[Docs] [txt|pdf|xml|html] [Tracker] [Email] [Diff1] [Diff2] [Nits]

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: August 29, 2013                               February 25, 2013


                             RTP Topologies
           draft-westerlund-avtcore-rtp-topologies-update-02

Abstract

   This document discusses point to point and 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 August 29, 2013.

Copyright Notice

   Copyright (c) 2013 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



Westerlund & Wenger      Expires August 29, 2013                [Page 1]


Internet-Draft               RTP Topologies                February 2013


   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
   3.  Topologies . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Point to Point . . . . . . . . . . . . . . . . . . . . . .  4
     3.2.  Point to Point via Middlebox . . . . . . . . . . . . . . .  5
       3.2.1.  Translators  . . . . . . . . . . . . . . . . . . . . .  5
       3.2.2.  Back to Back RTP sessions  . . . . . . . . . . . . . .  8
     3.3.  Point to Multipoint Using Multicast  . . . . . . . . . . .  9
       3.3.1.  Any Source Multicast (ASM) . . . . . . . . . . . . . .  9
       3.3.2.  Source Specific Multicast (SSM)  . . . . . . . . . . . 11
       3.3.3.  SSM with Local Unicast Resources . . . . . . . . . . . 12
     3.4.  Point to Multipoint Using Mesh . . . . . . . . . . . . . . 14
     3.5.  Point to Multipoint Using the RFC 3550 Translator  . . . . 15
       3.5.1.  Relay - Transport Translator . . . . . . . . . . . . . 15
       3.5.2.  Media Translator . . . . . . . . . . . . . . . . . . . 16
     3.6.  Point to Multipoint Using the RFC 3550 Mixer Model . . . . 16
       3.6.1.  Media Mixing . . . . . . . . . . . . . . . . . . . . . 18
       3.6.2.  Media Switching  . . . . . . . . . . . . . . . . . . . 21
     3.7.  Source Projecting Middlebox  . . . . . . . . . . . . . . . 23
     3.8.  Point to Multipoint Using Video Switching MCUs . . . . . . 26
     3.9.  Point to Multipoint Using RTCP-Terminating MCU . . . . . . 27
     3.10. De-composite Endpoint  . . . . . . . . . . . . . . . . . . 29
     3.11. Non-Symmetric Mixer/Translators  . . . . . . . . . . . . . 30
     3.12. Combining Topologies . . . . . . . . . . . . . . . . . . . 30
   4.  Comparing Topologies . . . . . . . . . . . . . . . . . . . . . 31
     4.1.  Topology Properties  . . . . . . . . . . . . . . . . . . . 31
       4.1.1.  All to All Media Transmission  . . . . . . . . . . . . 31
       4.1.2.  Transport or Media Interoperability  . . . . . . . . . 32
       4.1.3.  Per Domain Bit-Rate Adaptation . . . . . . . . . . . . 32
       4.1.4.  Aggregation of Media . . . . . . . . . . . . . . . . . 32
       4.1.5.  View of All Session Participants . . . . . . . . . . . 32
       4.1.6.  Loop Detection . . . . . . . . . . . . . . . . . . . . 33
     4.2.  Comparison of Topologies . . . . . . . . . . . . . . . . . 33
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 33
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 35
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 36
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 36
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 36
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 36
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 37



Westerlund & Wenger      Expires August 29, 2013                [Page 2]


Internet-Draft               RTP Topologies                February 2013


1.  Introduction

   Real-time Transport Protocol (RTP) [RFC3550] topologies describe
   methods for interconnecting RTP entities and their processing
   behavior of RTP and RTCP.  This document tries to address past and
   existing confusion, especially with respect to terms not defined in
   RTP but in common use in the conversational communication industry,
   such as MCU.  In doing so, this memo provides 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






Westerlund & Wenger      Expires August 29, 2013                [Page 3]


Internet-Draft               RTP Topologies                February 2013


   PtP:  Point to Point

   SSM:  Source-Specific Multicast

   SSRC:  Synchronization Source


3.  Topologies

   This subsection defines several topologies that are relevant for
   codec control but also RTP usage in other contexts.  The section
   starts with point to point cases, without and with middleboxes.  Then
   follows a number of different methods for establishing point to
   multipoint communication.  These are structure around the most
   fundamental enabler, i.e. multicast, a mesh of connections,
   translators, mixers and source projection middlebox, to finally
   discuss MCUs.  The section ends by discussing de-composed endpoints,
   asymmetric middlebox behaviors and combining topologies.

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

   For each of the RTP-defined topologies, we discuss how RTP, RTCP, and
   the carried media are handled.  With respect to RTCP, we also discuss
   the handling of RTCP feedback messages as defined in [RFC4585] and
   [RFC5104].  Any important differences between the two will be
   illuminated in the discussion.  At this stage we don't intended to
   discuss in detail how each and every feedback messages should be
   treated in the various topologies.

3.1.  Point to Point

   Shortcut name: Topo-Point-to-Point

   The Point to Point (PtP) topology (Figure 1) consists of two
   endpoints, communicating using unicast.  Both RTP and RTCP traffic
   are conveyed endpoint-to-endpoint, using unicast traffic only (even
   if, in exotic cases, this unicast traffic happens to be conveyed over
   an IP-multicast address).

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

                         Figure 1: Point to Point

   The main property of this topology is that A sends to B, and only B,
   while B sends to A, and only A. This avoids all complexities of



Westerlund & Wenger      Expires August 29, 2013                [Page 4]


Internet-Draft               RTP Topologies                February 2013


   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.  For RTP sessions
   which use multiple SSRC per endpoint it can be relevant to implement
   support for cross reporting suppression as defined in "Real-Time
   Transport Protocol (RTP) Considerations for Endpoints Sending
   Multiple Media Streams" [I-D.lennox-avtcore-rtp-multi-stream].

3.2.  Point to Point via Middlebox

   This section discusses cases where two endpoints communicate but have
   one or more middlebox involved in the RTP session.

3.2.1.  Translators

   Shortcut name: Topo-PtP-Translator

   Two main categories of Translators can be distinguished; Transport
   Translators and Media translators.  Both Translator types share
   common attributes that separate them from Mixers.  For each media
   stream that the Translator receives, it generates an individual
   stream in the other domain.  A translator keeps the SSRC for a stream
   across the translation, whereas a Mixer can select a single media
   stream, or send out multiple mixed media streams, but always under
   its own SSRC, possibly using the CSRC field to indicate the source(s)
   of the content.  Mixers are more common in point to multipoint cases
   than in PtP.  The reason is that in PtP use cases the primary focus
   is interoperability, such as transcoding to a codec the receiver
   supports, which can be done by a media translator.

   As specified in Section 7.1 of [RFC3550], the SSRC space is common
   for all participants in the RTP session, independent of on which side
   of the Translator the session resides.  Therefore, it is the
   responsibility of the participants to run SSRC collision detection,
   and the SSRC is thus a field the Translator cannot change.  Any SDES
   information associated with a SSRC or CSRC also needs to be forwarded
   between the domains for any SSRC/CSRC used in the different domains.

   A Translator commonly does not use an SSRC of its own, and is not
   visible as an active participant in the session.  One reason to have
   its own SSRC is when a Translator acts as a quality monitor that
   sends RTCP reports and therefore is required to have an SSRC.
   Another example is the case when a Translator is prepared to use RTCP



Westerlund & Wenger      Expires August 29, 2013                [Page 5]


Internet-Draft               RTP Topologies                February 2013


   feedback messages.  This may, for example, occur in a translator
   configured to detect packet loss of important video packets and wants
   to trigger repair by the media sender, by sending feedback messages.
   While such feedback could use the SSRC of the target for the
   translator, but this in turn would require translation of the targets
   RTCP reports to make them consistent.  It may be simpler to expose an
   additional SSRC in the session, the only concern are endpoints
   failing to support the full RTP specification, thus having issues
   with multiple SSRCs reporting on the RTP streams sent by that
   endpoint.

   In general, a Translator implementation should consider which RTCP
   feedback messages or codec-control messages it needs to understand in
   relation to the functionality of the Translator itself.  This is
   completely in line with the requirement to also translate RTCP
   messages between the domains.

3.2.1.1.  Transport Relay/Anchoring

   There exist a number of different types of middleboxes that might be
   inserted between two RTP endpoints on the transport level, e.g.
   perform changes on the IP/UDP headers, and are, therefore, basic
   transport translators.  These middleboxes come in many variations
   including NAT [RFC3022] traversal by pinning the media path to a
   public address domain relay, network topologies where the media flow
   is required to pass a particular point for audit by employing
   relaying, or preserving privacy by hiding each peers transport
   addresses to the other party.  Other protocols or functionalities
   that provide this behavior are TURN [RFC5766] servers, Session Border
   Gateways and Media Processing Nodes with media anchoring
   functionalities.

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

                 Figure 2: Point to Point with Translator

   What is common for these functions is that they are normally
   transparent on RTP level, i.e. they perform no changes on any RTP or
   RTCP packet fields, only on the lower layers.  However, they may
   effect the path the RTP and RTCP packets are routed between the
   endpoints in the RTP session, and thereby only indirectly affect the
   RTP session.  For this reason, one could believe that transport
   translator type middleboxes do not need to included in this document.
   However, this topology can raise additional requirements the RTP
   implementation and its interactions with the signalling solution.
   Both in signalling and in certain RTCP field other network addresses



Westerlund & Wenger      Expires August 29, 2013                [Page 6]


Internet-Draft               RTP Topologies                February 2013


   than those of the relay can occur, due to that B has different
   network address than the relay (T).  However, implementation not
   capable of this will neither not work when endpoints are subject to
   NAT.

3.2.1.2.  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 such unicast to
   multicast) 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.

   The most basic transport translators that operate below RTP level was
   already discussed in Section 3.2.1.1.

3.2.1.3.  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
   (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 in the media encoding.

   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).

   When media translation occurs, the Translator's task regarding
   handling of RTCP traffic becomes substantially more complex.  In this
   case, the Translator needs to rewrite B's RTCP Receiver Report before
   forwarding them to A. 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 A sends, due to the different media format and data rate.
   Therefore, if the Receiver Reports were forwarded without changes,



Westerlund & Wenger      Expires August 29, 2013                [Page 7]


Internet-Draft               RTP Topologies                February 2013


   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.

   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 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
   for A's media streams.  In this case, T needs 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 3: When De-composition is a Translator

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

3.2.2.  Back to Back RTP sessions

   There exist middleboxes that interconnect two endpoints through
   themselves not by being part of a common RTP session.  Instead they
   establish two different RTP sessions, one between A and the middlebox
   (MB) and another between the MB and B.

     |<--Session A-->|  |<--Session B-->|
   +------+        +------+         +------+
   |  A   |------->|  MB  |-------->|  B   |
   +------+        +------+         +------+




Westerlund & Wenger      Expires August 29, 2013                [Page 8]


Internet-Draft               RTP Topologies                February 2013


               Figure 4: When De-composition is a Translator

   The MB acts as a application level gateway and bridges the two RTP
   session.  This bridging can be as basic as forwarding the RTP
   payloads between the sessions, or more complex including media
   transcoding.  The difference with the single RTP session context is
   the handling of the SSRCs and the other session related identifiers,
   such as CNAMEs.  With two different RTP sessions these can be freely
   changed and it becomes the MB task to maintain the right relations.

   The signalling or other above-RTP level functionalities referencing
   RTP media streams may be what is most impacted by using two RTP
   sessions and changing identifiers.  The structure with two RTP
   sessions also puts a congestion control requirement on the middlebox,
   because it becomes fully responsible for the media stream it sources
   into each of the sessions.

   This can be solved locally or by bridging also statistics from the
   receiving endpoint.  However, from an implementation point this
   requires the implementation to support dealing with a number of
   inconsistencies.  First, packet loss must be detected for an RTP flow
   sent from A to the MB, and that loss must be reported through a
   skipped sequence number in the flow from the MB to B. This coupling
   and the resulting inconsistencies is conceptually easier to handle
   when considering the two flows as belonging to a single RTP session.

3.3.  Point to Multipoint Using Multicast

   Multicast is a IP layer functionality that is available in some
   networks.  Two main flavors can be distinguished: Any Source
   Multicast (ASM) where any multicast group participant can send to the
   group address and expect the packet to reach all group participants;
   and Source Specific Multicast (SSM), where only a particular IP host
   sends to the multicast group.  Both these models are discussed below
   in their respective section.

3.3.1.  Any Source Multicast (ASM)

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












Westerlund & Wenger      Expires August 29, 2013                [Page 9]


Internet-Draft               RTP Topologies                February 2013


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

               Figure 5: Point to Multipoint Using Multicast

   Point to Multipoint (PtM) is defined here as using a multicast
   topology as a transmission model, in which traffic from any
   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 scenario can occur, for
      example, where a multi-media session is distributed using two or
      more multicast groups and a participant is subscribed only to a
      subset of these sessions.

   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 in the
   range of billions).  The above can be realized using Any Source
   Multicast (ASM).

   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
   factors such as the connectivity is so good) that it allows the
   participants to use early or immediate feedback, as defined in AVPF
   [RFC4585].  Even when the environment would allow for the use of a
   small multicast group, some applications may still want to use the
   more limited options for RTCP feedback available to large multicast
   groups, for example when there is a likelyhood that the threshold of
   the small multicast group (in terms of participants) may be exceeded
   during the lifetime of a session.

   RTCP feedback messages in multicast reach, like media, every
   subscriber (subject to packet losses and multicast group
   subscription).  Therefore, the feedback suppression mechanism
   discussed in [RFC4585] is typically required.  Each individual node



Westerlund & Wenger      Expires August 29, 2013               [Page 10]


Internet-Draft               RTP Topologies                February 2013


   needs to process every feedback message it receives, not to determine
   if it is affected or if the feedback message applies only to some
   other participant, but also to derive timing restriction for the
   sending of its own feedback messages, if any.

3.3.2.  Source Specific Multicast (SSM)

   In Any Source Multicast, any of the participants can send to all the
   other participants, by sending a packet to the multicast group.  In
   contrast, Source Specific Multicast [RFC4607] refers to scenarios
   where only a single source (Distribution Source) can send to the
   multicast group, creating a topology that looks like the one below:

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

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

       Figure 6: Point to Multipoint using Source Specific Multicast

   In the SSM topology (Figure 6) a number of RTP sources (1 to M) are
   allowed to send media to the SSM group.  These send media to a
   dedicated distribution source, which then forwards the media streams
   to the multicast group on behalf of the original senders.  The media
   streams reach the Receivers (R(1) to R(n)).  The Receivers' RTCP
   cannot be sent to the multicast group, as the SSM multicast group by
   definition has only a single source.  To support RTCP, an RTP
   extension for SSM [RFC5760] was defined.  It uses unicast
   transmission to send RTCP from each of the receivers to one or more
   Feedback Targets (FT).  The feedback targets relay the RTCP
   unmodified, or provide summary of the participants RTCP reports



Westerlund & Wenger      Expires August 29, 2013               [Page 11]


Internet-Draft               RTP Topologies                February 2013


   towards the whole group by forwarding the RTCP traffic to the
   distribution source.  Figure 6 only shows a single feedback target
   integrated in the distribution source, but for scalability the FT can
   be many and have responsibility for sub-groups of the receivers.  For
   summary reports, however, there must be a single feedback aggregating
   all the summaries to a common message to the whole receiver group.

   The RTP extension for SSM specifies how feedback (both reception
   information and specific feedback events) are handled.  The more
   general problems associated with the use of multicast, where everyone
   receives what the distribution source sends 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 needs to be a member of most or all of these
       RTP sessions in order to perform well.

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

   3.  Multicast applications need 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 need to have the same RTP and payload type
   configuration.  Otherwise, A could, for example, be using payload
   type 97 to identify the video codec H.264, while B would identify it
   as MPEG-2.

   Security solutions for this type of group communications are also
   challenging.  First, 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].

3.3.3.  SSM with Local Unicast Resources

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






Westerlund & Wenger      Expires August 29, 2013               [Page 12]


Internet-Draft               RTP Topologies                February 2013


    -----------                                       --------------
   |           |------------------------------------>|              |
   |           |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->|              |
   |           |                                     |              |
   | Multicast |          ----------------           |              |
   |  Source   |         | Retransmission |          |              |
   |           |-------->|  Server  (RS)  |          |              |
   |           |.-.-.-.->|                |          |              |
   |           |         |  ------------  |          |              |
    -----------          | |  Feedback  | |<.=.=.=.=.|              |
                         | | Target (FT)| |<~~~~~~~~~| RTP Receiver |
   PRIMARY MULTICAST     |  ------------  |          |   (RTP_Rx)   |
   RTP SESSION with      |                |          |              |
   UNICAST FEEDBACK      |                |          |              |
                         |                |          |              |
   - - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- -
                         |                |          |              |
   UNICAST BURST         |  ------------  |          |              |
   (or RETRANSMISSION)   | |   Burst/   | |<~~~~~~~~>|              |
   RTP SESSION           | |  Retrans.  | |.........>|              |
                         | |Source (BRS)| |<.=.=.=.=>|              |
                         |  ------------  |          |              |
                         |                |          |              |
                          ----------------            --------------

      -------> Multicast RTP Flow
      .-.-.-.> Multicast RTCP Flow
      .=.=.=.> Unicast RTCP Reports
      ~~~~~~~> Unicast RTCP Feedback Messages
      .......> Unicast RTP Flow

                                 Figure 7

   The Rapid acquisition extension allows an endpoint joining an SSM
   multicast session to request media starting with the last sync-point
   (from where media can be decoded without prior packets) to be sent at
   high speed until such time where, after decoding of these bursted
   media packets, the correct media timing is established, i.e. media
   packets are received within adequate buffer intervals for this
   application.  This is accomplished by first establishing an unicast
   PtP RTP session between the BRS (Figure 7) and the RTP Receiver.
   That session is used to transmit cached packets from the multicast
   group at higher then nominal speed so to synchronize the receiver to
   the ongoing multicast packet flow.  Once the RTP receiver and its
   decoder have caught up with the multicast session's current delivery,
   the receiver switches over to receiving from the multicast group
   directly.  The (still existing) PtP RTP session can be used as a
   repair channel, i.e. for RTP Retransmission traffic of those packets



Westerlund & Wenger      Expires August 29, 2013               [Page 13]


Internet-Draft               RTP Topologies                February 2013


   that were not received from the multicast group.

3.4.  Point to Multipoint Using Mesh

   Shortcut name: Topo-Mesh

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

                 Figure 8: 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.

   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.

   An alternative structure for establishing the above topology is to
   use independent RTP sessions between each pair of peers, i.e. three
   different RTP sessions.  In some scenarios, the same RTP media stream
   is being sent from each sending endpoint.  In others, some form of
   local adaptation takes place in one or more of the RTP media streams,
   rendering them non-identical.  From a topologies viewpoint, a
   difference exists in the behaviours around RTCP.  For example, when a
   single RTP session spans all three endpoints and their connecting
   flows, a RTCP bandwidth is calculated and used for this single one
   joint session.  In contrast, when there are multiple independent RTP
   sessions, each has its local RTCP bandwidth allocation.  Also, when
   multiple sessions are used, endpoints not directly involved in these
   sessions do not have any awareness of the conditions occurring in
   sessions not involving that endpoint.  For example, in case of the
   three endpoint configuration above, endpoint A has no awareness of
   the conditions occurring in the session between endpoints B and C



Westerlund & Wenger      Expires August 29, 2013               [Page 14]


Internet-Draft               RTP Topologies                February 2013


   (whereas, if a single RTP session were used, it would have such
   awareness).  Loop detection is also affected.  With independent RTP
   sessions, the SSRC/CSRC can't be used to determine when a endpoint
   receives its own media stream or a mixed media stream including its
   own media stream a condition known as a loop.  The identification of
   loops and, in most cases, its avoidance, has to be achieved by other
   means, for example through signaling, or the use of an RTP external
   name space binding SSRC/CSRC among any communicating RTP sessions in
   the mesh.

3.5.  Point to Multipoint Using the RFC 3550 Translator

   This section discusses some additional usages related to point to
   multipoint of Translators compared to the point to point only cases
   in Section 3.2.1.

3.5.1.  Relay - Transport Translator

   Shortcut name: Topo-PtM-Trn-Translator

   This section discusses Transport Translator only usages to enable
   multipoint sessions.

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

               Figure 9: Point to Multipoint Using Multicast

   Figure 9 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.









Westerlund & Wenger      Expires August 29, 2013               [Page 15]


Internet-Draft               RTP Topologies                February 2013


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

         Figure 10: RTP Translator (Relay) with Only Unicast Paths

   Another Translator scenario is depicted in Figure 10.  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 set of considerations
   those described in the ASM RTP topology.  It also puts some
   additional signalling requirements onto the session establishment;
   for example, a common configuration of RTP payload types is required.

3.5.2.  Media Translator

   In the context of multipoint communications a Media Translator is not
   providing new mechanisms to establish a multipoint session.  It is
   much more an enabler or facilitator that ensures one or some sub-set
   of session participants can participate in the session.

   If B in Figure 9 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.  This
   transcoding can help the other participants in the Multicast part of
   the session, by not requiring the quality transmitted by A to be
   lowered to the nitrates that B is actually capable of receiving.

3.6.  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 generating a new RTP stream and, in most
   cases, by manipulation of the media data.  One common application for
   a Mixer is to allow a participant to receive a session with a reduced
   amount of resources.





Westerlund & Wenger      Expires August 29, 2013               [Page 16]


Internet-Draft               RTP Topologies                February 2013


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

       Figure 11: 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
   the uncompressed domain) and then encodes it for transmission to a
   participant.  The CSRC Count (CC) and CSRC fields in the RTP header
   can be 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 different conference participants 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.  This also
   implies that any SDES information normally needs to be forwarded
   across the mixer.

   The Mixer is responsible for generating RTCP packets in accordance
   with its role.  It is a receiver and should therefore send receiver
   reports for the media streams it receives.  In its role as a media
   sender, it should also generate sender reports for those media
   streams it sends.  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 11 is involved in three domains that
   need to be separated: the any source multicast network (including
   participants A and C), participant B, and participant D. Assuming all
   four participants in the conference are interested in receiving
   content from each other participant, the Mixer produces different
   mixed streams for B and D, as the one to B may contain content



Westerlund & Wenger      Expires August 29, 2013               [Page 17]


Internet-Draft               RTP Topologies                February 2013


   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 3.11 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 by the Mixer may result in the generation
   and transmission of RTCP feedback messages by the Mixer to the
   participants in the other domain(s).  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 11 (to the left of the
   Mixer) with individual unicast paths as depicted in Figure 12, the
   Mixer model is very similar to the one discussed in Section 3.9
   below.  Please see the discussion in Section 3.9 about the
   differences between these two models.

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

               Figure 12: RTP Mixer with Only Unicast Paths

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

3.6.1.  Media Mixing

   The media mixing mixer is likely the one that most think of when they
   hear the term "mixer".  Its basic pattern of operation is that it
   receives media streams from (typically several) participants.  Of
   those, it selects (either through static configuration or by dynamic,
   content dependent means such as voice activation) the stream(s) to be
   included in a media domain mix.  Then it creates a single outgoing



Westerlund & Wenger      Expires August 29, 2013               [Page 18]


Internet-Draft               RTP Topologies                February 2013


   stream from this mix.

   The most commonly deployed media mixer is probably the audio mixer,
   used in voice conferencing, where the output consists of a mixture of
   all the input streams; this needs minimal signalling to be
   successfully set up.  Audio mixing is relatively straightforward and
   commonly possible for a reasonable number of participants.  Lets
   assume that you want to mix N streams from different participants.
   The mixer needs to decode those N streams, typically into the sample
   domain.  Then it needs to produce N or N+1 mixes, the reasons that
   different mixes are needed being that each contributing source get a
   mix of all other sources except its own, 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 packetized and sent out.  In many cases, audio
   level normalization is also required before the actual mixing
   process.

   Video can't really be "mixed" and produce something particularly
   useful for the users, however creating an composition out of the
   contributed video streams is possible and known as "tiling".  For
   example the reconstructed, appropriately scaled down videos can be
   spatially arranged in a set of tiles, each tile containing the video
   from a participant.  Tiles can be of different sizes, so that, for
   example, a particularly important participant, or the loudest
   speaker, is being shown on in larger tile than other participants.  A
   self-picture can be included in the tiling, which can either be
   locally produced or be a feedback from a received and reconstructed
   video image (allowing for confidence monitoring, the participant sees
   himself/herself just as other participants see him/her).  The tiling
   normally operates on reconstructed video in the sample domain.  The
   tiled image is encoded, packetized, and sent by the mixer.  It is
   possible that a middlebox with media mixing duties contains only a
   single mixer of the aforementioned type, in which case all
   participants necessarily see the same tiled video, even if it is
   being sent over different RTP streams.  More common, however, are
   mixing arrangement where an individual mixer is available for each
   outgoing port of the middlebox, allowing individual compositions for
   each participant.

   One problem with media mixing is that it consumes both large amount
   of media processing (for the actual mixing process in the
   uncompressed domain) and encoding resources (for the encoding of the
   mixed signal).  Another problem is the quality degradation created by
   decoding and re-encoding the media that is encapsulated in the RTP
   media stream, which is the result of the lossy nature of most, if not
   all, commonly used media codecs.  A third problem is the latency



Westerlund & Wenger      Expires August 29, 2013               [Page 19]


Internet-Draft               RTP Topologies                February 2013


   introduced by the media mixing, which can be substantial and
   annoyingly noticeable in case of video.  The advantage of media
   mixing is that it is quite simplistic for the clients to handle the
   single media stream (which includes the mixed aggregate of many
   sources), as they don't need to handle multiple decodings, local
   mixing and composition.  In fact, mixers were introduced in pre-RTP
   times so that legacy, single stream receiving endpoints can
   successfully participate in what a user would recognize as a
   multiparty session.
           +-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 13: Session and SSRC details for Media Mixer

   From an RTP perspective media mixing can be very straightforward as
   can be seen in Figure 13.  The mixer presents one SSRC towards the
   receiving client, e.g.  MA1 to Peer A; the associated stream of which
   is the media mix of the other participants.  As, in this example,
   each peer receives a different version produced by the mixer, there
   is no actual relation between the different RTP sessions in the
   actual media or the transport level information.  There are, however,
   common relationships between RTP1-RTP3 namely SSRC space and identity



Westerlund & Wenger      Expires August 29, 2013               [Page 20]


Internet-Draft               RTP Topologies                February 2013


   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, allowing
   the receiver to identify the contributing sources even if this were
   not possible through the media itself or other signaling means.

   The CSRC has, in turn, utility in RTP extensions, like the Mixer to
   Client audio levels RTP header extension [RFC6465].  If the SSRC from
   endpoint to mixer leg are used as CSRC in another RTP session, then
   RTP1, RTP2 and RTP3 become one joint session as they have a common
   SSRC space.  At this stage, the mixer also need to consider which
   RTCP information it needs to expose in the different legs.  In the
   above scenario, commonly, a mixer would expose nothing more than the
   Source Description (SDES) information and RTCP BYE for CSRC leaving
   the session.  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.

3.6.2.  Media Switching

   Media switching mixers are commonly used in such limited
   functionality scenarios where no, or only very limited, concurrent
   presentation of multiple sources is required by the application.  An
   RTP Mixer based on media switching avoids the media decoding and
   encoding cycle in the mixer, as it conceptually forwards the encoded
   media stream as it was being sent to the mixer, but not the
   decryption and re-encryption cycle as it rewrites RTP headers.
   Forwarding media (in contrast to reconstructing-mixing-encoding
   media) reduces the amount of computational resources needed in the
   mixer and increases the media quality (both in terms of fidelity and
   reduced latency) per transmitted bit.

   A media switching mixer maintains a pool of SSRCs representing
   conceptual or functional streams the mixer can produce.  These
   streams are created by selecting media from one of RTP media streams
   received by the mixer and forwarded to the peer using the mixer's own
   SSRCs.  The mixer can switch between available sources if that is
   required by the concept for the source, like currently active
   speaker.  Note that the mixer, in most cases, still need to perform a
   certain amount of media processing, as many media formats do not
   allow to "tune" into the stream at arbitrary points of their
   bitstream.

   To achieve a coherent RTP media stream from the mixer's SSRC, the
   mixer needs 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



Westerlund & Wenger      Expires August 29, 2013               [Page 21]


Internet-Draft               RTP Topologies                February 2013


   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, and each peer is
   capable of locally handling two video streams simultaneously.
          +-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 14: Media Switching RTP Mixer




Westerlund & Wenger      Expires August 29, 2013               [Page 22]


Internet-Draft               RTP Topologies                February 2013


   The Media Switching RTP mixer can, similarly to the Media Mixing
   Mixer, reduce the bit-rate required for media transmission towards
   the different peers by selecting and forwarding only a sub-set of RTP
   media streams it receives from the conference participants.  In many
   practical cases, the link capacities of either direction between
   peers and mixer are the same, which effectively limits the subset to
   a single media stream.

   To ensure that a media receiver can correctly decode the RTP media
   stream after a switch, a state saving (frame-based) codec needs to
   start its decoding from independent refresh points or similar points
   in the bitstream.  For some codecs, for example frame based speech
   and audio codecs, this is easily achieved by starting the decoding at
   RTP packet boundaries (proper packetization on the encoder side
   assumed), as each packet boundary provides a refresh point.  For
   other (mostly video-) codecs, refresh points are less common in the
   bitstream or may not be present at all without an explicit request to
   the respective encoder.  For this purpose there exists the Full Intra
   Request [RFC5104] RTCP codec control message.

   Also in this type of mixer one could consider to terminate the RTP
   sessions fully between the different end-point and mixer legs.  The
   same arguments and considerations as discussed in Section 3.9 need to
   be taken into consideration and apply here.

3.7.  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 middlebox can select which of the potential sources
   that are currently actively transmitting media, despite that the
   middlebox, in another RTP session, may receive media from that end-
   point.  This is similar to the media switching Mixer but has some
   important differences in RTP details.

















Westerlund & Wenger      Expires August 29, 2013               [Page 23]


Internet-Draft               RTP Topologies                February 2013


          +-A---------+             +-Middlebox-----------------+
          | +-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---+ |       |     | |
          | | |    FV1|------------>|---------+-+------>|     | |
          | | |       |<------------|AV1 <----+-+-------|     | |
          | | |       | :    :    : |: :  : : : : :  : :|     | |
          | | |       |<------------|EV1 <----+-+-------|     | |
          | | +-------|             |---------+ |       |     | |
          | +---------|             |-----------+       +-----+ |
          +-----------+             +---------------------------+

                   Figure 15: Media Projecting Middlebox

   In the six participant conference depicted above in (Figure 15) one
   can see that end-point A is aware of five incoming SSRCs, BV1-FV1.
   If this middlebox intends to have a similar behaviour as in
   Section 3.6.2 where the mixer provides the end-points with the two
   latest speaking end-points, then only two out of these five SSRCs
   need concurrently transmit media to A. As the middlebox selects the
   source in the different RTP sessions that transmit media to the end-
   points, each RTP media stream requires some rewriting of RTP header
   fields when being projected from one session into another.  In



Westerlund & Wenger      Expires August 29, 2013               [Page 24]


Internet-Draft               RTP Topologies                February 2013


   particular, the sequence number needs to be consecutively incremented
   based on the packet actually being transmitted in each RTP session.
   Therefore, the RTP sequence number offset will change each time a
   source is turned on in a RTP session.  The timestamp (possibly
   offset) stays the same.

   As the RTP sessions are independent, the SSRC numbers used can also
   be handled independently, thereby bypassing the requirement for SSRC
   collision detection and avoidance.  On the other hand, tools such as
   remapping tables between the RTP sessions are required.  For example,
   the stream that is being sent by endpoint B to the middlebox (BV1)
   may use an SSRC value of 12345678.  When that media stream is sent to
   endpoint F by the middlebox, it can use any SSRC value, e.g.
   87654321.  As a result, each endpoint may have a different view of
   the application usage of a particular SSRC.  Any RTP level identity
   information, such as SDES items also needs to update the SSRC
   referenced, if the included SDES items are intended to be global.
   Thus the application must not use SSRC as references to RTP media
   streams when communicating with other peers directly.  This also
   affects loop detection which will fail to work, as there is no common
   namespace and identities across the different legs in the
   communication session on RTP level.  Instead this responsibility
   falls onto higher layers.

   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
   request locally or needs to translate the request into the RTP
   session that contains the media source.  Both end-points and the
   middlebox need to implement conference related codec control
   functionalities to provide a good experience.  Commonly used are Full
   Intra Request to request from the media source to provide switching
   points between the sources, and Temporary Maximum Media Bit-rate
   Request (TMMBR) to enable the middlebox to aggregate congestion
   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
   source to middlebox link).

   This version of the middlebox 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 be able to handle
   as many decoder instances as the middlebox received, or have
   efficient switching of decoder contexts in a more limited set of
   actual decoder instances to cope with the switches.  The application
   also gets more responsibility to update how the media provided is to
   be presented to the user.

   Note, this could potentially be seen as a media translator which



Westerlund & Wenger      Expires August 29, 2013               [Page 25]


Internet-Draft               RTP Topologies                February 2013


   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.  It also has
   mixer aspects, as the streams it provides are not basically
   translated version, but instead they have conceptual property
   assigned to them.  Thus this topology appears to be some hybrid
   between the translator and mixer model.

3.8.  Point to Multipoint Using Video Switching MCUs

   Shortcut name: Topo-Video-switch-MCU

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

        Figure 16: Point to Multipoint Using a Video Switching MCU

   This PtM topology was common before, although the RTCP-terminating
   MCUs, as discussed in the next section, where perhaps even more
   common.  This topology, as well as the following one, was a result of
   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.




Westerlund & Wenger      Expires August 29, 2013               [Page 26]


Internet-Draft               RTP Topologies                February 2013


   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.

   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
   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.

3.9.  Point to Multipoint Using RTCP-Terminating MCU

   Shortcut name: Topo-RTCP-terminating-MCU

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

        Figure 17: 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



Westerlund & Wenger      Expires August 29, 2013               [Page 27]


Internet-Draft               RTP Topologies                February 2013


   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
   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.






Westerlund & Wenger      Expires August 29, 2013               [Page 28]


Internet-Draft               RTP Topologies                February 2013


3.10.  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 its 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 18.

   Which decomposition scheme is possible is highly dependent on the RTP
   session usage.  It is not really feasible to decompose one logical
   end-point into two different transport nodes in one RTP session.  A
   third party monitor would report such an attempt as two entities
   being two different end-points with a CNAME collision.  As a result,
   a fully RTP conformant de-composited endpoint is one where the
   different decomposed parts use 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 18: De-composite End-Point

   In the above usage, let us assume that the different RTP sessions are
   used for audio and video.  The audio and video parts, however, use a
   common CNAME and also have a common clock to ensure that
   synchronization 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 requirement that can be derived from the above usage is that the
   transport flows for each RTP session might be under common control,



Westerlund & Wenger      Expires August 29, 2013               [Page 29]


Internet-Draft               RTP Topologies                February 2013


   but still are addressed 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.

3.11.  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 11,
   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 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. Therefore, 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 locally generated reports reflecting the situation
   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.

3.12.  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 does not forward RTCP from sources in other domains,
   but instead generates 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



Westerlund & Wenger      Expires August 29, 2013               [Page 30]


Internet-Draft               RTP Topologies                February 2013


   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 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.

   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 3.5, 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.




Westerlund & Wenger      Expires August 29, 2013               [Page 31]


Internet-Draft               RTP Topologies                February 2013


   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.

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).



Westerlund & Wenger      Expires August 29, 2013               [Page 32]


Internet-Draft               RTP Topologies                February 2013


   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 3.8 does not
   entirely fulfill this.  The one described in Section 3.9 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 3.8, at least as long as it forwards the
   RTCP between the participants.  However, the MCU in Section 3.9 will
   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



Westerlund & Wenger      Expires August 29, 2013               [Page 33]


Internet-Draft               RTP Topologies                February 2013


   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
   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:




Westerlund & Wenger      Expires August 29, 2013               [Page 34]


Internet-Draft               RTP Topologies                February 2013


   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,
   a Translator needs one security association per participant it
   translates towards, in the opposite domain.  Considering Figure 9,
   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.







Westerlund & Wenger      Expires August 29, 2013               [Page 35]


Internet-Draft               RTP Topologies                February 2013


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

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

   [I-D.ietf-avtcore-rtp-security-options]
              Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", draft-ietf-avtcore-rtp-security-options-01
              (work in progress), October 2012.

   [I-D.lennox-avtcore-rtp-multi-stream]
              Lennox, J. and M. Westerlund, "Real-Time Transport
              Protocol (RTP) Considerations for Endpoints Sending
              Multiple Media Streams",
              draft-lennox-avtcore-rtp-multi-stream-01 (work in
              progress), October 2012.

   [RFC3022]  Srisuresh, P. and K. Egevang, "Traditional IP Network
              Address Translator (Traditional NAT)", RFC 3022,
              January 2001.

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



Westerlund & Wenger      Expires August 29, 2013               [Page 36]


Internet-Draft               RTP Topologies                February 2013


   [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.

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              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
              Transport Protocol (RTP) Header Extension for Mixer-to-
              Client Audio Level Indication", RFC 6465, December 2011.


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











Westerlund & Wenger      Expires August 29, 2013               [Page 37]


Html markup produced by rfcmarkup 1.129c, available from https://tools.ietf.org/tools/rfcmarkup/