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Versions: 00 01 02 03 draft-ietf-avtcore-multiplex-guidelines

Network Working Group                                      M. Westerlund
Internet-Draft                                                 B. Burman
Intended status: BCP                                            Ericsson
Expires: April 26, 2012                                       C. Perkins
                                                   University of Glasgow
                                                        October 24, 2011


                     RTP Multiplexing Architecture
           draft-westerlund-avtcore-multiplex-architecture-00

Abstract

   RTP has always been a protocol that supports multiple participants
   each sending their own media streams in an RTP session.  Thus relying
   on the three main multiplexing points in RTP; RTP session, SSRC and
   Payload Type for their various needs.  However, most usages of RTP
   have been less complex often with a single SSRC in each direction,
   with a single RTP session per media type.  But the more complex
   usages start to be more common and thus guidance on how to use RTP in
   various complex cases are needed.  This document analyzes a number of
   cases and discusses the usage of the various multiplexing points and
   the need for functionality when defining RTP/RTCP extensions that
   utilize multiple RTP streams and multiple RTP sessions.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 26, 2012.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal



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   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.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . . .  5
     2.1.  Requirements Language  . . . . . . . . . . . . . . . . . .  5
     2.2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . .  5
   3.  RTP Multiplex Points . . . . . . . . . . . . . . . . . . . . .  6
     3.1.  Session  . . . . . . . . . . . . . . . . . . . . . . . . .  6
     3.2.  SSRC . . . . . . . . . . . . . . . . . . . . . . . . . . .  7
     3.3.  CSRC . . . . . . . . . . . . . . . . . . . . . . . . . . .  8
     3.4.  Payload Type . . . . . . . . . . . . . . . . . . . . . . .  8
   4.  Multiple Streams Alternatives  . . . . . . . . . . . . . . . .  9
   5.  RTP Topologies and Issues  . . . . . . . . . . . . . . . . . . 10
     5.1.  Point to Point . . . . . . . . . . . . . . . . . . . . . . 11
       5.1.1.  RTCP Reporting . . . . . . . . . . . . . . . . . . . . 11
       5.1.2.  Compound RTCP Packets  . . . . . . . . . . . . . . . . 12
     5.2.  Point to Multipoint Using Multicast  . . . . . . . . . . . 12
     5.3.  Point to Multipoint Using an RTP Translator  . . . . . . . 14
     5.4.  Point to Multipoint Using an RTP Mixer . . . . . . . . . . 15
     5.5.  Point to Multipoint using Multiple Unicast flows . . . . . 16
     5.6.  Decomposited End-Point . . . . . . . . . . . . . . . . . . 16
   6.  Dismissing Payload Type Multiplexing . . . . . . . . . . . . . 18
   7.  Multiple Streams Discussion  . . . . . . . . . . . . . . . . . 20
     7.1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . 20
     7.2.  RTP/RTCP Aspects . . . . . . . . . . . . . . . . . . . . . 20
       7.2.1.  The RTP Specification  . . . . . . . . . . . . . . . . 20
       7.2.2.  Multiple SSRC Legacy Considerations  . . . . . . . . . 22
       7.2.3.  RTP Specification Clarifications Needed  . . . . . . . 23
       7.2.4.  Handling Varying sets of Senders . . . . . . . . . . . 23
       7.2.5.  Cross Session RTCP requests  . . . . . . . . . . . . . 23
       7.2.6.  Binding Related Sources  . . . . . . . . . . . . . . . 23
       7.2.7.  Forward Error Correction . . . . . . . . . . . . . . . 25
       7.2.8.  Transport Translator Sessions  . . . . . . . . . . . . 26
       7.2.9.  Multiple Media Types in one RTP session  . . . . . . . 26
     7.3.  Signalling Aspects . . . . . . . . . . . . . . . . . . . . 28
       7.3.1.  Session Oriented Properties  . . . . . . . . . . . . . 28
       7.3.2.  SDP Prevents Multiple Media Types  . . . . . . . . . . 29
     7.4.  Network Apsects  . . . . . . . . . . . . . . . . . . . . . 29



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       7.4.1.  Quality of Service . . . . . . . . . . . . . . . . . . 29
       7.4.2.  NAT and Firewall Traversal . . . . . . . . . . . . . . 29
       7.4.3.  Multicast  . . . . . . . . . . . . . . . . . . . . . . 31
       7.4.4.  Multiplexing multiple RTP Session on a Single
               Transport  . . . . . . . . . . . . . . . . . . . . . . 32
     7.5.  Security Aspects . . . . . . . . . . . . . . . . . . . . . 32
       7.5.1.  Security Context Scope . . . . . . . . . . . . . . . . 32
       7.5.2.  Key-Management for Multi-party session . . . . . . . . 33
   8.  Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 33
   9.  RTP Specification Clarifications . . . . . . . . . . . . . . . 35
     9.1.  RTCP Reporting from all SSRCs  . . . . . . . . . . . . . . 35
     9.2.  RTCP Self-reporting  . . . . . . . . . . . . . . . . . . . 35
     9.3.  Combined RTCP Packets  . . . . . . . . . . . . . . . . . . 35
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 35
   11. Security Considerations  . . . . . . . . . . . . . . . . . . . 36
   12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 36
   13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 36
     13.1. Normative References . . . . . . . . . . . . . . . . . . . 36
     13.2. Informative References . . . . . . . . . . . . . . . . . . 36
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 39































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1.  Introduction

   This document focuses at issues of non-basic usage of RTP [RFC3550]
   where multiple media sources of the same media type are sent over
   RTP.  Separation of different media types is another issue that will
   be discussed in this document.  The intended uses include for example
   multiple sources from the same end-point, multiple streams from a
   single media source, multiple end-points each having a source, or an
   application that needs multiple representations (encodings) of a
   particular source.  It will be shown that these uses are inter-
   related and need a common discussion to ensure consistency.  In
   general, usage of the RTP session and media streams will be discussed
   in detail.

   RTP is already designed for multiple participants in a communication
   session.  This is not restricted to multicast, as many believe, but
   also provides functionality over unicast, using either multiple
   transport flows below RTP or a network node that re-distributes the
   RTP packets.  The node can for example be a transport translator
   (relay) that forwards the packets unchanged, a translator performing
   media translation in addition to forwarding, or an RTP mixer that
   creates new conceptual sources from the received streams.  In
   addition, multiple streams may occur when a single end-point have
   multiple media sources of the same media type, like multiple cameras
   or microphones that need to be sent simultaneously.

   Historically, the most common RTP use cases have been point to point
   Voice over IP (VoIP) or streaming applications, commonly with no more
   than one media source per end-point and media type (typically audio
   and video).  Even in conferencing applications, especially voice
   only, the conference focus or bridge has provided a single stream
   with a mix of the other participants to each participant.  It is also
   common to have individual RTP sessions between each end-point and the
   RTP mixer.

   SSRC is the RTP media stream identifier that helps to uniquely
   identify media sources in RTP sessions.  Even though available SSRC
   space can theoretically handle more than 4 billion simultaneous
   sources, the perceived need for handling multiple SSRCs in
   implementations has been small.  This has resulted in an installed
   legacy base that isn't fully RTP specification compliant and will
   have different issues if they receive multiple SSRCs of media, either
   simultaneously or in sequence.  These issues will manifest themselves
   in various ways, either by software crashes or simply in limited
   functionality, like only decoding and playing back the first or
   latest received SSRC and discarding media related to any other SSRCs.

   There have also arisen various cases where multiple SSRCs are used to



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   represent different aspects of what is in fact a single underlying
   media source.  A very basic case is RTP retransmission [RFC4588]
   which have one SSRC for the original stream, and a second SSRC either
   in the same session or in a different session to represent the
   retransmitted packets to ensure that the monitoring functions still
   function.  Another use case is scalable encoding, such as the RTP
   payload format for Scalable Video Coding (SVC) [RFC6190], which has
   an operation mode named Multiple Session Transmission (MST) that uses
   one SSRC in each RTP session to send one or more scalability layers.
   A third example is simulcast where a single media source is encoded
   in different versions and transmitted to an RTP mixer that picks
   which version to actually distribute to a given receiver part of the
   RTP session.

   This situation has created a need for a document that discusses the
   existing possibilities in the RTP protocol and how these can and
   should be used in applications.  A new set of applications needing
   more advanced functionalities from RTP is also emerging on the
   market, such as telepresence and advanced video conferencing.  Thus
   furthering the need for a more common understanding of how multiple
   streams are handled in RTP to ensure media plane interoperability.

   The document starts with some definitions and then goes into the
   existing RTP functionalities around multiplexing.  Both the desired
   behavior and the implications of a particular behavior depend on
   which topologies are used, which requires some consideration.  This
   is followed by a discussion of some choices in multiplexing behavior
   and their impacts.  Finally, some recommendations and examples are
   provided.


2.  Definitions

2.1.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

2.2.  Terminology

   The following terms and abbreviations are used in this document:

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




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   Media Stream:  A sequence of RTP packets using a single SSRC that
      together carry part or all of the content of a specific Media Type
      from a specific sender source within a given RTP session.

   Media Aggregate:  All Media Streams related to a particular Source.

   Media Type:  Audio, video, text or data whose form and meaning are
      defined by a specific real-time application.

   Source:  The source of a particular media stream.  Either a single
      media capturing device such as a video camera, or a microphone, or
      a specific output of a media production function, such as an audio
      mixer, or some video editing function.


3.  RTP Multiplex Points

   This section describes the existing RTP tools that enable
   multiplexing of different media streams and RTP functionalities.

3.1.  Session

   The RTP Session is the highest semantic level in RTP and contains all
   of the RTP functionality.

   RTP in itself does not contain any Session identifier, but relies on
   the underlying transport.  For example, when running RTP on top of
   UDP, an RTP endpoint can identify and delimit an RTP Session from
   other RTP Sessions through the UDP source and destination transport
   address, consisting of network address and port number(s).  Most
   commonly only the destination address, i.e. all traffic received on a
   particular port, is defined as belonging to a specific RTP Session.
   It is worth noting that in practice a more narrow definition of the
   transport flows that are related to a give RTP session is possible.
   An RTP session can for example be defined as one or more 5-tuples
   (Transport Protocol, Source Address, Source Port, Destination
   Address, Destination Port).  Any set of identifiers of RTP and RTCP
   packet flows are sufficient to determine if the flow belongs to a
   particular session or not.

   Commonly, RTP and RTCP use separate ports and the destination
   transport address is in fact an address pair, but in the case of RTP/
   RTCP multiplex [RFC5761] there is only a single port.

   A source that changes its source transport address during a session
   must also choose a new SSRC identifier to avoid being interpreted as
   a looped source.




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   The set of participants considered part of the same RTP Session is
   defined by[RFC3550] as those that share a single SSRC space.  That
   is, those participants that can see an SSRC identifier transmitted by
   any one of the other participants.  A participant can receive an SSRC
   either as SSRC or CSRC in RTP and RTCP packets.  Thus, the RTP
   Session scope is decided by the participants' network interconnection
   topology, in combination with RTP and RTCP forwarding strategies
   deployed by end-points and any interconnecting middle nodes.

3.2.  SSRC

   The Synchronization Source (SSRC) identifier is used to identify
   individual sources within an RTP Session.  The SSRC number is
   globally unique within an RTP Session and all RTP implementations
   must be prepared to use procedures for SSRC collision handling, which
   results in an SSRC number change.  The SSRC number is randomly
   chosen, carried in every RTP packet header and is not dependent on
   network address.  SSRC is also used as identifier to refer to
   separate media streams in RTCP.

   A media source having an SSRC identifier can be of different types:

   Real:  Connected to a "physical" media source, for example a camera
      or microphone.

   Conceptual:  A source with some attributed property generated by some
      network node, for example a filtering function in an RTP mixer
      that provides the most active speaker based on some criteria, or a
      mix representing a set of other sources.

   Virtual:  A source that does not generate any RTP media stream in
      itself (e.g. an end-point only receiving in an RTP session), but
      anyway need a sender SSRC for use as source in RTCP reports.

   Note that a "multimedia source" that generates more than one media
   type, e.g. a conference participant sending both audio and video,
   need not (and commonly should not) use the same SSRC value across RTP
   sessions.  RTCP Compound packets containing the CNAME SDES item is
   the designated method to bind an SSRC to a CNAME, effectively cross-
   correlating SSRCs within and between RTP Sessions as coming from the
   same end-point.  The main property attributed to SSRCs associated
   with the same CNAME is that they are from a particular
   synchronization context and may be synchronized at playback.  There
   exist a few other methods to relate different SSRC where use of CNAME
   is inappropriate, such as session-based RTP retransmission [RFC4588].

   Note also that RTP sequence number and RTP timestamp are scoped by
   SSRC and thus independent between different SSRCs.



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   An RTP receiver receiving a previously unseen SSRC value must
   interpret it as a new source.  It may in fact be a previously
   existing source that had to change SSRC number due to an SSRC
   conflict.  However, the originator of the previous SSRC should have
   ended the conflicting source by sending an RTCP BYE for it prior to
   starting to send with the new SSRC, so the new SSRC is anyway
   effectively a new source.

   Some RTP extension mechanisms already require the RTP stacks to
   handle additional SSRCs, like SSRC multiplexed RTP retransmission
   [RFC4588].  However, that still only requires handling a single media
   decoding chain per pair of SSRCs.

3.3.  CSRC

   The Contributing Source (CSRC) can arguably be seen as a sub-part of
   a specific SSRC and thus a multiplexing point.  It is optionally
   included in the RTP header, shares the SSRC number space and
   specifies which set of SSRCs that has contributed to the RTP payload.
   However, even though each RTP packet and SSRC can be tagged with the
   contained CSRCs, the media representation of an individual CSRC is in
   general not possible to extract from the RTP payload since it is
   typically the result of a media mixing (merge) operation (by an RTP
   mixer) on the individual media streams corresponding to the CSRC
   identifiers.  Due to these restrictions, CSRC will not be considered
   a fully qualified multiplex point and will be disregarded in the rest
   of this document.

3.4.  Payload Type

   The Payload Type number is also carried in every RTP packet header
   and identifies what format the RTP payload has.  The term "format"
   here includes whatever can be described by out-of-band signaling
   means for dynamic payload types, as well as the statically allocated
   payload types in [RFC3551].  In SDP the term "format" includes media
   type, RTP timestamp sampling rate, codec, codec configuration,
   payload format configurations, and various robustness mechanisms such
   as redundant encodings [RFC2198].

   The meaning of a Payload Type definition (the number) is re-used
   between all media streams within an RTP session, when the definition
   is either static or signaled through SDP.  There however do exist
   cases where each end-point have different sets of payload types due
   to SDP offer/answer.

   Although Payload Type definitions are commonly local to an RTP
   Session, there are some uses where Payload Type numbers need be
   unique across RTP Sessions.  This is for example the case in Media



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   Decoding Dependency [RFC5583] where Payload Types are used to
   describe media dependency across RTP Sessions.

   Given that multiple Payload Types are defined in an RTP Session, a
   media sender is free to change the Payload Type on a per packet
   basis.  One example of designed per-packet change of Payload Type is
   a speech codec that makes use of generic Comfort Noise [RFC3389].

   The RTP Payload Type in RTP is designed such that only a single
   Payload Type is valid at any time instant in the SSRC's timestamp
   time line, effectively time-multiplexing different Payload Types if
   any switch occurs.  Even when this constraint is met, having
   different rates on the RTP timestamp clock for the RTP Payload Types
   in use in the same RTP Session have issues such as loss of
   synchronization.  Payload Type clock rate switching requires some
   special consideration that is described in the multiple clock rates
   specification [I-D.ietf-avtext-multiple-clock-rates].

   If there is a true need to send multiple Payload Types for the same
   SSRC that are valid for the same RTP Timestamps, then redundant
   encodings [RFC2198] can be used.  Several additional constraints than
   the ones mentioned above need to be met to enable this use, one of
   which are that the combined payload sizes of the different Payload
   Types must not exceed the transport MTU.

   Other aspects of RTP payload format use are described in RTP Payload
   HowTo [I-D.ietf-payload-rtp-howto].


4.  Multiple Streams Alternatives

   This section reviews the alternatives to enable multi-stream
   handling.  Let's start with describing mechanisms that could enable
   multiple media streams, independent of the purpose for having
   multiple streams.

   SSRC Multiplexing:  Each additional Media Stream gets its own SSRC
      within a RTP Session.

   Session Multiplexing:  Using additional RTP Sessions to handle
      additional Media Streams

   Payload Type Multiplexing:  Using different RTP payload types for
      different additional streams.

   Independent of the reason to use additional media streams, achieving
   it using payload type multiplexing is not a good choice as can be
   seen in the below section (Section 6).  The RTP payload type alone is



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   not suitable for cases where additional media streams are required.
   Streams need their own SSRCs, so that they get their own sequence
   number space.  The SSRC itself is also important so that the media
   stream can be referenced and reported on.

   This leaves us with two choices, either using SSRC multiplexing to
   have multiple SSRCs from one end-point in one RTP session, or create
   additional RTP sessions to hold that additional SSRC.  As the below
   discussion will show, in reality we cannot choose a single one of the
   two solutions.  To utilize RTP well and as efficiently as possible,
   both are needed.  The real issue is finding the right guidance on
   when to create RTP sessions and when additional SSRCs in an RTP
   session is the right choice.

   In the below discussion, please keep in mind that the reasons for
   having multiple media streams vary and include but are not limited to
   the following:

   o  Multiple Media Sources of the same media type

   o  Retransmission streams

   o  FEC stream

   o  Alternative Encoding

   o  Scalability layer

   Thus the choice made due to one reason may not be the choice suitable
   for another reason.  In the above list, the different items have
   different levels of maturity in the discussion on how to solve them.
   The clearest understanding is associated with multiple media sources
   of the same media type.  However, all warrant discussion and
   clarification on how to deal with them.


5.  RTP Topologies and Issues

   The impact of how RTP Multiplex is performed will in general vary
   with how the RTP Session participants are interconnected; the RTP
   Topology [RFC5117].  This section describes the topologies and
   attempts to highlight the important behaviors concerning RTP
   multiplexing and multi-stream handling.  It lists any identified
   issues regarding RTP and RTCP handling, and introduces additional
   topologies that are supported by RTP beyond those included in RTP
   Topologies [RFC5117].  The RTP Topologies that do not follow the RTP
   specification or do not attempt to utilize the facilities of RTP are
   ignored in this document.



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5.1.  Point to Point

   This is the most basic use case with an RTP session containing of two
   end-points.  Each end-point has one or more SSRCs.

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

                              Point to Point

5.1.1.  RTCP Reporting

   In cases when an end-point uses multiple SSRCs, we have found two
   closely related issues.  The first is if every SSRC shall report on
   all other SSRC, even the ones originating from the same end-point.
   The reason for this would be ensure that no monitoring function
   should suspect a breakage in the RTP session.

   The second issue around RTCP reporting arise when an end-point
   receives one or more media streams, and when the receiving end-point
   itself sends multiple SSRC in the same RTP session.  As transport
   statistics are gathered per end-point and shared between the nodes,
   all the end-point's SSRC will report based on the same received data,
   the only difference will be which SSRCs sends the report.  This could
   be considered unnecessary overhead, but for consistency it might be
   simplest to always have all sending SSRCs send RTCP reports on all
   media streams the end-point receives.

   The current RTP text is silent about sending RTCP Receiver Reports
   for an endpoint's own sources, but does not preclude either sending
   or omitting them.  The uncertainty in the expected behavior in those
   cases have likely caused variations in the implementation strategy.
   This could cause an interoperability issue where it is not possible
   to determine if the lack of reports are a true transport issue, or
   simply a result of implementation.

   Although this issue is valid already for the simple point to point
   case, it needs to be considered in all topologies.  From the
   perspective of an end-point, any solution needs to take into account
   what a particular end-point can determine without explicit
   information of the topology.  For example, a Transport Translator
   (Relay) topology will look quite similar as point to point on an RTP
   level but is different.  The main difference between a point to point
   with two SSRC being sent from the remote end-point and a Transport
   Translator with two single SSRC remote clients are that the RTT may
   vary between the SSRCs (but it is not guaranteed), and that the SSRCs
   may have different CNAMEs.



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5.1.2.  Compound RTCP Packets

   When an end-point has multiple SSRCs and it needs to send RTCP
   packets on behalf of these SSRCs, the question arises if and how RTCP
   packets with different source SSRCs can be sent in the same compound
   packet.  If it is allowed, then some consideration of the
   transmission scheduling is needed.

5.2.  Point to Multipoint Using Multicast

   This section discusses the Point to Multi-point using Multicast to
   interconnect the session participants.  This needs to consider both
   Any Source Multicast (ASM) and Source-Specific Multicast (SSM).

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

              Point to Multipoint Using Any Source Multicast

   In Any Source Multicast, any of the participants can send to all the
   other participants, simply by sending a packet to the multicast
   group.  That is not possible in Source Specific Multicast [RFC4607]
   where only a single source (Distribution Source) can send to the
   multicast group, creating a topology that looks like the one below:




















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                                         Source-specific
          +--------+       +-----+          Multicast
          |Media   |       |     |     +----------------> R(1)
          |Sender 1|<----->| D S |     |                    |
          +--------+       | I O |  +--+                    |
                           | 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|<----->|   | |<-------------------------+
          +--------+       +-----+            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.

            Point to Multipoint using Source Specific Multicast

   In this topology a number of Media Senders (1 to M) are allowed to
   send media to the SSM group, sends media to the distribution source
   which then forwards the media streams to the multicast group.  The
   media streams reach the Receivers (R(1) to R(n)).  The Receiver's
   RTCP cannot be sent to the multicast group.  To support RTCP, an RTP
   extension for SSM [RFC5760] was defined that use unicast transmission
   to send RTCP from the receivers to one or more Feedback Targets (FT).

   As multicast is a one to many distribution system this must be taken
   into consideration.  For example, the only practical method for
   adapting the bit-rate sent towards a given receiver is to use a set
   of multicast groups, where each multicast group represents a
   particular bit-rate.  The media encoding is either scalable, where
   multiple layers can be combined, or simulcast where a single version
   is selected.  By either selecting or combing multicast groups, the
   receiver can control the bit-rate sent on the path to itself.  It is
   also common that transport robustification is sent in its own
   multicast group to allow for interworking with legacy or to support
   different levels of protection.

   The result of this is three common behaviors for RTP multicast:





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   1.  Use of multiple RTP sessions for the same media type.

   2.  The need for identifying RTP sessions that are related in one of
       several ways.

   3.  The need for binding related SSRCs in different RTP sessions
       together.

   This indicates that Multicast is an important consideration when
   working with the RTP multiplexing and multi stream architecture
   questions.  It is also important to note that so far there is no
   special mode for basic behavior between multicast and unicast usages
   of RTP.  Yes, there are extensions targeted to deal with multicast
   specific cases but the general applicability does need to be
   considered.

5.3.  Point to Multipoint Using an RTP Translator

   Transport Translators (Relays) are a very important consideration for
   this document as they result in an RTP session situation that is very
   similar to how an ASM group RTP session would behave.

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

                       Transport Translator (Relay)

   One of the most important aspects with the simple relay is that it is
   both easy to implement and require minimal amount of resources as
   only transport headers are rewritten, no RTP modifications nor media
   transcoding occur.  Thus it is most likely the cheapest and most
   generally deployable method for multi-point sessions.  The most
   obvious downside of this basic relaying is that the translator has no
   control over how many streams needs to be delivered to a receiver.
   Nor can it simply select to deliver only certain streams, at least
   not without new RTCP extensions to coherently handle the fact that
   some middlebox temporarily stops a stream, preventing some receivers
   from reporting on it.  This consistency problem in RTCP reporting
   needs to be handled.

   The Transport Translator does not need to have an SSRC of itself, nor
   need it send any RTCP reports on the flows that passes it, but it may
   choose to do that.



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   Use of a transport translator results in that any of the end-points
   will receive multiple SSRCs over a single unicast transport flow from
   the translator.  That is independent of the other end-points having
   only a single or several SSRCs.  End-points that have multiple SSRCs
   put further requirements on how SSRCs can be related or bound within
   and across RTP sessions and how they can be identified on an
   application level.

   A Media Translator can perform a large variety of media functions
   affecting the media stream passing the translator, coming from one
   source and destined to a particular end-point.  The media stream can
   be transcoded to a different bit-rate, change to another encoder,
   change the packetization of the media stream, add FEC streams, or
   terminate RTP retransmissions.  The latter behaviors require the
   translator to use SSRCs that only exist in a particular sub-domain of
   the RTP session, and it may also create additional sessions when the
   mechanism applied on one side so requires.

5.4.  Point to Multipoint Using an RTP Mixer

   The most commonly used topology in centralized conferencing is based
   on the RTP Mixer.  The main reason for this is that it provides a
   very consistent view of the RTP session towards each participant.
   That is accomplished through the mixer having its own SSRCs and any
   media sent to the participants will be sent using those SSRCs.  If
   the mixer wants to identify the underlying media sources for its
   conceptual streams, it can identify them using CSRC.  The media
   stream the mixer provides can be an actual media mixing of multiple
   media sources.  It might also be as simple as selecting one of the
   underlying sources based on some mixer policy or control signalling.

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

                                 RTP Mixer

   In the case where the mixer does stream selection, an application may
   in fact desire multiple simultaneous streams but only as many as the
   mixer can handle.  As long as the mixer and an end-point can agree on
   the maximum number of streams and how the streams that are delivered
   are selected, this provides very good functionality.  As these
   streams are forwarded using the mixer's SSRCs, there are no
   inconsistencies within the session.



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5.5.  Point to Multipoint using Multiple Unicast flows

   Based on the RTP session definition, it is clearly possible to have a
   joint RTP session over multiple transport flows like the below three
   end-point joint session.  In this case, A needs to send its' media
   streams and RTCP packets to both B and C over their respective
   transport flows.  As long as all participants do the same, everyone
   will have a joint view of the RTP session.

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

          Point to Multi-Point using Multiple Unicast Transprots

   This doesn't create any additional requirements beyond the need to
   have multiple transport flows associated with a single RTP session.
   Note that an end-point may use a single local port to receive all
   these transport flows, or it might have separate local reception
   ports for each of the end-points.

5.6.  Decomposited End-Point

   There is some possibility that an RTP end-point implementation in
   fact reside on multiple devices, each with their own network address.
   A very basic use case for this would be to separate audio and video
   processing for a particular end-point, like a conference room, into
   one device handling the audio and another handling the video being
   interconnected by some control functions allowing them to behave as a
   single end-point.














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                 +---------------------+
                 | End-point A         |
                 | Local Area Network  |
                 |      +------------+ |
                 |   +->| Audio      |<+----\
                 |   |  +------------+ |     \    +------+
                 |   |  +------------+ |      +-->|      |
                 |   +->| Video      |<+--------->|  B   |
                 |   |  +------------+ |      +-->|      |
                 |   |  +------------+ |     /    +------+
                 |   +->| Control    |<+----/
                 |      +------------+ |
                 +---------------------+

                          Decomposited End-Point

   In the above usage, let us assume that the RTP sessions are different
   for audio and video.  The audio and video parts will use a common
   CNAME and also have a common clock to ensure that synchronization and
   clock drift handling works despite the decomposition.  However, if
   the audio and video were in a single RTP session then this use case
   becomes problematic.  This as all transport flow receivers will need
   to receive all the other media streams that are part of the session.
   Thus the audio component will receive also all the video media
   streams, while the video component will receive all the audio ones,
   thus doubling the site's bandwidth requirements from all other
   session participants.  With a joint RTP session it also becomes
   evident that a given end-point, as interpreted from a CNAME
   perspective, has two sets of transport flows for receiving the
   streams and the decomposition isn't hidden.

   The requirements that can derived from the above usage is that the
   transport flows for each RTP session might be under common control
   but still go to what looks like different end-points based on
   addresses and ports.  A conclusion can also be reached that
   decomposition without using separate RTP sessions has downsides and
   potential for RTP/RTCP issues.

   There exist another use case which might be considered as a
   decomposited end-point.  However, as will be shown this should be
   considered a translator instead.  An example of this is when an end-
   point A sends a media flow to B. On the path there is a device C that
   on A's behalf does something with the media streams, for example adds
   an RTP session with FEC information for A's media streams.  C will in
   this case need to bind the new FEC streams to A's media stream by
   using the same CNAME as A.





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   +------+        +------+         +------+
   |      |        |      |         |      |
   |  A   |------->|  C   |-------->|  B   |
   |      |        |      |---FEC-->|      |
   +------+        +------+         +------+

                    When Decomposition is a Translator

   This type of functionality where C does something with the media
   stream on behalf of A is clearly covered under the media translator
   definition (Section 5.3).


6.  Dismissing Payload Type Multiplexing

   Before starting a discussion on when to use what alternative, we will
   first document a number of reasons why using the payload type as a
   multiplexing point for anything related to multiple streams is
   unsuitable and will not be considered further.

   If one attempts to use Payload type multiplexing beyond it's defined
   usage, that has well known negative effects on RTP.  To use Payload
   type as the single discriminator for multiple streams implies that
   all the different media streams are being sent with the same SSRC,
   thus using the same timestamp and sequence number space.  This has
   many effects:

   1.   Putting restraint on RTP timestamp rate for the multiplexed
        media.  For example, media streams that use different RTP
        timestamp rates cannot be combined, as the timestamp values need
        to be consistent across all multiplexed media frames.  Thus
        streams are forced to use the same rate.  When this is not
        possible, Payload Type multiplexing cannot be used.

   2.   Many RTP payload formats may fragment a media object over
        multiple packets, like parts of a video frame.  These payload
        formats need to determine the order of the fragments to
        correctly decode them.  Thus it is important to ensure that all
        fragments related to a frame or a similar media object are
        transmitted in sequence and without interruptions within the
        object.  This can relatively simple be solved on the sender side
        by ensuring that the fragments of each media stream are sent in
        sequence.

   3.   Some media formats require uninterrupted sequence number space
        between media parts.  These are media formats where any missing
        RTP sequence number will result in decoding failure or invoking
        of a repair mechanism within a single media context.  The text/



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        T140 payload format [RFC4103] is an example of such a format.
        These formats will need a sequence numbering abstraction
        function between RTP and the individual media stream before
        being used with Payload Type multiplexing.

   4.   Sending multiple streams in the same sequence number space makes
        it impossible to determine which Payload Type and thus which
        stream a packet loss relates to.

   5.   If RTP Retransmission [RFC4588] is used and there is a loss, it
        is possible to ask for the missing packet(s) by SSRC and
        sequence number, not by Payload Type.  If only some of the
        Payload Type multiplexed streams are of interest, there is no
        way of telling which missing packet(s) belong to the interesting
        stream(s) and all lost packets must be requested, wasting
        bandwidth.

   6.   The current RTCP feedback mechanisms are built around providing
        feedback on media streams based on stream ID (SSRC), packet
        (sequence numbers) and time interval (RTP Timestamps).  There is
        almost never a field to indicate which Payload Type is reported,
        so sending feedback for a specific media stream is difficult
        without extending existing RTCP reporting.

   7.   The current RTCP media control messages [RFC5104] specification
        is oriented around controlling particular media flows, i.e.
        requests are done addressing a particular SSRC.  Such mechanisms
        would need to be redefined to support Payload Type multiplexing.

   8.   The number of payload types are inherently limited.
        Accordingly, using Payload Type multiplexing limits the number
        of streams that can be multiplexed and does not scale.  This
        limitation is exacerbated if one uses solutions like RTP and
        RTCP multiplexing [RFC5761] where a number of payload types are
        blocked due to the overlap between RTP and RTCP.

   9.   At times, there is a need to group multiplexed streams and this
        is currently possible for RTP Sessions and for SSRC, but there
        is no defined way to group Payload Types.

   10.  It is currently not possible to signal bandwidth requirements
        per media stream when using Payload Type Multiplexing.

   11.  Most existing SDP media level attributes cannot be applied on a
        per Payload Type level and would require re-definition in that
        context.





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   12.  A legacy end-point that doesn't understand the indication that
        different RTP payload types are different media streams may be
        slightly confused by the large amount of possibly overlapping or
        identically defined RTP Payload Types.


7.  Multiple Streams Discussion

7.1.  Introduction

   Using multiple media streams is a well supported feature of RTP.
   However, what can be unclear for most implementors or people writing
   RTP/RTCP extensions attempting to apply multiple streams, is when it
   is most appropriate to add an additional SSRC in an existing RTP
   session and when it is better to use multiple RTP sessions.  This
   section tries to discuss the various considerations needed.  The next
   section then concludes with some guidelines.

7.2.  RTP/RTCP Aspects

   This section discusses RTP and RTCP aspects worth considering when
   selecting between SSRC multiplexing and Session multiplexing.

7.2.1.  The RTP Specification

   RFC 3550 contains some recommendations and a bullet list with 5
   arguments for different aspects of RTP multiplexing.  Let's review
   Section 5.2 of [RFC3550], reproduced below:

   "For efficient protocol processing, the number of multiplexing points
   should be minimized, as described in the integrated layer processing
   design principle [ALF].  In RTP, multiplexing is provided by the
   destination transport address (network address and port number) which
   is different for each RTP session.  For example, in a teleconference
   composed of audio and video media encoded separately, each medium
   SHOULD be carried in a separate RTP session with its own destination
   transport address.

   Separate audio and video streams SHOULD NOT be carried in a single
   RTP session and demultiplexed based on the payload type or SSRC
   fields.  Interleaving packets with different RTP media types but
   using the same SSRC would introduce several problems:

   1.  If, say, two audio streams shared the same RTP session and the
       same SSRC value, and one were to change encodings and thus
       acquire a different RTP payload type, there would be no general
       way of identifying which stream had changed encodings.




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   2.  An SSRC is defined to identify a single timing and sequence
       number space.  Interleaving multiple payload types would require
       different timing spaces if the media clock rates differ and would
       require different sequence number spaces to tell which payload
       type suffered packet loss.

   3.  The RTCP sender and receiver reports (see Section 6.4) can only
       describe one timing and sequence number space per SSRC and do not
       carry a payload type field.

   4.  An RTP mixer would not be able to combine interleaved streams of
       incompatible media into one stream.

   5.  Carrying multiple media in one RTP session precludes: the use of
       different network paths or network resource allocations if
       appropriate; reception of a subset of the media if desired, for
       example just audio if video would exceed the available bandwidth;
       and receiver implementations that use separate processes for the
       different media, whereas using separate RTP sessions permits
       either single- or multiple-process implementations.

   Using a different SSRC for each medium but sending them in the same
   RTP session would avoid the first three problems but not the last
   two.

   On the other hand, multiplexing multiple related sources of the same
   medium in one RTP session using different SSRC values is the norm for
   multicast sessions.  The problems listed above don't apply: an RTP
   mixer can combine multiple audio sources, for example, and the same
   treatment is applicable for all of them.  It may also be appropriate
   to multiplex streams of the same medium using different SSRC values
   in other scenarios where the last two problems do not apply."

   Let's consider one argument at a time.  The first is an argument for
   using different SSRC for each individual media stream, which still is
   very applicable.

   The second argument is advocating against using payload type
   multiplexing, which still stands as can been seen by the extensive
   list of issues found in Section 6.

   The third argument is yet another argument against payload type
   multiplexing.

   The fourth is an argument against multiplexing media streams that
   require different handling into the same session.  This is to
   simplify the processing at any receiver of the media stream.  If all
   media streams that exist in an RTP session is of one media type and



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   one particular purpose, there is no need for deeper inspection of the
   packets before processing them in both end-points and RTP aware
   middle nodes.

   The fifth argument discusses network aspects that we will discuss
   more below in Section 7.4.  It also goes into aspects of
   implementation, like decomposed end-points where different processes
   or inter-connected devices handle different aspects of the whole
   multi-media session.

   A summary of RFC 3550's view on multiplexing is to use unique SSRCs
   for anything that is its' own media/packet stream, and secondly use
   different RTP sessions for media streams that don't share media type
   and purpose, to maximize flexibility when it comes to processing and
   handling of the media streams.

   This mostly agrees with the discussion and recommendations in this
   document.  However, there has been an evolution of RTP since that
   text was written which needs to be reflected in the discussion.
   Additional clarifications for specific cases are also needed.

7.2.2.  Multiple SSRC Legacy Considerations

   When establishing RTP sessions that may contain end-points that
   aren't updated to handle multiple streams following these
   recommendations, a particular application can have issues with
   multiple SSRCs within a single session.  These issues include:

   1.  Need to handle more than one stream simultaneously rather than
       replacing an already existing stream with a new one.

   2.  Be capable of decoding multiple streams simultaneously.

   3.  Be capable of rendering multiple streams simultaneously.

   RTP Session multiplexing could potentially avoid these issues if
   there is only a single SSRC at each end-point, and in topologies
   which appears like point to point as seen the end-point.  However,
   forcing the usage of session multiplexing due to this reason would be
   a great mistake, as it is likely that a significant set of
   applications will need a combination of SSRC multiplexing of several
   media sources and session multiplexing for other aspects such as
   encoding alternatives, robustification or simply to support legacy.
   However, this issue does need consideration when deploying multiple
   media streams within an RTP session where legacy end-points may
   occur.





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7.2.3.  RTP Specification Clarifications Needed

   The RTP specification contains a few things that are potential
   interoperability issues when using multiple SSRCs within a session.
   These issues are described and discussed in Section 9.  These should
   not be considered strong arguments against using SSRC multiplexing
   when otherwise appropriate, and there are some issues we expect to be
   solved in the near future.

7.2.4.  Handling Varying sets of Senders

   Another potential issue that needs to be considered is where a
   limited set of simultaneously active sources varies within a larger
   set of session members.  As each media decoding chain may contain
   state, it is important that this type of usage ensures that a
   receiver can flush a decoding state for an inactive source and if
   that source becomes active again, it does not assume that this
   previous state exists.

   This behavior might in certain applications be possible to limit to a
   particular RTP Session and instead use multiple RTP sessions.  But in
   some cases it is likely unavoidable and the most appropriate thing is
   to SSRC multiplex.

7.2.5.  Cross Session RTCP requests

   There currently exist no functionality to make truly synchronized and
   atomic RTCP requests across multiple RTP Sessions.  Instead separate
   RTCP messages will have to be sent in each session.  This gives SSRC
   multiplexed streams a slight advantage as RTCP requests for different
   streams in the same session can be sent in a compound RTCP packet.
   Thus providing an atomic operation if different modifications of
   different streams are requested at the same time.

   In Session multiplexed cases, the RTCP timing rules in the sessions
   and the transport aspects, such as packet loss and jitter, prevents a
   receiver from relying on atomic operations, instead more robust and
   forgiving mechanisms need to be used.

7.2.6.  Binding Related Sources

   A common problem in a number of various RTP extensions has been how
   to bind together related sources.  This issue is common independent
   of SSRC multiplexing and Session Multiplexing, and any solution and
   recommendation to the problem should work equally well for both to
   avoid creating barriers between using session multiplexing and SSRC
   multiplexing.




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   The current solutions don't have these properties.  There exist one
   solution for grouping RTP session together in SDP [RFC5888] to know
   which RTP session contains for example the FEC data for the source
   data in another session.  However, this mechanism does not work on
   individual media flows and is thus not directly applicable to the
   problem.  The other solution is also SDP based and can group SSRCs
   within a single RTP session [RFC5576].  Thus this mechanism can bind
   media streams in SSRC multiplexed cases.  Both solutions have the
   shortcoming of being restricted to SDP based signalling and also do
   not work in cases where the session's dynamic properties are such
   that it is difficult or resource consuming to keep the list of
   related SSRCs up to date.

   One possible solution could be to mandate the same SSRC being used in
   all RTP session in case of session multiplexing.  We do note that
   Section 8.3 of the RTP Specification [RFC3550] recommends using a
   single SSRC space across all RTP sessions for layered coding.
   However this recommendation has some downsides and is less applicable
   beyond the field of layered coding.  To use the same sender SSRC in
   all RTP sessions from a particular end-point can cause issues if an
   SSRC collision occurs.  If the same SSRC is used as the required
   binding between the streams, then all streams in the related RTP
   sessions must change their SSRC.  This is extra likely to cause
   problems if the participant populations are different in the
   different sessions.  For example, in case of large number of
   receivers having selected totally random SSRC values in each RTP
   session as RFC 3550 specifies, a change due to a SSRC collision in
   one session can then cause a new collision in another session.  This
   cascading effect is not severe but there is an increased risk that
   this occurs for well populated sessions.  In addition, being forced
   to change the SSRC affects all the related media streams; instead of
   having to re-synchronize only the originally conflicting stream, all
   streams will suddenly need to be re-synchronized with each other.
   This will prevent also the media streams not having an actual
   collision from being usable during the re-synchronization and also
   increases the time until synchronization is finalized.  In addition,
   it requires exception handling in the SSRC generation.

   The above collision issue does not occur in case of having only one
   SSRC space across all sessions and all participants will be part of
   at least one session, like the base layer in layered encoding.  In
   that case the only downside is the special behavior that needs to be
   well defined by anyone using this.  But, having an exception behavior
   where the SSRC space is common across all session an that doesn't fit
   all the RTP extensions or payload formats present in the sessions is
   a issue.  It is possible to create a situation where the different
   mechanisms can't be combined due to the non standard SSRC allocation
   behavior.



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   Existing mechanisms with known issues:

   RTP Retransmission (RFC4588):  Has two modes, one for SSRC
      multiplexing and one for Session multiplexing.  The session
      multiplexing requires the same CNAME and mandates that the same
      SSRC is used in both sessions.  Using the same SSRC does work but
      will potentially have issues in certain cases.  In SSRC
      multiplexed mode the CNAME is used, and when the first
      retransmission request is sent, one must not have another
      retransmission request outstanding for an SSRC which don't have a
      the binding between the original SSRC and the retransmission
      stream's SSRC.  This works but creates some limitations that can
      be avoided by a more explicit mechanism.  The SDP based ssrc-group
      mechanism is sufficient in this case as long as the application
      can rely on the signalling based solution.

   Scalable Video Coding (RFC6190):  As an example of scalable coding,
      SVC [RFC6190] has various modes.  The Multi Session Transmission
      (MST) uses Session multiplexing to separate scalability layers.
      However, this specification has failed to explicit how these
      layers are bound together in cases where CNAME isn't sufficient.
      CNAME is no longer sufficient when more than one media source
      occur within a session that have the same CNAME, for example due
      to multiple video cameras capturing the same lecture hall.  This
      likely implies that a single SSRC space as recommend by Section
      8.3 of RTP [RFC3550] is to be used.

   Forward Error Correction:  If some type of FEC or redundancy stream
      is being sent, it needs it's own SSRC, with the exception of
      constructions like redundancy encoding [RFC2198].  Thus in case of
      transmitting the FEC in the same session as the source data, the
      inter SSRC relation within a session is needed.  In case of
      sending the redundant data in a separate session from the source,
      the SSRC in each session needs to be related.  This occurs for
      example in RFC5109 when using session separation of original and
      FEC data.  SSRC multiplexing is not supported, only using
      redundant encoding is supported.

   This issue appears to need action to harmonize and avoid future
   shortcomings in extension specifications.  A proposed solution for
   handling this issue is [I-D.westerlund-avtext-rtcp-sdes-srcname].

7.2.7.  Forward Error Correction

   There exist a number of Forward Error Correction (FEC) based schemes
   for how to reduce the packet loss of the original streams.  Most of
   the FEC schemes will protect a single source flow.  The protection is
   achieved by transmitting a certain amount of redundant information



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   that is encoded such that it can repair one or more packet loss over
   the set of packets they protect.  This sequence of redundant
   information also needs to be transmitted as its own media stream, or
   in some cases instead of the original media stream.  Thus many of
   these schemes creates a need for binding the related flows as
   discussed above.  They also create additional flows that need to be
   transported.  Looking at the history of these schemes, there is both
   SSRC multiplexed and Session multiplexed solutions and some schemes
   that support both.

   Using a Session multiplexed solution provides good support for legacy
   when deploying FEC or changing the scheme used so that some set of
   receivers may not be able to utilize the FEC information.  By placing
   it in a separate RTP session, it can easily be ignored.

   In usages involving multicast, having the FEC information on its own
   multicast group and RTP session allows for flexibility, for example
   when using Rapid Acquisition of Multicast Groups (RAMS) [RFC6285].
   During the RAMS burst where data is received over unicast and where
   it is possible to combine with unicast based retransmission
   [RFC4588], there is no need to burst the FEC data related to the
   burst of the source media streams needed to catch up with the
   multicast group.  This saves bandwidth to the receiver during the
   burst, enabling quicker catch up.  When the receiver has catched up
   and joins the multicast group(s) for the source, it can at the same
   time join the multicast group with the FEC information.  Having the
   source stream and the FEC in separate groups allow for easy
   separation in the Burst/Retransmission Source (BRS) without having to
   individually classify packets.

7.2.8.  Transport Translator Sessions

   A basic Transport Translator relays any incoming RTP and RTCP packets
   to the other participants.  The main difference between SSRC
   multiplexing and Session multiplexing resulting from this use case is
   that for SSRC multiplexing it is not possible for a particular
   session participant to decide to receive a subset of media streams.
   When using separate RTP sessions for the different sets of media
   streams, a single participant can choose to leave one of the sessions
   but not the other.

7.2.9.  Multiple Media Types in one RTP session

   Having different media types, like audio and video, in the same RTP
   sessions is not forbidden, only recommended against as can be seen in
   Section 7.2.1.  When using multiple media types, there are a number
   of considerations:




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   Payload Type gives Media Type:  This solution is dependent on getting
      the media type from the Payload Type.  Thus overloading this de-
      multiplexing point in a receiver for two purposes.  First for the
      main media type and determining the processing chain, then later
      for the exact configuration of the encoder and packetization.

   Payload Type field limiations:  The total number of Payload Types
      available to use in an RTP session is fairly limited, especially
      if Multiplexing RTP Data and Control Packets on a Single Port
      [RFC5761] is used.  For certain applications negotiating a large
      set of codes and configuration may become an issue.

   Don't switch media types for an SSRC:  The primary reasons to avoid
      switching from sending for example audio to sending video using
      the same SSRC is the implications on a receiver.  When this
      happens, the processing chain in the receiver will have to switch
      from one media type to another.  As the different media type's
      entire processing chains are different and are connected to
      different outputs it is difficult to reuse the decoding chain,
      which a normal codec change likely can.  Instead the entire
      processing chain has to be torn down and replaced.  In addition,
      there is likely a clock rate switching problem, possibly resulting
      in synchronization loss at the point of switching media type if
      some packet loss occurs.

   RTCP Bit-rate Issues:  If the media types are significantly different
      in bit-rate, the RTCP bandwidth rates assigned to each source in a
      session can result in interesting effects, like that the RTCP bit-
      rate share for an audio stream is larger than the actual audio
      bit-rate.  In itself this doesn't cause any conflicts, only
      potentially unnecessary overhead.  It is possible to avoid this
      using AVPF or SAVPF and setting trr-int parameter, which can bring
      down unnecessary regular reporting while still allowing for rapid
      feedback.

   Decomposited end-points:  Decomposited nodes that rely on the regular
      network to separate audio and video to different devices do not
      work well with this session setup.  If they are forced to work,
      all media receiver parts of a decomposited end-point will receive
      all media, thus doubling the bit-rate consumption for the end-
      point.

   RTP Mixers and Translators:  An RTP mixer or Media Translator will
      also have to support this particular session setup, where it
      before could rely on the RTP session to determine what processing
      options should be applied to the incoming packets.

   As can be seen, there is nothing in here that prevents using a single



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   RTP session for multiple media types, however it does create a number
   of limitations and special case implementation requirements.  So
   anyone considering to use this setup should carefully review if the
   reasons for using a single RTP session is sufficient to motivate this
   special case.

7.3.  Signalling Aspects

   There exist various signalling solutions for establishing RTP
   sessions.  Many are SDP [RFC4566] based, however SDP functionality is
   also dependent on the signalling protocols carrying the SDP.  Where
   RTSP [RFC2326] and SAP [RFC2974] both use SDP in a declarative
   fashion, SIP [RFC3261] uses SDP with the additional definition of
   Offer/Answer [RFC3264].  The impact on signalling and especially SDP
   needs to be considered as it can greatly affect how to deploy a
   certain multiplexing point choice.

7.3.1.  Session Oriented Properties

   One aspect of the existing signalling is that it is focused around
   sessions, or at least in the case of SDP the media description.
   There are a number of things that are signalled on a session level/
   media description but that are not necessarily strictly bound to an
   RTP session and could be of interest to signal specifically for a
   particular media stream within the session.  The following properties
   have been identified as being potentially useful to signal not only
   on RTP session level:

   o  Bitrate/Bandwidth exist today only at aggregate or a common any
      media stream limit

   o  Which SSRC that will use which RTP Payload Types

   Some of these issues are clearly SDP's problem rather than RTP
   limitations.  However, if the aim is to deploy an SSRC multiplexed
   solution that contains several sets of media streams with different
   properties (encoding/packetization parameter, bit-rate, etc), putting
   each set in a different RTP session would directly enable negotiation
   of the parameters for each set.  If insisting on SSRC multiplexing, a
   number of signalling extensions are needed to clarify that there are
   multiple sets of media streams with different properties and that
   they shall in fact be kept different, since a single set will not
   satisfy the applications requirements.

   This does in fact create a strong driver to use RTP session
   multiplexing for any case where different sets of media streams with
   different requirements exist.




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7.3.2.  SDP Prevents Multiple Media Types

   SDP encoded in its structure a prevention against using multiple
   media types in the same RTP session.  A media description in SDP can
   only have a single media type; audio, video, text, image,
   application.  This media type is used as the top-level media type for
   identifying the actual payload format bound to a particular payload
   type using the rtpmap attribute.  Thus a high fence against using
   multiple media types in the same session was created.

   There is a proposal in the MMUSIC WG for how one could allow multiple
   media lines describe a single underlying transport
   [I-D.holmberg-mmusic-sdp-bundle-negotiation] and thus support either
   one RTP session with multiple media types.  There is also a solution
   for multiplexing multiple RTP sessions onto the same transport
   [I-D.westerlund-avtcore-single-transport-multiplexing].

7.4.  Network Apsects

   The multiplexing choice has impact on network level mechanisms that
   need to be considered by the implementor.

7.4.1.  Quality of Service

   When it comes to Quality of Service mechanisms, they are either flow
   based or marking based.  RSVP [RFC2205] is an example of a flow based
   mechanism, while Diff-Serv [RFC2474] is an example of a Marking based
   one.  For a marking based scheme, the method of multiplexing will not
   affect the possibility to use QoS.

   However, for a flow based scheme there is a clear difference between
   the methods.  SSRC multiplexing will result in all media streams
   being part of the same 5-tuple (protocol, source address, destination
   address, source port, destination port) which is the most common
   selector for flow based QoS.  Thus, separation of the level of QoS
   between media streams is not possible.  That is however possible for
   session based multiplexing, where each different version can be in a
   different RTP session that can be sent over different 5-tuples.

7.4.2.  NAT and Firewall Traversal

   In today's network there exist a large number of middleboxes.  The
   ones that normally have most impact on RTP are Network Address
   Translators (NAT) and Firewalls (FW).

   Below we analyze and comment on the impact of requiring more
   underlying transport flows in the presence of NATs and Firewalls:




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   End-Point Port Consumption:  A given IP address only has 65536
      available local ports per transport protocol for all consumers of
      ports that exist on the machine.  This is normally never an issue
      for an end-user machine.  It can become an issue for servers that
      handle large number of simultaneous streams.  However, if the
      application uses ICE to authenticate STUN requests, a server can
      serve multiple end-points from the same local port, and use the
      whole 5-tuple (source and destination address, source and
      destination port, protocol) as identifier of flows after having
      securely bound them to the remote end-point address using the STUN
      request.  In theory the minimum number of media server ports
      needed are the maximum number of simultaneous RTP Sessions a
      single end-point may use.  In practice, implementation will
      probably benefit from using more server ports to simplify
      implementation or avoid performance bottlenecks.

   NAT State:  If an end-point sits behind a NAT, each flow it generates
      to an external address will result in a state that has to be kept
      in the NAT.  That state is a limited resource.  In home or Small
      Office/Home Office (SOHO) NATs, memory or processing are usually
      the most limited resources.  For large scale NATs serving many
      internal end-points, available external ports are typically the
      scarce resource.  Port limitations is primarily a problem for
      larger centralized NATs where end-point independent mapping
      requires each flow to use one port for the external IP address.
      This affects the maximum number of internal users per external IP
      address.  However, it is worth pointing out that a real-time video
      conference session with audio and video is likely using less than
      10 UDP flows, compared to certain web applications that can use
      100+ TCP flows to various servers from a single browser instance.

   NAT Traversal Excess Time:  Making the NAT/FW traversal takes a
      certain amount of time for each flow.  It also takes time in a
      phase of communication between accepting to communicate and the
      media path being established which is fairly critical.  The best
      case scenario for how much extra time it can take following the
      specified ICE procedures are: 1.5*RTT + Ta*(Additional_Flows-1),
      where Ta is the pacing timer, which ICE specifies to be no smaller
      than 20 ms.  That assumes a message in one direction, and then an
      immediate triggered check back.  This as ICE first finds one
      candidate pair that works prior to establish multiple flows.
      Thus, there are no extra time until one has found a working
      candidate pair.  Based on that working pair the extra time is to
      in parallel establish the, in most cases 2-3, additional flows.







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   NAT Traversal Failure Rate:  Due to the need to establish more than a
      single flow through the NAT, there is some risk that establishing
      the first flow succeeds but that one or more of the additional
      flows fail.  The risk that this happens is hard to quantify, but
      it should be fairly low as one flow from the same interfaces has
      just been successfully established .  Thus only rare events such
      as NAT resource overload, or selecting particular port numbers
      that are filtered etc, should be reasons for failure.

   SSRC multiplexing keeps additional media streams within one RTP
   Session and does not introduce any additional NAT traversal
   complexities per media stream.  In contrast, the session multiplexing
   is using one RTP session per media stream.  Thus additional lower
   layer transport flows will be required, unless an explicit de-
   multiplexing layer is added between RTP and the transport protocol.
   A proposal for how to multiplex multiple RTP sessions over the same
   single lower layer transport exist in
   [I-D.westerlund-avtcore-single-transport-multiplexing].

7.4.3.  Multicast

   Multicast groups provides a powerful semantics for a number of real-
   time applications, especially the ones that desire broadcast-like
   behaviors with one end-point transmitting to a large number of
   receivers, like in IPTV.  But that same semantics do result in a
   certain number of limitations.

   One limitation is that for any group, sender side adaptation to the
   actual receiver properties causes a degradation for all participants
   to what is supported by the receiver with the worst conditions among
   the group participants.  In most cases this is not acceptable.
   Instead various receiver based solutions are employed to ensure that
   the receivers achieve best possible performance.  By using scalable
   encoding and placing each scalability layer in a different multicast
   group, the receiver can control the amount of traffic it receives.
   To have each scalability layer on a different multicast group, one
   RTP session per multicast group is used.

   If instead a single RTP session over multiple transports were to be
   deployed, i.e. multicast groups with each layer as it's own SSRC,
   then very different views of the RTP session would exist.  That as
   one receiver may see only a single layer (SSRC), while another may
   see three SSRCs if it joined three multicast groups.  This would
   cause disjoint RTCP reports where a management system would not be
   able to determine if a receiver isn't reporting on a particular SSRC
   due to that it is not a member of that multicast group, or because it
   doesn't receive it as a result of a transport failure.




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   Thus it appears easiest and most straightforward to use multiple RTP
   sessions.  In addition, the transport flow considerations in
   multicast are a bit different from unicast.  First of all there is no
   shortage of port space, as each multicast group has its own port
   space.

7.4.4.  Multiplexing multiple RTP Session on a Single Transport

   For applications that doesn't need flow based QoS and like to save
   ports and NAT/FW traversal costs, there is a proposal for how to
   achieve multiplexing of multiple RTP sessions over the same lower
   layer transport
   [I-D.westerlund-avtcore-single-transport-multiplexing].  Using such a
   solution would allow session multiplexing without most of the
   perceived downsides of additional RTP sessions creating a need for
   additional transport flows.

7.5.  Security Aspects

   On the basic level there is no significant difference in security
   when having one RTP session and having multiple.  However, there are
   a few more detailed considerations that might need to be considered
   in certain usages.

7.5.1.  Security Context Scope

   When using SRTP [RFC3711] the security context scope is important and
   can be a necessary differentiation in some applications.  As SRTP's
   crypto suites (so far) is built around symmetric keys, the receiver
   will need to have the same key as the sender.  This results in that
   none in a multi-party session can be certain that a received packet
   really was sent by the claimed sender or by another party having
   access to the key.  In most cases this is a sufficient security
   property, but there are a few cases where this does create
   situations.

   The first case is when someone leaves a multi-party session and one
   wants to ensure that the party that left can no longer access the
   media streams.  This requires that everyone re-keys without
   disclosing the keys to the excluded party.

   A second case is when using security as an enforcing mechanism for
   differentiation.  Take for example a scalable layer or a high quality
   simulcast version which only premium users are allowed to access.
   The mechanism preventing a receiver from getting the high quality
   stream can be based on the stream being encrypted with a key that
   user can't access without paying premium, having the key-management
   limit access to the key.



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   In the latter case it is likely easiest from signalling, transport
   (if done over multicast) and security to use a different RTP session.
   That way the user(s) not intended to receive a particular stream can
   easily be excluded.  There is no need to have SSRC specific keys,
   which many of the key-management systems cannot handle.

7.5.2.  Key-Management for Multi-party session

   Performing key-management for Multi-party session can be a challenge.
   This section considers some of the issues.

   Transport translator based session cannot use Security Description
   [RFC4568] nor DTLS-SRTP [RFC5764] without an extension as each end-
   point provides it's set of keys.  In centralized conference, the
   signalling counterpart is a conference server and the media plane
   unicast counterpart (to which DTLS messages would be sent) is the
   translator.  Thus an extension like Encrypted Key Transport
   [I-D.ietf-avt-srtp-ekt] are needed or a MIKEY [RFC3830] based
   solution that allows for keying all session participants with the
   same master key.

   Keying of multicast transported SRTP face similar challenges as the
   transport translator case.


8.  Guidelines

   This section contains a number of recommendations for implementors or
   specification writers when it comes to handling multi-stream.

   Don't Require the same SSRC across Sessions:  As discussed in
      Section 7.2.6 there exist drawbacks in using the same SSRC in
      multiple RTP sessions as a mechanism to bind related media streams
      together.  Instead a mechanism to explicitly signal the relation
      SHOULD be used, either in RTP/RTCP or in the used signalling
      mechanism that establish the RTP session(s).

   Use SSRC multiplexing for additional Media Sources:  In the cases an
      RTP end-point needs to transmit additional media source(s) of the
      same media type and purpose in the application it is RECOMMENDED
      to send them as additional SSRCs in the same RTP session.  For
      example a telepresence room where there are three cameras, and
      each camera captures 2 persons sitting at the table, sending each
      camera as its own SSRC within a single RTP session is recommended.







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   Use additional RTP sessions for streams with different purposes:
      When media streams have different purpose or processing
      requirements it is RECOMMENDED that the different types of streams
      are put in different RTP sessions.

   When using Session Multiplexing use grouping:  When using Session
      Multiplexing solutions it is RECOMMENDED to be explicitly group
      the involved RTP sessions using the signalling mechanism, for
      example The Session Description Protocol (SDP) Grouping Framework.
      [RFC5888]

   RTP/RTCP Extensions May Support SSRC and Session Multiplexing:  When
      defining an RTP or RTCP extension, the creator needs to consider
      if this extension is applicable in both SSRC multiplexed and
      Session multiplexed usages.  If it is, then any generic extensions
      are RECOMMENDED to support both.  Applications that are not as
      generally applicable will have to consider if interoperability is
      better served by defining a single solution or providing both
      options.

   Transport Support Extensions:  When defining new RTP/RTCP extensions
      intended for transport support, like the retransmission or FEC
      mechanisms, they are RECOMMENDED to include support for both SSRC
      and Session multiplexing so that application developers can choose
      freely from the set of mechanisms without concerning themselves
      with if a particular solution only supports one of the
      multiplexing choices.

   This discussion and guidelines points out that a small set of
   extension mechanisms could greatly improve the situation when it
   comes to using multiple streams independently of Session multiplexing
   or SSRC multiplexing.  These extensions are:

   Media Source Identification:  A Media source identification that can
      be used to bind together media streams that are related to the
      same media source.  A proposal
      [I-D.westerlund-avtext-rtcp-sdes-srcname] exist for a new SDES
      item SRCNAME that also can be used with the a=ssrc SDP attribute
      to provide signalling layer binding information.

   SSRC limiations within RTP sessions:  By providing a signalling
      solution that allows the signalling peers to explicitly express
      both support and limitations on how many simultaneous media
      streams an end-point can handle within a given RTP Session.  That
      ensures that usage of SSRC multiplexing occurs when supported and
      without overloading an end-point.  This extension is proposed in
      [I-D.westerlund-avtcore-max-ssrc].




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9.  RTP Specification Clarifications

   This section describes a number of clarifications to the RTP
   specifications that are likely necessary for aligned behavior when
   RTP sessions contains more SSRCs than one local and one remote.

9.1.  RTCP Reporting from all SSRCs

   When one have multiple SSRC in an RTP node, then all these SSRC must
   send RTCP SR or RR as long as the SSRC exist.  It is not sufficient
   that only one SSRC in the node sends report blocks on the incoming
   RTP streams.  The reason for this is that a third party monitor may
   not necessarily be able to determine that all these SSRC are in fact
   co-located and originate from the same stack instance that gather
   report data.

9.2.  RTCP Self-reporting

   For any RTP node that sends more than one SSRC, there exist the
   question if SSRC1 needs to report its reception of SSRC2 and vice
   versa.  The reason that they in fact need to report on all other
   local streams as being received is report consistency.  A third party
   monitor that considers the full matrix of media streams and all known
   SSRC reports on these media streams would detect a gap in the reports
   which could be a transport issue unless identified as in fact being
   sources from same node.

9.3.  Combined RTCP Packets

   When a node contains multiple SSRCs, it is questionable if an RTCP
   compound packet can only contain RTCP packets from a single SSRC or
   if multiple SSRCs can include their packets in a joint compound
   packet.  The high level question is a matter for any receiver
   processing on what to expect.  In addition to that question there is
   the issue of how to use the RTCP timer rules in these cases, as the
   existing rules are focused on determining when a single SSRC can
   send.


10.  IANA Considerations

   This document makes no request of IANA.

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






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11.  Security Considerations


12.  Acknowledgements


13.  References

13.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

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

13.2.  Informative References

   [ALF]      Clark, D. and D. Tennenhouse, "Architectural
              Considerations for a New Generation of Protocols", SIGCOMM
              Symposium on         Communications Architectures and
              Protocols (Philadelphia, Pennsylvania), pp. 200--208, IEEE
              Computer Communications Review, Vol. 20(4),
              September 1990.

   [I-D.holmberg-mmusic-sdp-bundle-negotiation]
              Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
              Using Session Description Protocol (SDP) Port Numbers",
              draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
              progress), October 2011.

   [I-D.ietf-avt-srtp-ekt]
              McGrew, D., Andreasen, F., Wing, D., and K. Fischer,
              "Encrypted Key Transport for Secure RTP",
              draft-ietf-avt-srtp-ekt-02 (work in progress), March 2011.

   [I-D.ietf-avtext-multiple-clock-rates]
              Petit-Huguenin, M., "Support for multiple clock rates in
              an RTP session", draft-ietf-avtext-multiple-clock-rates-01
              (work in progress), July 2011.

   [I-D.ietf-payload-rtp-howto]
              Westerlund, M., "How to Write an RTP Payload Format",
              draft-ietf-payload-rtp-howto-01 (work in progress),
              July 2011.

   [I-D.westerlund-avtcore-max-ssrc]



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              Westerlund, M., Burman, B., and F. Jansson, "Multiple
              Synchronization sources (SSRC) in RTP Session Signaling",
              draft-westerlund-avtcore-max-ssrc (work in progress),
              October 2011.

   [I-D.westerlund-avtcore-single-transport-multiplexing]
              Westerlund, M., "Multiple RTP Session on a Single Lower-
              Layer Transport",
              draft-westerlund-avtcore-transport-multiplexing (work in
              progress), October 2011.

   [I-D.westerlund-avtext-rtcp-sdes-srcname]
              Westerlund, M., Burman, B., and P. Sandgren, "RTCP SDES
              Item SRCNAME to Label Individual Sources",
              draft-westerlund-avtext-rtcp-sdes-srcname (work in
              progress), October 2011.

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              September 1997.

   [RFC2205]  Braden, B., Zhang, L., Berson, S., Herzog, S., and S.
              Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1
              Functional Specification", RFC 2205, September 1997.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC2474]  Nichols, K., Blake, S., Baker, F., and D. Black,
              "Definition of the Differentiated Services Field (DS
              Field) in the IPv4 and IPv6 Headers", RFC 2474,
              December 1998.

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, October 2000.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, September 2002.



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   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

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

   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              August 2004.

   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, June 2005.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, July 2006.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

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

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

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

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.

   [RFC5583]  Schierl, T. and S. Wenger, "Signaling Media Decoding
              Dependency in the Session Description Protocol (SDP)",
              RFC 5583, July 2009.

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




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Internet-Draft         RTP Multiplex Architecture           October 2011


   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888, June 2010.

   [RFC6190]  Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              "RTP Payload Format for Scalable Video Coding", RFC 6190,
              May 2011.

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


Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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


   Bo Burman
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 13 11
   Email: bo.burman@ericsson.com












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   Colin Perkins
   University of Glasgow
   School of Computing Science
   Glasgow  G12 8QQ
   United Kingdom

   Email: csp@csperkins.org












































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