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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: Informational                                  Ericsson
Expires: September 13, 2012                                   C. Perkins
                                                   University of Glasgow
                                                          March 12, 2012


                     RTP Multiplexing Architecture
           draft-westerlund-avtcore-multiplex-architecture-01

Abstract

   Real-time Transport Protocol is a flexible protocol possible to use
   in a wide range of applications and network and system topologies.
   This flexibility and the implications of different choices should be
   understood by any application developer using RTP.  To facilitate
   that understanding, this document contains an in-depth discussion of
   the usage of RTP's multiplexing points; the RTP session, the
   Synchronization Source Identifier (SSRC), and the payload type.  The
   focus is put on the first two, trying to give guidance and source
   material for an analysis on the most suitable choices for the
   application being designed.

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 September 13, 2012.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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



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   (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 . . . . . . . . . . . . . . . . . . . . . . . . . . .  9
     3.4.  Payload Type . . . . . . . . . . . . . . . . . . . . . . .  9
   4.  Multiple Streams Alternatives  . . . . . . . . . . . . . . . . 10
   5.  RTP Topologies and Issues  . . . . . . . . . . . . . . . . . . 11
     5.1.  Point to Point . . . . . . . . . . . . . . . . . . . . . . 12
       5.1.1.  RTCP Reporting . . . . . . . . . . . . . . . . . . . . 12
       5.1.2.  Compound RTCP Packets  . . . . . . . . . . . . . . . . 13
     5.2.  Point to Multipoint Using Multicast  . . . . . . . . . . . 13
     5.3.  Point to Multipoint Using an RTP Translator  . . . . . . . 15
     5.4.  Point to Multipoint Using an RTP Mixer . . . . . . . . . . 16
     5.5.  Point to Multipoint using Multiple Unicast flows . . . . . 17
     5.6.  De-composite End-Point . . . . . . . . . . . . . . . . . . 18
   6.  Multiple Streams Discussion  . . . . . . . . . . . . . . . . . 19
     6.1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . 19
     6.2.  RTP/RTCP Aspects . . . . . . . . . . . . . . . . . . . . . 19
       6.2.1.  The RTP Specification  . . . . . . . . . . . . . . . . 19
       6.2.2.  Handling Varying sets of Senders . . . . . . . . . . . 22
       6.2.3.  Cross Session RTCP Requests  . . . . . . . . . . . . . 22
       6.2.4.  Binding Related Sources  . . . . . . . . . . . . . . . 22
       6.2.5.  Forward Error Correction . . . . . . . . . . . . . . . 24
       6.2.6.  Transport Translator Sessions  . . . . . . . . . . . . 25
     6.3.  Interworking . . . . . . . . . . . . . . . . . . . . . . . 25
       6.3.1.  Interworking Applications  . . . . . . . . . . . . . . 26
       6.3.2.  Multiple SSRC Legacy Considerations  . . . . . . . . . 27
     6.4.  Signalling Aspects . . . . . . . . . . . . . . . . . . . . 28
       6.4.1.  Session Oriented Properties  . . . . . . . . . . . . . 28
       6.4.2.  SDP Prevents Multiple Media Types  . . . . . . . . . . 29
       6.4.3.  Media Stream Usage . . . . . . . . . . . . . . . . . . 29
     6.5.  Network Aspects  . . . . . . . . . . . . . . . . . . . . . 30
       6.5.1.  Quality of Service . . . . . . . . . . . . . . . . . . 30



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       6.5.2.  NAT and Firewall Traversal . . . . . . . . . . . . . . 31
       6.5.3.  Multicast  . . . . . . . . . . . . . . . . . . . . . . 32
       6.5.4.  Multiplexing multiple RTP Session on a Single
               Transport  . . . . . . . . . . . . . . . . . . . . . . 33
     6.6.  Security Aspects . . . . . . . . . . . . . . . . . . . . . 33
       6.6.1.  Security Context Scope . . . . . . . . . . . . . . . . 33
       6.6.2.  Key-Management for Multi-party session . . . . . . . . 34
       6.6.3.  Complexity Implications  . . . . . . . . . . . . . . . 34
     6.7.  Multiple Media Types in one RTP session  . . . . . . . . . 35
   7.  Arch-Types . . . . . . . . . . . . . . . . . . . . . . . . . . 37
     7.1.  Single SSRC per Session  . . . . . . . . . . . . . . . . . 37
     7.2.  Multiple SSRCs of the Same Media Type  . . . . . . . . . . 39
     7.3.  Multiple Sessions for one Media type . . . . . . . . . . . 40
     7.4.  Multiple Media Types in one Session  . . . . . . . . . . . 41
     7.5.  Summary  . . . . . . . . . . . . . . . . . . . . . . . . . 43
   8.  Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 43
   9.  Proposal for Future Work . . . . . . . . . . . . . . . . . . . 44
   10. RTP Specification Clarifications . . . . . . . . . . . . . . . 45
     10.1. RTCP Reporting from all SSRCs  . . . . . . . . . . . . . . 45
     10.2. RTCP Self-reporting  . . . . . . . . . . . . . . . . . . . 45
     10.3. Combined RTCP Packets  . . . . . . . . . . . . . . . . . . 45
   11. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 46
   12. Security Considerations  . . . . . . . . . . . . . . . . . . . 46
   13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 46
   14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 46
     14.1. Normative References . . . . . . . . . . . . . . . . . . . 46
     14.2. Informative References . . . . . . . . . . . . . . . . . . 46
   Appendix A.  Dismissing Payload Type Multiplexing  . . . . . . . . 49
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 51






















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1.  Introduction

   Real-time Transport Protocol (RTP) [RFC3550] is a commonly used
   protocol for real-time media transport.  It is a protocol that
   provides great flexibility and can support a large set of different
   applications.  RTP has several multiplexing points designed for
   different purposes.  These enable support of multiple media streams
   and switching between different encoding or packetization of the
   media.  By using multiple RTP sessions, sets of media streams can be
   structured for efficient processing or identification.  Thus the
   question for any RTP application designer is how to best use the RTP
   session, the SSRC and the payload type to meet the application's
   needs.

   The purpose of this document is to provide clear information about
   the possibilities of RTP when it comes to multiplexing.  The RTP
   application designer should understand the implications that come
   from a particular choice of RTP multiplexing points.  The document
   will recommend against some usages as being unsuitable, in general or
   for particular purposes.

   RTP was from the beginning designed for multiple participants in a
   communication session.  This is not restricted to multicast, as some
   may 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 re-distributing 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, like multiple cameras
   or microphones that need to be sent simultaneously.

   This document has been written due to increased interest in more
   advanced usage of RTP, resulting in questions regarding the most
   appropriate RTP usage.  The limitations in some implementations, RTP/
   RTCP extensions, and signalling has also been exposed.  It is
   expected that some limitations will be addressed by updates or new
   extensions resolving the shortcomings.  The authors also hope that
   clarification on the usefulness of some functionalities in RTP will
   result in more complete implementations in the future.

   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.  Some arch-types of RTP usage are discussed.



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   Finally, some recommendations and examples are provided.

   This document is currently an individual contribution, but it is the
   intention of the authors that this should become a WG document that
   objectively describes and provides suitable recommendations for which
   there is WG consensus.  Currently this document only represents the
   views of the authors.  The authors gladly accept any feedback on the
   document and will be happy to discuss suitable recommendations.


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.

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

   Media Source:  The originator or source of a particular Media Stream.
      It can either be a single media capturing device such as a video
      camera, a microphone, or a specific output of a media production
      function, such as an audio mixer, or some video editing function.

   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.

   Multiplex:  The operation of taking multiple entities as input,
      aggregating them onto some common resource while keeping the
      individual entities addressable such that they can later be fully
      and unambiguously separated (de-multiplexed) again.






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

   Source:  See Media Source.


3.  RTP Multiplex Points

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

3.1.  Session

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

   Identifier:  RTP in itself does not contain any Session identifier,
      but relies either on the underlying transport or on the used
      signalling protocol, depending on in which context the identifier
      is used (e.g. transport or signalling).  Due to this, a single RTP
      Session may have multiple associated identifiers belonging to
      different contexts.

      Position:  Depending on underlying transport and signalling
         protocol.  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).
         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.
         Another example is SDP signalling [RFC4566], where the grouping
         framework [RFC5888] uses an identifier per "m="-line.  If there
         is a one-to-one mapping between "m="-line and RTP Session, that
         grouping framework identifier can identify a single RTP
         Session.

      Usage:  Identify separate RTP Sessions.






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      Uniqueness:  Globally unique within the general communication
         context for the specific end-point.

      Inter-relation:  Depending on the underlying transport and
         signalling protocol.

   Special Restrictions:  None.

   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.

   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

   An RTP Session serves one or more Media Sources, each sending a Media
   Stream.

   Identifier:  Synchronization Source (SSRC), 32-bit unsigned number.

      Position:  In every RTP and RTCP packet header.  May be present in
         RTCP payload.  May be present in SDP signalling.

      Usage:  Identify individual Media Sources within an RTP Session.
         Refer to individual Media Sources in RTCP messages and SDP
         signalling.

      Uniqueness:  Randomly chosen, globally unique within an RTP
         Session and not dependent on network address.

      Inter-relation:  SSRC belonging to the same synchronization
         context (originating from the same end-point), within or
         between RTP Sessions, are indicated through use of identical
         SDES CNAME items in RTCP compound packets with those SSRC as
         originating source.  SDP signalling can provide explicit SSRC
         grouping [RFC5576].  When CNAME is inappropriate or
         insufficient, there exist a few other methods to relate
         different SSRC.  One such case is session-based RTP
         retransmission [RFC4588].  In some cases, the same SSRC
         Identifier value is used to relate streams in two different RTP



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         Sessions, such as in Multi-Session Transmission of scalable
         video [RFC6190].

   Special Restrictions:  All RTP implementations must be prepared to
      use procedures for SSRC collision handling, which results in an
      SSRC number change.  A Media Source that changes its RTP Session
      identifier (e.g. source transport address) during a session must
      also choose a new SSRC identifier to avoid being interpreted as
      looped source.  Note that RTP sequence number and RTP timestamp
      are scoped by SSRC and thus independent between different SSRCs.

   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.

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

   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



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   [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

   Each Media Stream can be represented in various encoding formats.

   Identifier:  Payload Type number.

      Position:  In every RTP header and in SDP signalling.

      Usage:  Identify a specific Media Stream encoding format.  The
         format definition may be taken from [RFC3551] for statically
         allocated Payload Types, but should be explicitly defined in
         signalling, such as SDP, both for static and dynamic Payload
         Types.  The term "format" here includes whatever can be
         described by out-of-band signaling means.  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].

      Uniqueness:  Scoped by sending end-point within an RTP Session.
         To avoid any potential for ambiguity, it is desirable that
         payload types are unique across all sending end-points within
         an RTP session, but this is often not true in practice.  All
         SSRC in an RTP session sent from an single end-point share the
         same Payload Types definitions.  The RTP Payload Type is
         designed such that only a single Payload Type is valid at any
         time instant in the SSRC's RTP timestamp time line, effectively
         time-multiplexing different Payload Types if any change occurs.
         Used Payload Type may change on a per-packet basis for an SSRC,
         for example a speech codec making use of generic Comfort Noise



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         [RFC3389].

      Inter-relation:  There are some uses where Payload Type numbers
         need be unique across RTP Sessions.  This is for example the
         case in Media Decoding Dependency [RFC5583] where Payload Types
         are used to describe media dependency across RTP Sessions.
         Another example is session-based RTP retransmission [RFC4588].

   Special Restrictions:  Using different RTP timestamp clock rates 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 is 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 Appendix A.  The RTP payload type alone is 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



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   referenced and reported on.

   This leaves us with two main choices, either using SSRC multiplexing
   to have multiple SSRCs from one end-point in one RTP session, or
   create an additional RTP session 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

   o  Retransmission streams

   o  FEC stream

   o  Alternative Encodings

   o  Scalability layers

   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 two
   end-points.  Each end-point has one or more SSRCs.

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

                         Figure 1: 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 to 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 has 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 is 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 to point to point on a
   transport level but is different on RTP level.  Assume a first
   scenario with two SSRC being sent from an end-point to a Transport
   Translator, and a second scenario with two single SSRC remote end-
   points sending to the same Transport Translator.  The main
   differences between those two scenarios are that in the second



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   scenario, the RTT may vary between the SSRCs (but it is not
   guaranteed), and the SSRCs may also have different CNAMEs.

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).
   There are large commercial deployments of multicast for applications
   like IPTV.

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

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

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

       Figure 3: 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 to 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 for large groups
   is to use a set of multicast groups, where each multicast group
   represents a particular bit-rate.  Otherwise the whole group gets
   media adapted to the participant with the worst conditions.  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 streams
   that improve transport robustness 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 possible 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 |
                    +---+      +------------+      +---+

                  Figure 4: 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, as it
   creates session inconsistencies.  If some middlebox temporarily stops
   a stream, this prevents some receivers from reporting on it.  From
   the senders perspective it will look like a transport failure.
   Applications having needs to stop or switch streams in the central
   node should consider using an RTP mixer to avoid this issue.

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



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   choose to do that.

   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.  The transport translator has a signalling
   requirement that also exist in any source multicast; all of the
   participants will need to have the same RTP and payload type
   configuration.  Otherwise, A could for example be using payload type
   97 as the video codec H.264 while B thinks it is MPEG-2.  It should
   be noted that SDP offer/answer [RFC3264] has issues with ensuring
   this property.

   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 translator can
   transcode to a different bit-rate, transcode to use 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 |
                    +---+      +------------+      +---+

                            Figure 5: RTP Mixer



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   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.

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 |
                                   +---+

     Figure 6: Point to Multi-Point using Multiple Unicast Transports

   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.

   There exists an alternative structure for establishing the above
   communication scenario (Figure 6) which uses independent RTP sessions
   between each pair of peers, i.e. three different RTP sessions.
   Unless independently adapted the same RTP media stream could be sent
   in both of the RTP sessions an end-point has.  The difference exists
   in the behaviors around RTCP, for example common RTCP bandwidth for
   one joint session, rather than three independent pools, and the
   awareness based on RTCP reports between the peers of how that third
   leg is doing.






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5.6.  De-composite 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.

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

                     Figure 7: De-composite 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, 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 is not 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.



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   There exist another use case which might be considered as a de-
   composite 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.

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

               Figure 8: When De-composition 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.  Multiple Streams Discussion

6.1.  Introduction

   Using multiple media streams is a well supported feature of RTP.
   However, it can be unclear for most implementers or people writing
   RTP/RTCP applications or extensions attempting to apply multiple
   streams 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.

6.2.  RTP/RTCP Aspects

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

6.2.1.  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



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   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.

   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."



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   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 Appendix A.

   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 are of one media type and
   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 6.5.  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.

6.2.1.1.  Different Media Types Recommendations

   The above quote from RTP [RFC3550] includes a strong recommendation:

      "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."

   It has been identified in "Why RTP Sessions Should Be Content
   Neutral" [I-D.alvestrand-rtp-sess-neutral] that the above statement
   is poorly supported by any of the motivations provided in the RTP
   specification.  This document has a more detailed analysis of



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   potential issues in having multiple media types in the same RTP
   session in Section 6.7.  An important influence for underlying
   thinking for the RTP design and likely this statement can be found in
   the academic paper by David Clark and David Tennenhouse
   "Architectural considerations for a new generation of protocols"
   [ALF].

6.2.2.  Handling Varying sets of Senders

   A potential issue that some application designers may need to
   consider is the case where the 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 will cause similar issues independent of SSRC or
   Session multiplexing.  It might be possible in certain applications
   to limit the changes to a subset of communication session
   participants by have the sub-set use particular RTP Sessions.

6.2.3.  Cross Session RTCP Requests

   There currently exists no functionality to make truly synchronized
   and atomic RTCP messages with some type of request semantics 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 messages 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, forcing it to use more
   robust and forgiving mechanisms.

6.2.4.  Binding Related Sources

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

   The current solutions do not have these properties.  There exists one



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   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 is an issue as this
   behavior does not fit all the RTP extensions or payload formats.  It
   is possible to create a situation where the different mechanisms
   cannot be combined due to the non standard SSRC allocation behavior.

   Existing mechanisms with known issues:



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   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 to bind media and
      retransmission streams together.  However, if multiple media
      streams are sent from the same end-point in the same session this
      does not provide non-ambiguous binding.  Therefore when the first
      retransmission request for a media stream is sent, one must not
      have another retransmission request outstanding for an SSRC which
      don't have a 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 be explicit on how these
      layers are bound together in cases where CNAME is not sufficient.
      CNAME is no longer sufficient when more than one media source
      occur within a session that has 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 its 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].

6.2.5.  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



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   achieved by transmitting a certain amount of redundant information
   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 create 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, in the sense that it
   supports the case where 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 caught 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.

6.2.6.  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.

6.3.  Interworking

   There are several different kinds of interworking, and this section
   discusses two related ones.  The interworking between different
   applications and the implications of potentially different choices of



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   usage of RTP's multiplexing points.  The second topic relates to what
   limitations may have to be considered working with some legacy
   applications.

6.3.1.  Interworking Applications

   It is not uncommon that applications or services of similar usage,
   especially the ones intended for interactive communication, ends up
   in a situation where one want to interconnect two or more of these
   applications.  From an RTP perspective this could be problem free if
   all the applications have made the same multiplexing choices, have
   the same capabilities in number of simultaneous media streams
   combined with the same set of RTP/RTCP extensions being supported.
   Unfortunately this may not always be true.

   In these cases one ends up in a situation where one might use a
   gateway to interconnect applications.  This gateway then needs to
   change the multiplexing structure or adhere to limitations in each
   application.  If one's goal is to make minimal amount of work in such
   a gateway, there are some multiplexing choices that one should avoid.
   The lowest amount of work represents solutions where one can take an
   SSRC from one RTP session in one application and forward it into
   another RTP session.  For example if one has one application that has
   multiple SSRCs for one media type in one session and another
   application that instead has chosen to use multiple RTP sessions with
   only a single SSRC per end-point in each of these sessions.  Then
   mapping an SSRC from the side with one session into an RTP session is
   possible.  However mapping SSRC from different RTP sessions into a
   single RTP session has the potential of creating SSRC collisions,
   especially if an end-point has not generated independent random SSRC
   values in each RTP session.  This issue is even more likely in a case
   where one side uses a single RTP session with multiple media types
   and the other uses different RTP session for different media or
   robustness mechanism such as retransmission [RFC4588].  Then it is
   more likely or maybe even required to use the same SSRC in the
   different RTP sessions.

   In cases where the used structure is incompatible, the gateway will
   need to make SSRC translation.  Thus this incurs overhead and some
   potential loss of functionality.  First of all, if one translates the
   SSRC in an RTP header then one will be forced to decrypt and re-
   encrypt if one uses SRTP and thus also needs to be part of the
   security association.  Secondly, changing the SSRC also means that
   one needs to translate all RTCP messages.  This can be more complex,
   but important so that the gateway does not end up having to terminate
   the end-to-end RTCP chain.  In that case the gateway will need to be
   able to take the role of a true end-point in each session, which may
   include functions such as bit-rate adaptation and correctly respond



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   to whatever RTCP extensions are being used, and then translate them
   or locally respond to them.  Thirdly, an SSRC translation may require
   that one changes RTP payloads; for example, an RTP retransmission
   packet contains an original sequence number that must match the
   sequence number used in for the corresponding packet with the new
   SSRC.  And for FEC packets this is even worse, as the original SSRC
   is included as part of the data for which FEC redundant data is
   calculated.  A fourth issue is the potential for these gateways to
   block evolution of the applications by blocking unknown RTP and RTCP
   extensions that the regular application has been extended with.

   If one uses security functions, like SRTP, they can as seen above
   incur both additional risk due to the gateway needing to be in
   security association between the end-points, unless the gateway is on
   the transport level, and additional complexities in form of the
   decrypt-encrypt cycles needed for each forwarded packet.  SRTP, due
   to its keying structure, also makes it hard to move a flow from one
   RTP session to another as each RTP session will have one or more
   different master keys and these must not be the same in multiple RTP
   sessions as that can result in two-time pads that completely breaks
   the confidentiality of the packets.

   An additional issue around interworking is that for multi-party
   applications it can be impossible to judge which different RTP
   multiplexing behaviors that will be used by end-points that attempt
   to join a session.  Thus if one attempts to use a multiplexing choice
   that has poor interworking, one may have to switch at a later stage
   when someone wants to participate in a multi-party session using an
   RTP application supporting only another behavior.  It is likely
   difficult to implement the switch without some media disruption.

   To summarize, certain types of applications are likely to be inter-
   worked.  Sets of applications of similar type should strive to use
   the same multiplexing structure to avoid the need to make an RTP
   session level gateway.  This as it incurs complexity costs, can force
   the gateway to be part of security associations, force SSRC
   translation and even payload translation which is also a potential
   hinder to application evolution.

6.3.2.  Multiple SSRC Legacy Considerations

   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



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   RTP mixer.

   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, adding robustness 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.

6.4.  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, while 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.

6.4.1.  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 those are not necessarily strictly bound to an
   RTP session and could be of interest to signal specifically for a
   particular media stream (SSRC) within the session.  The following
   properties have been identified as being potentially useful to signal
   not only on RTP session level:




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   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
   only, 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 application's 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.

6.4.2.  SDP Prevents Multiple Media Types

   SDP encoded in its structure 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 an accepted WG item in the MMUSIC WG to define how multiple
   media lines describe a single underlying transport
   [I-D.holmberg-mmusic-sdp-bundle-negotiation] and thus it becomes
   possible in SDP to define one RTP session with multiple media types.

6.4.3.  Media Stream Usage

   Media streams being transported in RTP has some particular usage in
   an RTP application.  This usage of the media stream is in many
   applications so far implicitly signalled.  For example by having all
   audio media streams arriving in the only audio RTP session they are
   to be decoded, mixed and played out.  However, in more advanced
   applications that use multiple media streams there will be more than
   a single usage or purpose among the set of media streams being sent
   or received.  RTP applications will need to signal this usage
   somehow.  Here the choice of SSRC multiplexing versus session
   multiplexing will have significant impact.  If one uses SSRC
   multiplexing to its full extent one will have to explicitly indicate



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   for each SSRC what its' usage and purpose are using some signalling
   between the application instances.

   This SSRC usage signalling will have some impact on the application
   and also on any central RTP nodes.  It is important in the design to
   consider the implications of the need for additional signalling
   between the nodes.  One consideration is if a receiver can utilize
   the media stream at all before it has received the signalling message
   describing the media stream and its usage.  Another consideration is
   that any RTP central node, like an RTP mixer or translator that
   selects, mixes or processes streams, in most cases will need to
   receive the same signalling to know how to treat media streams with
   different usage in the right fashion.

   Application designers should consider putting media streams of the
   same usage and/or receiving the same treatment in middleboxes in the
   same RTP sessions and use the RTP session as an explicit indication
   of how to deal with media streams.  By having session level
   indication of usage and have different RTP sessions for different
   usages, the need for stream specific signalling can be reduced.
   Especially signalling of the type that is time critical and needs to
   be provided prior to the media stream being available.

6.5.  Network Aspects

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

6.5.1.  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.







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6.5.2.  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:

   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



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      immediate triggered check back.  This as ICE first finds one
      candidate pair that works prior to establish multiple flows.
      Thus, there is no extra time until one has found a working
      candidate pair.  Based on that working pair the needed extra time
      is to in parallel establish the, in most cases 2-3, additional
      flows.

   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.

   Deep Packet Inspection and Multiple Streams:  Firewalls differ in how
      deeply they inspect packets.  There exist some potential that
      deeply inspecting firewalls will have similar legacy issues with
      multiple SSRCs as some stack implementations.

   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].

6.5.3.  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 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



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   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.

   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.

6.5.4.  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 and where usage of multiple media
   types in one RTP session is not suitable, 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.

6.6.  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.

6.6.1.  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
   no one 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.



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   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.

   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.

6.6.2.  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 its 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] is 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.

6.6.3.  Complexity Implications

   The usage of security functions can surface complexity implications
   of the choice of multiplexing and topology.  This becomes especially
   evident in RTP topologies having any type of middlebox that processes
   or modifies RTP/RTCP packets.  Where there is very small overhead for
   a not secured RTP translator or mixer to rewrite an SSRC value in the
   RTP packet, the cost of doing it when using cryptographic security
   functions is higher.  For example if using SRTP [RFC3711], the actual
   security context and exact crypto key are determined by the SSRC
   field value.  If one changes it, the encryption and authentication



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   tag must be performed using another key.  Thus changing the SSRC
   value implies a decryption using the old SSRC and its security
   context followed by an encryption using the new one.

   There exist many valid cases where a middlebox will be forced to
   perform such cryptographic operations due to the intended purpose of
   the middlebox, for example a media transcoding RTP translator cannot
   avoid performing these operations as they will produce a different
   payload compared to the input.  However, there exist some cases where
   another topology and/or multiplexing choice could avoid the
   complexities.

6.7.  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 earlier
   discussed in Section 6.2.1.1.  When using multiple media types, there
   are a number of considerations:

   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 making it serve two purposes.
      First to provide the main media type and determining the
      processing chain, then later for the exact configuration of the
      encoder and packetization.

   Payload Type field limitations:  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 this may become an issue.

   An SSRC cannot use two clock rates simultaneously:  The used RTP
      clock rate for an SSRC is determined from the payload type.  As
      discussed in Appendix A it is not possible to simultaneously use
      two different clock rates for the same SSRC.  Even switching clock
      rate once has potential issues if packet loss occurs at the same
      time.  Different media types commonly have different clock rates
      preventing or creating issues to use two different media types for
      the same SSRC.

   Do not 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,



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      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.  So this is a behavior that shall be
      avoided.

   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.

   De-composite end-points:  De-composite 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 de-composite end-point will receive
      all media, thus doubling the bit-rate consumption for the end-
      point.

   Flow based QoS Separation:  Flow based QoS mechanisms will see all
      the media streams in the RTP session as part of a single flow.
      Therefore there is no possibility to provide separated QoS
      behavior for the different media types or flows.

   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.

   Legacy Implementations:  The use of multiple media types has the
      potential for even larger issues with legacy implementations than
      single media type SSRC multiplexing due to the occurrence of
      multiple media types among the payload type configurations.

   As can be seen, there is nothing in here that prevents using a single
   RTP session for multiple media types, however it does create a number
   of limitations and special case implementation requirements.  So
   anyone considering using this setup should carefully review if the
   reasons for using a single RTP session are sufficient to motivate the
   needed special handling.






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7.  Arch-Types

   This section discusses some arch-types of how RTP multiplexing can be
   used in applications to achieve certain goals and a summary of their
   implications.  For each arch-type there is discussion of benefits and
   downsides.

7.1.  Single SSRC per Session

   In this arch-type each end-point in a point-to-point session has only
   a single SSRC, thus the RTP session contains only two SSRCs, one
   local and one remote.  This session can be used both unidirectional,
   i.e. only a single media stream or bi-directional, i.e. both end-
   points have one media stream each.  If the application needs
   additional media flows between the end-points, they will have to
   establish additional RTP sessions.

   The Pros:

   1.  This arch-type has great legacy interoperability potential as it
       will not tax any RTP stack implementations.

   2.  The signalling has good possibilities to negotiate and describe
       the exact formats and bit-rates for each media stream, especially
       using today's tools in SDP.

   3.  It does not matter if usage or purpose of the media stream is
       signalled on media stream level or session level as there is no
       difference.

   4.  It is possible to control security association per RTP session
       with current key-management.

   The Cons:

   a.  The number of required RTP sessions cannot really be higher,
       which has the implications:

       *  Linear growth of the amount of NAT/FW state with number of
          media streams.

       *  Increased delay and resource consumption from NAT/FW
          traversal.

       *  Likely larger signalling message and signalling processing
          requirement due to the amount of session related information.





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       *  Higher potential for a single media stream to fail during
          transport between the end-points.

   b.  When the number of RTP sessions grows, the amount of explicit
       state for relating media stream also grows, linearly or possibly
       exponentially, depending on how the application needs to relate
       media streams.

   c.  The port consumption may become a problem for centralized
       services, where the central node's port consumption grows rapidly
       with the number of sessions.

   d.  For applications where the media streams are highly dynamic in
       their usage, i.e. entering and leaving, the amount of signalling
       can grow high.  Issues arising from the timely establishment of
       additional RTP sessions can also arise.

   e.  Cross session RTCP requests needs is likely to exist and may
       cause issues.

   f.  If the same SSRC value is reused in multiple RTP sessions rather
       than being randomly chosen, interworking with applications that
       uses another multiplexing structure than this application will
       have issues and require SSRC translation.

   g.  Cannot be used with Any Source Multicast (ASM) as one cannot
       guarantee that only two end-points participate as packet senders.
       Using SSM, it is possible to restrict to these requirements if no
       RTCP feedback is used.

   h.  For most security mechanisms, each RTP session or transport flow
       requires individual key-management and security association
       establishment thus increasing the overhead.

   i.  Does not support multiparty session within a session.  Instead
       each multi-party participant will require an individual RTP
       session to a given end-point, even if a central node is used.

   RTP applications that need to inter-work with legacy RTP
   applications, like VoIP and video conferencing, can potentially
   benefit from this structure.  However, a large number of media
   descriptions in SDP can also run into issues with existing
   implementations.  For any application needing a larger number of
   media flows, the overhead can become very significant.  This
   structure is also not suitable for multi-party sessions, as any given
   media stream from each participant, although having same usage in the
   application, must have its own RTP session.  In addition, the dynamic
   behavior that can arise in multi-party applications can tax the



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   signalling system and make timely media establishment more difficult.

7.2.  Multiple SSRCs of the Same Media Type

   In this arch-type, each RTP session serves only a single media type.
   The RTP session can contain multiple media streams, either from a
   single end-point or due to multiple end-points.  This commonly
   creates a low number of RTP sessions, typically only two one for
   audio and one for video with a corresponding need for two listening
   ports when using RTP and RTCP multiplexing.

   The Pros:

   1.  Low number of RTP sessions needed compared to single SSRC case.
       This implies:

       *  Reduced NAT/FW state

       *  Lower NAT/FW Traversal Cost in both processing and delay.

   2.  Allows for early de-multiplexing in the processing chain in RTP
       applications where all media streams of the same type have the
       same usage in the application.

   3.  Works well with media type de-composite end-points.

   4.  Enables Flow-based QoS with different prioritization between
       media types.

   5.  For applications with dynamic usage of media streams, i.e. they
       come and go frequently, having much of the state associated with
       the RTP session rather than an individual SSRC can avoid the need
       for in-session signalling of meta-information about each SSRC.

   6.  Low overhead for security association establishment.

   The Cons:

   a.  May have some need for cross session RTCP requests for things
       that affect both media types in an asynchronous way.

   b.  Some potential for concern with legacy implementations that does
       not support the RTP specification fully when it comes to handling
       multiple SSRC per end-point.

   c.  Will not be able to control security association for sets of
       media streams within the same media type with today's key-
       management mechanisms, only between SDP media descriptions.



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   For RTP applications where all media streams of the same media type
   share same usage, this structure provides efficiency gains in amount
   of network state used and provides more faith sharing with other
   media flows of the same type.  At the same time, it is still
   maintaining almost all functionalities when it comes to negotiation
   in the signalling of the properties for the individual media type and
   also enabling flow based QoS prioritization between media types.  It
   handles multi-party session well, independently of multicast or
   centralized transport distribution, as additional sources can
   dynamically enter and leave the session.

7.3.  Multiple Sessions for one Media type

   In this arch-type one goes one step further than in the above
   (Section 7.2) by using multiple RTP sessions also for a single media
   type.  The main reason for going in this direction is that the RTP
   application needs separation of the media streams due to their usage.
   Some typical reasons for going to this arch-type are scalability over
   multicast, simulcast, need for extended QoS prioritization of media
   streams due to their usage in the application, or the need for fine
   granular signalling using today's tools.

   The Pros:

   1.  More suitable for Multicast usage where receivers can
       individually select which RTP sessions they want to participate
       in, assuming each RTP session has its own multicast group.

   2.  Detailed indication of the application's usage of the media
       stream, where multiple different usages exist.

   3.  Less need for SSRC specific explicit signalling for each media
       stream and thus reduced need for explicit and timely signalling.

   4.  Enables detailed QoS prioritization for flow based mechanisms.

   5.  Works well with de-composite end-points.

   6.  Handles dynamic usage of media streams well.

   7.  For transport translator based multi-party sessions, this
       structure allows for improved control of which type of media
       streams an end-point receives.

   8.  The scope for who is included in a security association can be
       structured around the different RTP sessions, thus enabling such
       functionality with existing key-management.




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   The Cons:

   a.  Increases the amount of RTP sessions compared to Multiple SSRCs
       of the Same Media Type.

   b.  Increased amount of session configuration state.

   c.  May need synchronized cross-session RTCP requests and require
       some consideration due to this.

   d.  For media streams that are part of scalability, simulcast or
       transport robustness it will be needed to bind sources, which
       must support multiple RTP sessions.

   e.  Some potential for concern with legacy implementations that does
       not support the RTP specification fully when it comes to handling
       multiple SSRC per end-point.

   f.  Higher overhead for security association establishment.

   g.  If the applications need finer control than on media type level
       over which session participants that are included in different
       sets of security associations, most of today's key-management
       will have difficulties establishing such a session.

   For more complex RTP applications that have several different usages
   for media streams of the same media type and / or uses scalability or
   simulcast, this solution can enable those functions at the cost of
   increased overhead associated with the additional sessions.  This
   type of structure is suitable for more advanced applications as well
   as multicast based applications requiring differentiation to
   different participants.

7.4.  Multiple Media Types in one Session

   This arch-type is to use a single RTP session for multiple different
   media types, like audio and video, and possibly also transport
   robustness mechanisms like FEC or Retransmission.  Each media stream
   will use its own SSRC and a given SSRC value from a particular end-
   point will never use the SSRC for more than a single media type.

   The Pros:

   1.  Single RTP session which implies:

       *  Minimal NAT/FW state.





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       *  Minimal NAT/FW Traversal Cost.

       *  Fate-sharing for all media flows.

   2.  Enables separation of the different media types based on the
       payload types so media type specific end-point or central
       processing can still be supported despite single session.

   3.  Can handle dynamic allocations of media streams well on an RTP
       level.  Depends on the application's needs for explicit
       indication of the stream usage and how timely that can be
       signalled.

   4.  Minimal overhead for security association establishment.

   The Cons:

   a.  Not suitable for interworking with other applications that uses
       individual RTP sessions per media type or multiple sessions for a
       single media type, due to high risk of forced SSRC translation.

   b.  Negotiation of bandwidth for the different media types is
       currently not possible in SDP.  This requires SDP extensions to
       enable payload or source specific bandwidth.  Likely to be a
       problem due to media type asymmetry in required bandwidth.

   c.  Does enforce higher bandwidth and processing on de-composite end-
       points.

   d.  Flow based QoS cannot provide separate treatment to some media
       streams compared to other in the single RTP session.

   e.  If there is significant asymmetry between the media streams RTCP
       reporting needs, there are some challenges in configuration and
       usage to avoid wasting RTCP reporting on the media stream that
       does not need that frequent reporting.

   f.  Not suitable for applications where some receivers like to
       receive only a subset of the media streams, especially if
       multicast or transport translator is being used.

   g.  Additional concern with legacy implementations that does not
       support the RTP specification fully when it comes to handling
       multiple SSRC per end-point, as also multiple simultaneous media
       types needs to be handled.

   h.  If the applications need finer control over which session
       participants that are included in different sets of security



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       associations, most key-management will have difficulties
       establishing such a session.

   The analysis in this document and considerations in Section 6.7
   implies that this is suitable only in a set of restricted use cases.
   The aspect in the above list that can be most difficult to judge long
   term is likely the potential need for interworking with other
   applications and services.

7.5.  Summary

   There are some clear relations between these arch-types.  Both the
   "single SSRC per RTP session" and the "multiple media types in one
   session" are cases which require full explicit signalling of the
   media stream relations.  However, they operate on two different
   levels where the first primarily enables session level binding, and
   the second needs to do it all on SSRC level.  From another
   perspective, the two solutions are the two extreme points when it
   comes to number of RTP sessions required.

   The two other arch-types "Multiple SSRCs of the Same Media Type" and
   "Multiple Sessions for one Media Type" are examples of two other
   cases that first of all allows for some implicit mapping of the role
   or usage of the media streams based on which RTP session they appear
   in.  It thus potentially allows for less signalling and in particular
   reduced need for real-time signalling in dynamic sessions.  They also
   represent points in between the first two when it comes to amount of
   RTP sessions established, i.e. representing an attempt to reduce the
   amount of sessions as much as possible without compromising the
   functionality the session provides both on network level and on
   signalling level.


8.  Guidelines

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

   Do not Require the same SSRC across Sessions:  As discussed in
      Section 6.2.4 there exist drawbacks in using the same SSRC in
      multiple RTP sessions as a mechanism to bind related media streams
      together.  It is instead recommended that a mechanism to
      explicitly signal the relation is used, either in RTP/RTCP or in
      the used signalling mechanism that establishes the RTP session(s).







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   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 tele-presence 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.

   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], using some appropriate grouping semantics.

   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.  Any extension intended to be generic
      is 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 which of the multiplexing choices a particular solution
      supports.


9.  Proposal for Future Work

   The above discussion and guidelines indicates 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



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      item SRCNAME that also can be used with the a=ssrc SDP attribute
      to provide signalling layer binding information.

   SSRC limitations 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].


10.  RTP Specification Clarifications

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

10.1.  RTCP Reporting from all SSRCs

   When one have multiple SSRC in an RTP node, 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.

10.2.  RTCP Self-reporting

   For any RTP node that sends more than one SSRC, there is 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.

10.3.  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



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   send.


11.  IANA Considerations

   This document makes no request of IANA.

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


12.  Security Considerations

   There is discussion of the security implications of choosing SSRC vs
   Session multiplexing in Section 6.6.


13.  Acknowledgements

   The authors would like to thanks Harald Alvestrand for providing
   input into the discussion regarding multiple media types in a single
   RTP session.


14.  References

14.1.  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.

14.2.  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.alvestrand-rtp-sess-neutral]
              Alvestrand, H., "Why RTP Sessions Should Be Content
              Neutral", draft-alvestrand-rtp-sess-neutral-00 (work in
              progress), December 2011.



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   [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]
              Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
              Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
              (work in progress), October 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-02
              (work in progress), January 2012.

   [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]
              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.




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   [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.

   [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.



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   [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.

   [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.


Appendix A.  Dismissing Payload Type Multiplexing

   This section documents a number of reasons why using the payload type
   as a multiplexing point for most things related to multiple streams
   is unsuitable.  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



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   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/
        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.



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   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.

   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.


Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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












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   Bo Burman
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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


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

   Email: csp@csperkins.org


































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