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Versions: 01 02 draft-ietf-avtcore-rtp-topologies-update

Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Obsoletes: 5117 (if approved)                                  S. Wenger
Intended status: Informational                                     Vidyo
Expires: April 18, 2013                                 October 15, 2012


                             RTP Topologies
           draft-westerlund-avtcore-rtp-topologies-update-01

Abstract

   This document discusses multi-endpoint topologies used in Real-time
   Transport Protocol (RTP)-based environments.  In particular,
   centralized topologies commonly employed in the video conferencing
   industry are mapped to the RTP terminology.

   This document is updated with additional topologies and are intended
   to replace RFC 5117.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 18, 2013.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must



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   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . . .  3
     2.1.  Glossary . . . . . . . . . . . . . . . . . . . . . . . . .  3
     2.2.  Indicating Requirement Levels  . . . . . . . . . . . . . .  4
   3.  Topologies . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Point to Point . . . . . . . . . . . . . . . . . . . . . .  4
     3.2.  Point to Multipoint Using Multicast  . . . . . . . . . . .  5
       3.2.1.  Any Source Multicast (ASM) . . . . . . . . . . . . . .  5
       3.2.2.  Source Specific Multicast (SSM)  . . . . . . . . . . .  6
       3.2.3.  SSM with Local Unicast Resources . . . . . . . . . . .  8
     3.3.  Point to Multipoint Using Mesh . . . . . . . . . . . . . .  8
     3.4.  Point to Multipoint Using the RFC 3550 Translator  . . . .  9
       3.4.1.  Relay - Transport Translator . . . . . . . . . . . . . 10
       3.4.2.  Media Translator . . . . . . . . . . . . . . . . . . . 11
     3.5.  Point to Multipoint Using the RFC 3550 Mixer Model . . . . 13
       3.5.1.  Media Mixing . . . . . . . . . . . . . . . . . . . . . 15
       3.5.2.  Media Switching  . . . . . . . . . . . . . . . . . . . 17
     3.6.  Source Projecting Middlebox  . . . . . . . . . . . . . . . 19
     3.7.  Point to Multipoint Using Video Switching MCUs . . . . . . 21
     3.8.  Point to Multipoint Using RTCP-Terminating MCU . . . . . . 23
     3.9.  De-composite Endpoint  . . . . . . . . . . . . . . . . . . 24
     3.10. Non-Symmetric Mixer/Translators  . . . . . . . . . . . . . 25
     3.11. Combining Topologies . . . . . . . . . . . . . . . . . . . 26
   4.  Comparing Topologies . . . . . . . . . . . . . . . . . . . . . 26
     4.1.  Topology Properties  . . . . . . . . . . . . . . . . . . . 27
       4.1.1.  All to All Media Transmission  . . . . . . . . . . . . 27
       4.1.2.  Transport or Media Interoperability  . . . . . . . . . 27
       4.1.3.  Per Domain Bit-Rate Adaptation . . . . . . . . . . . . 28
       4.1.4.  Aggregation of Media . . . . . . . . . . . . . . . . . 28
       4.1.5.  View of All Session Participants . . . . . . . . . . . 28
       4.1.6.  Loop Detection . . . . . . . . . . . . . . . . . . . . 28
     4.2.  Comparison of Topologies . . . . . . . . . . . . . . . . . 29
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 29
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 31
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 31
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 31
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 31
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 32
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 32





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1.  Introduction

   When working on the Codec Control Messages [RFC5104], considerable
   confusion was noticed in the community with respect to terms such as
   Multipoint Control Unit (MCU), Mixer, and Translator, and their usage
   in various topologies.  This document tries to address this confusion
   by providing a common information basis for future discussion and
   specification work.  It attempts to clarify and explain sections of
   the Real-time Transport Protocol (RTP) spec [RFC3550] in an informal
   way.  It is not intended to update or change what is normatively
   specified within RFC 3550.

   When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was
   developed the main emphasis lay in the efficient support of point to
   point and small multipoint scenarios without centralized multipoint
   control.  However, in practice, many small multipoint conferences
   operate utilizing devices known as Multipoint Control Units (MCUs).
   MCUs may implement Mixer or Translator (in RTP [RFC3550] terminology)
   functionality and signalling support.  They may also contain
   additional application functionality.  This document focuses on the
   media transport aspects of the MCU that can be realized using RTP, as
   discussed below.  Further considered are the properties of Mixers and
   Translators, and how some types of deployed MCUs deviate from these
   properties.


2.  Definitions

2.1.  Glossary

   ASM:  Any Source Multicast

   AVPF:  The Extended RTP Profile for RTCP-based Feedback

   CSRC:  Contributing Source

   Link:  The data transport to the next IP hop

   MCU:  Multipoint Control Unit

   Path:  The concatenation of multiple links, resulting in an end-to-
      end data transfer.

   PtM:  Point to Multipoint







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   PtP:  Point to Point

   SSM:  Source-Specific Multicast

   SSRC:  Synchronization Source

2.2.  Indicating Requirement Levels

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

   The RFC 2119 language is used in this document to highlight those
   important requirements and/or resulting solutions that are necessary
   to address the issues raised in this document.


3.  Topologies

   This subsection defines several topologies that are relevant for
   codec control but also RTP usage in other contexts.  The first relate
   to the RTP system model utilizing multicast and/or unicast, as
   envisioned in RFC 3550.  Two later topologies (MCU and RTCP
   terminating), in contrast, describe the deployed system models as
   used in many H.323 [H323] video conferences, where both the media
   streams and the RTP Control Protocol (RTCP) control traffic terminate
   at the MCU.  In these two cases, the media sender does not receive
   the (unmodified or Translator-modified) Receiver Reports from all
   sources (which it needs to interpret based on Synchronization Source
   (SSRC) values) and therefore has no full information about all the
   endpoint's situation as reported in RTCP Receiver Reports (RRs).
   More topologies can be constructed by combining any of the models;
   see Section 3.11.

   The topologies may be referenced in other documents by a shortcut
   name, indicated by the prefix "Topo-".

   For each of the RTP-defined topologies, we discuss how RTP, RTCP, and
   the carried media are handled.  With respect to RTCP, we also
   introduce the handling of RTCP feedback messages as defined in
   [RFC4585] and [RFC5104].  Any important differences between the two
   will be illuminated in the discussion.

3.1.  Point to Point

   Shortcut name: Topo-Point-to-Point

   The Point to Point (PtP) topology (Figure 1) consists of two



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   endpoints, communicating using unicast.  Both RTP and RTCP traffic
   are conveyed endpoint-to-endpoint, using unicast traffic only (even
   if, in exotic cases, this unicast traffic happens to be conveyed over
   an IP-multicast address).

                            +---+         +---+
                            | A |<------->| B |
                            +---+         +---+

                         Figure 1: Point to Point

   The main property of this topology is that A sends to B, and only B,
   while B sends to A, and only A. This avoids all complexities of
   handling multiple endpoints and combining the requirements from them.
   Note that an endpoint can still use multiple RTP Synchronization
   Sources (SSRCs) in an RTP session.  The number of RTP sessions in use
   between A and B can also be of any number.

   RTCP feedback messages for the indicated SSRCs are communicated
   directly between the endpoints.  Therefore, this topology poses
   minimal (if any) issues for any feedback messages.

3.2.  Point to Multipoint Using Multicast

   Multicast is a IP layer functionality that is available in some
   networks.  It comes in two main flavors, Any Source Multicast (ASM)
   where any multicast group participant can send to the group address
   and expect the packet to reach all group participants.  The other
   model is Source Specific Multicast (SSM) where only a particular IP
   host is allowed to send to the multicast group.  Both these models
   are discussed below in their respective section.

3.2.1.  Any Source Multicast (ASM)

   Shortcut name: Topo-ASM (was Topo-Multicast)

                                   +-----+
                        +---+     /       \    +---+
                        | A |----/         \---| B |
                        +---+   /   Multi-  \  +---+
                               +    Cast     +
                        +---+   \  Network  /  +---+
                        | C |----\         /---| D |
                        +---+     \       /    +---+
                                   +-----+

               Figure 2: Point to Multipoint Using Multicast




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   Point to Multipoint (PtM) is defined here as using a multicast
   topology as a transmission model, in which traffic from any
   participant reaches all the other participants, except for cases such
   as:

   o  packet loss, or

   o  when a participant does not wish to receive the traffic for a
      specific multicast group and therefore has not subscribed to the
      IP-multicast group in question.  This is for the cases where a
      multi-media session is distributed using two or more multicast
      groups.

   In the above context, "traffic" encompasses both RTP and RTCP
   traffic.  The number of participants can vary between one and many,
   as RTP and RTCP scale to very large multicast groups (the theoretical
   limit of the number of participants in a single RTP session is
   approximately two billion).  The above can be realized using Any
   Source Multicast (ASM).

   For feedback usage it is relevant to make distinction of that subset
   of multicast sessions wherein the number of participants in the
   multicast group is so low that it allows the participants to use
   early or immediate feedback, as defined in AVPF [RFC4585].  This
   document refers to those groups as "small multicast groups".  Some
   applications may still want to use larger multicast groups where the
   RTCP feedback possibilities are more limited.

   RTCP feedback messages in multicast will, like media, reach everyone
   (subject to packet losses and multicast group subscription).
   Therefore, the feedback suppression mechanism discussed in [RFC4585]
   is required.  Each individual node needs to process every feedback
   message it receives to determine if it is affected or if the feedback
   message applies only to some other participant.

3.2.2.  Source Specific Multicast (SSM)

   In Any Source Multicast, any of the participants can send to all the
   other participants, simply by sending a packet to the multicast
   group.  That is not possible in Source Specific Multicast [RFC4607]
   where only a single source (Distribution Source) can send to the
   multicast group, creating a topology that looks like the one below:









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          +--------+       +-----+
          |Media   |       |     |       Source-specific
          |Sender 1|<----->| D S |          Multicast
          +--------+       | I O |  +--+----------------> R(1)
                           | S U |  |  |                    |
          +--------+       | T R |  |  +-----------> R(2)   |
          |Media   |<----->| R C |->+  |           :   |    |
          |Sender 2|       | I E |  |  +------> R(n-1) |    |
          +--------+       | B   |  |  |          |    |    |
              :            | U   |  +--+--> R(n)  |    |    |
              :            | T +-|          |     |    |    |
              :            | I | |<---------+     |    |    |
          +--------+       | O |F|<---------------+    |    |
          |Media   |       | N |T|<--------------------+    |
          |Sender M|<----->|   | |<-------------------------+
          +--------+       +-----+       RTCP Unicast

          FT = Feedback Target
          Transport from the Feedback Target to the Distribution
          Source is via unicast or multicast RTCP if they are not
          co-located.

       Figure 3: Point to Multipoint using Source Specific Multicast

   In the SSM topology (Figure 3) a number of RTP sources (1 to M) are
   allowed to send media to the SSM group.  These send media to the
   distribution source which then forwards the media streams to the
   multicast group.  The media streams reach the Receivers (R(1) to
   R(n)).  The Receivers' RTCP cannot be sent to the multicast group.
   To support RTCP, an RTP extension for SSM [RFC5760] was defined to
   use unicast transmission to send RTCP from the receivers to one or
   more Feedback Targets (FT).

   The RTP extension for SSM deals with how feedback both general
   reception information and specific feedback events are generally
   handled.  The general problems of multicast that everyone will
   receive what the distribution source sends needs to be accounted for.

   The result of this is some common behaviours for RTP multicast:

   1.  Multicast applications often use a group of RTP sessions, not
       one.  Each endpoint will need to be a member of a number of RTP
       sessions in order to perform well.

   2.  Within each RTP session, the number of media sinks is likely to
       be much larger than the number of RTP sources.





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   3.  Multicast applications need signalling functions to identify the
       relationships between RTP sessions.

   4.  Multicast applications need signalling functions to identify the
       relationships between SSRCs in different RTP sessions.

   All multicast configurations share a signalling requirement; all of
   the participants will need to have the same RTP and payload type
   configuration.  Otherwise, A could for example be using payload type
   97 as the video codec H.264 while B thinks it is MPEG-2.

   Security solutions for this type of group communications are also
   challenging.  First of all the key-management and the security
   protocol must support group communication.  Source authentication
   becomes more difficult and requires special solutions.  For more
   discussion on this please review Options for Securing RTP Sessions
   [I-D.ietf-avtcore-rtp-security-options].

3.2.3.  SSM with Local Unicast Resources

   [RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions"
   results in additional extensions to SSM Topology.  Should be
   described.

3.3.  Point to Multipoint Using Mesh

   Shortcut name: Topo-Mesh

                              +---+      +---+
                              | A |<---->| B |
                              +---+      +---+
                                ^         ^
                                 \       /
                                  \     /
                                   v   v
                                   +---+
                                   | C |
                                   +---+

                 Figure 4: Point to Multi-Point using Mesh

   Based on the RTP session definition, it is clearly possible to have a
   joint RTP session over multiple unicast transport flows like the
   above three endpoint joint session.  In this case, A needs to send
   its' media streams and RTCP packets to both B and C over their
   respective transport flows.  As long as all participants do the same,
   everyone will have a joint view of the RTP session.




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   This doesn't create any additional requirements beyond the need to
   have multiple transport flows associated with a single RTP session.
   Note that an endpoint may use a single local port to receive all
   these transport flows, or it might have separate local reception
   ports for each of the endpoints.

   There exists an alternative structure for establishing the above
   topology which uses independent RTP sessions between each pair of
   peers, i.e. three different RTP sessions.  Unless independently
   adapted the same RTP media stream could be sent in both of the RTP
   sessions an endpoint has.  The difference exists in the behaviours
   around RTCP, for example common RTCP bandwidth for one joint session,
   rather than three independent pools, and the awareness based on RTCP
   reports between the peers of how that third leg is doing.

3.4.  Point to Multipoint Using the RFC 3550 Translator

   Shortcut name: Topo-Translator

   Two main categories of Translators can be distinguished; Transport
   Translators and Media translators.  Both Translator types share
   common attributes that separate them from Mixers.  For each media
   stream that the Translator receives, it generates an individual
   stream in the other domain.  A Translator always keeps the SSRC for a
   stream across the translation, where a Mixer can select a media
   stream, or send them out mixed, always under its own SSRC, using the
   CSRC field to indicate the source(s) of the content.

   As specified in Section 7.1 of [RFC3550], the SSRC space is common
   for all participants in the session, independent of on which side
   they are of the Translator.  Therefore, it is the responsibility of
   the participants to run SSRC collision detection, and the SSRC is a
   field the Translator should not change.

   A Translator commonly does not use an SSRC of its own, and is not
   visible as an active participant in the session.  One reason is when
   a Translator acts as a quality monitor that sends RTCP reports and
   therefore is required to have an SSRC.  Another example is the case
   when a Translator is prepared to use RTCP feedback messages.  This
   may, for example, occur when it suffers packet loss of important
   video packets and wants to trigger repair by the media sender, by
   sending feedback messages.  This can be done using the SSRC of the
   target for the translator, but this requires translation of the
   targets RTCP reports to make them consistent, so it is likely simpler
   to expose an additional SSRC in the session.

   In general, a Translator implementation should consider which RTCP
   feedback messages or codec-control messages it needs to understand in



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   relation to the functionality of the Translator itself.  This is
   completely in line with the requirement to also translate RTCP
   messages between the domains.

   The RTCP translation process can be trivial, for example, when
   Transport Translators just need to adjust IP addresses and transport
   protocol ports, or they can be quite complex as in the case of media
   Translators.  See Section 7.2 of [RFC3550].

3.4.1.  Relay - Transport Translator

   Transport Translators (Topo-Trn-Translator) do not modify the media
   stream itself, but are concerned with transport parameters.
   Transport parameters, in the sense of this section, comprise the
   transport addresses (to bridge different domains) and the media
   packetization to allow other transport protocols to be interconnected
   to a session (in gateways).  Of the 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 concerned about the mapping of the SSRC/
   CSRC (Contributing Source) concept to the non-RTP protocol.  When
   designing a Translator to a non-RTP-based media transport, one
   crucial factor lies in how to handle different sources and their
   identities.  This problem space is not discussed henceforth.

                        +-----+
             +---+     /       \     +------------+      +---+
             | A |<---/         \    |            |<---->| B |
             +---+   /   Multi-  \   |            |      +---+
                    +    Cast     +->| Translator |
             +---+   \  Network  /   |            |      +---+
             | C |<---\         /    |            |<---->| D |
             +---+     \       /     +------------+      +---+
                        +-----+

               Figure 5: Point to Multipoint Using Multicast

   Figure 5 depicts an example of a Transport Translator performing at
   least IP address translation.  It allows the (non-multicast-capable)
   participants B and D to take part in an any source multicast session
   by having the Translator forward their unicast traffic to the
   multicast addresses in use, and vice versa.  It must also forward B's
   traffic to D, and vice versa, to provide each of B and D with a
   complete view of the session.

   Also a point to point communication can end up in a situation when
   the peer it is communicating with needs basic transport translators
   functions.  This include NAT traversal by pinning the media path to a



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   public address domain relay, network topologies where the media flow
   is required to pass a particular point by employing relaying or
   preserving privacy by hiding each peers transport addresses to the
   other party.

                      +---+        +---+         +---+
                      | A |<------>| T |<------->| B |
                      +---+        +---+         +---+

                      Point to Point with Translator

   This type of very basic relay service should in most case need to
   have no RTP functionality.  Thus one can believe that they do not
   need to included in this document.  However, due to that the network
   level addressing and the RTP identifier view of the RTP session and
   who the peer is doesn't match as in the PtP unicast scenario depicted
   above this topology can raise additional requirements.


                    +---+      +------------+      +---+
                    | A |<---->|            |<---->| B |
                    +---+      |            |      +---+
                               | Translator |
                    +---+      |            |      +---+
                    | C |<---->|            |<---->| D |
                    +---+      +------------+      +---+

         Figure 6: RTP Translator (Relay) with Only Unicast Paths

   Another Translator scenario is depicted in Figure 6.  Herein, the
   Translator connects multiple users of a conference through unicast.
   This can be implemented using a very simple transport Translator,
   which in this document is called a relay.  The relay forwards all
   traffic it receives, both RTP and RTCP, to all other participants.
   In doing so, a multicast network is emulated without relying on a
   multicast-capable network infrastructure.

   For RTCP feedback this results in a similar considerations that arise
   for the ASM RTP topology.  It also puts some special signalling
   requirements where common configuration of RTP payload types for
   example are required.

3.4.2.  Media Translator

   Media Translators (Topo-Media-Translator), in contrast, modify the
   media stream itself.  This process is commonly known as transcoding.
   The modification of the media stream can be as small as removing
   parts of the stream, and it can go all the way to a full transcoding



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   (down to the sample level or equivalent) utilizing a different media
   codec.  Media Translators are commonly used to connect entities
   without a common interoperability point.

   Stand-alone Media Translators are rare.  Most commonly, a combination
   of Transport and Media Translators are used to translate both the
   media stream and the transport aspects of a stream between two
   transport domains (or clouds).

   If B in Figure 5 were behind a limited network path, the Translator
   may perform media transcoding to allow the traffic received from the
   other participants to reach B without overloading the path.

   When, in the example depicted in Figure 5, the Translator acts only
   as a Transport Translator, then the RTCP traffic can simply be
   forwarded, similar to the media traffic.  However, when media
   translation occurs, the Translator's task becomes substantially more
   complex, even with respect to the RTCP traffic.  In this case, the
   Translator needs to rewrite B's RTCP Receiver Report before
   forwarding them to D and the multicast network.  The rewriting is
   needed as the stream received by B is not the same stream as the
   other participants receive.  For example, the number of packets
   transmitted to B may be lower than what D receives, due to the
   different media format and data rate.  Therefore, if the Receiver
   Reports were forwarded without changes, the extended highest sequence
   number would indicate that B were substantially behind in reception,
   while it most likely it would not be.  Therefore, the Translator must
   translate that number to a corresponding sequence number for the
   stream the Translator received.  Similar arguments can be made for
   most other fields in the RTCP Receiver Reports.

   A media Translator may in some cases act on behalf of the "real"
   source and respond to RTCP feedback messages.  This may occur, for
   example, when a receiver requests a bandwidth reduction, and the
   media Translator has not detected any congestion or other reasons for
   bandwidth reduction between the media source and itself.  In that
   case, it is sensible that the media Translator reacts to the codec
   control messages itself, for example, by transcoding to a lower media
   rate.  If it were not reacting, the media quality in the media
   sender's domain may suffer, as a result of the media sender adjusting
   its media rate (and quality) according to the needs of the slow past-
   Translator endpoint, at the expense of the rate and quality of all
   other session participants.

   A variant of translator behaviour worth pointing out is the one
   depicted in Figure 7 of an endpoint A sends a media flow to B. On the
   path there is a device T that on A's behalf does something with the
   media streams, for example adds an RTP session with FEC information



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   for A's media streams.  T will in this case need to bind the new FEC
   streams to A's media stream, for example by using the same CNAME as
   A.

   +------+        +------+         +------+
   |      |        |      |         |      |
   |  A   |------->|  T   |-------->|  B   |
   |      |        |      |---FEC-->|      |
   +------+        +------+         +------+

               Figure 7: When De-composition is a Translator

   This type of functionality where T does something with the media
   stream on behalf of A is clearly covered under the media translator
   definition.

3.5.  Point to Multipoint Using the RFC 3550 Mixer Model

   Shortcut name: Topo-Mixer

   A Mixer is a middlebox that aggregates multiple RTP streams, which
   are part of a session, by manipulation of the media data and
   generating a new RTP stream.  One common application for a Mixer is
   to allow a participant to receive a session with a reduced amount of
   resources.

                         +-----+
              +---+     /       \     +-----------+      +---+
              | A |<---/         \    |           |<---->| B |
              +---+   /   Multi-  \   |           |      +---+
                     +    Cast     +->|   Mixer   |
              +---+   \  Network  /   |           |      +---+
              | C |<---\         /    |           |<---->| D |
              +---+     \       /     +-----------+      +---+
                         +-----+

       Figure 8: Point to Multipoint Using the RFC 3550 Mixer Model

   A Mixer can be viewed as a device terminating the media streams
   received from other session participants.  Using the media data from
   the received media streams, a Mixer generates a media stream that is
   sent to the session participant.

   The content that the Mixer provides is the mixed aggregate of what
   the Mixer receives over the PtP or PtM paths, which are part of the
   same conference session.

   The Mixer is the content source, as it mixes the content (often in



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   the uncompressed domain) and then encodes it for transmission to a
   participant.  The CSRC Count (CC) and CSRC fields in the RTP header
   are used to indicate the contributors of to the newly generated
   stream.  The SSRCs of the to-be-mixed streams on the Mixer input
   appear as the CSRCs at the Mixer output.  That output stream uses a
   unique SSRC that identifies the Mixer's stream.  The CSRC should be
   forwarded between the two domains to allow for loop detection and
   identification of sources that are part of the global session.  Note
   that Section 7.1 of RFC 3550 requires the SSRC space to be shared
   between domains for these reasons.

   The Mixer is responsible for generating RTCP packets in accordance
   with its role.  It is a receiver and should therefore send reception
   reports for the media streams it receives.  In its role as a media
   sender, it should also generate Sender Reports for those media
   streams sent.  As specified in Section 7.3 of RFC 3550, a Mixer must
   not forward RTCP unaltered between the two domains.

   The Mixer depicted in Figure 8 is involved in three domains that need
   to be separated: the any source multicast network, participant B, and
   participant D. The Mixer produces different mixed streams to B and D,
   as the one to B may contain content received from D, and vice versa.
   However, the Mixer may only need one SSRC per media type in each
   domain that is the receiving entity and transmitter of mixed content.

   In the multicast domain, a Mixer still needs to provide a mixed view
   of the other domains.  This makes the Mixer simpler to implement and
   avoids any issues with advanced RTCP handling or loop detection,
   which would be problematic if the Mixer were providing non-symmetric
   behavior.  Please see Section 3.10 for more discussion on this topic.
   However, the mixing operation in each domain could potentially be
   different.

   A Mixer is responsible for receiving RTCP feedback messages and
   handling them appropriately.  The definition of "appropriate" depends
   on the message itself and the context.  In some cases, the reception
   of a codec-control message may result in the generation and
   transmission of RTCP feedback messages by the Mixer to the
   participants in the other domain.  In other cases, a message is
   handled by the Mixer itself and therefore not forwarded to any other
   domain.

   When replacing the multicast network in Figure 8 (to the left of the
   Mixer) with individual unicast paths as depicted in Figure 9, the
   Mixer model is very similar to the one discussed in Section 3.8
   below.  Please see the discussion in Section 3.8 about the
   differences between these two models.




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                    +---+      +------------+      +---+
                    | A |<---->|            |<---->| B |
                    +---+      |            |      +---+
                               |   Mixer    |
                    +---+      |            |      +---+
                    | C |<---->|            |<---->| D |
                    +---+      +------------+      +---+

                Figure 9: RTP Mixer with Only Unicast Paths

   Lets now discuss in more detail different mixing operations that a
   mixer can perform and how that can affect the RTP and RTCP.

3.5.1.  Media Mixing

   The media mixing mixer is likely the one that most thinks of when
   they hear the term mixer.  Its basic pattern of operation is that it
   will receive the different participants RTP media stream.  Select
   which that are to be included in a media domain mix of the incoming
   RTP media streams.  Then create a single outgoing stream from this
   mix.

   The most commonly deployed media mixer is probably the audio mixer,
   used in voice conferencing, where the output consists of some mixture
   of all the input streams; this needs minimal signalling to be
   successful.  Audio mixing is straight forward and commonly possible
   to do for a number of participants.  Lets assume that you want to mix
   N number of streams from different participants.  Then the mixer need
   to perform N decodings.  Then it needs to produce N or N+1 mixes, the
   reasons that different mixes are needed are so that each contributing
   source get a mix which don't contain themselves, as this would result
   in an echo.  When N is lower than the number of all participants one
   may produce a Mix of all N streams for the group that are currently
   not included in the mix, thus N+1 mixes.  These audio streams are
   then encoded again, RTP packetised and sent out.

   Video can't really be "mixed" and produce something particular useful
   for the users, however creating an composition out of the contributed
   video streams can be done.  In fact it can be done in a number of
   ways, tiling the different streams creating a chessboard, selecting
   someone as more important and showing them large and a number of
   other sources as smaller overlays is another.  Also here one commonly
   need to produce a number of different compositions so that the
   contributing part doesn't need to see themselves.  Then the mixer re-
   encodes the created video stream, RTP packetise it and send it out

   The problem with media mixing is that it both consume large amount of
   media processing and encoding resources.  The second is the quality



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   degradation created by decoding and re-encoding the RTP media stream.
   Its advantage is that it is quite simplistic for the clients to
   handle as they don't need to handle local mixing and composition.
           +-A---------+          +-MIXER----------------------+
           | +-RTP1----|          |-RTP1------+        +-----+ |
           | | +-Audio-|          |-Audio---+ | +---+  |     | |
           | | |    AA1|--------->|---------+-+-|DEC|->|     | |
           | | |       |<---------|MA1 <----+ | +---+  |     | |
           | | |       |          |(BA1+CA1)|\| +---+  |     | |
           | | +-------|          |---------+ +-|ENC|<-| B+C | |
           | +---------|          |-----------+ +---+  |     | |
           +-----------+          |                    |     | |
                                  |                    |  M  | |
           +-B---------+          |                    |  E  | |
           | +-RTP2----|          |-RTP2------+        |  D  | |
           | | +-Audio-|          |-Audio---+ | +---+  |  I  | |
           | | |    BA1|--------->|---------+-+-|DEC|->|  A  | |
           | | |       |<---------|MA2 <----+ | +---+  |     | |
           | | +-------|          |(BA1+CA1)|\| +---+  |     | |
           | +---------|          |---------+ +-|ENC|<-| A+C | |
           +-----------+          |-----------+ +---+  |     | |
                                  |                    |  M  | |
           +-C---------+          |                    |  I  | |
           | +-RTP3----|          |-RTP3------+        |  X  | |
           | | +-Audio-|          |-Audio---+ | +---+  |  E  | |
           | | |    CA1|--------->|---------+-+-|DEC|->|  R  | |
           | | |       |<---------|MA3 <----+ | +---+  |     | |
           | | +-------|          |(BA1+CA1)|\| +---+  |     | |
           | +---------|          |---------+ +-|ENC|<-| A+B | |
           +-----------+          |-----------+ +---+  +-----+ |
                                  +----------------------------+

            Figure 10: Session and SSRC details for Media Mixer

   From an RTP perspective media mixing can be very straight forward as
   can be seen in Figure 10.  The mixer present one SSRC towards the
   peer client, e.g.  MA1 to Peer A, which is the media mix of the other
   participants.  As each peer receives a different version produced by
   the mixer there are no actual relation between the different RTP
   sessions in the actual media or the transport level information.
   There is however one connection between RTP1-RTP3 in this figure.  It
   has to do with the SSRC space and the identity information.  When A
   receives the MA1 stream which is a combination of BA1 and CA1
   streams, the mixer may include CSRC information in the MA1 stream to
   identify the contributing source BA1 and CA1.

   The CSRC has in its turn utility in RTP extensions, like the in Mixer
   to Client audio levels RTP header extension [RFC6465].  If the SSRC



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   from endpoint to mixer leg are used as CSRC in another RTP session
   then RTP1, RTP2 and RTP3 becomes one joint session as they have a
   common SSRC space.  At this stage the mixer also need to consider
   which RTCP information it need to expose in the different legs.  For
   the above situation commonly nothing more than the Source Description
   (SDES) information and RTCP BYE for CSRC need to be exposed.  The
   main goal would be to enable the correct binding against the
   application logic and other information sources.  This also enables
   loop detection in the RTP session.

3.5.2.  Media Switching

   An RTP Mixer based on media switching avoids the media decoding and
   encoding cycle in the mixer, but not the decryption and re-encryption
   cycle as it rewrites RTP headers.  This both reduces the amount of
   computational resources needed in the mixer and increases the media
   quality per transmitted bit.  This is achieve by letting the mixer
   have a number of SSRCs that represents conceptual or functional
   streams the mixer produces.  These streams are created by selecting
   media from one of the by the mixer received RTP media streams and
   forward the media using the mixers own SSRCs.  The mixer can then
   switch between available sources if that is required by the concept
   for the source, like currently active speaker.

   To achieve a coherent RTP media stream from the mixer's SSRC the
   mixer is forced to rewrite the incoming RTP packet's header.  First
   the SSRC field must be set to the value of the Mixer's SSRC.
   Secondly, the sequence number must be the next in the sequence of
   outgoing packets it sent.  Thirdly the RTP timestamp value needs to
   be adjusted using an offset that changes each time one switch media
   source.  Finally depending on the negotiation the RTP payload type
   value representing this particular RTP payload configuration may have
   to be changed if the different endpoint mixer legs have not arrived
   on the same numbering for a given configuration.  This also requires
   that the different end-points do support a common set of codecs,
   otherwise media transcoding for codec compatibility is still
   required.

   Lets consider the operation of media switching mixer that supports a
   video conference with six participants (A-F) where the two latest
   speakers in the conference are shown to each participants.  Thus the
   mixer has two SSRCs sending video to each peer.









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          +-A---------+             +-MIXER----------------------+
          | +-RTP1----|             |-RTP1------+        +-----+ |
          | | +-Video-|             |-Video---+ |        |     | |
          | | |    AV1|------------>|---------+-+------->|  S  | |
          | | |       |<------------|MV1 <----+-+-BV1----|  W  | |
          | | |       |<------------|MV2 <----+-+-EV1----|  I  | |
          | | +-------|             |---------+ |        |  T  | |
          | +---------|             |-----------+        |  C  | |
          +-----------+             |                    |  H  | |
                                    |                    |     | |
          +-B---------+             |                    |  M  | |
          | +-RTP2----|             |-RTP2------+        |  A  | |
          | | +-Video-|             |-Video---+ |        |  T  | |
          | | |    BV1|------------>|---------+-+------->|  R  | |
          | | |       |<------------|MV3 <----+-+-AV1----|  I  | |
          | | |       |<------------|MV4 <----+-+-EV1----|  X  | |
          | | +-------|             |---------+ |        |     | |
          | +---------|             |-----------+        |     | |
          +-----------+             |                    |     | |
                                    :                    :     : :
                                    :                    :     : :
          +-F---------+             |                    |     | |
          | +-RTP6----|             |-RTP6------+        |     | |
          | | +-Video-|             |-Video---+ |        |     | |
          | | |    CV1|------------>|---------+-+------->|     | |
          | | |       |<------------|MV11 <---+-+-AV1----|     | |
          | | |       |<------------|MV12 <---+-+-EV1----|     | |
          | | +-------|             |---------+ |        |     | |
          | +---------|             |-----------+        +-----+ |
          +-----------+             +----------------------------+


                   Figure 11: Media Switching RTP Mixer

   The Media Switching RTP mixer can similar to the Media Mixing one
   reduce the bit-rate needed towards the different peers by selecting
   and switching in a sub-set of RTP media streams out of the ones it
   receives from the conference participants.

   To ensure that a media receiver can correctly decode the RTP media
   stream after a switch, it becomes necessary to ensure for state
   saving codecs that they start from default state at the point of
   switching.  Thus one common tool for video is to request that the
   encoding creates an intra picture, something that isn't dependent on
   earlier state.  This can be done using Full Intra Request [RFC5104]
   RTCP codec control message.

   Also in this type of mixer one could consider to terminate the RTP



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   sessions fully between the different end-point and mixer legs.  The
   same arguments and considerations as discussed in Section 3.8 applies
   here.

3.6.  Source Projecting Middlebox

   Another method for handling media in the RTP mixer is to project all
   potential RTP sources (SSRCs) into a per end-point independent RTP
   session.  The mixer can then select which of the potential sources
   that are currently actively transmitting media, despite that the
   mixer in another RTP session receives media from that end-point.
   This is similar to the media switching Mixer but have some important
   differences in RTP details.






































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          +-A---------+             +-MIXER---------------------+
          | +-RTP1----|             |-RTP1------+       +-----+ |
          | | +-Video-|             |-Video---+ |       |     | |
          | | |    AV1|------------>|---------+-+------>|     | |
          | | |       |<------------|BV1 <----+-+-------|  S  | |
          | | |       |<------------|CV1 <----+-+-------|  W  | |
          | | |       |<------------|DV1 <----+-+-------|  I  | |
          | | |       |<------------|EV1 <----+-+-------|  T  | |
          | | |       |<------------|FV1 <----+-+-------|  C  | |
          | | +-------|             |---------+ |       |  H  | |
          | +---------|             |-----------+       |     | |
          +-----------+             |                   |  M  | |
                                    |                   |  A  | |
          +-B---------+             |                   |  T  | |
          | +-RTP2----|             |-RTP2------+       |  R  | |
          | | +-Video-|             |-Video---+ |       |  I  | |
          | | |    BV1|------------>|---------+-+------>|  X  | |
          | | |       |<------------|AV1 <----+-+-------|     | |
          | | |       |<------------|CV1 <----+-+-------|     | |
          | | |       | :    :    : |: :  : : : : :  : :|     | |
          | | |       |<------------|FV1 <----+-+-------|     | |
          | | +-------|             |---------+ |       |     | |
          | +---------|             |-----------+       |     | |
          +-----------+             |                   |     | |
                                    :                   :     : :
                                    :                   :     : :
          +-F---------+             |                   |     | |
          | +-RTP6----|             |-RTP6------+       |     | |
          | | +-Video-|             |-Video---+ |       |     | |
          | | |    CV1|------------>|---------+-+------>|     | |
          | | |       |<------------|AV1 <----+-+-------|     | |
          | | |       | :    :    : |: :  : : : : :  : :|     | |
          | | |       |<------------|EV1 <----+-+-------|     | |
          | | +-------|             |---------+ |       |     | |
          | +---------|             |-----------+       +-----+ |
          +-----------+             +---------------------------+

                     Figure 12: Media Projecting Mixer

   So in this six participant conference depicted above in (Figure 12)
   one can see that end-point A will in this case be aware of 5 incoming
   SSRCs, BV1-FV1.  If this mixer intend to have the same behaviour as
   in Section 3.5.2 where the mixer provides the end-points with the two
   latest speaking end-points, then only two out of these five SSRCs
   will concurrently transmit media to A. As the mixer selects which
   source in the different RTP sessions that transmit media to the end-
   points each RTP media stream will require some rewriting when being
   projected from one session into another.  The main thing is that the



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   sequence number will need to be consecutively incremented based on
   the packet actually being transmitted in each RTP session.  Thus the
   RTP sequence number offset will change each time a source is turned
   on in a RTP session.

   As the RTP sessions are independent the SSRC numbers used can be
   handled independently also thus working around any SSRC collisions by
   having remapping tables between the RTP sessions.  This will result
   that each endpoint may have a different view of the application usage
   of a particular SSRC.  Thus the application must not use SSRC as
   references to RTP media streams when communicating with other peers
   directly.

   The mixer will also be responsible to act on any RTCP codec control
   requests coming from an end-point and decide if it can act on it
   locally or needs to translate the request into the RTP session that
   contains the media source.  Both end-points and the mixer will need
   to implement conference related codec control functionalities to
   provide a good experience.  Full Intra Request to request from the
   media source to provide switching points between the sources,
   Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
   to aggregate congestion control response towards the media source and
   have it adjust its bit-rate in case the limitation is not in the
   source to mixer link.

   This version of the mixer also puts different requirements on the
   end-point when it comes to decoder instances and handling of the RTP
   media streams providing media.  As each projected SSRC can at any
   time provide media the end-point either needs to handle having thus
   many allocated decoder instances or have efficient switching of
   decoder contexts in a more limited set of actual decoder instances to
   cope with the switches.  The WebRTC application also gets more
   responsibility to update how the media provides is to be presented to
   the user.

   Note, this could potentially be seen as a media translator which
   include an on/off logic as part of its media 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 above.

3.7.  Point to Multipoint Using Video Switching MCUs

   Shortcut name: Topo-Video-switch-MCU








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                    +---+      +------------+      +---+
                    | A |------| Multipoint |------| B |
                    +---+      |  Control   |      +---+
                               |   Unit     |
                    +---+      |   (MCU)    |      +---+
                    | C |------|            |------| D |
                    +---+      +------------+      +---+

        Figure 13: Point to Multipoint Using a Video Switching MCU

   This PtM topology is still deployed today, although the RTCP-
   terminating MCUs, as discussed in the next section, are perhaps more
   common.  This topology, as well as the following one, reflect today's
   lack of wide availability of IP multicast technologies, as well as
   the simplicity of content switching when compared to content mixing.
   The technology is commonly implemented in what is known as "Video
   Switching MCUs".

   A video switching MCU forwards to a participant a single media
   stream, selected from the available streams.  The criteria for
   selection are often based on voice activity in the audio-visual
   conference, but other conference management mechanisms (like
   presentation mode or explicit floor control) are known to exist as
   well.

   The video switching MCU may also perform media translation to modify
   the content in bit-rate, encoding, or resolution.  However, it still
   may indicate the original sender of the content through the SSRC.  In
   this case, the values of the CC and CSRC fields are retained.

   If not terminating RTP, the RTCP Sender Reports are forwarded for the
   currently selected sender.  All RTCP Receiver Reports are freely
   forwarded between the participants.  In addition, the MCU may also
   originate RTCP control traffic in order to control the session and/or
   report on status from its viewpoint.

   The video switching MCU has most of the attributes of a Translator.
   However, its stream selection is a mixing behavior.  This behavior
   has some RTP and RTCP issues associated with it.  The suppression of
   all but one media stream results in most participants seeing only a
   subset of the sent media streams at any given time, often a single
   stream per conference.  Therefore, RTCP Receiver Reports only report
   on these streams.  Consequently, the media senders that are not
   currently forwarded receive a view of the session that indicates
   their media streams disappear somewhere en route.  This makes the use
   of RTCP for congestion control, or any type of quality reporting,
   very problematic.




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   To avoid the aforementioned issues, the MCU needs to implement two
   features.  First, it needs to act as a Mixer (see Section 3.5) and
   forward the selected media stream under its own SSRC and with the
   appropriate CSRC values.  Second, the MCU needs to modify the RTCP
   RRs it forwards between the domains.  As a result, it is RECOMMENDED
   that one implement a centralized video switching conference using a
   Mixer according to RFC 3550, instead of the shortcut implementation
   described here.

3.8.  Point to Multipoint Using RTCP-Terminating MCU

   Shortcut name: Topo-RTCP-terminating-MCU

                    +---+      +------------+      +---+
                    | A |<---->| Multipoint |<---->| B |
                    +---+      |  Control   |      +---+
                               |   Unit     |
                    +---+      |   (MCU)    |      +---+
                    | C |<---->|            |<---->| D |
                    +---+      +------------+      +---+

        Figure 14: Point to Multipoint Using Content Modifying MCUs

   In this PtM scenario, each participant runs an RTP point-to-point
   session between itself and the MCU.  This is a very commonly deployed
   topology in multipoint video conferencing.  The content that the MCU
   provides to each participant is either:

   a.  a selection of the content received from the other participants,
       or

   b.  the mixed aggregate of what the MCU receives from the other PtP
       paths, which are part of the same conference session.

   In case a), the MCU may modify the content in bit-rate, encoding, or
   resolution.  No explicit RTP mechanism is used to establish the
   relationship between the original media sender and the version the
   MCU sends.  In other words, the outgoing sessions typically use a
   different SSRC, and may well use a different payload type (PT), even
   if this different PT happens to be mapped to the same media type.
   This is a result of the individually negotiated session for each
   participant.

   In case b), the MCU is the content source as it mixes the content and
   then encodes it for transmission to a participant.  According to RTP
   [RFC3550], the SSRC of the contributors are to be signalled using the
   CSRC/CC mechanism.  In practice, today, most deployed MCUs do not
   implement this feature.  Instead, the identification of the



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   participants whose content is included in the Mixer's output is not
   indicated through any explicit RTP mechanism.  That is, most deployed
   MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby
   indicating no available CSRC information, even if they could identify
   the content sources as suggested in RTP.

   The main feature that sets this topology apart from what RFC 3550
   describes is the breaking of the common RTP session across the
   centralized device, such as the MCU.  This results in the loss of
   explicit RTP-level indication of all participants.  If one were using
   the mechanisms available in RTP and RTCP to signal this explicitly,
   the topology would follow the approach of an RTP Mixer.  The lack of
   explicit indication has at least the following potential problems:

   1.  Loop detection cannot be performed on the RTP level.  When
       carelessly connecting two misconfigured MCUs, a loop could be
       generated.

   2.  There is no information about active media senders available in
       the RTP packet.  As this information is missing, receivers cannot
       use it.  It also deprives the client of information related to
       currently active senders in a machine-usable way, thus preventing
       clients from indicating currently active speakers in user
       interfaces, etc.

   Note that deployed MCUs (and endpoints) rely on signalling layer
   mechanisms for the identification of the contributing sources, for
   example, a SIP conferencing package [RFC4575].  This alleviates, to
   some extent, the aforementioned issues resulting from ignoring RTP's
   CSRC mechanism.

   As a result of the shortcomings of this topology, it is RECOMMENDED
   to instead implement the Mixer concept as specified by RFC 3550.

3.9.  De-composite Endpoint

   The implementation of an application may desire to send a subset of
   the application's data to each of multiple devices, each with their
   own network address.  A very basic use case for this would be to
   separate audio and video processing for a particular endpoint, like a
   conference room, into one device handling the audio and another
   handling the video, being interconnected by some control functions
   allowing them to behave as a single endpoint in all aspects except
   for transport Figure 15.

   Which decomposition that is possible is highly dependent on the RTP
   session usage.  It is not really feasible to decomposed one logical
   end-point into two different transport node in one RTP session.  From



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   a third party monitor of such an attempt the two entities would look
   like two different end-points with a CNAME collision.  This put a
   requirement on that the only type of de-composited endpoint that RTP
   really supports is one where the different parts have separate RTP
   sessions to send and/or receive media streams intended for them.

                +---------------------+
                | Endpoint A          |
                | Local Area Network  |
                |      +------------+ |
                |   +->| Audio      |<+-RTP---\
                |   |  +------------+ |        \    +------+
                |   |  +------------+ |         +-->|      |
                |   +->| Video      |<+-RTP-------->|  B   |
                |   |  +------------+ |         +-->|      |
                |   |  +------------+ |        /    +------+
                |   +->| Control    |<+-SIP---/
                |      +------------+ |
                +---------------------+

                     Figure 15: De-composite End-Point

   In the above usage, let us assume that the RTP sessions are different
   for audio and video.  The audio and video parts will use a common
   CNAME and also have a common clock to ensure that synchronisation and
   clock drift handling works despite the decomposition.  Also the RTCP
   handling works correctly as long as only one part of the de-composite
   is part of each RTP session.  That way any differences in the path
   between A's audio entity and B and A's video and B are related to
   different SSRCs in different RTP sessions.

   The requirements that can derived from the above usage is that the
   transport flows for each RTP session might be under common control
   but still go to what looks like different endpoints based on
   addresses and ports.  This geometry cannot be accomplished using one
   RTP session, so in this case, multiple RTP sessions are needed.

3.10.  Non-Symmetric Mixer/Translators

   Shortcut name: Topo-Asymmetric

   It is theoretically possible to construct an MCU that is a Mixer in
   one direction and a Translator in another.  The main reason to
   consider this would be to allow topologies similar to Figure 8, 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 streams from B
   and D are forwarded without changes.  Avoiding this mixing would save
   media processing resources that perform the mixing in cases where it



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   isn't needed.  However, there would still be a need to mix B's stream
   towards D. Only in the direction B -> multicast domain or D ->
   multicast domain would it be possible to work as a Translator.  In
   all other directions, it would function as a Mixer.

   The Mixer/Translator would still need to process and change the RTCP
   before forwarding it in the directions of B or D to the multicast
   domain.  One issue is that A and C do not know about the mixed-media
   stream the Mixer sends to either B or D. Thus, any 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 thinking
   that B and D aren't receiving A and C at all, the Mixer needs to
   insert its Receiver Reports 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 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 for the stream, all RTCP from A and C must be
   suppressed by the Mixer.

   This topology is so problematic and it is so easy to get the RTCP
   processing wrong, that it is NOT RECOMMENDED to implement this
   topology.

3.11.  Combining Topologies

   Topologies can be combined and linked to each other using Mixers or
   Translators.  However, care must be taken in handling the SSRC/CSRC
   space.  A Mixer will not forward RTCP from sources in other domains,
   but will instead generate its own RTCP packets for each domain it
   mixes into, including the necessary Source Description (SDES)
   information for both the CSRCs and the SSRCs.  Thus, in a mixed
   domain, the only SSRCs seen will be the ones present in the domain,
   while there can be CSRCs from all the domains connected together with
   a combination of Mixers and Translators.  The combined SSRC and CSRC
   space is common over any Translator or Mixer.  This is important to
   facilitate loop detection, something that is likely to be even more
   important in combined topologies due to the mixed behavior between
   the domains.  Any hybrid, like the Topo-Video-switch-MCU or Topo-
   Asymmetric, requires considerable thought on how RTCP is dealt with.


4.  Comparing Topologies

   The topologies discussed in Section 3 have different properties.
   This section first lists these properties and then maps the different
   topologies to them.  Please note that even if a certain property is
   supported within a particular topology concept, the necessary
   functionality may, in many cases, be optional to implement.



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   Note: This section has not yet been updated with the new additions of
   topologies.

4.1.  Topology Properties

4.1.1.  All to All Media Transmission

   Multicast, at least Any Source Multicast (ASM), provides the
   functionality that everyone may send to, or receive from, everyone
   else within the session.  MCUs, Mixers, and Translators may all
   provide that functionality at least on some basic level.  However,
   there are some differences in which type of reachability they
   provide.

   The transport Translator function called "relay", in Section 3.4, is
   the one that provides the emulation of ASM that is closest to true
   IP-multicast-based, all to all transmission.  Media Translators,
   Mixers, and the MCU variants do not provide a fully meshed forwarding
   on the transport level; instead, they only allow limited forwarding
   of content from the other session participants.

   The "all to all media transmission" requires that any media
   transmitting entity considers the path to the least capable receiver.
   Otherwise, the media transmissions may overload that path.
   Therefore, a media sender needs to monitor the path from itself to
   any of the participants, to detect the currently least capable
   receiver, and adapt its sending rate accordingly.  As multiple
   participants may send simultaneously, the available resources may
   vary.  RTCP's Receiver Reports help performing this monitoring, at
   least on a medium time scale.

   The transmission of RTCP automatically adapts to any changes in the
   number of participants due to the transmission algorithm, defined in
   the RTP specification [RFC3550], and the extensions in AVPF [RFC4585]
   (when applicable).  That way, the resources utilized for RTCP stay
   within the bounds configured for the session.

4.1.2.  Transport or Media Interoperability

   Translators, Mixers, and RTCP-terminating MCU all allow changing the
   media encoding or the transport to other properties of the other
   domain, thereby providing extended interoperability in cases where
   the participants lack a common set of media codecs and/or transport
   protocols.







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4.1.3.  Per Domain Bit-Rate Adaptation

   Participants are most likely to be connected to each other with a
   heterogeneous set of paths.  This makes congestion control in a Point
   to Multipoint set problematic.  For the ASM and "relay" scenario,
   each individual sender has to adapt to the receiver with the least
   capable path.  This is no longer necessary when Media Translators,
   Mixers, or MCUs are involved, as each participant only needs to adapt
   to the slowest path within its own domain.  The Translator, Mixer, or
   MCU topologies all require their respective outgoing streams to
   adjust the bit-rate, packet-rate, etc., to adapt to the least capable
   path in each of the other domains.  That way one can avoid lowering
   the quality to the least-capable participant in all the domains at
   the cost (complexity, delay, equipment) of the Mixer or Translator.

4.1.4.  Aggregation of Media

   In the all to all media property mentioned above and provided by ASM,
   all simultaneous media transmissions share the available bit-rate.
   For participants with limited reception capabilities, this may result
   in a situation where even a minimal acceptable media quality cannot
   be accomplished.  This is the result of multiple media streams
   needing to share the available resources.  The solution to this
   problem is to provide for a Mixer or MCU to aggregate the multiple
   streams into a single one.  This aggregation can be performed
   according to different methods.  Mixing or selection are two common
   methods.

4.1.5.  View of All Session Participants

   The RTP protocol includes functionality to identify the session
   participants through the use of the SSRC and CSRC fields.  In
   addition, it is capable of carrying some further identity information
   about these participants using the RTCP Source Descriptors (SDES).
   To maintain this functionality, it is necessary that RTCP is handled
   correctly in domain bridging function.  This is specified for
   Translators and Mixers.  The MCU described in Section 3.7 does not
   entirely fulfill this.  The one described in Section 3.8 does not
   support this at all.

4.1.6.  Loop Detection

   In complex topologies with multiple interconnected domains, it is
   possible to form media loops.  RTP and RTCP support detecting such
   loops, as long as the SSRC and CSRC identities are correctly set in
   forwarded packets.  It is likely that loop detection works for the
   MCU, described in Section 3.7, at least as long as it forwards the
   RTCP between the participants.  However, the MCU in Section 3.8 will



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   definitely break the loop detection mechanism.

4.2.  Comparison of Topologies

   The table below attempts to summarize the properties of the different
   topologies.  The legend to the topology abbreviations are: Topo-
   Point-to-Point (PtP), Topo-Multicast (Multic), Topo-Trns-Translator
   (TTrn), Topo-Media-Translator (including Transport Translator)
   (MTrn), Topo-Mixer (Mixer), Topo-Asymmetric (ASY), Topo-Video-switch-
   MCU (MCUs), and Topo-RTCP-terminating-MCU (MCUt).  In the table
   below, Y indicates Yes or full support, N indicates No support, (Y)
   indicates partial support, and N/A indicates not applicable.

   Property               PtP  Multic TTrn MTrn Mixer ASY MCUs MCUt
   ------------------------------------------------------------------
   All to All media        N    Y      Y    Y   (Y)   (Y) (Y)  (Y)
   Interoperability        N/A  N      Y    Y    Y     Y   N    Y
   Per Domain Adaptation   N/A  N      N    Y    Y     Y   N    Y
   Aggregation of media    N    N      N    N    Y    (Y)  Y    Y
   Full Session View       Y    Y      Y    Y    Y     Y  (Y)   N
   Loop Detection          Y    Y      Y    Y    Y     Y  (Y)   N

   Please note that the Media Translator also includes the transport
   Translator functionality.


5.  Security Considerations

   The use of Mixers and Translators has impact on security and the
   security functions used.  The primary issue is that both Mixers and
   Translators modify packets, thus preventing the use of integrity and
   source authentication, unless they are trusted devices that take part
   in the security context, e.g., the device can send Secure Realtime
   Transport Protocol (SRTP) and Secure Realtime Transport Control
   Protocol (SRTCP) [RFC3711] packets to session endpoints.  If
   encryption is employed, the media Translator and Mixer need to be
   able to decrypt the media to perform its function.  A transport
   Translator may be used without access to the encrypted payload in
   cases where it translates parts that are not included in the
   encryption and integrity protection, for example, IP address and UDP
   port numbers in a media stream using SRTP [RFC3711].  However, in
   general, the Translator or Mixer needs to be part of the signalling
   context and get the necessary security associations (e.g., SRTP
   crypto contexts) established with its RTP session participants.

   Including the Mixer and Translator in the security context allows the
   entity, if subverted or misbehaving, to perform a number of very
   serious attacks as it has full access.  It can perform all the



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   attacks possible (see RFC 3550 and any applicable profiles) as if the
   media session were not protected at all, while giving the impression
   to the session participants that they are protected.

   Transport Translators have no interactions with cryptography that
   works above the transport layer, such as SRTP, since that sort of
   Translator leaves the RTP header and payload unaltered.  Media
   Translators, on the other hand, have strong interactions with
   cryptography, since they alter the RTP payload.  A media Translator
   in a session that uses cryptographic protection needs to perform
   cryptographic processing to both inbound and outbound packets.

   A media Translator may need to use different cryptographic keys for
   the inbound and outbound processing.  For SRTP, different keys are
   required, because an RFC 3550 media Translator leaves the SSRC
   unchanged during its packet processing, and SRTP key sharing is only
   allowed when distinct SSRCs can be used to protect distinct packet
   streams.

   When the media Translator uses different keys to process inbound and
   outbound packets, each session participant needs to be provided with
   the appropriate key, depending on whether they are listening to the
   Translator or the original source.  (Note that there is an
   architectural difference between RTP media translation, in which
   participants can rely on the RTP Payload Type field of a packet to
   determine appropriate processing, and cryptographically protected
   media translation, in which participants must use information that is
   not carried in the packet.)

   When using security mechanisms with Translators and Mixers, it is
   possible that the Translator or Mixer could create different security
   associations for the different domains they are working in.  Doing so
   has some implications:

   First, it might weaken security if the Mixer/Translator accepts a
   weaker algorithm or key in one domain than in another.  Therefore,
   care should be taken that appropriately strong security parameters
   are negotiated in all domains.  In many cases, "appropriate"
   translates to "similar" strength.  If a key management system does
   allow the negotiation of security parameters resulting in a different
   strength of the security, then this system SHOULD notify the
   participants in the other domains about this.

   Second, the number of crypto contexts (keys and security related
   state) needed (for example, in SRTP [RFC3711]) may vary between
   Mixers and Translators.  A Mixer normally needs to represent only a
   single SSRC per domain and therefore needs to create only one
   security association (SRTP crypto context) per domain.  In contrast,



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   a Translator needs one security association per participant it
   translates towards, in the opposite domain.  Considering Figure 5,
   the Translator needs two security associations towards the multicast
   domain, one for B and one for D. It may be forced to maintain a set
   of totally independent security associations between itself and B and
   D respectively, so as to avoid two-time pad occurrences.  These
   contexts must also be capable of handling all the sources present in
   the other domains.  Hence, using completely independent security
   associations (for certain keying mechanisms) may force a Translator
   to handle N*DM keys and related state; where N is the total number of
   SSRCs used over all domains and DM is the total number of domains.

   There exist a number of different mechanisms to provide keys to the
   different participants.  One example is the choice between group keys
   and unique keys per SSRC.  The appropriate keying model is impacted
   by the topologies one intends to use.  The final security properties
   are dependent on both the topologies in use and the keying
   mechanisms' properties, and need to be considered by the application.
   Exactly which mechanisms are used is outside of the scope of this
   document.  Please review RTP Security Options
   [I-D.ietf-avtcore-rtp-security-options] to get a better understanding
   of most of the available options.


6.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.


7.  Acknowledgements

   The authors would like to thank Bo Burman, Umesh Chandra, Roni Even,
   Keith Lantz, Ladan Gharai, Geoff Hunt, and Mark Baugher for their
   help in reviewing this document.


8.  References

8.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time



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              Applications", STD 64, RFC 3550, July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC4575]  Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
              Initiation Protocol (SIP) Event Package for Conference
              State", RFC 4575, August 2006.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              July 2006.

8.2.  Informative References

   [H323]     ITU-T Recommendation H.323, "Packet-based multimedia
              communications systems", June 2006.

   [I-D.ietf-avtcore-rtp-security-options]
              Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", draft-ietf-avtcore-rtp-security-options-00
              (work in progress), July 2012.

   [RFC4607]  Holbrook, H. and B. Cain, "Source-Specific Multicast for
              IP", RFC 4607, August 2006.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, February 2008.

   [RFC5760]  Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
              Protocol (RTCP) Extensions for Single-Source Multicast
              Sessions with Unicast Feedback", RFC 5760, February 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
              Transport Protocol (RTP) Header Extension for Mixer-to-
              Client Audio Level Indication", RFC 6465, December 2011.








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Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com


   Stephan Wenger
   Vidyo
   433 Hackensack Ave
   Hackensack, NJ  07601
   USA

   Email: stewe@stewe.org
































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