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RFC 7667

RTP Topologies

Pages: 48
Obsoletes:  5117
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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


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

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

2.1. Glossary

ASM: Any-Source Multicast AVPF: The extended RTP profile for RTCP-based feedback CSRC: Contributing Source Link: The data transport to the next IP hop Middlebox: A device that is on the Path that media travel between two endpoints MCU: Multipoint Control Unit Path: The concatenation of multiple links, resulting in an end-to-end data transfer. PtM: Point to Multipoint PtP: Point to Point 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.
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   Multimedia Session:   A Multimedia Session is an association among a
      group of participants engaged in communication via one or more RTP

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.
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   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
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   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. 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
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   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. 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 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
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   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

   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.
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                 +------+        +------+         +------+
                 |      |        |      |         |      |
                 |  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.
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   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
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   Point to Multipoint (PtM) is defined here as using a multicast
   topology as a transmission model, in which traffic from any 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.
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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
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   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

   The aforementioned situation results in common behavior for RTP

   1.  Multicast applications often use a group of RTP sessions, not
       one.  Each endpoint needs to be a member of most or all of these
       RTP sessions in order to perform well.

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

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

      -------> Multicast RTP 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
Top   ToC   RFC7667 - Page 17
   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.

(page 17 continued on part 2)

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