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rfc:rfc7667

Internet Engineering Task Force (IETF) M. Westerlund Request for Comments: 7667 Ericsson Obsoletes: 5117 S. Wenger Category: Informational Vidyo ISSN: 2070-1721 November 2015

                           RTP Topologies

Abstract

 This document discusses point-to-point and multi-endpoint topologies
 used in environments based on the Real-time Transport Protocol (RTP).
 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
 5117.

Status of This Memo

 This document is not an Internet Standards Track specification; it is
 published for informational purposes.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It represents the consensus of the IETF community.  It has
 received public review and has been approved for publication by the
 Internet Engineering Steering Group (IESG).  Not all documents
 approved by the IESG are a candidate for any level of Internet
 Standard; see Section 2 of RFC 5741.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 http://www.rfc-editor.org/info/rfc7667.

Westerlund & Wenger Informational [Page 1] RFC 7667 RTP Topologies November 2015

Copyright Notice

 Copyright (c) 2015 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (http://trustee.ietf.org/license-info) in effect on the date of
 publication of this document.  Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document.  Code Components extracted from this document must
 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.

Westerlund & Wenger Informational [Page 2] RFC 7667 RTP Topologies November 2015

Table of Contents

 1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
 2.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   5
   2.1.  Glossary  . . . . . . . . . . . . . . . . . . . . . . . .   5
   2.2.  Definitions Related to RTP Grouping Taxonomy  . . . . . .   5
 3.  Topologies  . . . . . . . . . . . . . . . . . . . . . . . . .   6
   3.1.  Point to Point  . . . . . . . . . . . . . . . . . . . . .   6
   3.2.  Point to Point via Middlebox  . . . . . . . . . . . . . .   7
     3.2.1.  Translators . . . . . . . . . . . . . . . . . . . . .   7
     3.2.2.  Back-to-Back RTP sessions . . . . . . . . . . . . . .  11
   3.3.  Point to Multipoint Using Multicast . . . . . . . . . . .  12
     3.3.1.  Any-Source Multicast (ASM)  . . . . . . . . . . . . .  12
     3.3.2.  Source-Specific Multicast (SSM) . . . . . . . . . . .  14
     3.3.3.  SSM with Local Unicast Resources  . . . . . . . . . .  15
   3.4.  Point to Multipoint Using Mesh  . . . . . . . . . . . . .  17
   3.5.  Point to Multipoint Using the RFC 3550 Translator . . . .  20
     3.5.1.  Relay - Transport Translator  . . . . . . . . . . . .  20
     3.5.2.  Media Translator  . . . . . . . . . . . . . . . . . .  21
   3.6.  Point to Multipoint Using the RFC 3550 Mixer Model  . . .  22
     3.6.1.  Media-Mixing Mixer  . . . . . . . . . . . . . . . . .  24
     3.6.2.  Media-Switching Mixer . . . . . . . . . . . . . . . .  27
   3.7.  Selective Forwarding Middlebox  . . . . . . . . . . . . .  29
   3.8.  Point to Multipoint Using Video-Switching MCUs  . . . . .  33
   3.9.  Point to Multipoint Using RTCP-Terminating MCU  . . . . .  34
   3.10. Split Component Terminal  . . . . . . . . . . . . . . . .  35
   3.11. Non-symmetric Mixer/Translators . . . . . . . . . . . . .  38
   3.12. Combining Topologies  . . . . . . . . . . . . . . . . . .  38
 4.  Topology Properties . . . . . . . . . . . . . . . . . . . . .  39
   4.1.  All-to-All Media Transmission . . . . . . . . . . . . . .  39
   4.2.  Transport or Media Interoperability . . . . . . . . . . .  40
   4.3.  Per-Domain Bitrate Adaptation . . . . . . . . . . . . . .  40
   4.4.  Aggregation of Media  . . . . . . . . . . . . . . . . . .  41
   4.5.  View of All Session Participants  . . . . . . . . . . . .  41
   4.6.  Loop Detection  . . . . . . . . . . . . . . . . . . . . .  42
   4.7.  Consistency between Header Extensions and RTCP  . . . . .  42
 5.  Comparison of Topologies  . . . . . . . . . . . . . . . . . .  42
 6.  Security Considerations . . . . . . . . . . . . . . . . . . .  43
 7.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  45
   7.1.  Normative References  . . . . . . . . . . . . . . . . . .  45
   7.2.  Informative References  . . . . . . . . . . . . . . . . .  45
 Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  48
 Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  48

Westerlund & Wenger Informational [Page 3] RFC 7667 RTP Topologies November 2015

1. Introduction

 Real-time Transport Protocol (RTP) [RFC3550] topologies describe
 methods for interconnecting RTP entities and their processing
 behavior for RTP and the RTP Control Protocol (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 signaling support.  They may also contain
 additional application-layer functionality.  This document focuses on
 the media transport aspects of the MCU that can be realized using
 RTP, as discussed below.  Further considered are the properties of
 mixers and translators, and how some types of deployed MCUs deviate
 from these properties.
 This document also codifies new multipoint architectures that have
 recently been introduced and that 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 Middleboxes
 (SFMs).  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 Mixer (Section 3.6.2).
 The document's attempt to clarify and explain sections of the RTP
 spec [RFC3550] is informal.  It is not intended to update or change
 what is normatively specified within RFC 3550.

Westerlund & Wenger Informational [Page 4] RFC 7667 RTP Topologies November 2015

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
 SFM:  Selective Forwarding Middlebox
 SSM:  Source-Specific Multicast
 SSRC:  Synchronization Source

2.2. Definitions Related to RTP Grouping Taxonomy

 The following definitions have been taken from [RFC7656].
 Communication Session:  A Communication Session is an association
    among two or more Participants communicating with each other via
    one or more 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 mentity, it is
    classified as a single "endpoint".
 Media Source:  A Media Source is the logical source of a time
    progressing digital media stream synchronized to a reference
    clock.  This stream is called a Source Stream.

Westerlund & Wenger Informational [Page 5] RFC 7667 RTP Topologies November 2015

 Multimedia Session:   A Multimedia Session is an association among a
    group of participants engaged in 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
 it 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 SFMs.  The section ends by
 discussing decomposited 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
 [RFC5104].

3.1. Point to Point

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

Westerlund & Wenger Informational [Page 6] RFC 7667 RTP Topologies November 2015

 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
 that use multiple SSRCs 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" [MULTI-STREAM-OPT].

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.
 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 Source Description (SDES) information
 associated with an 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

Westerlund & Wenger Informational [Page 7] RFC 7667 RTP Topologies November 2015

 configured to detect packet loss of important video packets, and it
 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 target RTCP reports to make them
 consistent.  It may be simpler to expose an additional SSRC in the
 session.  The only concern is that 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 exotic by implementers.
 In general, a translator implementation should consider which RTCP
 feedback messages or codec-control messages it needs to understand in
 relation to the functionality of the translator itself.  This is
 completely in line with the requirement to also translate RTCP
 messages between the domains.

3.2.1.1. Transport Relay/Anchoring

 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 and 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 Traversal Using Relays around NAT
 (TURN) [RFC5766] servers, Session Border Gateways, and Media
 Processing Nodes with media anchoring functionalities.
                   +---+        +---+         +---+
                   | 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 since the RTP and RTCP packets are routed
 between the endpoints in the RTP session, and thereby they 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

Westerlund & Wenger Informational [Page 8] RFC 7667 RTP Topologies November 2015

 requirements in the RTP implementation and its interactions with the
 signaling solution.  Both in signaling 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
 from injecting traffic into a session, which one peer may, in the
 absence of adequate security in the relay, think it comes from the
 other peer.

3.2.1.2. 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 as 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.
 The most basic Transport Translators that operate below the RTP level
 were already discussed in Section 3.2.1.1.

3.2.1.3. Media Translator

 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

Westerlund & Wenger Informational [Page 9] RFC 7667 RTP Topologies November 2015

 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 was substantially behind in reception, while it most
 likely would not be.  Therefore, the translator must translate that
 number to a corresponding sequence number for the stream the
 translator received.  Similar requirements exist 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
 rate.
 A variant of translator behavior worth pointing out is the one
 depicted in Figure 3 of an endpoint A sending an RTP stream
 containing media (only) to B.  On the path, there is a device T that
 manipulates the RTP streams on A's behalf.  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.

Westerlund & Wenger Informational [Page 10] RFC 7667 RTP Topologies November 2015

               +------+        +------+         +------+
               |      |        |      |         |      |
               |  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 metadata 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.  Instead, they establish 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
 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
 responsibility to maintain the correct relations.

Westerlund & Wenger Informational [Page 11] RFC 7667 RTP Topologies November 2015

 The signaling 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)
                                 +-----+
                      +---+     /       \    +---+
                      | A |----/         \---| B |
                      +---+   /   Multi-  \  +---+
                             +    cast     +
                      +---+   \  Network  /  +---+
                      | C |----\         /---| D |
                      +---+     \       /    +---+
                                 +-----+
             Figure 5: Point to Multipoint Using Multicast

Westerlund & Wenger Informational [Page 12] RFC 7667 RTP Topologies November 2015

 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 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
 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.

Westerlund & Wenger Informational [Page 13] RFC 7667 RTP Topologies November 2015

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
        co-located.
     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 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

Westerlund & Wenger Informational [Page 14] RFC 7667 RTP Topologies November 2015

 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
 subgroups 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, need to be accounted
 for.
 The aforementioned situation results in common behavior for RTP
 multicast:
 1.  Multicast applications often use a group of RTP sessions, not
     one.  Each endpoint needs to be a member of most or all of these
     RTP sessions in order to perform well.
 2.  Within each RTP session, the number of media sinks is likely to
     be much larger than the number of RTP sources.
 3.  Multicast applications need signaling functions to identify the
     relationships between RTP sessions.
 4.  Multicast applications need signaling functions to identify the
     relationships between SSRCs in different RTP sessions.
 All multicast configurations share a signaling 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 unpredictable but almost certainly
 not visually pleasing results.
 Security solutions for this type of group communication 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].

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.

Westerlund & Wenger Informational [Page 15] RFC 7667 RTP Topologies November 2015

  1. ———- ————–

| |————————————>| |

 |           |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->|              |
 |           |                                     |              |
 | 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)| |<.=.=.=.=>|              |
                       |  ------------  |          |              |
                       |                |          |              |
                        ----------------            --------------
  1. ——> 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 the decoding of these burst-
 delivered media packets, the correct media timing is established,
 i.e., media packets are received within adequate buffer intervals for
 this application.  This is accomplished by first establishing a
 unicast PtP RTP session between the Burst/Retransmission Source (BRS)
 (Figure 7) and the RTP Receiver.  The unicast session is used to
 transmit cached packets from the multicast group at higher then
 normal speed in order to synchronize the receiver to the ongoing
 multicast 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

Westerlund & Wenger Informational [Page 16] RFC 7667 RTP Topologies November 2015

 many deployed applications, 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 Multipoint 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.

Westerlund & Wenger Informational [Page 17] RFC 7667 RTP Topologies November 2015

       +-A--------------------+
       |+---+                 |
       ||CAM|                 |                 +-B-----------+
       |+---+     +-UDP1------|                 |-UDP1------+ |
       |  |       | +-RTP1----|                 |-RTP1----+ | |
       |  V       | | +-Video-|                 |-Video-+ | | |
       |+----+    | | |       |<----------------|BV1    | | | |
       ||ENC |----+-+-+--->AV1|---------------->|       | | | |
       |+----+    | | +-------|                 |-------+ | | |
       |  |       | +---------|                 |---------+ | |
       |  |       +-----------|                 |-----------+ |
       |  |                   |                 +-------------+
       |  |                   |
       |  |                   |                 +-C-----------+
       |  |       +-UDP2------|                 |-UDP2------+ |
       |  |       | +-RTP1----|                 |-RTP1----+ | |
       |  |       | | +-Video-|                 |-Video-+ | | |
       |  +-------+-+-+--->AV1|---------------->|       | | | |
       |          | | |       |<----------------|CV1    | | | |
       |          | | +-------|                 |-------+ | | |
       |          | +---------|                 |---------+ | |
       |          +-----------|                 |-----------+ |
       +----------------------+                 +-------------+
        Figure 9: A Multi-Unicast Mesh with a Joint RTP Session
 Figure 9 depicts endpoint 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, a 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.

Westerlund & Wenger Informational [Page 18] RFC 7667 RTP Topologies November 2015

        +-A----------------------+              +-B-----------+
        |+---+                   |              |             |
        ||MIC|       +-UDP1------|              |-UDP1------+ |
        |+---+       | +-RTP1----|              |-RTP1----+ | |
        | |  +----+  | | +-Audio-|              |-Audio-+ | | |
        | +->|ENC1|--+-+-+--->AA1|------------->|       | | | |
        | |  +----+  | | |       |<-------------|BA1    | | | |
        | |          | | +-------|              |-------+ | | |
        | |          | +---------|              |---------+ | |
        | |          +-----------|              |-----------+ |
        | |          ------------|              |-------------|
        | |                      |              |-------------+
        | |                      |
        | |                      |              +-C-----------+
        | |                      |              |             |
        | |          +-UDP2------|              |-UDP2------+ |
        | |          | +-RTP2----|              |-RTP2----+ | |
        | |  +----+  | | +-Audio-|              |-Audio-+ | | |
        | +->|ENC2|--+-+-+--->AA2|------------->|       | | | |
        |    +----+  | | |       |<-------------|CA1    | | | |
        |            | | +-------|              |-------+ | | |
        |            | +---------|              |---------+ | |
        |            +-----------|              |-----------+ |
        +------------------------+              +-------------+
    Figure 10: A Multi-Unicast Mesh with an Independent RTP Session
 Let's 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 how the RTP sessions 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 bitrate
 produced by the encoders can also be tuned differently to address any
 congestion control requirements differing for the paths A to B
 compared to A to C.
 From a topologies viewpoint, an important difference exists in the
 behavior around RTCP.  First, when a single RTP session spans all
 three endpoints 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

Westerlund & Wenger Informational [Page 19] RFC 7667 RTP Topologies November 2015

 configuration in Figure 8, endpoint A has no awareness of the
 conditions occurring in the session between endpoints B and C
 (whereas if a single RTP session were used, it would have such
 awareness).
 Loop detection is also affected.  With independent RTP sessions, the
 SSRC/CSRC cannot be used to determine when an endpoint receives its
 own media stream, or a mixed media stream including its own media
 stream (a condition known as a loop).  The identification of loops
 and, in most cases, their avoidance, has to be achieved by other
 means, for example, through signaling or the use of an RTP external
 namespace 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 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 |
           +---+   /           \   |            |      +---+
                  +  Multicast  +->| 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
 both B and D with a complete view of the session.

Westerlund & Wenger Informational [Page 20] RFC 7667 RTP Topologies November 2015

                 +---+      +------------+      +---+
                 | 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 signaling 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 from using the
 listening ports on the translator to inject traffic.  The translator
 can, however, go one step further, especially if explicit SSRC
 signaling is used, to prevent endpoints from sending 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 a cryptographic level allows
 anyone to impersonate another in the same RTP session.
 A translator that doesn't change the RTP/RTCP packet content can be
 operated without requiring it to have access to the security contexts
 used to protect the RTP/RTCP traffic between the participants.

3.5.2. Media Translator

 In the context of multipoint communications, a Media Translator is
 not providing new mechanisms to establish a multipoint session.  It
 is more of an enabler, or facilitator, that ensures a given endpoint
 or a defined subset 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
 path.  This transcoding can help the other endpoints in the multicast

Westerlund & Wenger Informational [Page 21] RFC 7667 RTP Topologies November 2015

 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

Westerlund & Wenger Informational [Page 22] RFC 7667 RTP Topologies November 2015

 the SSRC space to be shared between domains for these reasons.  This
 also implies that any SDES information normally needs to be forwarded
 across the mixer.
 The mixer is responsible for generating RTCP packets in accordance
 with its role.  It is 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
 domains.
 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
 different.
 A mixer is responsible for receiving RTCP feedback messages and
 handling them appropriately.  The definition of "appropriate" depends
 on the message itself and the context.  In some cases, the reception
 of a codec-control message by the mixer may result in the generation
 and transmission of RTCP feedback messages by the mixer to the
 endpoints in the other domain(s).  In other cases, a message is
 handled by the mixer locally and therefore not forwarded to any other
 domain.
 When replacing the multicast network in Figure 13 (to the left of the
 mixer) with individual unicast paths as depicted in Figure 14, the
 mixer model is very similar to the one discussed in Section 3.9
 below.  Please see the discussion in Section 3.9 about the
 differences between these two models.

Westerlund & Wenger Informational [Page 23] RFC 7667 RTP Topologies November 2015

                 +---+      +------------+      +---+
                 | 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-Mixing 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 signaling
 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
 endpoint 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, with 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

Westerlund & Wenger Informational [Page 24] RFC 7667 RTP Topologies November 2015

 participant, or the loudest speaker, is being shown in a larger tile
 than other participants.  A self-view picture can be included in the
 tiling, which can be either locally produced or 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 arrangements 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 the 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-synchronized 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.

Westerlund & Wenger Informational [Page 25] RFC 7667 RTP Topologies November 2015

         +-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 Sources
 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 RTP header
 extension for Mixer-to-Client Audio Level Indication [RFC6465].  If

Westerlund & Wenger Informational [Page 26] RFC 7667 RTP Topologies November 2015

 the SSRCs 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 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 Mixer

 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 and also in 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
 bitstream.
 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 the 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

Westerlund & Wenger Informational [Page 27] RFC 7667 RTP Topologies November 2015

 requires that the different endpoints support a common set of codecs,
 otherwise media transcoding for codec compatibility would still be
 required.
 We now consider the operation of a Media-Switching Mixer that
 supports a video conference with six participating endpoints (A-F)
 where the two most recent speakers in the conference are shown to
 each receiving endpoint.  Thus, the mixer has two SSRCs sending video
 to each peer, and each peer is capable of locally handling two video
 streams simultaneously.
       +-A---------+             +-MIXER----------------------+
       | +-RTP1----|             |-RTP1------+        +-----+ |
       | | +-Video-|             |-Video---+ |        |     | |
       | | |    AV1|------------>|---------+-+------->|  S  | |
       | | |       |<------------|MV1 <----+-+-BV1----|  W  | |
       | | |       |<------------|MV2 <----+-+-EV1----|  I  | |
       | | +-------|             |---------+ |        |  T  | |
       | +---------|             |-----------+        |  C  | |
       +-----------+             |                    |  H  | |
                                 |                    |     | |
       +-B---------+             |                    |  M  | |
       | +-RTP2----|             |-RTP2------+        |  A  | |
       | | +-Video-|             |-Video---+ |        |  T  | |
       | | |    BV1|------------>|---------+-+------->|  R  | |
       | | |       |<------------|MV3 <----+-+-AV1----|  I  | |
       | | |       |<------------|MV4 <----+-+-EV1----|  X  | |
       | | +-------|             |---------+ |        |     | |
       | +---------|             |-----------+        |     | |
       +-----------+             |                    |     | |
                                 :                    :     : :
                                 :                    :     : :
       +-F---------+             |                    |     | |
       | +-RTP6----|             |-RTP6------+        |     | |
       | | +-Video-|             |-Video---+ |        |     | |
       | | |    FV1|------------>|---------+-+------->|     | |
       | | |       |<------------|MV11 <---+-+-AV1----|     | |
       | | |       |<------------|MV12 <---+-+-EV1----|     | |
       | | +-------|             |---------+ |        |     | |
       | +---------|             |-----------+        +-----+ |
       +-----------+             +----------------------------+
                 Figure 16: Media-Switching RTP Mixer

Westerlund & Wenger Informational [Page 28] RFC 7667 RTP Topologies November 2015

 The Media-Switching Mixer can, similarly to the Media-Mixing Mixer,
 reduce the bitrate required for media transmission towards the
 different peers by selecting and forwarding only a subset of RTP
 streams it receives from the sending endpoints.  In case the mixer
 receives simulcast transmissions or a scalable encoding of the Media
 Source, the mixer has more degrees of freedom to select streams or
 subsets of streams 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 fully terminating 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.

Westerlund & Wenger Informational [Page 29] RFC 7667 RTP Topologies November 2015

        +-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 the rewriting of certain RTP header fields when being
 projected from one session into another.  In particular, the sequence

Westerlund & Wenger Informational [Page 30] RFC 7667 RTP Topologies November 2015

 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
 an 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 requires tools such as
 remapping tables between the RTP sessions.  Using independent RTP
 sessions is not required, as it is possible for the switching
 behavior to also perform 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 the 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 deciding 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 that switching points
 be provided between the sources and Temporary Maximum Media Bitrate
 Request (TMMBR) to enable the middlebox to aggregate congestion
 control responses towards the Media Source so to enable it to adjust
 its bitrate (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

Westerlund & Wenger Informational [Page 31] RFC 7667 RTP Topologies November 2015

 only in the temporal dimension.  In both scalable coding and
 simulcast cases, the video signal is represented by a set of two or
 more bitstreams, providing a corresponding number of distinct
 fidelity points.  The middlebox selects which parts of a scalable
 bitstream (or which bitstream, in the case of simulcasting) to
 forward to each of the receiving endpoints.  The decision may be
 driven by a number of factors, such as available bitrate, desired
 layout, etc.  Contrary to transcoding MCUs, SFMs 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 that includes 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 versions, but
 instead they have conceptual property assigned to them and can be
 both turned on/off as well as fully or partially delivered.  Thus,
 this topology appears to be some hybrid between the translator and
 mixer model.
 The differences between a Selective Forwarding Middlebox and a
 Switching-Media 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
 represents 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.

Westerlund & Wenger Informational [Page 32] RFC 7667 RTP Topologies November 2015

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
 well.
 The video-switching MCU may also perform media translation to modify
 the content in bitrate, 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

Westerlund & Wenger Informational [Page 33] RFC 7667 RTP Topologies November 2015

 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 bitrate,
 encoding format, or resolution.  No explicit RTP mechanism is used to
 establish the relationship between the original RTP stream of the
 media being sent and the 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

Westerlund & Wenger Informational [Page 34] RFC 7667 RTP Topologies November 2015

 [RFC3550], the SSRC of the contributors are to be signaled 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 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
     generated.
 2.  There is no information about active media senders available in
     the RTP packet.  As this information is missing, receivers cannot
     use it.  It also deprives the client of information related to
     currently active senders in a machine-usable way, thus preventing
     clients from indicating currently active speakers in user
     interfaces, etc.
 Note that many/most deployed MCUs (and video conferencing endpoints)
 rely on signaling-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 not be 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.

Westerlund & Wenger Informational [Page 35] RFC 7667 RTP Topologies November 2015

 The various (media processing) subunits need (logically and
 physically) to be interconnected by control functionality, but their
 media plane functionality may be split.  These types of terminals are
 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
 1990s.
 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
 CNAME.
             +---------------------+
             | 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 subunits 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,
 middleboxes, and other tools should have no problem with this.

Westerlund & Wenger Informational [Page 36] RFC 7667 RTP Topologies November 2015

 However, outside of the split component terminal scenario (and
 perhaps a multihomed 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 gracefully endpoints that use a common
 CNAME across multiple transport-layer flows: they report an error
 condition in which two separate endpoints are using the same CNAME.
 Depending on the sophistication of the support staff, such erroneous
 reports can lead to support issues.
 The 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 Figure 20, B will send RTCP reports to each of the
 subunits of A.  If the RTCP packets that B sends to the audio subunit
 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 subunit 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 subunit of A only
 report on the received audio, and the RTCP packets B sends to the
 video subunit 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 subunits of a split terminal gives each subunit visibility
 into reception quality of RTP streams sent by the other subunits.

Westerlund & Wenger Informational [Page 37] RFC 7667 RTP Topologies November 2015

 This information can help diagnose reception quality problems, but at
 the cost of increased RTCP bandwidth use.
 RTP streams sent by the subunits 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's 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's and D's sender reports.  In the
 opposite direction, the receiver reports from A and C about B's and
 D's streams also need to be aggregated into the mixer's receiver
 reports sent to B and D.  Since B and D only have the mixer as source
 for the stream, all RTCP from A and C must be suppressed by the
 mixer.
 This topology is so problematic, and it is so easy to get the RTCP
 processing wrong, that it is not recommended for implementation.

3.12. Combining Topologies

 Topologies can be combined and linked to each other using mixers or
 translators.  However, care must be taken in handling the SSRC/CSRC
 space.  A mixer does not forward RTCP from sources in other domains,
 but instead generates its own RTCP packets for each domain it mixes
 into, including the necessary SDES information for both the CSRCs and

Westerlund & Wenger Informational [Page 38] RFC 7667 RTP Topologies November 2015

 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 ASM, provides the
 functionality that everyone may send to, or receive from, everyone
 else within the session.  SSM can provide a similar functionality by
 having anyone intending to participate as a 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, Selective Forwarding Middleboxes (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 come 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 perform
 this monitoring, at least on a medium time scale.

Westerlund & Wenger Informational [Page 39] RFC 7667 RTP Topologies November 2015

 The resource consumption for performing all-to-all transmission
 varies depending on 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
 topologies causes N-1 streams of transmitted packets to traverse the
 first-hop link from the endpoint, in a mesh with N endpoints.  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, RTCP-terminating MCUs, 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
 capabilities.

4.3. Per-Domain Bitrate 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.  In the ASM, SSM, Mesh with common RTP session, and
 Transport Relay scenarios, 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,
 SFMs, 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 bitrate, 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

Westerlund & Wenger Informational [Page 40] RFC 7667 RTP Topologies November 2015

 translator, and potentially the 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 bitrate.  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 use 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 bitrate.  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 impair 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 SDES.  In topologies that
 provide a full all-to-all functionality, i.e., ASM, Mesh with common
 RTP session, and 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 functions.  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 a Back-to-Back RTP session, MESH with independent
 RTP sessions, and the 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 namespace is not necessarily joint.
 Instead, identification will require knowledge of SSRC/CSRC mappings
 that the SFM performed; see Section 3.7.

Westerlund & Wenger Informational [Page 41] RFC 7667 RTP Topologies November 2015

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, and SFMs 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 Media
 ID (MID) in [SDP-BUNDLE].  At the time of writing, there was also a
 proposal for how to include some SDES into an RTP header extension
 [RTCP-SDES].
 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 the 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 MID 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-Trn-
 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), Topo-RTCP-terminating-MCU (RTM), and
 Selective Forwarding Middlebox (SFM).  In the table below, Y

Westerlund & Wenger Informational [Page 42] RFC 7667 RTP Topologies November 2015

 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 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 Real-
 time Transport Protocol (SRTP) and Secure Real-time 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
 signaling 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
 work above the transport layer, such as SRTP, since that sort of
 translator leaves the RTP header and payload unaltered.  Media
 Translators, on the other hand, have strong interactions with
 cryptography, since they alter the RTP payload.  A Media Translator
 in a session that uses cryptographic protection needs to perform
 cryptographic processing to both inbound and outbound packets.

Westerlund & Wenger Informational [Page 43] RFC 7667 RTP Topologies November 2015

 A Media Translator may need to use different cryptographic keys for
 the inbound and outbound processing.  For SRTP, different keys are
 required, because an RFC 3550 Media Translator leaves the SSRC
 unchanged during its packet processing, and SRTP key sharing is only
 allowed when distinct SSRCs can be used to protect distinct packet
 streams.
 When the Media Translator uses different keys to process inbound and
 outbound packets, each session participant needs to be provided with
 the appropriate key, depending on whether they are listening to the
 translator or the original source.  (Note that there is an
 architectural difference between RTP media translation, in which
 participants can rely on the RTP payload type field of a packet to
 determine appropriate processing, and cryptographically protected
 media translation, in which participants must use information that is
 not carried in the packet.)
 When using security mechanisms with translators, SFMs, and mixers, it
 is possible that the translator, SFM, or mixer could create different
 security associations for the different domains they are working in.
 Doing so has some implications:
 First, it might weaken security if the mixer/translator accepts a
 weaker algorithm or key in one domain rather 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 SSRC per domain and therefore needs to create only one
 security association (SRTP crypto context) per domain.  In contrast,
 a translator needs one security association per participant it
 translates towards, in the opposite domain.  Considering Figure 11,
 the translator needs two security associations towards the multicast
 domain: one for B and one for D.  It may be forced to maintain a set
 of totally independent security associations between itself and B and
 D, respectively, so as to avoid two-time pad occurrences.  These
 contexts must also be capable of handling all the sources present in
 the other domains.  Hence, using completely independent security
 associations (for certain keying mechanisms) may force a translator
 to handle N*DM keys and related state, where N is the total number of
 SSRCs used over all domains and DM is the total number of domains.

Westerlund & Wenger Informational [Page 44] RFC 7667 RTP Topologies November 2015

 The ASM, SSM, Relay, and Mesh (with common RTP session) topologies
 each have multiple endpoints that require shared knowledge about the
 different crypto contexts for the endpoints.  These multiparty
 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. References

7.1. Normative References

 [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
            July 2003, <http://www.rfc-editor.org/info/rfc3550>.
 [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,
            DOI 10.17487/RFC4585, July 2006,
            <http://www.rfc-editor.org/info/rfc4585>.
 [RFC7656]  Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
            B. Burman, Ed., "A Taxonomy of Grouping Semantics and
            Mechanisms for Real-Time Transport Protocol (RTP)
            Sources", RFC 7656, November 2015,
            <http://www.rfc-editor.org/info/rfc7656>.

7.2. Informative References

 [MULTI-STREAM-OPT]
            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",
            Work in Progress, draft-ietf-avtcore-rtp-multi-stream-
            optimisation-08, October 2015.

Westerlund & Wenger Informational [Page 45] RFC 7667 RTP Topologies November 2015

 [RFC1112]  Deering, S., "Host extensions for IP multicasting", STD 5,
            RFC 1112, DOI 10.17487/RFC1112, August 1989,
            <http://www.rfc-editor.org/info/rfc1112>.
 [RFC3022]  Srisuresh, P. and K. Egevang, "Traditional IP Network
            Address Translator (Traditional NAT)", RFC 3022,
            DOI 10.17487/RFC3022, January 2001,
            <http://www.rfc-editor.org/info/rfc3022>.
 [RFC3569]  Bhattacharyya, S., Ed., "An Overview of Source-Specific
            Multicast (SSM)", RFC 3569, DOI 10.17487/RFC3569, July
            2003, <http://www.rfc-editor.org/info/rfc3569>.
 [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
            Norrman, "The Secure Real-time Transport Protocol (SRTP)",
            RFC 3711, DOI 10.17487/RFC3711, March 2004,
            <http://www.rfc-editor.org/info/rfc3711>.
 [RFC4575]  Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
            Session Initiation Protocol (SIP) Event Package for
            Conference State", RFC 4575, DOI 10.17487/RFC4575, August
            2006, <http://www.rfc-editor.org/info/rfc4575>.
 [RFC4607]  Holbrook, H. and B. Cain, "Source-Specific Multicast for
            IP", RFC 4607, DOI 10.17487/RFC4607, August 2006,
            <http://www.rfc-editor.org/info/rfc4607>.
 [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
            "Codec Control Messages in the RTP Audio-Visual Profile
            with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
            February 2008, <http://www.rfc-editor.org/info/rfc5104>.
 [RFC5117]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
            DOI 10.17487/RFC5117, January 2008,
            <http://www.rfc-editor.org/info/rfc5117>.
 [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
            Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
            2008, <http://www.rfc-editor.org/info/rfc5285>.
 [RFC5760]  Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
            Protocol (RTCP) Extensions for Single-Source Multicast
            Sessions with Unicast Feedback", RFC 5760,
            DOI 10.17487/RFC5760, February 2010,
            <http://www.rfc-editor.org/info/rfc5760>.

Westerlund & Wenger Informational [Page 46] RFC 7667 RTP Topologies November 2015

 [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,
            DOI 10.17487/RFC5766, April 2010,
            <http://www.rfc-editor.org/info/rfc5766>.
 [RFC6285]  Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
            "Unicast-Based Rapid Acquisition of Multicast RTP
            Sessions", RFC 6285, DOI 10.17487/RFC6285, June 2011,
            <http://www.rfc-editor.org/info/rfc6285>.
 [RFC6465]  Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
            time Transport Protocol (RTP) Header Extension for Mixer-
            to-Client Audio Level Indication", RFC 6465,
            DOI 10.17487/RFC6465, December 2011,
            <http://www.rfc-editor.org/info/rfc6465>.
 [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
            Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
            <http://www.rfc-editor.org/info/rfc7201>.
 [RTCP-SDES]
            Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
            Header Extension for RTCP Source Description Items", Work
            in Progress, draft-ietf-avtext-sdes-hdr-ext-02, July 2015.
 [SDP-BUNDLE]
            Holmberg, C., Alvestrand, H., and C. Jennings,
            "Negotiating Media Multiplexing Using the Session
            Description Protocol (SDP)", Work in Progress,
            draft-ietf-mmusic-sdp-bundle-negotiation-23, July 2015.

Westerlund & Wenger Informational [Page 47] RFC 7667 RTP Topologies November 2015

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.

Authors' Addresses

 Magnus Westerlund
 Ericsson
 Farogatan 2
 SE-164 80 Kista
 Sweden
 Phone: +46 10 714 82 87
 Email: magnus.westerlund@ericsson.com
 Stephan Wenger
 Vidyo
 433 Hackensack Ave
 Hackensack, NJ  07601
 United States
 Email: stewe@stewe.org

Westerlund & Wenger Informational [Page 48]

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