These sources send media to a dedicated distribution
the multicast group on behalf of the original senders. The media source, which forwards the RTP streams to the multicast group on
streams reach the Receivers (R(1) to R(n)). The Receivers' RTCP behalf of the original RTP sources. The RTP streams reach the
messages cannot be sent to the multicast group, as the SSM multicast receiving End Points (Receivers henceforth) (R(1) to R(n)). The
group by definition has only a single IP sender. To support RTCP, an Receivers' RTCP messages cannot be sent to the multicast group, as
RTP extension for SSM [RFC5760] was defined. It uses unicast the SSM multicast group by definition has only a single IP sender.
transmission to send RTCP from each of the receivers to one or more To support RTCP, an RTP extension for SSM [RFC5760] was defined. It
Feedback Targets (FT). The feedback targets relay the RTCP uses unicast transmission to send RTCP from each of the receivers to
unmodified, or provide a summary of the participants RTCP reports one or more Feedback Targets (FT). The feedback targets relay the
towards the whole group by forwarding the RTCP traffic to the RTCP unmodified, or provide a summary of the participants RTCP
reports towards the whole group by forwarding the RTCP traffic to the
distribution source. Figure 6 only shows a single feedback target 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 many and have responsibility for sub-groups of the receivers. For be distributed and each instance can have responsibility for sub-
summary reports, however, there must be a single feedback aggregating groups of the receivers. For summary reports, however, there
all the summaries to a common message to the whole receiver group. typically must be a single feedback target aggregating all the
summaries to a common message to the whole receiver group.
The RTP extension for SSM specifies how feedback (both reception The RTP extension for SSM specifies how feedback (both reception
information and specific feedback events) are handled. The more information and specific feedback events) are handled. The more
general problems associated with the use of multicast, where everyone general problems associated with the use of multicast, where everyone
receives what the distribution source sends needs to be accounted receives what the distribution source sends needs to be accounted
for. for.
Aforementioned situation results in common behavior for RTP Aforementioned situation results in common behavior for RTP
multicast: multicast:
1. Multicast applications often use a group of RTP sessions, not 1. Multicast applications often use a group of RTP sessions, not
one. Each endpoint needs to be a member of most or all of these one. Each End Point needs to be a member of most or all of these
RTP sessions in order to perform well. RTP sessions in order to perform well.
2. Within each RTP session, the number of media sinks is likely to 2. Within each RTP session, the number of media sinks is likely to
be much larger than the number of RTP sources. be much larger than the number of RTP sources.
3. Multicast applications need signalling functions to identify the 3. Multicast applications need signalling functions to identify the
relationships between RTP sessions. relationships between RTP sessions.
4. Multicast applications need signalling functions to identify the 4. Multicast applications need signalling functions to identify the
relationships between SSRCs in different RTP sessions. relationships between SSRCs in different RTP sessions.
All multicast configurations share a signalling requirement: all of All multicast configurations share a signalling requirement: all of
the participants need to have the same RTP and payload type the End Points need to have the same RTP and payload type
configuration. Otherwise, A could, for example, be using payload configuration. Otherwise, End Point A could, for example, be using
type 97 to identify the video codec H.264, while B would identify it payload type 97 to identify the video codec H.264, while End Point B
as MPEG-2. would identify it as MPEG-2, with unpredicatble but almost certainly
not visually pleasing results.
Security solutions for this type of group communications are also Security solutions for this type of group communications are also
challenging. First, the key-management and the security protocol challenging. First, the key-management and the security protocol
must support group communication. Source authentication becomes more must support group communication. Source authentication becomes more
difficult and requires special 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
[RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions" [RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions"
results in additional extensions to SSM Topology. results in additional extensions to SSM Topology.
----------- -------------- ----------- --------------
| |------------------------------------>| | | |------------------------------------>| |
| |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| | | |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| |
skipping to change at page 14, line 30 skipping to change at page 15, line 30
- - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- - - - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- -
| | | | | | | |
UNICAST BURST | ------------ | | | UNICAST BURST | ------------ | | |
(or RETRANSMISSION) | | Burst/ | |<~~~~~~~~>| | (or RETRANSMISSION) | | Burst/ | |<~~~~~~~~>| |
RTP SESSION | | Retrans. | |.........>| | RTP SESSION | | Retrans. | |.........>| |
| |Source (BRS)| |<.=.=.=.=>| | | |Source (BRS)| |<.=.=.=.=>| |
| ------------ | | | | ------------ | | |
| | | | | | | |
---------------- -------------- ---------------- --------------
-------> Multicast RTP Flow -------> Multicast RTP Stream
.-.-.-.> Multicast RTCP Flow .-.-.-.> Multicast RTCP Stream
.=.=.=.> Unicast RTCP Reports .=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages ~~~~~~~> Unicast RTCP Feedback Messages
.......> Unicast RTP Flow .......> Unicast RTP Stream
Figure 7 Figure 7
The Rapid acquisition extension allows an endpoint joining an SSM The Rapid acquisition extension allows an End Point joining an SSM
multicast session to request media starting with the last sync-point multicast session to request media starting with the last sync-point
(from where media can be decoded without requiring context (from where media can be decoded without requiring context
established by the decoding of prior packets) to be sent at high established by the decoding of prior packets) to be sent at high
speed until such time where, after decoding of these burst-delivered speed until such time where, after decoding of these burst-delivered
media packets, the correct media timing is established, i.e. media media packets, the correct media timing is established, i.e. media
packets are received within adequate buffer intervals for this packets are received within adequate buffer intervals for this
application. This is accomplished by first establishing a unicast application. This is accomplished by first establishing a unicast
PtP RTP session between the Burst/Retransmission Source (BRS, PtP RTP session between the Burst/Retransmission Source (BRS,
Figure 7) and the RTP Receiver. The unicast session is used to Figure 7) and the RTP Receiver. The unicast session is used to
transmit cached packets from the multicast group at higher then transmit cached packets from the multicast group at higher then
normal speed in order to synchronize the receiver to the ongoing normal speed in order to synchronize the receiver to the ongoing
multicast packet flow. 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. The switches over to receiving directly from the multicast group. The
(still existing) PtP RTP session is, in many deployed applications, (still existing) PtP RTP session is, in many deployed applications,
be used as a repair channel, i.e., for RTP Retransmission traffic of be 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
skipping to change at page 15, line 26 skipping to change at page 16, line 26
\ / \ /
\ / \ /
v v v v
+---+ +---+
| C | | C |
+---+ +---+
Figure 8: Point to Multi-Point using Mesh Figure 8: Point to Multi-Point using Mesh
Based on the RTP session definition, it is clearly possible to have a Based on the RTP session definition, it is clearly possible to have a
joint RTP session over multiple unicast transport flows like the joint RTP session involving three or more End Points over multiple
above joint three endpoint session. In this case, A needs to send unicast transport flows, like the joint three End point session
its' media streams and RTCP packets to both B and C over their depicted above. In this case, A needs to send its RTP streams and
respective transport flows. As long as all participants do the same, RTCP packets to both B and C over their respective transport flows.
everyone will have a joint view of the RTP session. As long as all End Points do the same, everyone will have a joint
view of the RTP session.
This does not create any additional requirements beyond the need to This topology does not create any additional requirements beyond the
have multiple transport flows associated with a single RTP session. need to have multiple transport flows associated with a single RTP
Note that an endpoint may use a single local port to receive all session. Note that an End Point may use a single local port to
these transport flows, or it might have separate local reception receive all these transport flows (in which case the sending port, IP
ports for each of the endpoints. address, or SSRC can be used to demultiplex), or it might have
separate local reception ports for each of the End Points.
+-A--------------------+ +-A--------------------+
|+---+ | |+---+ |
||CAM| | +-B-----------+ ||CAM| | +-B-----------+
|+---+ +-UDP1------| |-UDP1------+ | |+---+ +-UDP1------| |-UDP1------+ |
| | | +-RTP1----| |-RTP1----+ | | | | | +-RTP1----| |-RTP1----+ | |
| V | | +-Video-| |-Video-+ | | | | V | | +-Video-| |-Video-+ | | |
|+----+ | | | |<----------------|BV1 | | | | |+----+ | | | |<----------------|BV1 | | | |
||ENC |----+-+-+--->AV1|---------------->| | | | | ||ENC |----+-+-+--->AV1|---------------->| | | | |
|+----+ | | +-------| |-------+ | | | |+----+ | | +-------| |-------+ | | |
skipping to change at page 16, line 31 skipping to change at page 17, line 31
| | | | +-Video-| |-Video-+ | | | | | | | +-Video-| |-Video-+ | | |
| +-------+-+-+--->AV1|---------------->| | | | | | +-------+-+-+--->AV1|---------------->| | | | |
| | | | |<----------------|CV1 | | | | | | | | |<----------------|CV1 | | | |
| | | +-------| |-------+ | | | | | | +-------| |-------+ | | |
| | +---------| |---------+ | | | | +---------| |---------+ | |
| +-----------| |-----------+ | | +-----------| |-----------+ |
+----------------------+ +-------------+ +----------------------+ +-------------+
Figure 9: An Multi-unicast Mesh with a joint RTP session Figure 9: An Multi-unicast Mesh with a joint RTP session
A joint RTP session from A's perspective for the Mesh depicted in A joint RTP session from End Point A's perspective for the Mesh
Figure 8 with a joint RTP session have multiple transport flows, here depicted in Figure 8 with a joint RTP session have multiple transport
enumerated as UDP1 and UDP2. However, there is only one RTP session flows, here enumerated as UDP1 and UDP2. However, there is only one
(RTP1). The media source (CAM) is encoded and transmitted over the RTP session (RTP1). The Media Source (CAM) is encoded and
SSRC (AV1) across both transport layers. However, as this is a joint transmitted over the SSRC (AV1) across both transport layers.
RTP session, the two streams must be the same. Thus, an congestion However, as this is a joint RTP session, the two streams must be the
control adaptation needed for the paths A to B and A to C needs to same. Thus, an congestion control adaptation needed for the paths A
use the most restricting path's properties. to B and A to C needs to use the most restricting path's properties.
An alternative structure for establishing the above topology is to An alternative structure for establishing the above topology is to
use independent RTP sessions between each pair of peers, i.e., three use independent RTP sessions between each pair of peers, i.e., three
different RTP sessions. In some scenarios, the same RTP media stream different RTP sessions. In some scenarios, the same RTP stream may
may be sent from transmitting endpoint, however it also supports be sent from the transmitting End Point, however it also supports
local adaptation taking place in one or more of the RTP media local adaptation taking place in one or more of the RTP streams,
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 17, line 35 skipping to change at page 18, line 35
| | | +-------| |-------+ | | | | | | +-------| |-------+ | | |
| | +---------| |---------+ | | | | +---------| |---------+ | |
| +-----------| |-----------+ | | +-----------| |-----------+ |
+------------------------+ +-------------+ +------------------------+ +-------------+
Figure 10: An Multi-unicast Mesh with independent RTP session Figure 10: An Multi-unicast Mesh with independent RTP session
Lets review the topology when independent RTP sessions are used, from Lets review the topology when independent RTP sessions are used, from
A's perspective in Figure 8 by considering both how the media is a A's perspective in Figure 8 by considering both how the media is a
handled and the RTP sessions that are set-up in Figure 10. A's handled and the RTP sessions that are set-up in Figure 10. A's
microphone is captured and the digital audio can then be feed into microphone is captured and the digital audio can then be fed into two
two different encoder instances, as each beeing associated with two different encoder instances, as each being associated with two
independent RTP sessions (RTP1 and RTP2). The SSRCs (AA1 and AA2) in independent RTP sessions (RTP1 and RTP2). The SSRCs (AA1 and AA2) in
each RTP session will be completely independent and the media bit- each RTP session are completely independent and the media bit-rate
rate produced by the encoders can also be tuned differently to produced by the encoders can also be tuned differently to address any
address any congestion control requirements differing for the paths A congestion control requirements differing for the paths A to B
to B compared to A to 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 and their connecting flows, an common RTCP bandwidth three End Points A, B, and C, and their connecting RTP streams, a
is calculated and used for this single joint session. In contrast, common RTCP bandwidth is calculated and used for this single joint
when there are multiple independent RTP sessions, each RTP session session. In contrast, when there are multiple independent RTP
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, End Points 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 End Point
configuration in Figure 8, endpoint A has no awareness of the configuration in Figure 8, End Point A has no awareness of the
conditions occurring in the session between endpoints B and C conditions occurring in the session between End Points 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 End Point receives its
own media stream, or a mixed media stream including its own media own media stream, or a mixed media stream including its own media
stream (a condition known as a loop). The identification of loops stream (a condition known as a loop). The identification of loops
and, in most cases, their avoidance, has to be achieved by other and, in most cases, their avoidance, has to be achieved by other
means, for example through signaling or the use of an RTP external means, for example through signaling or the use of an RTP external
name space binding SSRC/CSRC among any communicating RTP sessions in name space binding SSRC/CSRC among any communicating RTP sessions in
the mesh. the mesh.
3.5. Point to Multipoint Using the RFC 3550 Translator 3.5. Point to Multipoint Using the RFC 3550 Translator
This section discusses some additional usages related to point to This section discusses some additional usages related to point to
skipping to change at page 18, line 45 skipping to change at page 19, line 45
+ cast +->| Translator | + cast +->| Translator |
+---+ \ Network / | | +---+ +---+ \ Network / | | +---+
| C |<---\ / | |<---->| D | | C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+ +---+ \ / +------------+ +---+
+-----+ +-----+
Figure 11: Point to Multipoint Using Multicast Figure 11: Point to Multipoint Using Multicast
Figure 11 depicts an example of a Transport Translator performing at Figure 11 depicts an example of a Transport Translator performing at
least IP address translation. It allows the (non-multicast-capable) least IP address translation. It allows the (non-multicast-capable)
participants B and D to take part in an any source multicast session End Points B and D to take part in an any source multicast session
by having the Translator forward their unicast traffic to the involving End Points A and C, by having the Translator forward their
multicast addresses in use, and vice versa. It must also forward B's unicast traffic to the multicast addresses in use, and vice versa.
traffic to D, and vice versa, to provide each of B and D with a It must also forward B's traffic to D, and vice versa, to provide
complete view of the session. each of B and D with a complete view of the session.
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Translator | | Translator |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 12: RTP Translator (Relay) with Only Unicast Paths Figure 12: RTP Translator (Relay) with Only Unicast Paths
Another Translator scenario is depicted in Figure 12. The Translator Another Translator scenario is depicted in Figure 12. The Translator
in this case connects multiple users of a conference through unicast. in this case connects multiple End Points through unicast. This can
This can be implemented using a very simple transport Translator be implemented using a very simple transport Translator which, in
which, in this document, is called a relay. The relay forwards all this document, is called a relay. The relay forwards all traffic it
traffic it receives, both RTP and RTCP, to all other participants. receives, both RTP and RTCP, to all other End Points. In doing so, a
In doing so, a multicast network is emulated without relying on a multicast network is emulated without relying on a multicast-capable
multicast-capable network infrastructure. network infrastructure.
For RTCP feedback this results in a similar set of considerations to For RTCP feedback this results in a similar set of considerations to
those described in the ASM RTP topology. It also puts some those described in the ASM RTP topology. It also puts some
additional signalling requirements onto the session establishment; additional signalling requirements onto the session establishment;
for example, a common configuration of RTP payload types is required. for example, a common configuration of RTP payload types is required.
Transport translators and relays should always consider doing source Transport translators and relays should always consider implementing
address filtering, to prevent attackers to inject traffic using the source address filtering, to prevent attackers to inject traffic
listening ports on the translator. The translator can however go one using the listening ports on the translator. The translator can,
step further, and especially if explicit SSRC signalling is used, however, go one step further, and especially if explicit SSRC
prevent other session participants to send SSRCs that are used by signalling is used, prevent End points to send SSRCs other than its
other participants in the session. This can improve the security own (that are, for example, used by other participants in the
properties of the session, despite the use of group keys that on session). This can improve the security properties of the session,
cryptographic level allows anyone to impersonate another in the same despite the use of group keys that on cryptographic level allows
RTP session. anyone to impersonate another in the same RTP session.
A Translator that doesn't change the RTP/RTCP packets content can be A Translator that doesn't change the RTP/RTCP packets content can be
operated without the requiring the translator to have access to the operated without the requiring it to have access to the security
security contexts used to protect the RTP/RTCP traffic between the contexts used to protect the RTP/RTCP traffic between the
participants. participants.
3.5.2. Media Translator 3.5.2. Media Translator
In the context of multipoint communications a Media Translator is not In the context of multipoint communications a Media Translator is not
providing new mechanisms to establish a multipoint session. It is providing new mechanisms to establish a multipoint session. It is
more of an enabler, or facilitator, that ensures one or some sub-set more of an enabler, or facilitator, that ensures a given End Point or
of session participants can participate in the session. a defined sub-set of End Points can participate in the session.
If B in Figure 11 were behind a limited network path, the Translator If End Point B in Figure 11 were behind a limited network path, the
may perform media transcoding to allow the traffic received from the Translator may perform media transcoding to allow the traffic
other participants to reach B without overloading the path. This received from the other End Points to reach B without overloading the
transcoding can help the other participants in the Multicast part of path. This transcoding can help the other End Points in the
the session, by not requiring the quality transmitted by A to be multicast part of the session, by not requiring the quality
lowered to the bitrates that B is actually capable of receiving. transmitted by A to be lowered to the bitrates that B is actually
capable of receiving (and vice versa).
3.6. Point to Multipoint Using the RFC 3550 Mixer Model 3.6. Point to Multipoint Using the RFC 3550 Mixer Model
Shortcut name: Topo-Mixer Shortcut name: Topo-Mixer
A Mixer is a middlebox that aggregates multiple RTP streams that are A Mixer is a middlebox that aggregates multiple RTP streams that are
part of a session by generating one or more new RTP streams and, in part of a session by generating one or more new RTP streams and, in
most cases, by manipulating the media data. One common application most cases, by manipulating the media data. One common application
for a Mixer is to allow a participant to receive a session with a for a Mixer is to allow a participant to receive a session with a
reduced amount of resources. reduced amount of resources.
skipping to change at page 20, line 30 skipping to change at page 21, line 31
| 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 media streams A Mixer can be viewed as a device terminating the RTP streams
received from other session participants. Using the media data from received from other End Points in the same RTP session. Using the
the received media streams, a Mixer generates media streams that are media data carried in the received RTP streams, a Mixer generates
sent to the session participant. derived RTP streams that are sent to the receiving End Points.
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 conference session. same Communication Session.
The Mixer is the content source, as it mixes the content (often in The Mixer creates the Media Source and the source RTP stream just
the uncompressed domain) and then encodes it for transmission to a like an End Point, as it mixes the content (often in the uncompressed
participant. The CSRC Count (CC) and CSRC fields in the RTP header domain) and then encodes and packetizes it for transmission to a
can be used to indicate the contributors to the newly generated receiving endpoint. The CSRC Count (CC) and CSRC fields in the RTP
stream. The SSRCs of the to-be-mixed streams on the Mixer input header can be used to indicate the contributors to the newly
appear as the CSRCs at the Mixer output. That output stream uses a generated RTP stream. The SSRCs of the to-be-mixed streams on the
unique SSRC that identifies the Mixer's stream. The CSRC should be Mixer input appear as the CSRCs at the Mixer output. That output
forwarded between the different conference participants to allow for stream uses a unique SSRC that identifies the Mixer's stream. The
loop detection and identification of sources that are part of the CSRC should be forwarded between the different End Points to allow
global session. Note that Section 7.1 of RFC 3550 requires the SSRC for loop detection and identification of sources that are part of the
space to be shared between domains for these reasons. This also Communication Session. Note that Section 7.1 of RFC 3550 requires
implies that any SDES information normally needs to be forwarded the SSRC space to be shared between domains for these reasons. This
also implies that any SDES information normally needs to be forwarded
across the mixer. 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 a receiver and should therefore send receiver with its role. It is an RTP receiver and should therefore send RTCP
reports for the media streams it receives. In its role as a media receiver reports for the RTP streams it receives and terminates. In
sender, it should also generate sender reports for those media its role as an RTP sender, it should also generate RTCP sender
streams it sends. As specified in Section 7.3 of RFC 3550, a Mixer reports for those RTP streams it sends. As specified in Section 7.3
must not forward RTCP unaltered between the two domains. of RFC 3550, a Mixer must not forward RTCP unaltered between the two
domains.
The Mixer depicted in Figure 13 is involved in three domains that 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 End
participants A and C), participant B, and participant D. Assuming Points A and C), End Point B, and End Point D. Assuming all four End
all four participants in the conference are interested in receiving Points in the conference are interested in receiving content from
content from each other participant, the Mixer produces different each other End Point, the Mixer produces different mixed RTP streams
mixed streams for B and D, as the one to B may contain content for B and D, as the one to B may contain content received from D, and
received from D, and vice versa. However, the Mixer may only need vice versa. However, the Mixer may only need one SSRC per media type
one SSRC per media type in each domain where it is the receiving in each domain where it is the receiving entity and transmitter of
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 End
participants in the other domain(s). In other cases, a message is Points in the other domain(s). In other cases, a message is handled
handled by the Mixer itself and therefore not forwarded to any other 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 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 media streams from several participants and selects the receives RTP streams from several End Points and selects the
stream(s) to be included in a media-domain mix. The selection can be stream(s) to be included in a media-domain mix. The selection can be
through static configuration or by dynamic, content dependent means through static configuration or by dynamic, content dependent means
such as voice activation. The mixer then creates a single outgoing such as voice activation. The mixer then creates a single outgoing
stream from this mix. RTP stream from this mix.
The most commonly deployed media mixer is probably the audio mixer, The most commonly deployed media mixer is probably the audio mixer,
used in voice conferencing, where the output consists of a mixture of used in voice conferencing, where the output consists of a mixture of
all the input streams; this needs minimal signalling to be all the input audio signals; this needs minimal signalling to be
successfully set up. Audio mixing is relatively straightforward and successfully set up. From a signal processing viewpoint, audio
commonly possible for a reasonable number of participants. Assume, mixing is relatively straightforward and commonly possible for a
for example, that one wants to mix N streams from different reasonable number of End Points. Assume, for example, that one wants
participants. The mixer needs to decode those N streams, typically to mix N streams from N different End Points. The mixer needs to
into the sample domain, and then produce N or N+1 mixes. Different decode those N streams, typically into the sample domain, and then
mixes are needed so that each contributing source gets a mix of all produce N or N+1 mixes. Different mixes are needed so that each
other sources except its own, as this would result in an echo. When contributing source gets a mix of all other sources except its own,
N is lower than the number of all participants one may produce a Mix as this would result in an echo. When N is lower than the number of
of all N streams for the group that are currently not included in the all End points, one may produce a mix of all N streams for the group
mix, thus N+1 mixes. These audio streams are then encoded again, RTP that are currently not included in the mix, thus N+1 mixes. These
packetized and sent out. In many cases, audio level normalization is audio streams are then encoded again, RTP packetized and sent out.
also required before the actual mixing process. In many cases, audio level normalization, noise suppression, and
similar signal processing steps are also required or desirable before
the actual mixing process commences.
In video, the term "mixing" has a different interpretation than 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 is known as "tiling". The combining contributed video streams, which is also known as "tiling".
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 a arranged in a set of tiles, each tile containing the video from an
participant. Tiles can be of different sizes, so that, for example, End Point (typically showing a human participant). Tiles can be of
a particularly important participant, or the loudest speaker, is different sizes, so that, for example, a particularly important
being shown on in larger tile than other participants. A self-view participant, or the loudest speaker, is being shown on in larger tile
picture can be included in the tiling, which can either be locally than other participants. A self-view picture can be included in the
produced or be a feedback from a received and reconstructed video tiling, which can either be locally produced or be a feedback from a
image. Such remote loopback allows for confidence monitoring, i.e., mixer-received and reconstructed video image. Such remote loopback
it enables the participant to see himself/herself just as other allows for confidence monitoring, i.e., it enables the participant to
participants see him/her. The tiling normally operates on see himself/herself in the same quality as other participants see
reconstructed video in the sample domain. The tiled image is him/her. The tiling normally operates on reconstructed video in the
encoded, packetized, and sent by the mixer. It is possible that a sample domain. The tiled image is encoded, packetized, and sent by
the mixer to the receiving End Points. It is possible that a
middlebox with media mixing duties contains only a single mixer of middlebox with media mixing duties contains only a single mixer of
the aforementioned type, in which case all participants necessarily the aforementioned type, in which case all participants necessarily
see the same tiled video, even if it is being sent over different RTP see the same tiled video, even if it is being sent over different RTP
streams. More common, however, are mixing arrangement where an streams. More common, however, are mixing arrangement where an
individual mixer is available for each outgoing port of the individual mixer is available for each outgoing port of the
middlebox, allowing individual compositions for each participant (a middlebox, allowing individual compositions for each receiving End
feature referred to as personalized layout). Point (a feature commonly referred to as personalized layout).
One problem with media mixing is that it consumes both large amount One problem with media mixing is that it consumes both large amounts
of media processing (for the actual mixing process in the of media processing resources (for the decoding and mixing process in
uncompressed domain) and encoding resources (for the encoding of the the uncompressed domain) and encoding resources (for the encoding of
mixed signal). Another problem is the quality degradation created by the mixed signal). Another problem is the quality degradation
decoding and re-encoding the media that is encapsulated in the RTP created by decoding and re-encoding the media, which is the result of
media stream, which is the result of the lossy nature of most the lossy nature of most commonly used media codecs. A third problem
commonly used media codecs. A third problem is the latency is the latency introduced by the media mixing, which can be
introduced by the media mixing, which can be substantial and substantial and annoyingly noticeable in case of video, or in case of
annoyingly noticeable in case of video, or in case of audio if that audio if that mixed audio is lip-sychronized with high latency video.
mixed audio is lip-sychronized with high latency video. The The advantage of media mixing is that it is straightforward for the
advantage of media mixing is that it is straightforward for the End Points to handle the single media stream (which includes the
clients to handle the single media stream (which includes the mixed mixed aggregate of many sources), as they don't need to handle
aggregate of many sources), as they don't need to handle multiple multiple decodings, local mixing and composition. In fact, mixers
decodings, local mixing and composition. In fact, mixers were were introduced in pre-RTP times so that legacy, single stream
introduced in pre-RTP times so that legacy, single stream receiving receiving endpoints (that, in some protocol environments, actually
endpoints could successfully participate in what a user would didn't need to be aware of the multipoint nature of teh conference)
recognize as a multiparty video conference. could successfully participate in what a user would recognize as a
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 24, line 38 skipping to change at page 25, line 38
| | | |<---------|MA3 <----+ | +---+ | | | | | | |<---------|MA3 <----+ | +---+ | | |
| | +-------| |(BA1+CA1)|\| +---+ | | | | | +-------| |(BA1+CA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+B | | | +---------| |---------+ +-|ENC|<-| A+B | |
+-----------+ |-----------+ +---+ +-----+ | +-----------+ |-----------+ +---+ +-----+ |
+----------------------------+ +----------------------------+
Figure 15: Session and SSRC details for Media Mixer Figure 15: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be a very simple process, as From an RTP perspective media mixing can be a very simple process, as
can be seen in Figure 15. The mixer presents one SSRC towards the can be seen in Figure 15. The mixer presents one SSRC towards the
receiving client, e.g., MA1 to Peer A, where the associated stream is receiving End Point, e.g., MA1 to Peer A, where the associated stream
the media mix of the other participants. As each peer, in this is the media mix of the other End Points. As each peer, in this
example, receives a different version of a mix from the mixer, there example, receives a different version of a mix from the mixer, there
is no actual relation between the different RTP sessions in terms of is no actual relation between the different RTP sessions in terms of
actual media or transport level information. There are, however, actual media or transport level information. There are, however,
common relationships between RTP1-RTP3, namely SSRC space and common relationships between RTP1-RTP3, namely SSRC space and
identity information. When A receives the MA1 stream which is a identity information. When A receives the MA1 stream which is a
combination of BA1 and CA1 streams, the mixer may include CSRC combination of BA1 and CA1 streams, the mixer may include CSRC
information in the MA1 stream to identify the contributing source BA1 information in the MA1 stream to identify the contributing source BA1
and CA1, allowing the receiver to identify the contributing sources and CA1, allowing the receiver to identify the contributing sources
even if this were not possible through the media itself or through even if this were not possible through the media itself or through
other signaling means. other signaling means.
The CSRC has, in turn, utility in RTP extensions, like the Mixer to The CSRC has, in turn, utility in RTP extensions, like the Mixer to
Client audio levels RTP header extension [RFC6465]. If the SSRCs Client audio levels RTP header extension [RFC6465]. If the SSRCs
from the endpoint to mixer paths are used as CSRCs in another RTP from the End Point to mixer paths are used as CSRCs in another RTP
session, then RTP1, RTP2 and RTP3 become one joint session as they session, then RTP1, RTP2 and RTP3 become one joint session as they
have a common SSRC space. At this stage, the mixer also needs to have a common SSRC space. At this stage, the mixer also needs to
consider which RTCP information it needs to expose in the different consider which RTCP information it needs to expose in the different
paths. In the above scenario, a mixer would normally expose nothing paths. In the above scenario, a mixer would normally expose nothing
more than the Source Description (SDES) information and RTCP BYE for more than the Source Description (SDES) information and RTCP BYE for
a CSRC leaving the session. The main goal would be to enable the a CSRC leaving the session. The main goal would be to enable the
correct binding against the application logic and other information correct binding against the application logic and other information
sources. This also enables loop detection in the RTP session. sources. This also enables loop detection in the RTP session.
3.6.2. Media Switching 3.6.2. Media Switching
Media switching mixers are used from 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, to more complex multi-stream
usages with receiver mixing or tiling, including combined with usages with receiver mixing or tiling, including combined with
simulcast and/or scalability between source and mixer. An RTP Mixer simulcast and/or scalability between source and mixer. An RTP Mixer
based on media switching avoids the media decoding and encoding based on media switching avoids the media decoding and encoding
operations in the mixer, as it conceptually forwards the encoded operations in the mixer, as it conceptually forwards the encoded
media stream as it was being sent to the mixer. It does not avoid, media stream as it was being sent to the mixer. It does not avoid,
however, the decryption and re-encryption cycle as it rewrites RTP however, the decryption and re-encryption cycle as it rewrites RTP
headers. Forwarding media (in contrast to reconstructing-mixing- 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 streams that the mixer can produce. These conceptual or functional RTP streams that the mixer can produce.
streams are created by selecting media from one of the RTP media 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 of their do not allow to "tune into" the stream at arbitrary points in their
bitstream. bitstream.
To achieve a coherent RTP media stream from the mixer's SSRC, the To achieve a coherent RTP stream from the mixer's SSRC, the mixer
mixer needs to rewrite the incoming RTP packet's header. First the needs to rewrite the incoming RTP packet's header. First the SSRC
SSRC field must be set to the value of the Mixer's SSRC. Second, the field must be set to the value of the Mixer's SSRC. Second, the
sequence number must be the next in the sequence of outgoing packets sequence number must be the next in the sequence of outgoing packets
it sent. Third, the RTP timestamp value needs to be adjusted using it sent. Third, the RTP timestamp value needs to be adjusted using
an offset that changes each time one switches media source. Finally, an offset that changes each time one switches media source. Finally,
depending on the negotiation of the RTP payload type, the value depending on the negotiation of the RTP payload type, the value
representing this particular RTP payload configuration may have to be representing this particular RTP payload configuration may have to be
changed if the different endpoint mixer paths have not arrived on the changed if the different End Point-to-mixer paths have not arrived on
same numbering for a given configuration. This also requires that the same numbering for a given configuration. This also requires
the different endpoints support a common set of codecs, otherwise that the different End Points support a common set of codecs,
media transcoding for codec compatibility would still be required. otherwise media transcoding for codec compatibility would still be
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 participants (A-F) where the two supports a video conference with six participating End Points (A-F)
most recent speakers in the conference are shown to each participant. where the two most recent speakers in the conference are shown to
The mixer has thus two SSRCs sending video to each peer, and each each receiving End Point. The mixer has thus two SSRCs sending video
peer is capable of locally handling two video streams simultaneously. to each peer, and each peer is capable of locally handling two video
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 | |
| +---------| |-----------+ | C | | | +---------| |-----------+ | C | |
+-----------+ | | H | | +-----------+ | | H | |
skipping to change at page 26, line 49 skipping to change at page 27, line 50
| | | |<------------|MV12 <---+-+-EV1----| | | | | | |<------------|MV12 <---+-+-EV1----| | |
| | +-------| |---------+ | | | | | | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ | | +---------| |-----------+ +-----+ |
+-----------+ +----------------------------+ +-----------+ +----------------------------+
Figure 16: Media Switching RTP Mixer Figure 16: Media Switching RTP Mixer
The Media Switching RTP mixer can, similarly to the Media Mixing The Media Switching RTP mixer can, similarly to the Media Mixing
Mixer, reduce the bit-rate required for media transmission towards Mixer, reduce the bit-rate required for media transmission towards
the different peers by selecting and forwarding only a sub-set of RTP the different peers by selecting and forwarding only a sub-set of RTP
media streams it receives from the conference participants. In cases streams it receives from the sending End Points. In cases the mixer
the mixer receives simulcast transmissions or a scalable encoding of receives simulcast transmissions or a scalable encoding of the media
the media source, the mixer has more degrees of freedom to select source, the mixer has more degrees of freedom to select streams or
streams or sub-sets of stream to forward to a receiver, both based on sub-sets of stream to forward to a receiving End Point, both based on
transport or client restrictions as well as application logic. transport or End Point restrictions as well as application logic.
To ensure that a media receiver can correctly decode the RTP media To ensure that a media receiver in an End Point can correctly decode
stream after a switch, a codec that uses temporal prediction needs to the media in the RTP stream after a switch, a codec that uses
start its decoding from independent refresh points, or similar points temporal prediction needs to start its decoding from independent
in the bitstream. For some codecs, for example frame based speech refresh points, or points in the bitstream offering similar
and audio codecs, this is easily achieved by starting the decoding at functionality (like "dirty refresh points"). For some codecs, for
RTP packet boundaries, as each packet boundary provides a refresh example frame based speech and audio codecs, this is easily achieved
point (assuming proper packetization on the encoder side). For other by starting the decoding at RTP packet boundaries, as each packet
codecs, particularly in video, refresh points are less common in the boundary provides a refresh point (assuming proper packetization on
bitstream or may not be present at all without an explicit request to the encoder side). For other codecs, particularly in video, refresh
the respective encoder. The Full Intra Request [RFC5104] RTCP codec points are less common in the bitstream or may not be present at all
control message has been defined for this purpose. without an explicit request to the respective encoder. The Full
Intra Request [RFC5104] RTCP codec control message has been defined
for this purpose.
In this type of mixer one could consider to fully terminate the RTP In this type of mixer one could consider to fully terminate the RTP
sessions between the different endpoint and mixer paths. The same sessions between the different End Point 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-End
endpoint, independent RTP session. The middlebox can select which of Point, 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 End Points. This is similar to the media
switching Mixer but has some important differences in RTP details. switching Mixer but has some important differences in RTP details.
+-A---------+ +-Middlebox-----------------+ +-A---------+ +-Middlebox-----------------+
| +-RTP1----| |-RTP1------+ +-----+ | | +-RTP1----| |-RTP1------+ +-----+ |
| | +-Video-| |-Video---+ | | | | | | +-Video-| |-Video---+ | | | |
| | | AV1|------------>|---------+-+------>| | | | | | AV1|------------>|---------+-+------>| | |
| | | |<------------|BV1 <----+-+-------| S | | | | | |<------------|BV1 <----+-+-------| S | |
| | | |<------------|CV1 <----+-+-------| W | | | | | |<------------|CV1 <----+-+-------| W | |
| | | |<------------|DV1 <----+-+-------| I | | | | | |<------------|DV1 <----+-+-------| I | |
| | | |<------------|EV1 <----+-+-------| T | | | | | |<------------|EV1 <----+-+-------| T | |
skipping to change at page 28, line 44 skipping to change at page 29, line 44
| | | FV1|------------>|---------+-+------>| | | | | | FV1|------------>|---------+-+------>| | |
| | | |<------------|AV1 <----+-+-------| | | | | | |<------------|AV1 <----+-+-------| | |
| | | | : : : |: : : : : : : : :| | | | | | | : : : |: : : : : : : : :| | |
| | | |<------------|EV1 <----+-+-------| | | | | | |<------------|EV1 <----+-+-------| | |
| | +-------| |---------+ | | | | | | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ | | +---------| |-----------+ +-----+ |
+-----------+ +---------------------------+ +-----------+ +---------------------------+
Figure 17: Selective Forwarding Middlebox Figure 17: Selective Forwarding Middlebox
In the six participant conference depicted above in (Figure 17) one In the six End Point conference depicted above in (Figure 17) one can
can see that end-point A is aware of five incoming SSRCs, BV1-FV1. see that End Point A is aware of five incoming SSRCs, BV1-FV1. If
If this middlebox intends to have a similar behavior as in this middlebox intends to have a similar behavior as in Section 3.6.2
Section 3.6.2 where the mixer provides the end-points with the two where the mixer provides the End Points with the two latest speaking
latest speaking end-points, then only two out of these five SSRCs End Points, then only two out of these five SSRCs need concurrently
need concurrently transmit media to A. As the middlebox selects the transmit media to A. As the middlebox selects the source in the
source in the different RTP sessions that transmit media to the end- different RTP sessions that transmit media to the End points, each
points, each RTP media stream requires some rewriting of RTP header RTP stream requires rewriting of certain RTP header fields when being
fields when being projected from one session into another. In projected from one session into another. In particular, the sequence
particular, the sequence number needs to be consecutively incremented number needs to be consecutively incremented based on the packet
based on the packet actually being transmitted in each RTP session. actually being transmitted in each RTP session. Therefore, the RTP
Therefore, the RTP sequence number offset will change each time a sequence number offset will change each time a source is turned on in
source is turned on in a RTP session. The timestamp (possibly a RTP session. The timestamp (possibly offset) stays the same.
offset) stays the same.
As the RTP sessions are independent, the SSRC numbers used can also As the RTP sessions are independent, the SSRC numbers used can also
be handled independently, thereby bypassing the requirement for SSRC be handled independently, thereby bypassing the requirement for SSRC
collision detection and avoidance. On the other hand, tools such as collision detection and avoidance. On the other hand, tools such as
remapping tables between the RTP sessions are required. For example, remapping tables between the RTP sessions are required. For example,
the stream that is being sent by endpoint B to the middlebox (BV1) the RTP stream that is being sent by End Point B to the middlebox
may use an SSRC value of 12345678. When that media stream is sent to (BV1) may use an SSRC value of 12345678. When that RTP stream is
endpoint F by the middlebox, it can use any SSRC value, e.g. sent to End Point 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 End Point 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 media Thus the application must not use SSRC as references to RTP streams
streams when communicating with other peers directly. This also when communicating with other peers directly. This also affects loop
affects loop detection which will fail to work, as there is no common detection which will fail to work, as there is no common namespace
namespace and identities across the different legs in the and identities across the different legs in the communication session
communication session on RTP level. Instead this responsibility on RTP level. Instead this responsibility falls onto higher layers.
falls onto higher layers.
The middlebox is also responsible to receive any RTCP codec control The middlebox is also responsible to receive any RTCP codec control
requests coming from an end-point, and decide if it can act on the requests coming from an End Point, and decide if it can act on the
request locally or needs to translate the request into the RTP request locally or needs to translate the request into the RTP
session that contains the media source. Both end-points and the session that contains the media source. Both End Points and the
middlebox need to implement conference related codec control middlebox need to implement conference related codec control
functionalities to provide a good experience. Commonly used are Full functionalities to provide a good experience. Commonly used are Full
Intra Request to request from the media source to provide switching Intra Request to request from the media source to provide switching
points between the sources, and Temporary Maximum Media Bit-rate points between the sources, and Temporary Maximum Media Bit-rate
Request (TMMBR) to enable the middlebox to aggregate congestion Request (TMMBR) to enable the middlebox to aggregate congestion
control responses towards the media source so to enable it to adjust control responses towards the media source so to enable it to adjust
its bit-rate (obviously only in case the limitation is not in the its bit-rate (obviously only in case the limitation is not in the
source to middlebox link). source to middlebox link).
The selective forwarding middlebox has been introduced in recently The selective forwarding middlebox has been introduced in recently
developed videoconferencing systems in conjunction with, and to developed videoconferencing systems in conjunction with, and to
capitalize on, scalable video coding as well as simulcasting. An capitalize on, scalable video coding as well as simulcasting. An
example of scalable video coding is Annex G of H.264, but other example of scalable video coding is Annex G of H.264, but other
codecs, including H.264 AVC and VP8 also exhibit scalability, albeit codecs, including H.264 AVC and VP8 also exhibit scalability, albeit
only in the temporal dimension. In both scalable coding and only in the temporal dimension. In both scalable coding and
simulcast cases the video signal is represented by a set of two or simulcast cases the video signal is represented by a set of two or
more bitstreams, providing a corresponding number of distinct more bitstreams, providing a corresponding number of distinct
fidelity points. The middlebox selects which parts of a scalable fidelity points. The middlebox selects which parts of a scalable
bitstream (or which bitstream, in the case of simulcasting) to bitstream (or which bitstream, in the case of simulcasting) to
forward to each of the receiving endpoints. The decision may be forward to each of the receiving End Points. The decision may be
driven by a number of factors, such as available bit rate, desired driven by a number of factors, such as available bit rate, desired
layout, etc. Contrary to transcoding MCUs, these "Selective layout, etc. Contrary to transcoding MCUs, these "Selective
Forwarding Units" (SFUs) have extremely low delay, and provide Forwarding Units" (SFUs) have extremely low delay, and provide
features that are typically associated with high-end systems features that are typically associated with high-end systems
(personalized layout, error localization) without any signal (personalized layout, error localization) without any signal
processing at the middlebox. They are also capable of scaling to a processing at the middlebox. They are also capable of scaling to a
large number of concurrent users, and--due to their very low delay-- large number of concurrent users, and--due to their very low delay--
can also be cascaded. can also be cascaded.
This version of the middlebox also puts different requirements on the This version of the middlebox also puts different requirements on the
endpoint when it comes to decoder instances and handling of the RTP End Point when it comes to decoder instances and handling of the RTP
media streams providing media. As each projected SSRC can, at any streams providing media. As each projected SSRC can, at any time,
time, provide media, the endpoint either needs to be able to handle provide media, the End Point either needs to be able to handle as
as many decoder instances as the middlebox received, or have many decoder instances as the middlebox received, or have efficient
efficient switching of decoder contexts in a more limited set of switching of decoder contexts in a more limited set of actual decoder
actual decoder instances to cope with the switches. The application instances to cope with the switches. The application also gets more
also gets more responsibility to update how the media provided is to responsibility to update how the media provided is to be presented to
be presented to the user. the user.
Note that this topology could potentially be seen as a media Note that this topology could potentially be seen as a media
translator which include an on/off logic as part of its media translator which include an on/off logic as part of its media
translation. The main difference would be a common global SSRC space translation. The main difference would be a common global SSRC space
in the case of the Media Translator and the mapped one used in the in the case of the Media Translator and the mapped one used in the
above. It also has mixer aspects, as the streams it provides are not above. It also has mixer aspects, as the streams it provides are not
basically translated version, but instead they have conceptual basically translated version, but instead they have conceptual
property assigned to them. Thus this topology appears to be some property assigned to them. Thus this topology appears to be some
hybrid between the translator and mixer model. hybrid between the translator and mixer model.
The differences between selective forwarding middlebox and a The differences between selective forwarding middlebox and a
switching mixer (Section 3.6.2) are minor, and they share most switching 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
info what is included in a particular RTP packet stream that information on what is included in a particular RTP stream that
represent a particular concept. Selective forwarding gets the source represent a particular concept. Selective forwarding gets the source
information through the SSRC, and instead have to use other mechanism information through the SSRC, and instead have to use other mechanism
to make clear the streams current purpose. to make clear the streams current purpose.
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 | +---+
skipping to change at page 31, line 35 skipping to change at page 32, line 35
presentation mode or explicit floor control) are known to exist as presentation mode or explicit floor control) are known to exist as
well. well.
The video switching MCU may also perform media translation to modify The video switching MCU may also perform media translation to modify
the content in bit-rate, encoding, or resolution. However, it still the content in bit-rate, encoding, or resolution. However, it still
may indicate the original sender of the content through the SSRC. In may indicate the original sender of the content through the SSRC. In
this case, the values of the CC and CSRC fields are retained. this case, the values of the CC and CSRC fields are retained.
If not terminating RTP, the RTCP Sender Reports are forwarded for the If not terminating RTP, the RTCP Sender Reports are forwarded for the
currently selected sender. All RTCP Receiver Reports are freely currently selected sender. All RTCP Receiver Reports are freely
forwarded between the participants. In addition, the MCU may also forwarded between the End points. 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 media stream results in most participants seeing only a all but one RTP stream results in most participants seeing only a
subset of the sent media streams at any given time, often a single 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 streams. Consequently, the media senders that are not on these RTP streams. Consequently, the End Points emitting RTP
currently forwarded receive a view of the session that indicates streams that are not currently forwarded receive a view of the
their media streams disappear somewhere en route. This makes the use session that indicates their RTP streams disappear somewhere en
of RTCP for congestion control, or any type of quality reporting, route. This makes the use of RTCP for congestion control, or any
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 media 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 |
+---+ | (MCU) | +---+ +---+ | (MCU) | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 19: Point to Multipoint Using Content Modifying MCUs Figure 19: Point to Multipoint Using Content Modifying MCUs
In this PtM scenario, each participant runs an RTP point-to-point In this PtM scenario, each End Point 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 participants, a. a selection of the content received from the other End Points, or
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 conference session. paths, which are part of the same Communication Session.
In case (a), the MCU may modify the content in terms of bit-rate, In case (a), the MCU may modify the content in terms of bit-rate,
encoding format, or resolution. No explicit RTP mechanism is used to encoding format, or resolution. No explicit RTP mechanism is used to
establish the relationship between the original media sender and the establish the relationship between the original RTP stream of the
version the MCU sends. In other words, the outgoing sessions media being sent RTP stream the MCU sends. In other words, the
typically use a different SSRC, and may well use a different payload outgoing RTP streams typically use a different SSRC, and may well use
type (PT), even if this different PT happens to be mapped to the same a different payload type (PT), even if this different PT happens to
media type. This is a result of the individually negotiated session be mapped to the same media type. This is a result of the
for each participant. individually negotiated RTP session for each End Point.
In case (b), the MCU is the content source as it mixes the content In case (b), the MCU is the Media Source and generates the Source RTP
and then encodes it for transmission to a participant. According to Stream as it mixes the received content and then encodes and
RTP [RFC3550], the SSRC of the contributors are to be signalled using packetizes it for transmission to an End Point. According to RTP
the CSRC/CC mechanism. In practice, today, most deployed MCUs do not [RFC3550], the SSRC of the contributors are to be signalled using the
implement this feature. Instead, the identification of the CSRC/CC mechanism. In practice, today, most deployed MCUs do not
participants whose content is included in the Mixer's output is not implement this feature. Instead, the identification of the End
Points whose content is included in the Mixer's output is not
indicated through any explicit RTP mechanism. That is, most deployed indicated through any explicit RTP mechanism. That is, most deployed
MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby
indicating no available CSRC information, even if they could identify indicating no available CSRC information, even if they could identify
the content sources as suggested in RTP. the original sending End Points 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 deployed MCUs (and endpoints) rely on signalling layer Note that many/most deployed MCUs (and video conferencing endpoints)
mechanisms for the identification of the contributing sources, for rely on signalling layer mechanisms for the identification of the
example, a SIP conferencing package [RFC4575]. This alleviates, to contributing sources, for example, a SIP conferencing package
some extent, the aforementioned issues resulting from ignoring RTP's [RFC4575]. This alleviates, to some extent, the aforementioned
CSRC mechanism. issues resulting from ignoring RTP's CSRC mechanism.
3.10. Split Component Endpoint 3.10. Split Component Terminal
Shortcut name: Topo-Split-Endpoint Shortcut name: Topo-Split-Terminal
The implementation of an application may desire to send a subset of In some applications, for example in some telepresence systems,
the application's data to each of multiple devices, each with its own terminals may be not integrated into a single functional unit, but
network address. A very basic use case for this would be to separate composed of more than one subunits. For example, a telepresence room
audio and video processing for a particular endpoint into different terminal employing multiple cameras and monitors may consist of
components. For example, in a video conference room system the multiple video conferencing subunits, each capable of handling a
endpoint could be considered as being composed of one device handling single camera and monitor. Another example would be a video
the audio and another handling the video, interconnected by some conferencing terminal in which audio is handled by one subunit, and
control functions allowing them to behave as a single endpoint in all video by another. Each of these subunits uses its own network
aspects except for transport as depicted in Figure 20. address. The various (media processing) subunits need (logically and
physically) to be interconnected by control functionality, but their
media plane functionality may be split. This type of terminals is
referred to as split component terminals.
Which decomposition scheme is possible is highly dependent on the RTP An example for such a split component terminal is depicted in
session usage. It is not really feasible to decompose one logical Figure 20. Within split component terminal A, at least audio and
end-point into two different transport nodes in one RTP session. A video subunits are addressed by their own network addresses. In some
third party monitor would report such an attempt as two entities of these systems, the control stack subunit may also have its own
being two different end-points with a CNAME collision. As a result, network address.
a fully RTP conformant de-composited endpoint is one where the
different decomposed parts use separate RTP sessions to send and/or From an RTP viewpoint, each of the subunits terminate RTP, and act as
receive media streams intended for them. an End Point in the sense that each subunit includes its own,
independent RTP stack. However, as the subunits are semantically
part of the same terminal, it is appropriate that this semantic
relationship is expressed in RTCP protocol elements, namely in the
CNAME.
+---------------------+ +---------------------+
| Endpoint A | | Endpoint A |
| Local Area Network | | Local Area Network |
| +------------+ | | +------------+ |
| +->| Audio |<+-RTP---\ | +->| Audio |<+-RTP---\
| | +------------+ | \ +------+ | | +------------+ | \ +------+
| | +------------+ | +-->| | | | +------------+ | +-->| |
| +->| Video |<+-RTP-------->| B | | +->| Video |<+-RTP-------->| B |
| | +------------+ | +-->| | | | +------------+ | +-->| |
| | +------------+ | / +------+ | | +------------+ | / +------+
| +->| Control |<+-SIP---/ | +->| Control |<+-SIP---/
| +------------+ | | +------------+ |
+---------------------+ +---------------------+
Figure 20: Split Component Endpoint Figure 20: Split Component Terminal
In the above usage, let us assume that the different RTP sessions are In a compliant RTP implementation, it is not feasible for more than
used for audio and video. The audio and video parts, however, use a one subunit to be part of a given RTP session. When attempting to do
common CNAME and also have a common clock to ensure that so, a third party monitor would report the two subunits as two
synchronization and clock drift handling works, despite the fact that separate End Points with a CNAME collision. As a result, a fully RTP
the components are separated. Also, RTCP handling works correctly as conformant split component terminal is one where the subunits use
long as only one part of the split endpoint is part of each RTP separate RTP sessions to send and/or receive RTP streams intended for
session. That way any differences in the path between A's audio them.
entity and B and A's video and B are related to different SSRCs in
different RTP sessions.
The requirement that can be derived from the above usage is that the In addition to the use of a common CNAME and the use of independent
transport flows for each RTP session might be under common control, RTP sessions for the RTP streams generated or consumed by the various
but still are addressed to what looks like different endpoints (based subunits, it is sensible that the subunits share a common clock, to
on addresses and ports). This connection diagram cannot be ensure that synchronization and clock drift handling works, despite
accomplished using one RTP session and thus multiple RTP sessions are the fact that the components are separated. RTCP handling works
needed. correctly as long as each subunit participates in its own RTP
session.
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 media towards the multicast domains with A and C. Instead, the RTP streams
streams from B and D are forwarded without changes. Avoiding this from B and D are forwarded without changes. Avoiding this mixing
mixing would save media processing resources that perform the mixing would save media processing resources that perform the mixing in
in cases where it isn't needed. However, there would still be a need cases where it isn't needed. However, there would still be a need to
to mix B's stream towards D. Only in the direction B -> multicast mix B's media towards D. Only in the direction B -> multicast domain
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 media stream would be missing. To avoid A and C related to A and C's RTP streams would be missing. To avoid A and C
thinking that B and D aren't receiving A and C at all, the Mixer 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 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 stream also need to be aggregated into the Mixer's Receiver
Reports sent to B and D. Since B and D only have the Mixer as source 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
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(ASM), provides the functionality that everyone may send to, or (ASM), provides the functionality that everyone may send to, or
receive from, everyone else within the session. Source-specific receive from, everyone else within the session. Source-specific
Multicast (SSM) can provide a similar functionality by having anyone Multicast (SSM) can provide a similar functionality by having anyone
intending to participate as sender to send its media to the SSM intending to participate as sender to send its media to the SSM
distribution source. The SSM distribution source forwards the media distribution source. The SSM distribution source forwards the media
to all receivers subscribed to the multicast group. Mesh, MCUs, to all receivers subscribed to the multicast group. Mesh, MCUs,
Mixers, SFMs and Translators may all provide that functionality at Mixers, SFMs and Translators may all provide that functionality at
least on some basic level. However, there are some differences in least on some basic level. However, there are some differences in
which type of reachability they provide. which type of reachability they provide.
Closest to true IP-multicast-based, all to all transmission comes Closest to true IP-multicast-based, all-to-all transmission comes
perhaps the transport Translator function called "relay" in in perhaps the transport Translator function called "relay" in in
Section 3.5, as well as the Mesh with joint RTP sessions. Media Section 3.5, as well as the Mesh with joint RTP sessions. Media
Translators, Mesh with independent RTP Sessions, Mixers, SFUs and the Translators, Mesh with independent RTP Sessions, Mixers, SFUs and the
MCU variants do not provide a fully meshed forwarding on the MCU variants do not provide a fully meshed forwarding on the
transport level; instead, they only allow limited forwarding of transport level; instead, they only allow limited forwarding of
content from the other session participants. content from the other session participants.
The "all to all media transmission" requires that any media The "all to all media transmission" requires that any media
transmitting entity considers the path to the least capable receiver. transmitting End Point considers the path to the least capable
Otherwise, the media transmissions may overload that path. receiving End Point. Otherwise, the media transmissions may overload
Therefore, a media sender needs to monitor the path from itself to that path. Therefore, a sending End Point needs to monitor the path
any of the participants, to detect the currently least capable from itself to any of the receiving End Points, to detect the
receiver, and adapt its sending rate accordingly. As multiple currently least capable receiver, and adapt its sending rate
participants may send simultaneously, the available resources may accordingly. As multiple End Points may send simultaneously, the
vary. RTCP's Receiver Reports help performing this monitoring, at available resources may vary. RTCP's Receiver Reports help
least on a medium time scale. performing this monitoring, at least on a medium time scale.
The resource consumption for performing all to all transmission The resource consumption for performing all to all transmission
Varies depending with the topology. Both ASM and SSM have the varies depending with the topology. Both ASM and SSM have the
benefit that only one copy of each packet traverses a particular benefit that only one copy of each packet traverses a particular
link. Using a relay causes the transmission of one copy of a packet link. Using a relay causes the transmission of one copy of a packet
per client-to-relay path and packet transmitted. However, in most per End Point-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 clients (which may be behind slower access links). The Mesh the End Points (which may be behind slower access links). The Mesh
causes N-1 streams of transmitted packets to traverse the first hop causes N-1 streams of transmitted packets to traverse the first hop
link from the client, in an N client mesh. How long the different link from the End Point, in an N End Point mesh. How long the
paths are common, is highly situation dependent. different paths are common, is highly situation dependent.
The transmission of RTCP by design adapts to any changes in the The transmission of RTCP by design adapts to any changes in the
number of participants due to the transmission algorithm, defined in number of participants due to the transmission algorithm, defined in
the RTP specification [RFC3550], and the extensions in AVPF [RFC4585] the RTP specification [RFC3550], and the extensions in AVPF [RFC4585]
(when applicable). That way, the resources utilized for RTCP stay (when applicable). That way, the resources utilized for RTCP stay
within the bounds configured for the session. within the bounds configured for the session.
4.1.2. Transport or Media Interoperability 4.1.2. Transport or Media Interoperability
All Translators, Mixers, and RTCP-terminating MCU, and Mesh with All Translators, Mixers, and RTCP-terminating MCU, and Mesh with
individual RTP sessions, allow changing the media encoding or the individual RTP sessions, allow changing the media encoding or the
transport to other properties of the other domain, thereby providing transport to other properties of the other domain, thereby providing
extended interoperability in cases where the participants lack a extended interoperability in cases where the End Points lack a common
common set of media codecs and/or transport protocols. Selective set of media codecs and/or transport protocols. Selective Forwarding
Forwarding Middleboxes can adopt the transport, and (at least) Middleboxes can adopt the transport, and (at least) selectively
selectively forward the encoded streams that match a receiver's forward the encoded streams that match a receiving End Point'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 receiver's encoding if the encoded streams do not match the receiving End
capabilities. Point's capabilities.
4.1.3. Per Domain Bit-Rate Adaptation 4.1.3. Per Domain Bit-Rate Adaptation
Participants are most likely to be connected to each other with a End Points are often connected to each other with a heterogeneous set
heterogeneous set of paths. This makes congestion control in a Point of paths. This makes congestion control in a Point to Multipoint set
to Multipoint set problematic. For the ASM, SSM, Mesh with common problematic. For the ASM, SSM, Mesh with common RTP session, and
RTP session, and Transport Relay scenario, each individual sender has Transport Relay scenario, each individual sending End Point has to
to adapt to the receiver with the least capable path, yielding adapt to the receiving End Point behind the least capable path,
suboptimal quality for the receivers behind the more capable paths. yielding suboptimal quality for the End Points behind the more
This is no longer necessary when Media Translators, Mixers, SFM or capable paths. This is no longer an issue when Media Translators,
MCUs are involved, as each participant only needs to adapt to the Mixers, SFM or MCUs are involved, as each End Point only needs to
slowest path within its own domain. The Translator, Mixer, SFM, or adapt to the slowest path within its own domain. The Translator,
MCU topologies all require their respective outgoing streams to Mixer, SFM, or MCU topologies all require their respective outgoing
adjust the bit-rate, packet-rate, etc., to adapt to the least capable RTP streams to adjust the bit-rate, packet-rate, etc., to adapt to
path in each of the other domains. That way one can avoid lowering the least capable path in each of the other domains. That way one
the quality to the least-capable participant in all the domains at can avoid lowering the quality to the least-capable End Point in all
the cost (complexity, delay, equipment) of the Mixer, SFM or the domains at the cost (complexity, delay, equipment) of the Mixer,
Translator, and potentially media sender (multicast/layered encoding SFM or Translator, and potentially media sender (multicast/layered
and sending the different representations). encoding and sending the different representations).
4.1.4. Aggregation of Media 4.1.4. 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 participants with transmissions share the available bit-rate. For End Points 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 media streams need to share the same resources. One because multiple RTP streams need to share the same resources. One
solution to this problem is to provide for a Mixer, or MCU to solution to this problem is to provide for a Mixer, or MCU to
aggregate the multiple streams into a single one, where the single aggregate the multiple RTP streams into a single one, where the
stream takes up less resources in terms of bit-rate. This single RTP stream takes up less resources in terms of bit-rate. This
aggregation can be performed according to different methods. Mixing aggregation can be performed according to different methods. Mixing
or selection are two common methods. Selection is almost always or selection are two common methods. Selection is almost always
possible and easy to implement. Mixing requires resources in the possible and easy to implement. Mixing requires resources in the
mixer, and may be relatively easy and not impairing the quality to mixer, and may be relatively easy and not impairing the quality too
badly (audio) or quite difficult (video tiling, which is not only badly (audio) or quite difficult (video tiling, which is not only
computationally complex but also reduces the pixel count per stream, computationally complex but also reduces the pixel count per stream,
with corresponding less in perceptual quality). with corresponding loss in perceptual quality).
4.1.5. View of All Session Participants 4.1.5. View of All Session Participants
The RTP protocol includes functionality to identify the session The RTP protocol includes functionality to identify the session
participants through the use of the SSRC and CSRC fields. In participants through the use of the SSRC and CSRC fields. In
addition, it is capable of carrying some further identity information addition, it is capable of carrying some further identity information
about these participants using the RTCP Source Descriptors (SDES). about these participants using the RTCP Source Descriptors (SDES).
In topologies that provide an full all to all functionality, i.e. In topologies that provide a full all-to-all functionality, i.e.
ASM, Mesh with common RTP session, Relay a compliant RTP ASM, Mesh with common RTP session, Relay a compliant RTP
implementation offers the functionality directly as specified in RTP. implementation offers the functionality directly as specified in RTP.
In topologies that do not offer all-to-all communication, it is In topologies that do not offer all-to-all communication, it is
necessary that RTCP is handled correctly in domain bridging function. necessary that RTCP is handled correctly in domain bridging function.
RTP includes explicit specification text for Translators and Mixers, RTP includes explicit specification text for Translators and Mixers,
and for SFMs the required functionality can be derived from that and for SFMs the required functionality can be derived from that
text. However, the MCU described in Section 3.8 cannot offer the text. However, the MCU described in Section 3.8 cannot offer the
full functionality for session participant identification through RTP full functionality for session participant identification through RTP
means. The topologies that create independent RTP sessions per means. The topologies that create independent RTP sessions per End
endpoint or pair of endpoints, like Back to Back RTP session, MESH Point or pair of End Points, like Back-to-Back RTP session, MESH with
with independent RTP sessions, and the RTCP terminating MCU RTCP independent RTP sessions, and the RTCP terminating MCU RTCP
terminating MCU (Section 3.9) do not support RTP based identification terminating MCU (Section 3.9) do not support RTP based identification
of session participants. In all those cases, other non-RTP based of session participants. In all those cases, other non-RTP based
mechanisms need to be implemented if such knowledge is required or mechanisms need to be implemented if such knowledge is required or
desirable. desirable.
4.1.6. Loop Detection 4.1.6. 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 End
participants. However, the Back to Back RTP sessions, Mesh with Points. However, the Back-to-Back RTP sessions, Mesh with
independent RTP sessions, SFM, will definitely break the loop independent RTP sessions, SFM, will definitely break the loop
detection mechanism. detection mechanism.
4.2. Comparison of Topologies 4.2. 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: Topo-
Point-to-Point (PtP), Topo-ASM (ASM), Topo-SSM (SSM), Topo-Trns- Point-to-Point (PtP), Topo-ASM (ASM), Topo-SSM (SSM), Topo-Trns-
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
skipping to change at page 39, line 38 skipping to change at page 40, line 44
Translator functionality. Translator functionality.
5. Security Considerations 5. Security Considerations
The use of Mixers, SFMs and Translators has impact on security and The use of Mixers, SFMs and Translators has impact on security and
the security functions used. The primary issue is that both Mixers, the security functions used. The primary issue is that both Mixers,
SFMs and Translators modify packets, thus preventing the use of SFMs and Translators modify packets, thus preventing the use of
integrity and source authentication, unless they are trusted devices integrity and source authentication, unless they are trusted devices
that take part in the security context, e.g., the device can send that take part in the security context, e.g., the device can send
Secure Realtime Transport Protocol (SRTP) and Secure Realtime Secure Realtime Transport Protocol (SRTP) and Secure Realtime
Transport Control Protocol (SRTCP) [RFC3711] packets to session Transport Control Protocol (SRTCP) [RFC3711] packets to End Points in
endpoints. If encryption is employed, the media Translator, SFM and the Communication Session. If encryption is employed, the media
Mixer need to be able to decrypt the media to perform its function. Translator, SFM and Mixer need to be able to decrypt the media to
A transport Translator may be used without access to the encrypted perform its function. A transport Translator may be used without
payload in cases where it translates parts that are not included in access to the encrypted payload in cases where it translates parts
the encryption and integrity protection, for example, IP address and that are not included in the encryption and integrity protection, for
UDP port numbers in a media stream using SRTP [RFC3711]. However, in example, IP address and UDP port numbers in a media stream using SRTP
general, the Translator, SFM or Mixer needs to be part of the [RFC3711]. However, in general, the Translator, SFM or Mixer needs
signalling context and get the necessary security associations (e.g., to be part of the signalling context and get the necessary security
SRTP crypto contexts) established with its RTP session participants. associations (e.g., SRTP crypto contexts) established with its RTP
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 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 works 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
skipping to change at page 41, line 16 skipping to change at page 42, line 25
domain, one for B and one for D. It may be forced to maintain a set domain, one for B and one for D. It may be forced to maintain a set
of totally independent security associations between itself and B and of totally independent security associations between itself and B and
D respectively, so as to avoid two-time pad occurrences. These D respectively, so as to avoid two-time pad occurrences. These
contexts must also be capable of handling all the sources present in contexts must also be capable of handling all the sources present in
the other domains. Hence, using completely independent security the other domains. Hence, using completely independent security
associations (for certain keying mechanisms) may force a Translator associations (for certain keying mechanisms) may force a Translator
to handle N*DM keys and related state; where N is the total number of to handle N*DM keys and related state; where N is the total number of
SSRCs used over all domains and DM is the total number of domains. SSRCs used over all domains and DM is the total number of domains.
The multicast based (ASM and SSM), Relay and Mesh with common RTP The multicast based (ASM and SSM), Relay and Mesh with common RTP
session are all topologies with multiple endpoints that requires session are all topologies with multiple End Points that require
knowledge about the different crypto contexts for the endpoints. shared knowledge about the different crypto contexts for the End
These multi-party topologies have special requirements on the key- Points. These multi-party topologies have special requirements on
management as well as the security functions. Specifically source- the key-management as well as the security functions. Specifically
authentication in these environments has special requirements. source-authentication in these environments has special requirements.
There exist a number of different mechanisms to provide keys to the There exist a number of different mechanisms to provide keys to the
different participants. One example is the choice between group keys different participants. One example is the choice between group keys
and unique keys per SSRC. The appropriate keying model is impacted and unique keys per SSRC. The appropriate keying model is impacted
by the topologies one intends to use. The final security properties by the topologies one intends to use. The final security properties
are dependent on both the topologies in use and the keying are dependent on both the topologies in use and the keying
mechanisms' properties, and need to be considered by the application. mechanisms' properties, and need to be considered by the application.
Exactly which mechanisms are used is outside of the scope of this Exactly which mechanisms are used is outside of the scope of this
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.
skipping to change at page 42, line 24 skipping to change at page 43, line 38
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006. 2006.
8.2. Informative References 8.2. Informative References
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session: "Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback", Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-02 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-03 (work
in progress), February 2014. in progress), July 2014.
[RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5, [RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5,
RFC 1112, August 1989. RFC 1112, August 1989.
[RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network [RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network
Address Translator (Traditional NAT)", RFC 3022, January Address Translator (Traditional NAT)", RFC 3022, January
2001. 2001.
[RFC3569] Bhattacharyya, S., "An Overview of Source-Specific [RFC3569] Bhattacharyya, S., "An Overview of Source-Specific
Multicast (SSM)", RFC 3569, July 2003. Multicast (SSM)", RFC 3569, July 2003.
skipping to change at page 43, line 5 skipping to change at page 44, line 17
with Feedback (AVPF)", RFC 5104, February 2008. with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control [RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760, February 2010. Sessions with Unicast Feedback", RFC 5760, February 2010.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to- Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 2011. Client Audio Level Indication", RFC 6465, December 2011.
[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, April 2014.
Authors' Addresses Authors' Addresses
Magnus Westerlund Magnus Westerlund
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