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Versions: (draft-westerlund-avtcore-rtp-topologies-update) 00 01 02 03 04 05 06 07 08 09 10 RFC 7667

Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Obsoletes: 5117 (if approved)                                  S. Wenger
Intended status: Informational                                     Vidyo
Expires: November 28, 2014                                  May 27, 2014


                             RTP Topologies
              draft-ietf-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 is 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 November 28, 2014.

Copyright Notice

   Copyright (c) 2014 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
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   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must



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

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   3
     2.1.  Glossary  . . . . . . . . . . . . . . . . . . . . . . . .   3
   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 . . . . . . . . . . . . . .   9
     3.3.  Point to Multipoint Using Multicast . . . . . . . . . . .  10
       3.3.1.  Any Source Multicast (ASM)  . . . . . . . . . . . . .  10
       3.3.2.  Source Specific Multicast (SSM) . . . . . . . . . . .  11
       3.3.3.  SSM with Local Unicast Resources  . . . . . . . . . .  13
     3.4.  Point to Multipoint Using Mesh  . . . . . . . . . . . . .  15
     3.5.  Point to Multipoint Using the RFC 3550 Translator . . . .  18
       3.5.1.  Relay - Transport Translator  . . . . . . . . . . . .  18
       3.5.2.  Media Translator  . . . . . . . . . . . . . . . . . .  19
     3.6.  Point to Multipoint Using the RFC 3550 Mixer Model  . . .  20
       3.6.1.  Media Mixing  . . . . . . . . . . . . . . . . . . . .  22
       3.6.2.  Media Switching . . . . . . . . . . . . . . . . . . .  25
     3.7.  Selective Forwarding Middlebox  . . . . . . . . . . . . .  27
     3.8.  Point to Multipoint Using Video Switching MCUs  . . . . .  30
     3.9.  Point to Multipoint Using RTCP-Terminating MCU  . . . . .  32
     3.10. Split Component Endpoint  . . . . . . . . . . . . . . . .  33
     3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . .  34
     3.12. Combining Topologies  . . . . . . . . . . . . . . . . . .  35
   4.  Comparing Topologies  . . . . . . . . . . . . . . . . . . . .  35
     4.1.  Topology Properties . . . . . . . . . . . . . . . . . . .  36
       4.1.1.  All to All Media Transmission . . . . . . . . . . . .  36
       4.1.2.  Transport or Media Interoperability . . . . . . . . .  37
       4.1.3.  Per Domain Bit-Rate Adaptation  . . . . . . . . . . .  37
       4.1.4.  Aggregation of Media  . . . . . . . . . . . . . . . .  37
       4.1.5.  View of All Session Participants  . . . . . . . . . .  38
       4.1.6.  Loop Detection  . . . . . . . . . . . . . . . . . . .  38
     4.2.  Comparison of Topologies  . . . . . . . . . . . . . . . .  39
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .  39
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  41
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  41
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  41
     8.1.  Normative References  . . . . . . . . . . . . . . . . . .  41
     8.2.  Informative References  . . . . . . . . . . . . . . . . .  42
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  43




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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 the Multipoint Control Unit or MCU.

   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.  In practice, however, most multipoint conferences operate
   utilizing centralized units referred to as MCUs.  MCUs may implement
   Mixer or Translator functionality (in RTP [RFC3550] terminology), and
   signalling support.  They may also contain additional application
   layer 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.

   This document also codifies new multipoint architectures that have
   recently been introduced and which were not anticipated in RFC 5117.
   These architectures use scalable video coding and simulcasting, and
   their associated centralized units are referred to as Selective
   Forwarding Units (SFU).  This codification provides a common
   information basis for future discussion and specification work.

   The document's attempt to clarify and explain sections of the Real-
   time Transport Protocol (RTP) spec [RFC3550] is informal.  It is not
   intended to update or change what is normatively specified within RFC
   3550.

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

   Middlebox:  A device that is on the Path that media travel between
      two Endpoints



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   MCU:  Multipoint Control Unit

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

   PtM:  Point to Multipoint

   PtP:  Point to Point

   SFU:  Selective Forwarding Unit

   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, with or without middleboxes.  Then
   follows a number of different methods for establishing point to
   multipoint communication.  These are structured around the most
   fundamental enabler, i.e., multicast, a mesh of connections,
   translators, mixers and finally MCUs and SFUs.  The section ends by
   discussing de-composited 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].

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).







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

                         Figure 1: Point to Point

   The main property of this topology is that A sends to B, and only B,
   while B sends to A, and only A.  This avoids all complexities of
   handling multiple endpoints and combining the requirements stemming
   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, subject
   only to system level limitations like the number range of ports.

   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 "Sending
   Multiple Media Streams in a Single RTP Session"
   [I-D.ietf-avtcore-rtp-multi-stream-optimisation].

3.2.  Point to Point via Middlebox

   This section discusses cases where two endpoints communicate but have
   one or more middleboxes 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,



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   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
   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, 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 is 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., to
   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 peer's 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





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   A common element in these functions is that they are normally
   transparent at the RTP level, i.e., they perform no changes on any
   RTP or RTCP packet fields and only affect the lower layers.  They may
   affect, however, 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 be included in this
   document.  This topology, however, can raise additional requirements
   in the RTP implementation and its interactions with the signalling
   solution.  Both in signalling and in certain RTCP fields, network
   addresses other than those of the relay can occur since B has a
   different network address than the relay (T).  Implementations that
   can not support this will also not work correctly when endpoints are
   subject to NAT.

   The transport relay implementation also have some considerations,
   where security considerations are an important aspect.  Source
   address filtering of incoming packets are usually important in
   relays, to prevent attackers to inject traffic into a session, which
   one peer will think comes from the other peer.

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, an important consideration is how to
   handle different sources and their identities.  This problem space is
   not discussed henceforth.

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

3.2.1.3.  Media Translator

   Media Translators (Topo-Media-Translator) 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 decoding and re-



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   encoding (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 Translator is 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,
   the extended highest sequence number would indicate that B were
   substantially behind in reception, while 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.










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   +------+        +------+         +------+
   |      |        |      |         |      |
   |  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, but not by being part of a common RTP session.  They
   establish instead two different RTP sessions, one between A and the
   middlebox and another between the middlebox and B.  This topology is
   called Topo-Back-To-Back

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

               Figure 4: When De-composition is a Translator

   The middlebox acts as an application-level gateway and bridges the
   two RTP sessions.  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 middlebox's task to maintain the correct
   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.

   Adherence to congestion control can be solved locally or by bridging
   also statistics from the receiving endpoint.  From an implementation
   point, however, this requires dealing with a number of
   inconsistencies.  First, packet loss must be detected for an RTP flow
   sent from A to the middlebox, and that loss must be reported through



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   a skipped sequence number in the flow from the middlebox 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 an IP layer functionality that is available in some
   networks.  Two main flavors can be distinguished: Any Source
   Multicast (ASM) [RFC1112] 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) [RFC3569], where
   only a particular IP host sends to the multicast group.  Both these
   models are discussed below in their respective sections.

3.3.1.  Any Source Multicast (ASM)

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

               +-----+
    +---+     /       \    +---+
    | 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 multimedia 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



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   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 likelihood 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 data, every
   subscriber (subject to packet losses and multicast group
   subscription).  Therefore, the feedback suppression mechanism
   discussed in [RFC4585] is typically required.  Each individual node
   needs to process every feedback message it receives, not only to
   determine if it is affected or if the feedback message applies only
   to some other participant, but also to derive timing restrictions 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 [RFC3569][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:



















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

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

       Figure 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 sources send media to
   a dedicated distribution source, which 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
   messages cannot be sent to the multicast group, as the SSM multicast
   group by definition has only a single IP sender.  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 a summary of the participants RTCP reports
   towards the whole group by forwarding the RTCP traffic to the
   distribution source.  Figure 6 only shows a single feedback target
   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.




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   Aforementioned situation results in common behavior 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 [RFC7201].

3.3.3.  SSM with Local Unicast Resources

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



















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    -----------                                       --------------
   |           |------------------------------------>|              |
   |           |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->|              |
   |           |                                     |              |
   | 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 requiring context
   established by the decoding of prior packets) to be sent at high
   speed until such time where, after decoding of these burst-delivered
   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 a unicast
   PtP RTP session between the Burst/Retransmission Source (BRS,
   Figure 7) and the RTP Receiver.  The unicast session is used to
   transmit cached packets from the multicast group at higher then
   normal speed in order 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 directly from the multicast group.  The



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   (still existing) PtP RTP session is, in many deployed applications,
   be used as a repair channel, i.e., for RTP Retransmission traffic of
   those packets 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 joint three endpoint 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 does not 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.


















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   +-A--------------------+
   |+---+                 |
   ||CAM|                 |                 +-B-----------+
   |+---+     +-UDP1------|                 |-UDP1------+ |
   |  |       | +-RTP1----|                 |-RTP1----+ | |
   |  V       | | +-Video-|                 |-Video-+ | | |
   |+----+    | | |       |<----------------|BV1    | | | |
   ||ENC |----+-+-+--->AV1|---------------->|       | | | |
   |+----+    | | +-------|                 |-------+ | | |
   |  |       | +---------|                 |---------+ | |
   |  |       +-----------|                 |-----------+ |
   |  |                   |                 +-------------+
   |  |                   |
   |  |                   |                 +-C-----------+
   |  |       +-UDP2------|                 |-UDP2------+ |
   |  |       | +-RTP1----|                 |-RTP1----+ | |
   |  |       | | +-Video-|                 |-Video-+ | | |
   |  +-------+-+-+--->AV1|---------------->|       | | | |
   |          | | |       |<----------------|CV1    | | | |
   |          | | +-------|                 |-------+ | | |
   |          | +---------|                 |---------+ | |
   |          +-----------|                 |-----------+ |
   +----------------------+                 +-------------+

         Figure 9: An Multi-unicast Mesh with a joint RTP session

   A joint RTP session from A's perspective for the Mesh depicted in
   Figure 8 with a joint RTP session have multiple transport flows, here
   enumerated as UDP1 and UDP2.  However, there is only one RTP session
   (RTP1).  The media source (CAM) is encoded and transmitted over the
   SSRC (AV1) across both transport layers.  However, as this is a joint
   RTP session, the two streams must be the same.  Thus, an congestion
   control adaptation needed for the paths A to B and A to C needs to
   use the most restricting path's properties.

   An alternative structure for establishing the above topology is to
   use independent RTP sessions between each pair of peers, i.e., three
   different RTP sessions.  In some scenarios, the same RTP media stream
   may be sent from transmitting endpoint, however it also supports
   local adaptation taking place in one or more of the RTP media
   streams, rendering them non-identical.










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   +-A----------------------+              +-B-----------+
   |+---+                   |              |             |
   ||MIC|       +-UDP1------|              |-UDP1------+ |
   |+---+       | +-RTP1----|              |-RTP1----+ | |
   | |  +----+  | | +-Audio-|              |-Audio-+ | | |
   | +->|ENC1|--+-+-+--->AA1|------------->|       | | | |
   | |  +----+  | | |       |<-------------|BA1    | | | |
   | |          | | +-------|              |-------+ | | |
   | |          | +---------|              |---------+ | |
   | |          +-----------|              |-----------+ |
   | |          ------------|              |-------------|
   | |                      |              |-------------+
   | |                      |
   | |                      |              +-C-----------+
   | |                      |              |             |
   | |          +-UDP2------|              |-UDP2------+ |
   | |          | +-RTP2----|              |-RTP2----+ | |
   | |  +----+  | | +-Audio-|              |-Audio-+ | | |
   | +->|ENC2|--+-+-+--->AA2|------------->|       | | | |
   |    +----+  | | |       |<-------------|CA1    | | | |
   |            | | +-------|              |-------+ | | |
   |            | +---------|              |---------+ | |
   |            +-----------|              |-----------+ |
   +------------------------+              +-------------+

       Figure 10: An Multi-unicast Mesh with independent RTP session

   Lets review the topology when independent RTP sessions are used, from
   A's perspective in Figure 8 by considering both how the media is a
   handled and the RTP sessions that are set-up in Figure 10.  A's
   microphone is captured and the digital audio can then be feed into
   two different encoder instances, as each beeing associated with two
   independent RTP sessions (RTP1 and RTP2).  The SSRCs (AA1 and AA2) in
   each RTP session will be completely independent and the media bit-
   rate produced by the encoders can also be tuned differently to
   address any congestion control requirements differing for the paths A
   to B compared to A to C.

   From a topologies viewpoint, an important difference exists in the
   behavior around RTCP.  First, when a single RTP session spans all
   three endpoints and their connecting flows, an common RTCP bandwidth
   is calculated and used for this single joint session.  In contrast,
   when there are multiple independent RTP sessions, each RTP session
   has its local RTCP bandwidth allocation.

   Further, when multiple sessions are used, endpoints not directly
   involved in a session, do not have any awareness of the conditions in
   those sessions.  For example, in the case of the three endpoint



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   configuration in Figure 8, endpoint A has no awareness of the
   conditions occurring in the session between endpoints B and C
   (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 cannot be used to determine when an 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, their 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 11: Point to Multipoint Using Multicast

   Figure 11 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.





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

         Figure 12: RTP Translator (Relay) with Only Unicast Paths

   Another Translator scenario is depicted in Figure 12.  The Translator
   in this case 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 to
   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.

   Transport translators and relays should always consider doing source
   address filtering, to prevent attackers to inject traffic using the
   listening ports on the translator.  The translator can however go one
   step further, and especially if explicit SSRC signalling is used,
   prevent other session participants to send SSRCs that are used by
   other participants in the session.  This can improve the security
   properties of the session, despite the use of group keys that on
   cryptographic level allows anyone to impersonate another in the same
   RTP session.

   A Translator that doesn't change the RTP/RTCP packets content can be
   operated without the requiring the translator to have access to the
   security contexts used to protect the RTP/RTCP traffic between the
   participants.

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
   more of an enabler, or facilitator, that ensures one or some sub-set
   of session participants can participate in the session.

   If B in Figure 11 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



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   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 bitrates 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 that are
   part of a session by generating one or more new RTP streams and, in
   most cases, by manipulating the media data.  One common application
   for a Mixer is to allow a participant to receive a session with a
   reduced amount of resources.

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

       Figure 13: 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 media streams that are
   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 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.



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   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 13 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
   received from D, and vice versa.  However, the Mixer may only need
   one SSRC per media type in each domain where it is the receiving
   entity and transmitter of 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.
   The mixing operation, however, 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 13 (to the left of the
   Mixer) with individual unicast paths as depicted in Figure 14, the
   Mixer model is very similar to the one discussed in Section 3.9
   below.  Please see the discussion in Section 3.9 about the
   differences between these two models.












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

               Figure 14: RTP Mixer with Only Unicast Paths

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

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 mode of operation is that it
   receives media streams from several participants and selects the
   stream(s) to be included in a media-domain mix.  The selection can be
   through static configuration or by dynamic, content dependent means
   such as voice activation.  The mixer then creates a single outgoing
   stream from this mix.

   The most commonly deployed media mixer is probably the audio mixer,
   used in voice conferencing, where the output consists of 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.  Assume,
   for example, that one wants to mix N streams from different
   participants.  The mixer needs to decode those N streams, typically
   into the sample domain, and then produce N or N+1 mixes.  Different
   mixes are needed so that each contributing source gets a mix of all
   other sources except its own, 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.

   In video, the term "mixing" has a different interpretation than
   audio.  It is commonly used to refer to the process of spatially
   combining contributed video streams is known as "tiling".  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-view
   picture can be included in the tiling, which can either be locally



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   produced or be a feedback from a received and reconstructed video
   image.  Such remote loopback allows for confidence monitoring, i.e.,
   it enables the participant to see 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 (a
   feature referred to as personalized layout).

   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
   commonly used media codecs.  A third problem is the latency
   introduced by the media mixing, which can be substantial and
   annoyingly noticeable in case of video, or in case of audio if that
   mixed audio is lip-sychronized with high latency video.  The
   advantage of media mixing is that it is straightforward 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 could successfully participate in what a user would
   recognize as a multiparty video conference.




















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   +-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 15: Session and SSRC details for Media Mixer

   From an RTP perspective media mixing can be a very simple process, as
   can be seen in Figure 15.  The mixer presents one SSRC towards the
   receiving client, e.g., MA1 to Peer A, where the associated stream is
   the media mix of the other participants.  As each peer, in this
   example, receives a different version of a mix from the mixer, there
   is no actual relation between the different RTP sessions in terms of
   actual media or transport level information.  There are, however,
   common relationships between RTP1-RTP3, namely SSRC space and
   identity information.  When A receives the MA1 stream which is a
   combination of BA1 and CA1 streams, the mixer may include CSRC
   information in the MA1 stream to identify the contributing source BA1
   and CA1, allowing the receiver to identify the contributing sources
   even if this were not possible through the media itself or through
   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 SSRCs



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   from the endpoint to mixer paths are used as CSRCs in another RTP
   session, then RTP1, RTP2 and RTP3 become one joint session as they
   have a common SSRC space.  At this stage, the mixer also needs to
   consider which RTCP information it needs to expose in the different
   paths.  In the above scenario, a mixer would normally expose nothing
   more than the Source Description (SDES) information and RTCP BYE for
   a 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 used from limited functionality scenarios
   where no, or only very limited, concurrent presentation of multiple
   sources is required by the application to more complex multi-stream
   usages with receiver mixing or tiling, including combined with
   simulcast and/or scalability between source and mixer.  An RTP Mixer
   based on media switching avoids the media decoding and encoding
   operations in the mixer, as it conceptually forwards the encoded
   media stream as it was being sent to the mixer.  It does not avoid,
   however, the decryption and re-encryption cycle as it rewrites RTP
   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).

   A media switching mixer maintains a pool of SSRCs representing
   conceptual or functional streams that the mixer can produce.  These
   streams are created by selecting media from one of the 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 the currently
   active speaker.  Note that the mixer, in most cases, still needs 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.  Second, the
   sequence number must be the next in the sequence of outgoing packets
   it sent.  Third, the RTP timestamp value needs to be adjusted using
   an offset that changes each time one switches media source.  Finally,
   depending on the negotiation of the RTP payload type, the value
   representing this particular RTP payload configuration may have to be
   changed if the different endpoint mixer paths have not arrived on the
   same numbering for a given configuration.  This also requires that




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   the different endpoints support a common set of codecs, otherwise
   media transcoding for codec compatibility would still be required.

   We now consider the operation of a media switching mixer that
   supports a video conference with six participants (A-F) where the two
   most recent speakers in the conference are shown to each participant.
   The mixer has thus 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 16: Media Switching RTP Mixer

   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 cases
   the mixer receives simulcast transmissions or a scalable encoding of



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   the media source, the mixer has more degrees of freedom to select
   streams or sub-sets of stream to forward to a receiver, both based on
   transport or client restrictions as well as application logic.

   To ensure that a media receiver can correctly decode the RTP media
   stream after a switch, a codec that uses temporal prediction 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, as each packet boundary provides a refresh
   point (assuming proper packetization on the encoder side).  For other
   codecs, particularly in video, refresh points are less common in the
   bitstream or may not be present at all without an explicit request to
   the respective encoder.  The Full Intra Request [RFC5104] RTCP codec
   control message has been defined for this purpose.

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

3.7.  Selective Forwarding Middlebox

   Another method for handling media in the RTP mixer is to "project",
   or make available, all potential RTP sources (SSRCs) into a per-
   endpoint, independent RTP session.  The middlebox can select which of
   the potential sources that are currently actively transmitting media
   will be sent to each of the endpoints.  This is similar to the media
   switching Mixer but has some important differences in RTP details.






















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   +-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 17: Selective Forwarding Middlebox

   In the six participant conference depicted above in (Figure 17) one
   can see that end-point A is aware of five incoming SSRCs, BV1-FV1.
   If this middlebox intends to have a similar behavior as in
   Section 3.6.2 where the mixer provides the end-points with the two
   latest speaking 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



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   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).

   The selective forwarding middlebox has been introduced in recently
   developed videoconferencing systems in conjunction with, and to
   capitalize on, scalable video coding as well as simulcasting.  An
   example of scalable video coding is Annex G of H.264, but other
   codecs, including H.264 AVC and VP8 also exhibit scalability, albeit
   only in the temporal dimension.  In both scalable coding and
   simulcast cases the video signal is represented by a set of two or
   more bitstreams, providing a corresponding number of distinct
   fidelity points.  The middlebox selects which parts of a scalable
   bitstream (or which bitstream, in the case of simulcasting) to
   forward to each of the receiving endpoints.  The decision may be



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   driven by a number of factors, such as available bit rate, desired
   layout, etc.  Contrary to transcoding MCUs, these "Selective
   Forwarding Units" (SFUs) have extremely low delay, and provide
   features that are typically associated with high-end systems
   (personalized layout, error localization) without any signal
   processing at the middlebox.  They are also capable of scaling to a
   large number of concurrent users, and--due to their very low delay--
   can also be cascaded.

   This version of the middlebox also puts different requirements on the
   endpoint 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 endpoint 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 that this topology could potentially be seen as a media
   translator which include an on/off logic as part of its media
   translation.  The main difference would be a common global SSRC space
   in the case of the Media Translator and the mapped one used in the
   above.  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.

   The differences between selective forwarding middlebox and a
   switching mixer (Section 3.6.2) are minor, and they share most
   properties.  The above requirement on having a large number of
   decoding instances or requiring efficient switching of decoder
   contexts, are one point of difference.  The other is how the
   identification is performed, where the Mixer uses CSRC to provide
   info what is included in a particular RTP packet stream that
   represent a particular concept.  Selective forwarding gets the source
   information through the SSRC, and instead have to use other mechanism
   to make clear the streams current purpose.

3.8.  Point to Multipoint Using Video Switching MCUs

   Shortcut name: Topo-Video-switch-MCU









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

        Figure 18: Point to Multipoint Using a Video Switching MCU

   This PtM topology was popular in early implementations of multipoint
   videoconferencing systems due to its simplicity, and the
   corresponding middlebox design has been known as a "video switching
   MCU".  The more complex RTCP-terminating MCUs, discussed in the next
   section, became the norm, however, when technology allowed
   implementations at acceptable costs.

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

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

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

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





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   To avoid the aforementioned issues, the MCU needs to implement two
   features.  First, it needs to act as a Mixer (see Section 3.6) 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 19: 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 terms of bit-rate,
   encoding format, or resolution.  No explicit RTP mechanism is used to
   establish the relationship between the original media sender and the
   version the MCU sends.  In other words, the outgoing sessions
   typically use a different SSRC, and may well use a different payload
   type (PT), even if this different PT happens to be mapped to the same
   media type.  This is a result of the individually negotiated session
   for each participant.

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



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

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

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

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

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

3.10.  Split Component Endpoint

   Shortcut name: Topo-Split-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 into different
   components.  For example, in a video conference room system the
   endpoint could be considered as being composed of one device handling
   the audio and another handling the video, interconnected by some
   control functions allowing them to behave as a single endpoint in all
   aspects except for transport as depicted in Figure 20.

   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



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   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 20: Split Component Endpoint

   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 fact that
   the components are separated.  Also, RTCP handling works correctly as
   long as only one part of the split endpoint 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,
   but still are addressed to what looks like different endpoints (based
   on addresses and ports).  This connection diagram cannot be
   accomplished using one RTP session and thus 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 13,
   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



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   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 for implementation.

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
   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.  It 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 describes these properties and then analyzes how
   these properties are supported by the different topologies.  Note
   that, even if a certain property is supported within a particular



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   topology concept, the necessary functionality may be optional to
   implement.

4.1.  Topology Properties

4.1.1.  All to All Media Transmission

   To recapitulate, multicast, and in particular Any Source Multicast
   (ASM), provides the functionality that everyone may send to, or
   receive from, everyone else within the session.  Source-specific
   Multicast (SSM) can provide a similar functionality by having anyone
   intending to participate as sender to send its media to the SSM
   distribution source.  The SSM distribution source forwards the media
   to all receivers subscribed to the multicast group.  Mesh, MCUs,
   Mixers, SFMs 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.

   Closest to true IP-multicast-based, all to all transmission comes
   perhaps the transport Translator function called "relay" in in
   Section 3.5, as well as the Mesh with joint RTP sessions.  Media
   Translators, Mesh with independent RTP Sessions, Mixers, SFUs and the
   MCU variants do not provide a fully meshed forwarding on the
   transport level; instead, they only allow limited forwarding of
   content from the other session participants.

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

   The resource consumption for performing all to all transmission
   Varies depending with the topology.  Both ASM and SSM have the
   benefit that only one copy of each packet traverses a particular
   link.  Using a relay causes the transmission of one copy of a packet
   per client-to-relay path and packet transmitted.  However, in most
   cases the links carrying the multiple copies will be the ones close
   to the relay (which can be assumed to be part of the network
   infrastructure with good connectivity to the backbone), rather than
   the clients (which may be behind slower access links).  The Mesh
   causes N-1 streams of transmitted packets to traverse the first hop
   link from the client, in an N client mesh.  How long the different
   paths are common, is highly situation dependent.



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   The transmission of RTCP by design 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

   All Translators, Mixers, and RTCP-terminating MCU, and Mesh with
   individual RTP sessions, 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.  Selective
   Forwarding Middleboxes can adopt the transport, and (at least)
   selectively forward the encoded streams that match a receiver's
   capability.  It requires an additional translator to change the media
   encoding if the encoded streams do not match the receiver's
   capabilities.

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, SSM, Mesh with common
   RTP session, and Transport Relay scenario, each individual sender has
   to adapt to the receiver with the least capable path, yielding
   suboptimal quality for the receivers behind the more capable paths.
   This is no longer necessary when Media Translators, Mixers, SFM or
   MCUs are involved, as each participant only needs to adapt to the
   slowest path within its own domain.  The Translator, Mixer, SFM, 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, SFM or
   Translator, and potentially media sender (multicast/layered encoding
   and sending the different representations).

4.1.4.  Aggregation of Media

   In the all to all media property mentioned above and provided by ASM,
   SSM, Mesh with common RTP session, and relay, 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,
   because multiple media streams need to share the same resources.  One
   solution to this problem is to provide for a Mixer, or MCU to
   aggregate the multiple streams into a single one, where the single



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   stream takes up less resources in terms of bit-rate.  This
   aggregation can be performed according to different methods.  Mixing
   or selection are two common methods.  Selection is almost always
   possible and easy to implement.  Mixing requires resources in the
   mixer, and may be relatively easy and not impairing the quality to
   badly (audio) or quite difficult (video tiling, which is not only
   computationally complex but also reduces the pixel count per stream,
   with corresponding less in perceptual quality).

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).
   In topologies that provide an full all to all functionality, i.e.
   ASM, Mesh with common RTP session, Relay a compliant RTP
   implementation offers the functionality directly as specified in RTP.
   In topologies that do not offer all-to-all communication, it is
   necessary that RTCP is handled correctly in domain bridging function.
   RTP includes explicit specification text for Translators and Mixers,
   and for SFMs the required functionality can be derived from that
   text.  However, the MCU described in Section 3.8 cannot offer the
   full functionality for session participant identification through RTP
   means.  The topologies that create independent RTP sessions per
   endpoint or pair of endpoints, like Back to Back RTP session, MESH
   with independent RTP sessions, and the RTCP terminating MCU RTCP
   terminating MCU (Section 3.9) do not support RTP based identification
   of session participants.  In all those cases, other non-RTP based
   mechanisms need to be implemented if such knowledge is required or
   desirable.

4.1.6.  Loop Detection

   In complex topologies with multiple interconnected domains, it is
   possible to unintentionally form media loops.  RTP and RTCP support
   detecting such loops, as long as the SSRC and CSRC identities are
   maintained and correctly set in forwarded packets.  Loop detection
   will work in ASM, SSM, Mesh with joint RTP session, and Relay.  It is
   likely that loop detection works for the video switching MCU
   Section 3.8, at least as long as it forwards the RTCP between the
   participants.  However, the Back to Back RTP sessions, Mesh with
   independent RTP sessions, SFM, will definitely break the loop
   detection mechanism.







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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-ASM (ASM), Topo-SSM (SSM), Topo-Trns-
   Translator (TT), Topo-Media-Translator (including Transport
   Translator) (MT), Topo-Mesh with joint session (MJS), Topo-Mesh with
   individual sessions (MIS), Topo-Mixer (Mix), Topo-Asymmetric (ASY),
   Topo-Video-switch-MCU (VSM), and Topo-RTCP-terminating-MCU (RTM),
   Selective Forwarding Middlebox (SFM).  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  ASM SSM  TT MT MJS MIS Mix ASY VSM RTM SFM
   ---------------------------------------------------------------------
   All to All media      N    Y  (Y)  Y  Y   Y  (Y) (Y) (Y) (Y) (Y) (Y)
   Interoperability      N/A  N   N   Y  Y   Y   Y   Y   Y   N   Y   Y
   Per Domain Adaptation N/A  N   N   N  Y   N   Y   Y   Y   N   Y   Y
   Aggregation of media  N    N   N   N  N   N   N   Y  (Y)  Y   Y   N
   Full Session View     Y    Y   Y   Y  Y   Y   N   Y   Y  (Y)  N   Y
   Loop Detection        Y    Y   Y   Y  Y   Y   N   Y   Y  (Y)  N   N

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

5.  Security Considerations

   The use of Mixers, SFMs and Translators has impact on security and
   the security functions used.  The primary issue is that both Mixers,
   SFMs and Translators modify packets, thus preventing the use of
   integrity and source authentication, unless they are trusted devices
   that take part in the security context, e.g., the device can send
   Secure Realtime Transport Protocol (SRTP) and Secure Realtime
   Transport Control Protocol (SRTCP) [RFC3711] packets to session
   endpoints.  If encryption is employed, the media Translator, SFM 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, SFM 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, SFM 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



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

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

   Second, the number of crypto contexts (keys and security related
   state) needed (for example, in SRTP [RFC3711]) may vary between
   Mixers, SFMs and Translators.  A Mixer normally needs to represent
   only a single SSRCs 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



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   translates towards, in the opposite domain.  Considering Figure 11,
   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.

   The multicast based (ASM and SSM), Relay and Mesh with common RTP
   session are all topologies with multiple endpoints that requires
   knowledge about the different crypto contexts for the endpoints.
   These multi-party topologies have special requirements on the key-
   management as well as the security functions.  Specifically source-
   authentication in these environments has special requirements.

   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 [RFC7201] to get a
   better understanding of most of the available options.

6.  IANA Considerations

   This document makes no request of IANA.

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

7.  Acknowledgements

   The authors would like to thank Bo Burman, Umesh Chandra, Roni Even,
   Keith Lantz, Ladan Gharai, Geoff Hunt, Mark Baugher, and Alex
   Eleftheriadis 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.



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   [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-multi-stream-optimisation]
              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session:
              Grouping RTCP Reception Statistics and Other Feedback",
              draft-ietf-avtcore-rtp-multi-stream-optimisation-02 (work
              in progress), February 2014.

   [RFC1112]  Deering, S., "Host extensions for IP multicasting", STD 5,
              RFC 1112, August 1989.

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

   [RFC3569]  Bhattacharyya, S., "An Overview of Source-Specific
              Multicast (SSM)", RFC 3569, July 2003.

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

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

   [RFC5760]  Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
              Protocol (RTCP) Extensions for Single-Source Multicast
              Sessions with Unicast Feedback", RFC 5760, February 2010.

   [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.





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   [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.

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, April 2014.

Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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


   Stephan Wenger
   Vidyo
   433 Hackensack Ave
   Hackensack, NJ  07601
   USA

   Email: stewe@stewe.org





















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