Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Obsoletes: 5117 (if approved)                                  S. Wenger
Intended status: Informational                                     Vidyo
Expires: January 3, 2016                                    July 2, 2015

                             RTP Topologies


   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 replaces RFC

Status of This Memo

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   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on January 3, 2016.

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   document authors.  All rights reserved.

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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 . . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.1.  Glossary  . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.2.  Definitions related to RTP grouping taxonomy  . . . . . .   4
   3.  Topologies  . . . . . . . . . . . . . . . . . . . . . . . . .   5
     3.1.  Point to Point  . . . . . . . . . . . . . . . . . . . . .   5
     3.2.  Point to Point via Middlebox  . . . . . . . . . . . . . .   6
       3.2.1.  Translators . . . . . . . . . . . . . . . . . . . . .   6
       3.2.2.  Back to Back RTP sessions . . . . . . . . . . . . . .  10
     3.3.  Point to Multipoint Using Multicast . . . . . . . . . . .  11
       3.3.1.  Any Source Multicast (ASM)  . . . . . . . . . . . . .  11
       3.3.2.  Source Specific Multicast (SSM) . . . . . . . . . . .  13
       3.3.3.  SSM with Local Unicast Resources  . . . . . . . . . .  15
     3.4.  Point to Multipoint Using Mesh  . . . . . . . . . . . . .  16
     3.5.  Point to Multipoint Using the RFC 3550 Translator . . . .  19
       3.5.1.  Relay - Transport Translator  . . . . . . . . . . . .  19
       3.5.2.  Media Translator  . . . . . . . . . . . . . . . . . .  20
     3.6.  Point to Multipoint Using the RFC 3550 Mixer Model  . . .  21
       3.6.1.  Media Mixing Mixer  . . . . . . . . . . . . . . . . .  23
       3.6.2.  Media Switching . . . . . . . . . . . . . . . . . . .  26
     3.7.  Selective Forwarding Middlebox  . . . . . . . . . . . . .  28
     3.8.  Point to Multipoint Using Video Switching MCUs  . . . . .  32
     3.9.  Point to Multipoint Using RTCP-Terminating MCU  . . . . .  33
     3.10. Split Component Terminal  . . . . . . . . . . . . . . . .  34
     3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . .  37
     3.12. Combining Topologies  . . . . . . . . . . . . . . . . . .  37
   4.  Topology Properties . . . . . . . . . . . . . . . . . . . . .  38
     4.1.  All to All Media Transmission . . . . . . . . . . . . . .  38
     4.2.  Transport or Media Interoperability . . . . . . . . . . .  39
     4.3.  Per Domain Bit-Rate Adaptation  . . . . . . . . . . . . .  39
     4.4.  Aggregation of Media  . . . . . . . . . . . . . . . . . .  40
     4.5.  View of All Session Participants  . . . . . . . . . . . .  40
     4.6.  Loop Detection  . . . . . . . . . . . . . . . . . . . . .  41
     4.7.  Consistency between header extensions and RTCP  . . . . .  41
   5.  Comparison of Topologies  . . . . . . . . . . . . . . . . . .  41
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  42
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  44
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  44
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  44
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  44
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  45
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  46

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1.  Introduction

   Real-time Transport Protocol (RTP) [RFC3550] topologies describe
   methods for interconnecting RTP entities and their processing
   behavior for 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 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

   This document also codifies new multipoint architectures that have
   recently been introduced and which were not anticipated in RFC 5117,
   thus this document replaces [RFC5117].  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 new topologies are Point to Point via Middlebox (Section 3.2),
   Source Specific Multicast (Section 3.3.2), SSM with Local Unicast
   Resources (Section 3.3.3), Point to Multipoint Using Mesh
   (Section 3.4), Selective Forwarding Middlebox (Section 3.7), and
   Split Component Terminal (Section 3.10).  The Point to Multipoint
   Using the RFC 3550 Mixer Model (Section 3.6) has been significantly
   expanded to cover two different versions namely Media Mixing Mixer
   (Section 3.6.1), and Media Switching (Section 3.6.2).

   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

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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

   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

2.2.  Definitions related to RTP grouping taxonomy

   [Note to RFC editor: The following definitions have been taken from
   draft-ietf-avtext-rtp-grouping-taxonomy-02 (taxonomy draft
   henceforth).  It is avtcore working group agreement to not delay the
   publication of the topologies-update document through a dependency to
   the taxonomy draft.  If, however, the taxonomy draft and this draft
   are in your work queue at the same time and there would be no
   significant additional delay (through your schedule, normative
   reference citations, or similar) in publishing both documents roughly
   in parallel, it would be preferable to replace the definition
   language with something like "as in [RFC YYYY]" where YYYY would be
   the RFC number of the published taxonomy draft.]

   The following definitions have been taken from draft-ietf-avtext-rtp-
   grouping-taxonomy-02, and are used in capitalized form throughout the

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   Communication Session:  A Communication Session is an association
      among group of participants communicating with each other via a
      set of Multimedia Sessions.

   Endpoint:  A single addressable entity sending or receiving RTP
      packets.  It may be decomposed into several functional blocks, but
      as long as it behaves as a single RTP stack entity it is
      classified as a single "Endpoint".

   Media Source:  A Media Source is the logical source of a reference
      clock synchronized, time progressing, digital media stream, called
      a Source Stream.

   Multimedia Session:   A multimedia session is an association among a
      group of participants engaged in the communication via one or more
      RTP Sessions.

3.  Topologies

   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 terminals, 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

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"

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 RTP
   stream that the Translator receives, it generates an individual RTP
   stream in the other domain.  A translator keeps the SSRC for an RTP
   stream across the translation, whereas a Mixer can select a single
   RTP stream from multiple received RTP streams (in cases like audio/
   video switching), or send out an RTP stream composed of multiple
   mixed media received in multiple RTP streams (in cases like audio
   mixing or video tiling), 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 of a middlebox is
   enabling interoperability, between otherwise non-interoperable
   endpoints, such as transcoding to a codec the receiver supports,
   which can be done by a media translator.

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   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 endpoints (as the RTP session participants) to
   run SSRC collision detection, and the SSRC is thus a field the
   Translator cannot change.  Any SDES information associated with a
   SSRC or CSRC also needs to be forwarded between the domains for any
   SSRC/CSRC used in the different domains.

   A Translator commonly does not use an SSRC of its own, and is not
   visible as an active participant in the RTP 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 sending endpoint, by sending feedback
   messages.  While such feedback could use the SSRC of the target for
   the translator (the receiving endpoint), 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
   may have issues with multiple SSRCs reporting on the RTP streams sent
   by that endpoint, as this use case may be viewed as excotic by

   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.  Transport Relay/Anchoring

   Shortcut name: Topo-PtP-Relay

   There exist a number of different types of middleboxes that might be
   inserted between two 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 RTP stream
   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

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

                 Figure 2: Point to Point with Translator

   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 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
   cannot support this will also not work correctly when endpoints are
   subject to NAT.

   The transport relay implementations also have to take into account
   security considerations.  In particular, source address filtering of
   incoming packets is usually important in relays, to prevent attackers
   to inject traffic into a session, which one peer may, in the absence
   fo adequate security in the relay, think it comes from the other
   peer.  Transport Translator

   Shortcut name: Topo-Trn-Translator

   Transport Translators (Topo-Trn-Translator) do not modify the RTP
   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).

   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.

   Of the transport Translators, this memo is primarily interested in
   those that use RTP on both sides, and this is assumed henceforth.

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

   Shortcut name: Topo-Media-Translator

   Media Translators (Topo-Media-Translator) modify the media inside the
   RTP stream.  This process is commonly known as transcoding.  The
   modification of the media can be as small as removing parts of the
   stream, and it can go all the way to a full decoding and re-encoding
   (down to the sample level or equivalent) utilizing a different media
   codec.  Media Translators are commonly used to connect endpoints
   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
   and the transport aspects of the RTP stream carrying the media
   between two transport domains.

   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 endpoint B's RTCP Receiver
   Report before forwarding them to endpoint A.  The rewriting is needed
   as the RTP stream received by B is not the same RTP stream as the
   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 requirements exists for most other
   fields in the RTCP Receiver Reports.

   A media Translator may in some cases act on behalf of the "real"
   source (the endpoint originally sending the media to the Translator)
   and respond to RTCP feedback messages.  This may occur, for example,
   when a receiving endpoint requests a bandwidth reduction, and the
   media Translator has not detected any congestion or other reasons for
   bandwidth reduction between the sending endpoint and itself.  In that
   case, it is sensible that the media Translator reacts to codec
   control messages itself, for example, by transcoding to a lower media

   A variant of translator behaviour worth pointing out is the one
   depicted in Figure 3 of an endpoint A sending a RTP stream containing
   media (only) to B.  On the path there is a device T that on A's

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   behalf manipulates the RTP streams.  One common example is that T
   adds a second RTP stream containing Forward Error Correction (FEC)
   information in order to protect A's (non FEC-protected) RTP stream.
   In this case, T needs to semantically bind the new FEC RTP stream to
   A's media-carrying RTP stream, for example by using the same CNAME as

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

                   Figure 3: Media Translator adding FEC

   there may also be cases where information is added into the original
   RTP stream, while leaving most or all of the original RTP packets
   intact (with the exception of certain RTP header fields, such as the
   sequence number).  One example is the injection of meta-data into the
   RTP stream, carried in their own RTP packets.

   Similarly, a Media Translator can sometimes remove information from
   the RTP stream, while otherwise leaving the remaining RTP packets
   unchanged (again with the exception of certain RTP header fields).

   Either type of functionality where T manipulates the RTP stream, or
   adds an accompanying RTP stream, on behalf of A is also covered under
   the media translator definition.

3.2.2.  Back to Back RTP sessions

   Shortcut name: Topo-Back-To-Back

   There exist middleboxes that interconnect two endpoints A and B
   through themselves (MB), 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: Back-to-back RTP sessions through Middlebox

   The middlebox acts as an application-level gateway and bridges the
   two RTP sessions.  This bridging can be as basic as forwarding the

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   RTP payloads between the sessions, or more complex including media
   transcoding.  The difference of this topology relative to 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
   respnsibility to maintain the correct relations.

   The signalling or other above-RTP level functionalities referencing
   RTP 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 on each of the
   two segments, or by bridging statistics from the receiving endpoint
   through the middlebox to the sending endpoint.  From an
   implementation point, however, the latter requires dealing with a
   number of inconsistencies.  First, packet loss must be detected for
   an RTP stream sent from A to the middlebox, and that loss must be
   reported through a skipped sequence number in the RTP stream from the
   middlebox to B.  This coupling and the resulting inconsistencies are
   conceptually easier to handle when considering the two RTP streams 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.  Each of
   these models are discussed below in their respective sections.

3.3.1.  Any Source Multicast (ASM)

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

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                        +---+     /       \    +---+
                        | 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 multicast
   group participant reaches all the other multicast group participants,
   except for cases such as:

   o  packet loss, or

   o  when a multicast group 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 multicast group
      participant is subscribed only to a subset of these sessions.

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

   For feedback usage, it is useful to define a "small multicast group"
   as a group where the number of multicast group 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 multicast group
   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

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   endpoint that is a multicast group participant 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 endpoint, but
   also to derive timing restrictions for the sending of its own
   feedback messages, if any.

3.3.2.  Source Specific Multicast (SSM)

   Shortcut name: Topo-SSM

   In Any Source Multicast, any of the multicast group participants can
   send to all the other multicast group 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:

          +--------+       +-----+
          |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

       Figure 6: Point to Multipoint using Source Specific Multicast

   In the SSM topology (Figure 6) a number of RTP sending endpoints (RTP
   sources henceforth) (1 to M) are allowed to send media to the SSM
   group.  These sources send media to a dedicated distribution source,
   which forwards the RTP streams to the multicast group on behalf of
   the original RTP sources.  The RTP streams reach the receiving
   endpoints (Receivers henceforth) (R(1) to R(n)).  The Receivers' RTCP
   messages cannot be sent to the multicast group, as the SSM multicast

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   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 distributed and each instance can have responsibility for sub-
   groups of the receivers.  For summary reports, however, there
   typically must be a single feedback target aggregating all the
   summaries to a common message to the whole receiver group.

   The RTP extension for SSM specifies how feedback (both reception
   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

   Aforementioned situation results in common behavior for RTP

   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 endpoints need to have the same RTP and payload type
   configuration.  Otherwise, endpoint A could, for example, be using
   payload type 97 to identify the video codec H.264, while endpoint B
   would identify it as MPEG-2, with unpredicatble but almost certainly
   not visually pleasing results.

   Security solutions for this type of group communications are also
   challenging.  First, the key-management and the security protocol
   must support group communication.  Source authentication becomes more
   difficult and requires specialized solutions.  For more discussion on
   this please review Options for Securing RTP Sessions [RFC7201].

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3.3.3.  SSM with Local Unicast Resources

   Shortcut name: Topo-SSM-RAMS

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

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

             Figure 7: SSM with Local Unicast Resources (RAMS)

   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

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   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 RTP stream.  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.  In
   many deployed application, the (still existing) PtP RTP session is
   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 involving three or more endpoints over multiple
   unicast transport flows, like the joint three endpoint session
   depicted above.  In this case, A needs to send its RTP streams and
   RTCP packets to both B and C over their respective transport flows.
   As long as all endpoints do the same, everyone will have a joint view
   of the RTP session.

   This topology 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 (in which case the sending port, IP
   address, or SSRC can be used to demultiplex), or it might have
   separate local reception ports for each of the endpoints.

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         |+---+                 |
         ||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

   Figure 9 depicts endpoints A's view of using a common RTP session
   when establishing the mesh as shown in Figure 8.  There is only one
   RTP session (RTP1) but two transport flows (UDP1 and UDP2).  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 stream may
   be sent from the transmitting endpoint, however it also supports
   local adaptation taking place in one or more of the RTP 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 10 by considering both how the media is
   handled and the RTP sessions that are set-up in Figure 10.  A's
   microphone is captured and the audio is fed into two different
   encoder instances, each with a different independent RTP session,
   i.e. RTP1 and RTP2 respectively.  The SSRCs (AA1 and AA2) in each RTP
   session are 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

   From a topologies viewpoint, an important difference exists in the
   behavior around RTCP.  First, when a single RTP session spans all
   three endpoints A, B, and C, and their connecting RTP streams, a
   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

   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)
   endpoints B and D to take part in an any source multicast session
   involving endpoints A and C, 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 endpoints 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 endpoints.  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 implementing
   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 endpoints to send SSRCs other than its
   own (that are, for example, 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 it to have access to the security
   contexts used to protect the RTP/RTCP traffic between the

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 a given endpoint or
   a defined sub-set of endpoints can participate in the session.

   If endpoint B in Figure 11 were behind a limited network path, the
   Translator may perform media transcoding to allow the traffic
   received from the other endpoints to reach B without overloading the

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   path.  This transcoding can help the other endpoints 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
   (and vice versa).

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 RTP streams
   received from other endpoints in the same RTP session.  Using the
   media data carried in the received RTP streams, a Mixer generates
   derived RTP streams that are sent to the receiving endpoints.

   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 Communication Session.

   The Mixer creates the Media Source and the source RTP stream just
   like an endpoint, as it mixes the content (often in the uncompressed
   domain) and then encodes and packetizes it for transmission to a
   receiving endpoint.  The CSRC Count (CC) and CSRC fields in the RTP
   header can be used to indicate the contributors to the newly
   generated RTP stream.  The SSRCs of the to-be-mixed streams on the
   Mixer input appear as the CSRCs at the Mixer output.  That output
   stream uses a unique SSRC that identifies the Mixer's stream.  The
   CSRC should be forwarded between the different endpoints to allow for
   loop detection and identification of sources that are part of the
   Communication Session.  Note that Section 7.1 of RFC 3550 requires
   the SSRC space to be shared between domains for these reasons.  This

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   also implies that any SDES information normally needs to be forwarded
   across the mixer.

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

   The Mixer depicted in Figure 13 is involved in three domains that
   need to be separated: the any source multicast network (including
   endpoints A and C), endpoint B, and endpoint D.  Assuming all four
   endpoints in the conference are interested in receiving content from
   all other endpoints, the Mixer produces different mixed RTP streams
   for B and D, as the one to B may contain content received from D, and
   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

   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
   endpoints in the other domain(s).  In other cases, a message is
   handled by the Mixer locally and therefore not forwarded to any other

   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 Mixer

   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 RTP streams from several endpoints 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 RTP
   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 audio signals; this needs minimal signalling to be
   successfully set up.  From a signal processing viewpoint, audio
   mixing is relatively straightforward and commonly possible for a
   reasonable number of endpoints.  Assume, for example, that one wants
   to mix N streams from N different endpoints.  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 endpoints, 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, noise suppression, and
   similar signal processing steps are also required or desirable before
   the actual mixing process commences.

   In video, the term "mixing" has a different interpretation than
   audio.  It is commonly used to refer to the process of spatially
   combining contributed video streams, which is also known as "tiling".
   The reconstructed, appropriately scaled down videos can be spatially
   arranged in a set of tiles, each tile containing the video from an
   endpoint (typically showing a human participant).  Tiles can be of
   different sizes, so that, for example, a particularly important

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   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 produced or be a feedback from a
   mixer-received and reconstructed video image.  Such remote loopback
   allows for confidence monitoring, i.e., it enables the participant to
   see himself/herself in the same quality 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 to the receiving endpoints.  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 receiving
   endpoint (a feature commonly referred to as personalized layout).

   One problem with media mixing is that it consumes both large amounts
   of media processing resources (for the decoding and 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, 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
   endpoints 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 (that, in some protocol environments, actually didn't need
   to be aware of the multipoint nature of the conference) 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 <----+ | +---+  |     | |
           | | +-------|          |(AA1+CA1)|\| +---+  |     | |
           | +---------|          |---------+ +-|ENC|<-| A+C | |
           +-----------+          |-----------+ +---+  |     | |
                                  |                    |  M  | |
           +-C---------+          |                    |  I  | |
           | +-RTP3----|          |-RTP3------+        |  X  | |
           | | +-Audio-|          |-Audio---+ | +---+  |  E  | |
           | | |    CA1|--------->|---------+-+-|DEC|->|  R  | |
           | | |       |<---------|MA3 <----+ | +---+  |     | |
           | | +-------|          |(AA1+BA1)|\| +---+  |     | |
           | +---------|          |---------+ +-|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 endpoint, e.g., MA1 to Peer A, where the associated stream
   is the media mix of the other endpoints.  As each peer, in this
   example, receives a different version of a mix from the mixer, there
   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 in 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 RTP streams that the mixer can produce.
   These RTP streams are created by selecting media from one of the RTP
   streams received by the mixer and forwarded to the peer using the
   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 in their

   To achieve a coherent RTP stream from the mixer's SSRC, the mixer
   needs to rewrite the incoming RTP packet's header.  First the SSRC
   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-to-mixer paths have not arrived on
   the same numbering for a given configuration.  This also requires
   that the different endpoints support a common set of codecs,

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   otherwise media transcoding for codec compatibility would still be

   We now consider the operation of a media switching mixer that
   supports a video conference with six participating endpoints (A-F)
   where the two most recent speakers in the conference are shown to
   each receiving endpoint.  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---+ |        |     | |
         | | |    FV1|------------>|---------+-+------->|     | |
         | | |       |<------------|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
   streams it receives from the sending endpoints.  In cases the mixer

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   receives simulcast transmissions or a scalable encoding of the media
   source, the mixer has more degrees of freedom to select streams or
   sub-sets of stream to forward to a receiving endpoint, both based on
   transport or endpoint restrictions as well as application logic.

   To ensure that a media receiver in an endpoint can correctly decode
   the media in the RTP stream after a switch, a codec that uses
   temporal prediction needs to start its decoding from independent
   refresh points, or points in the bitstream offering similar
   functionality (like "dirty refresh points").  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 endpoint conference depicted above in (Figure 17) one can
   see that endpoint A is aware of five incoming SSRCs, BV1-FV1.  If
   this middlebox intends to have a similar behavior as in Section 3.6.2
   where the mixer provides the endpoints with the two latest speaking
   endpoints, 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 endpoints, each RTP
   stream requires rewriting of certain RTP header fields when being
   projected from one session into another.  In particular, the sequence

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   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.

   The RTP sessions can be considered independent, resulting in that the
   SSRC numbers used can also be handled independently.  This simplifies
   the SSRC collision detection and avoidance, but require tools such as
   remapping tables between the RTP sessions.  Using independent RTP
   sessions are not required, as the switching behavior is possible to
   perform also with a common SSRC space.  However, in this case
   collision detection and handling becomes a different problem.  It is
   up to the implementation to use a single common SSRC space or
   separate ones.

   Using separate SSRC spaces has some implications.  For example, the
   RTP stream that is being sent by endpoint B to the middlebox (BV1)
   may use an SSRC value of 12345678.  When that RTP stream is sent to
   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 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 for receiving any RTCP codec
   control requests coming from an endpoint, and decide if it can act on
   the request locally or needs to translate the request into the RTP
   session/transport leg that contains the media source.  Both endpoints
   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

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   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
   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
   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 topology has the property that all SSRCs present in
   the session are visible to an endpoint.  It also has mixer aspects,
   as the streams it provides are not basically translated version, but
   instead they have conceptual property assigned to them and can be
   both turned on/off as well as being fully or partially delivered.
   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
   information on what is included in a particular RTP stream that
   represent a particular concept.  Selective forwarding gets the source
   information through the SSRC, and instead uses other mechanisms to
   indicate the streams intended usage, if needed.

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3.8.  Point to Multipoint Using Video Switching MCUs

   Shortcut name: Topo-Video-switch-MCU

                   +---+      +------------+      +---+
                   | 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

   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 endpoints.  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 RTP stream results in most participants seeing only a
   subset of the sent RTP streams at any given time, often a single RTP
   stream per conference.  Therefore, RTCP Receiver Reports only report
   on these RTP streams.  Consequently, the endpoints emitting RTP
   streams that are not currently forwarded receive a view of the
   session that indicates their RTP streams disappear somewhere en

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   route.  This makes the use of RTCP for congestion control, or any
   type of quality reporting, very problematic.

   To avoid the aforementioned issues, the MCU needs to implement two
   features.  First, it needs to act as a Mixer (see Section 3.6) and
   forward the selected RTP 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 endpoint 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 endpoints, or

   b.  the mixed aggregate of what the MCU receives from the other PtP
       paths, which are part of the same Communication 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 RTP stream of the
   media being sent RTP stream the MCU sends.  In other words, the
   outgoing RTP streams 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 RTP session for each endpoint.

   In case (b), the MCU is the Media Source and generates the Source RTP
   Stream as it mixes the received content and then encodes and
   packetizes it for transmission to an endpoint.  According to RTP

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   [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 endpoints
   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 original sending endpoints 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

   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 many/most deployed MCUs (and video conferencing 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 Terminal

   Shortcut name: Topo-Split-Terminal

   In some applications, for example in some telepresence systems,
   terminals may be not integrated into a single functional unit, but
   composed of more than one subunits.  For example, a telepresence room
   terminal employing multiple cameras and monitors may consist of
   multiple video conferencing subunits, each capable of handling a
   single camera and monitor.  Another example would be a video
   conferencing terminal in which audio is handled by one subunit, and
   video by another.  Each of these subunits uses its own physical
   network interface (for example: Ethernet jack) and network address.

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   The various (media processing) subunits need (logically and
   physically) to be interconnected by control functionality, but their
   media plane functionality may be split.  This type of terminals is
   referred to as split component terminals.  Historically, the earliest
   split component terminals were perhaps the independent audio and
   video conference software tools used over the MBONE in the late

   An example for such a split component terminal is depicted in
   Figure 20.  Within split component terminal A, at least audio and
   video subunits are addressed by their own network addresses.  In some
   of these systems, the control stack subunit may also have its own
   network address.

   From an RTP viewpoint, each of the subunits terminates RTP, and acts
   as an endpoint in the sense that each subunit includes its own,
   independent RTP stack.  However, as the subunits are semantically
   part of the same terminal, it is appropriate that this semantic
   relationship is expressed in RTCP protocol elements, namely in the

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

                    Figure 20: Split Component Terminal

   It is further sensible that the subunits share a common clock from
   which RTP and RTCP clocks are derived, to facilitate synchronization
   and avoid clock drift.

   To indicate that audio and video Source Streams generated by
   different sub-units share a common clock, and can be synchronized,
   the RTP streams generated from those Source Streams need to include
   the same CNAME in their RTCP SDES packets.  The use of a common CNAME
   for RTP flows carried in different transport-layer flows is entirely
   normal for RTP and RTCP senders, and fully compliant RTP endpoints,
   middle-boxes, and other tools should have no problem with this.

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   However, outside of the split component terminal scenario (and
   perhaps a multi-homed endpoint scenario, which is not further
   discussed herein), the use of a common CNAME in RTP streams sent from
   separate endpoints (as opposed to a common CNAME for RTP streams sent
   on different transport layer flows between two endpoints) is rare.
   It has been reported that at least some third party tools like some
   network monitors do not handle endpoints that use of a common CNAME
   across multiple transport layer flows gracefully: they report an
   error condition that two separate endpoints are using the same CNAME.
   Depending on the sophistication of the support staff, such erroneous
   reports can lead to support issues.

   Aforementioned support issue can sometimes be avoided if each of the
   subunits of a split component terminal is configured to use a
   different CNAME, with the synchronization between the RTP streams
   being indicated by some non-RTP signaling channel rather than using a
   common CNAME sent in RTCP.  This complicates the signaling,
   especially in cases where there are multiple SSRCs in use with
   complex synchronization requirements, as is the same in many current
   telepresence systems.  Unless one uses RTCP terminating topologies
   such as Topo-RTCP-terminating-MCU, sessions involving more than one
   video subunit with a common CNAME are close to unavoidable.

   The different RTP streams comprising a split terminal system can form
   a single RTP session or they can form multiple RTP sessions,
   depending on the visibility of their SSRC values in RTCP reports.  If
   the receiver of the RTP streams sent by the split terminal sends
   reports relating to all of the RTP flows (i.e., to each SSRC) in each
   RTCP report then a single RTP session is formed.  Alternatively, if
   the receiver of the RTP streams sent by the split terminal does not
   send cross-reports in RTCP, then the audio and video form separate
   RTP sessions.

   For example, in the Figure 20, B will send RTCP reports to each of
   the sub-units of A.  If the RTCP packets that B sends to the audio
   sub-unit of A include reports on the reception quality of the video
   as well as the audio, and similarly if the RTCP packets that B sends
   to the video sub-unit of A include reports on the reception quality
   of the audio as well as video, then a single RTP session is formed.
   However, if the RTCP packets B sends to the audio sub-unit of A only
   report on the received audio, and the RTCP packet B sends to the
   video sub-unit of A only report on the received video, then there are
   two separate RTP sessions.

   Forming a single RTP session across the RTP streams sent by the
   different sub-units of a split terminal gives each sub-unit
   visibility into reception quality of RTP streams sent by the other

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   sub-units.  This information can help diagnose reception quality
   problems, but at the cost of increased RTCP bandwidth use.

   RTP streams sent by the sub-units of a split terminal need to use the
   same CNAME in their RTCP packets if they are to be synchronized,
   irrespective of whether a single RTP session is formed or not.

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 RTP 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 media 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 RTP streams would be missing.  To avoid A and C
   thinking that B and D aren't receiving A and C at all, the Mixer
   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

   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

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   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.  Topology Properties

   The topologies discussed in Section 3 have different properties.
   This section describes these properties.  Note that, even if a
   certain property is supported within a particular topology concept,
   the necessary functionality may be optional to implement.

4.1.  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.

   The topologies that comes closest to emulating Any Source IP
   Multicast, with all-to-all transmission capabilities, are the
   transport Translator function called "relay" in Section 3.5, as well
   as the Mesh with joint RTP sessions (Section 3.4).  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 endpoint considers the path to the least capable
   receiving endpoint.  Otherwise, the media transmissions may overload
   that path.  Therefore, a sending endpoint needs to monitor the path
   from itself to any of the receiving endpoints, to detect the
   currently least capable receiver, and adapt its sending rate
   accordingly.  As multiple endpoints may send simultaneously, the
   available resources may vary.  RTCP's Receiver Reports help
   performing this monitoring, at least on a medium time scale.

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   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 endpoint-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 endpoints (which may be behind slower access links).  The Mesh
   causes N-1 streams of transmitted packets to traverse the first hop
   link from the endpoint, in an N endpoint mesh.  How long the
   different paths are common, is highly situation dependent.

   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.

4.2.  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 endpoints 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 receiving endpoint's
   capability.  It requires an additional translator to change the media
   encoding if the encoded streams do not match the receiving endpoint's

4.3.  Per Domain Bit-Rate Adaptation

   Endpoints are often 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 sending endpoint has to
   adapt to the receiving endpoint behind the least capable path,
   yielding suboptimal quality for the endpoints behind the more capable
   paths.  This is no longer an issue when Media Translators, Mixers,
   SFM or MCUs are involved, as each endpoint only needs to adapt to the
   slowest path within its own domain.  The Translator, Mixer, SFM, or
   MCU topologies all require their respective outgoing RTP 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 endpoint in all the domains at the
   cost (complexity, delay, equipment) of the Mixer, SFM or Translator,

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   and potentially media sender (multicast/layered encoding and sending
   the different representations).

4.4.  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 endpoints with
   limited reception capabilities, this may result in a situation where
   even a minimal acceptable media quality cannot be accomplished,
   because multiple RTP streams need to share the same resources.  One
   solution to this problem is to provide for a Mixer, or MCU to
   aggregate the multiple RTP streams into a single one, where the
   single RTP 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 too
   badly (audio) or quite difficult (video tiling, which is not only
   computationally complex but also reduces the pixel count per stream,
   with corresponding loss in perceptual quality).

4.5.  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 a 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), with an exception of SFM, 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.  When it comes to SFM the SSRC name space is
   not necessarily joint, instead identification will require knowledge
   of SSRC/CSRC mappings that the SFM performed, see Section 3.7.

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4.6.  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
   endpoints.  However, the Back-to-Back RTP sessions, Mesh with
   independent RTP sessions, SFM, will definitely break the loop
   detection mechanism.

4.7.  Consistency between header extensions and RTCP

   Some RTP header extensions have relevance not only end-to-end, but
   also hop-to-hop, meaning at least some of the middleboxes in the path
   are aware of their potential presence through signaling, intercept
   and interpret such header extensions and potentially also rewrite or
   generate them.  Modern header extensions generally follow "A General
   Mechanism for RTP Header Extensions" [RFC5285], which allows for all
   of the above.  Examples for such header extensions include the mid
   (media ID) in [I-D.ietf-mmusic-sdp-bundle-negotiation].  At the time
   of writing there was also a proposal for how to include any SDES into
   an RTP header extension [I-D.westerlund-avtext-sdes-hdr-ext].

   When such header extensions are in use, any middlebox that
   understands it must ensure consistency between the extensions it sees
   and/or generates, and the RTCP it receives and generates.  For
   example, the mid of bundle is sent in an RTP header extension and
   also in an RTCP SDES message.  This apparent redundancy was
   introduced as unaware middleboxes may choose to discard RTP header
   extensions.  Obviously, inconsistency between the media ID sent in
   the RTP header extension and in the RTCP SDES message could lead to
   undesirable results, and, therefore, consistency is needed.
   Middleboxes unaware of the nature of a header extension, as specified
   in [RFC5285], are free to forward or discard header extensions.

5.  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

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   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.

6.  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 endpoints in
   the Communication Session.  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
   the media session were not protected at all, while giving the
   impression to the human 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

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   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

   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
   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

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   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 require
   shared 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.

7.  IANA Considerations

   This document makes no request of IANA.

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

8.  Acknowledgements

   The authors would like to thank Mark Baugher, Bo Burman, Ben
   Campbell, Umesh Chandra, Alex Eleftheriadis, Roni Even, Ladan Gharai,
   Geoff Hunt, Suresh Krishnan, Keith Lantz, Jonathan Lennox, Scarlet
   Liuyan, Suhas Nandakumar, Colin Perkins, and Dan Wing for their help
   in reviewing and improving this document.

9.  References

9.1.  Normative References

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [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

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9.2.  Informative References

              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-05 (work
              in progress), February 2015.

              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-22 (work in progress), June 2015.

              Westerlund, M., Even, R., and M. Zanaty, "RTP Header
              Extension for RTCP Source Description Items", draft-
              westerlund-avtext-sdes-hdr-ext-03 (work in progress),
              November 2014.

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

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

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

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

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

   [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.

   [RFC5117]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
              January 2008.

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   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

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

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC6285]  Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
              "Unicast-Based Rapid Acquisition of Multicast RTP
              Sessions", RFC 6285, June 2011.

   [RFC6465]  Ivov, E., Marocco, E., and J. Lennox, "A Real-time
              Transport Protocol (RTP) Header Extension for Mixer-to-
              Client Audio Level Indication", RFC 6465, December 2011.

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

Authors' Addresses

   Magnus Westerlund
   Farogatan 6
   SE-164 80 Kista

   Phone: +46 10 714 82 87

   Stephan Wenger
   433 Hackensack Ave
   Hackensack, NJ  07601


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