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Versions: (draft-westerlund-avtcore-multiplex-architecture) 00 01 02 03

Network Working Group                                      M. Westerlund
Internet-Draft                                                 B. Burman
Intended status: Informational                                  Ericsson
Expires: April 11, 2015                                       C. Perkins
                                                   University of Glasgow
                                                           H. Alvestrand
                                                                  Google
                                                        October 08, 2014


    Guidelines for using the Multiplexing Features of RTP to Support
                         Multiple Media Streams
               draft-ietf-avtcore-multiplex-guidelines-03

Abstract

   The Real-time Transport Protocol (RTP) is a flexible protocol that
   can be used in a wide range of applications, networks, and system
   topologies.  That flexibility makes for wide applicability, but can
   complicate the application design process.  One particular design
   question that has received much attention is how to support multiple
   media streams in RTP.  This memo discusses the available options and
   design trade-offs, and provides guidelines on how to use the
   multiplexing features of RTP to support multiple media streams.

Status of This Memo

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

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

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

   This Internet-Draft will expire on April 11, 2015.

Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.





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   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.1.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   4
     2.2.  Subjects Out of Scope . . . . . . . . . . . . . . . . . .   6
   3.  Reasons for Multiplexing and Grouping RTP Media Streams . . .   6
   4.  RTP Multiplexing Points . . . . . . . . . . . . . . . . . . .   7
     4.1.  RTP Session . . . . . . . . . . . . . . . . . . . . . . .   8
     4.2.  Synchronisation Source (SSRC) . . . . . . . . . . . . . .   9
     4.3.  Contributing Source (CSRC)  . . . . . . . . . . . . . . .  10
     4.4.  RTP Payload Type  . . . . . . . . . . . . . . . . . . . .  11
   5.  RTP Topologies and Issues . . . . . . . . . . . . . . . . . .  12
     5.1.  Point to Point  . . . . . . . . . . . . . . . . . . . . .  12
     5.2.  Translators & Gateways  . . . . . . . . . . . . . . . . .  13
     5.3.  Point to Multipoint Using Multicast . . . . . . . . . . .  13
     5.4.  Point to Multipoint Using an RTP Transport Translator . .  14
     5.5.  Point to Multipoint Using an RTP Mixer  . . . . . . . . .  15
   6.  RTP Multiplexing: When to Use Multiple RTP Sessions . . . . .  15
     6.1.  RTP and RTCP Protocol Considerations  . . . . . . . . . .  16
       6.1.1.  The RTP Specification . . . . . . . . . . . . . . . .  16
       6.1.2.  Multiple SSRCs in a Session . . . . . . . . . . . . .  18
       6.1.3.  Handling Varying Sets of Senders  . . . . . . . . . .  19
       6.1.4.  Cross Session RTCP Requests . . . . . . . . . . . . .  19
       6.1.5.  Binding Related Sources . . . . . . . . . . . . . . .  19
       6.1.6.  Forward Error Correction  . . . . . . . . . . . . . .  21
       6.1.7.  Transport Translator Sessions . . . . . . . . . . . .  21
     6.2.  Interworking Considerations . . . . . . . . . . . . . . .  21
       6.2.1.  Types of Interworking . . . . . . . . . . . . . . . .  22
       6.2.2.  RTP Translator Interworking . . . . . . . . . . . . .  22
       6.2.3.  Gateway Interworking  . . . . . . . . . . . . . . . .  22
       6.2.4.  Multiple SSRC Legacy Considerations . . . . . . . . .  23
     6.3.  Network Considerations  . . . . . . . . . . . . . . . . .  24
       6.3.1.  Quality of Service  . . . . . . . . . . . . . . . . .  24
       6.3.2.  NAT and Firewall Traversal  . . . . . . . . . . . . .  25
       6.3.3.  Multicast . . . . . . . . . . . . . . . . . . . . . .  27
       6.3.4.  Multiplexing multiple RTP Session on a Single
               Transport . . . . . . . . . . . . . . . . . . . . . .  27



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     6.4.  Security and Key Management Considerations  . . . . . . .  28
       6.4.1.  Security Context Scope  . . . . . . . . . . . . . . .  28
       6.4.2.  Key Management for Multi-party session  . . . . . . .  28
       6.4.3.  Complexity Implications . . . . . . . . . . . . . . .  29
   7.  Archetypes  . . . . . . . . . . . . . . . . . . . . . . . . .  29
     7.1.  Single SSRC per Session . . . . . . . . . . . . . . . . .  29
     7.2.  Multiple SSRCs of the Same Media Type . . . . . . . . . .  31
     7.3.  Multiple Sessions for one Media type  . . . . . . . . . .  32
     7.4.  Multiple Media Types in one Session . . . . . . . . . . .  34
     7.5.  Summary . . . . . . . . . . . . . . . . . . . . . . . . .  35
   8.  Summary considerations and guidelines . . . . . . . . . . . .  36
     8.1.  Guidelines  . . . . . . . . . . . . . . . . . . . . . . .  36
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  37
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  37
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  37
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  37
     11.2.  Informative References . . . . . . . . . . . . . . . . .  37
   Appendix A.  Dismissing Payload Type Multiplexing . . . . . . . .  41
   Appendix B.  Proposals for Future Work  . . . . . . . . . . . . .  43
   Appendix C.  Signalling considerations  . . . . . . . . . . . . .  44
     C.1.  Signalling Aspects  . . . . . . . . . . . . . . . . . . .  44
       C.1.1.  Session Oriented Properties . . . . . . . . . . . . .  44
       C.1.2.  SDP Prevents Multiple Media Types . . . . . . . . . .  45
       C.1.3.  Signalling Media Stream Usage . . . . . . . . . . . .  45
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  46

1.  Introduction

   The 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 needs to understand the implications that come
   from a particular usage of the RTP multiplexing points.  The document
   will recommend against some usages as being unsuitable, in general or
   for particular purposes.





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   RTP was from the beginning designed for multiple participants in a
   communication session.  This is not restricted to multicast, as some
   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 or protocol translation in
   addition to forwarding, or an RTP mixer that creates new sources from
   the received streams.  In addition, multiple streams can occur when a
   single endpoint 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
   behaviour and the implications of a particular behaviour depend on
   which topologies are used, which requires some consideration.  This
   is followed by a discussion of some choices in multiplexing behaviour
   and their impacts.  Some archetypes of RTP usage are discussed.
   Finally, some recommendations and examples are provided.

2.  Definitions

2.1.  Terminology

   The following terms and abbreviations are used in this document:

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

   Multiparty:  A communication situation including multiple endpoints.
      In this document it will be used to refer to situations where more
      than two endpoints communicate.

   Media Source:  The source of a stream of data of one Media Type, 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.



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      Sending data from a Media Source can cause multiple RTP sources to
      send multiple Media Streams.

   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.

   RTP Source:  The originator or source of a particular Media Stream.
      Identified using an SSRC in a particular RTP session.  An RTP
      source is the source of a single media stream, and is associated
      with a single endpoint and a single Media Source.  An RTP Source
      is just called a Source in RFC 3550.

   RTP Sink:  A recipient of a Media Stream.  The Media Sink is
      identified using one or more SSRCs.  There can be more than one
      RTP Sink for one RTP source.

   CNAME:  "Canonical name" - identifier associated with one or more RTP
      sources from a single endpoint.  Defined in the RTP specification
      [RFC3550].  A CNAME identifies a synchronisation context.  A CNAME
      is associated with a single endpoint, although some RTP nodes will
      use an endpoint's CNAME on that endpoints behalf.  An endpoint can
      use multiple CNAMEs.  A CNAME is intended to be globally unique
      and stable for the full duration of a communication session.
      [RFC6222][I-D.ietf-avtcore-6222bis] gives updated guidelines for
      choosing CNAMEs.

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

   Multiplexing:  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.

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

   RTP Session Group:  One or more RTP sessions that are used together
      to perform some function.  Examples are multiple RTP sessions used
      to carry different layers of a layered encoding.  In an RTP
      Session Group, CNAMEs are assumed to be valid across all RTP



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      sessions, and designate synchronisation contexts that can cross
      RTP sessions.

   Source:  Term that ought not be used alone.  An RTP Source, as
      identified by its SSRC, is the source of a single Media Stream; a
      Media Source can be the source of mutiple Media Streams.

   SSRC:  A 32-bit unsigned integer used as identifier for a RTP Source.

   CSRC:  Contributing Source, A SSRC identifier used in a context, like
      the RTP headers CSRC list, where it is clear that the Media Source
      is not the source of the media stream, instead only a contributor
      to the Media Stream.

   Signalling:  The process of configuring endpoints to participate in
      one or more RTP sessions.

2.2.  Subjects Out of Scope

   This document is focused on issues that affect RTP.  Thus, issues
   that involve signalling protocols, such as whether SIP, Jingle or
   some other protocol is in use for session configuration, the
   particular syntaxes used to define RTP session properties, or the
   constraints imposed by particular choices in the signalling
   protocols, are mentioned only as examples in order to describe the
   RTP issues more precisely.

   This document assumes the applications will use RTCP.  While there
   are such applications that don't send RTCP, they do not conform to
   the RTP specification, and thus can be regarded as reusing the RTP
   packet format but not implementing the RTP protocol.

3.  Reasons for Multiplexing and Grouping RTP Media Streams

   The reasons why an endpoint might choose to send multiple media
   streams are widespread.  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  Multiple Media Streams might be needed to represent one Media
      Source (for instance when using layered encodings)

   o  A Retransmission stream might repeat the content of another Media
      Stream





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   o  An FEC stream might provide material that can be used to repair
      another Media Stream

   o  Alternative Encodings, for instance different codecs for the same
      audio stream

   o  Alternative formats, for instance multiple resolutions of the same
      video stream

   For each of these, it is necessary to decide if each additional media
   stream gets its own SSRC multiplexed within a RTP Session, or if it
   is necessary to use additional RTP sessions to group the media
   streams.  The choice between these made due to one reason might 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.  As the
   discussion below will show, in reality we cannot choose a single one
   of the two solutions.  To utilise 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.

4.  RTP Multiplexing Points

   This section describes the multiplexing points present in the RTP
   protocol that can be used to distinguish media streams and groups of
   media streams.  Figure 1 outlines the process of demultiplexing
   incoming RTP streams:

                           |
                           | packets
           +--             v
           |        +------------+
           |        |   Socket   |
           |        +------------+
           |            ||  ||
      RTP  |       RTP/ ||  |+-----> SCTP ( ...and any other protocols)
   Session |       RTCP ||  +------> STUN (multiplexed using same port)
           +--          ||
           +--          ||
           |      (split by SSRC)
           |       ||   ||   ||
           |       ||   ||   ||
     Media |      +--+ +--+ +--+
   Streams |      |PB| |PB| |PB| Jitter buffer, process RTCP, FEC, etc.
           |      +--+ +--+ +--+



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           +--      |   |     |
           (pick rending context based on PT)
           +--      |  /      |
           |        +---+     |
           |         /  |     |
   Payload |      +--+ +--+ +--+
   Formats |      |CR| |CR| |CR| Codecs and rendering
           |      +--+ +--+ +--+
           +--

                   Figure 1: RTP Demultiplexing Process

4.1.  RTP Session

   An RTP Session is the highest semantic layer in the RTP protocol, and
   represents an association between a group of communicating endpoints.
   The set of participants that form an RTP session is defined as those
   that share a single synchronisation source space [RFC3550].  That is,
   if a group of participants are each aware of the synchronisation
   source identifiers belonging to the other participants, then those
   participants are in a single RTP session.  A participant can become
   aware of a synchronisation source identifier by receiving an RTP
   packet containing it in the SSRC field or CSRC list, by receiving an
   RTCP packet mentioning it in an SSRC field, or through signalling
   (e.g., the SDP "a=ssrc:" attribute).  Thus, the scope of an RTP
   session is determined by the participants' network interconnection
   topology, in combination with RTP and RTCP forwarding strategies
   deployed by the endpoints and any middleboxes, and by the signalling.

   RTP does not contain a session identifier.  Rather, it relies on the
   underlying transport layer to separate different sessions, and on the
   signalling to identify sessions in a manner that is meaningful to the
   application.  The signalling layer might give sessions an explicit
   identifier, or their identification might be implicit based on the
   addresses and ports used.  Accordingly, a single RTP Session can have
   multiple associated identifiers, explicit and implicit, belonging to
   different contexts.  For example, when running RTP on top of UDP/IP,
   an RTP endpoint can identify and delimit an RTP Session from other
   RTP Sessions using the UDP source and destination IP addresses and
   UDP port numbers.  Another example is when using SDP grouping
   framework [RFC5888] which uses an identifier per "m="-line; if there
   is a one-to-one mapping between "m="-lines and RTP sessions, that
   grouping framework identifier will identify an RTP Session.

   RTP sessions are globally unique, but their identity can only be
   determined by the communication context at an endpoint of the
   session, or by a middlebox that is aware of the session context.  The
   relationship between RTP sessions depending on the underlying



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   application, transport, and signalling protocol.  The RTP protocol
   makes no normative statements about the relationship between
   different RTP sessions, however the applications that use more than
   one RTP session will have some higher layer understanding of the
   relationship between the sessions they create.

4.2.  Synchronisation Source (SSRC)

   A synchronisation source (SSRC) identifies an RTP source or an RTP
   sink.  Every endpoint will have at least one synchronisation source
   identifier, even if it does not send media (endpoints that are only
   RTP sinks still send RTCP, and use their synchronisation source
   identifier in the RTCP packets they send).  An endpoint can have
   multiple synchronisation sources identifiers if it contains multiple
   RTP sources (i.e., if it sends multiple media streams).  Endpoints
   that are both RTP sources and RTP sinks use the same synchronisation
   sources in both roles.  At any given time, a RTP source has one and
   only one SSRC - although that can change over the lifetime of the RTP
   source or sink.

   The synchronisation Source identifier is a 32-bit unsigned integer.
   It is present in every RTP and RTCP packet header, and in the payload
   of some RTCP packet types.  It can also be present in SDP signalling.
   Unless pre-signalled using the SDP "a=ssrc:" attribute [RFC5576], the
   synchronisation source identifier is chosen at random.  It is not
   dependent on the network address of the endpoint, and is intended to
   be unique within an RTP session.  Synchronisation source identifier
   collisions can occur, and are handled as specified in [RFC3550] and
   [RFC5576], resulting in the synchronisation source identifier of the
   affecting RTP sources and/or sinks changing.  An RTP source that
   changes its RTP Session identifier (e.g.  source transport address)
   during a session has to choose a new SSRC identifier to avoid being
   interpreted as looped source.

   Synchronisation source identifiers that belong to the same
   synchronisation context (i.e., that represent media streams that can
   be synchronised using information in RTCP SR packets) are indicated
   by use of identical CNAME chunks in corresponding RTCP SDES packets.
   SDP signalling can also be used to provide explicit grouping of
   synchronisation sources [RFC5576].

      In some cases, the same SSRC Identifier value is used to relate
      streams in two different RTP Sessions, such as in Multi-Session
      Transmission of scalable video [RFC6190].  This is NOT RECOMMENDED
      since there is no guarantee of uniqueness in SSRC values across
      RTP sessions.





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   Note that RTP sequence number and RTP timestamp are scoped by the
   synchronisation source.  Each RTP source will have a different
   synchronisation source, and the corresponding media stream will have
   a separate RTP sequence number and timestamp space.

   An SSRC identifier is used by different type of sources as well as
   sinks:

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

   Processed Media Source:  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.

   RTP Sink:  A source that does not generate any RTP media stream in
      itself (e.g.  an endpoint or middlebox only receiving in an RTP
      session).  It still needs a sender SSRC for use as source in RTCP
      reports.

   Note that a endpoint that generates more than one media type, e.g.  a
   conference participant sending both audio and video, need not (and
   commonly does 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
   endpoint.  The main property attributed to SSRCs associated with the
   same CNAME is that they are from a particular synchronisation context
   and can be synchronised at playback.

   An RTP receiver receiving a previously unseen SSRC value will
   interpret it as a new source.  It might 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 ought to 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.

4.3.  Contributing Source (CSRC)

   The Contributing Source (CSRC) is not a separate identifier.  Rather
   a synchronisation source identifier is listed as a CSRC in the RTP
   header of a packet generated by an RTP mixer if the corresponding
   SSRC was in the header of one of the packets that contributed to the
   mix.





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   It is not possible, in general, to extract media represented by an
   individual CSRC 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.  The exception is the case
   when only a single CSRC is indicated as this represent forwarding of
   a media stream, possibly modified.  The RTP header extension for
   Mixer-to-Client Audio Level Indication [RFC6465] expands on the
   receivers information about a packet with a CSRC list.  Due to these
   restrictions, CSRC will not be considered a fully qualified
   multiplexing point and will be disregarded in the rest of this
   document.

4.4.  RTP Payload Type

   Each Media Stream utilises one or more RTP payload formats.  An RTP
   payload format describes how the output of a particular media codec
   is framed and encoded into RTP packets.  The payload format used is
   identified by the payload type field in the RTP data packet header.
   The combination therefore identifies a specific Media Stream encoding
   format.  The format definition can be taken from [RFC3551] for
   statically allocated payload types, but ought to 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 signalling 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].

   The payload type is scoped by sending endpoint within an RTP Session.
   All synchronisation sources sent from an single endpoint 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 RTP source's RTP timestamp time line, effectively time-
   multiplexing different Payload Types if any change occurs.  The
   payload type used can change on a per-packet basis for an SSRC, for
   example a speech codec making use of generic comfort noise [RFC3389].
   If there is a true need to send multiple Payload Types for the same
   SSRC that are valid for the same instant, 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 ought
   not exceed the transport MTU.

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

   The payload type is not a multiplexing point at the RTP layer (see
   Appendix A for a detailed discussion of why using the payload type as



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   an RTP multiplexing point does not work).  The RTP payload type is,
   however, used to determine how to render a media stream, and so can
   be viewed as selecting a rendering context.  The rendering context
   can be defined by the signalling, and the RTP payload type number is
   sometimes used to associate an RTP media stream with the signalling.
   This association is possible provided unique RTP payload type numbers
   are used in each context.  For example, an RTP media stream can be
   associated with an SDP "m=" line by comparing the RTP payload type
   numbers used by the media stream with payload types signalled in the
   "a=rtpmap:" lines in the media sections of the SDP.  If RTP media
   streams are being associated with signalling contexts based on the
   RTP payload type, then the assignment of RTP payload type numbers
   MUST be unique across signalling contexts; if the same RTP payload
   format configuration is used in multiple contexts, then a different
   RTP payload type number has to be assigned in each context to ensure
   uniqueness.  If the RTP payload type number is not being used to
   associated RTP media streams with a signalling context, then the same
   RTP payload type number can be used to indicate the exact same RTP
   payload format configuration in multiple contexts.

5.  RTP Topologies and Issues

   The impact of how RTP multiplexing is performed will in general vary
   with how the RTP Session participants are interconnected, described
   by RTP Topology [RFC5117] and its intended successor
   [I-D.westerlund-avtcore-rtp-topologies-update].

5.1.  Point to Point

   Even the most basic use case, denoted Topo-Point-to-Point in
   [I-D.westerlund-avtcore-rtp-topologies-update], raises a number of
   considerations that are discussed in detail below (Section 6).  They
   range over such aspects as:

   o  Does my communication peer support RTP as defined with multiple
      SSRCs?

   o  Do I need network differentiation in form of QoS?

   o  Can the application more easily process and handle the media
      streams if they are in different RTP sessions?

   o  Do I need to use additional media streams for RTP retransmission
      or FEC.

   o  etc.





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   The point to point topology can contain one to many RTP sessions with
   one to many media sources per session, each having one or more RTP
   sources per media source.

5.2.  Translators & Gateways

   A point to point communication can end up in a situation when the
   peer it is communicating with is not compatible with the other peer
   for various reasons:

   o  No common media codec for a media type thus requiring transcoding

   o  Different support for multiple RTP sources and RTP sessions

   o  Usage of different media transport protocols, i.e RTP or other.

   o  Usage of different transport protocols, e.g.  UDP, DCCP, TCP

   o  Different security solutions, e.g.  IPsec, TLS, DTLS, SRTP with
      different keying mechanisms.

   In many situations this is resolved by the inclusion of a translator
   between the two peers, as described by Topo-PtP-Translator in
   [I-D.westerlund-avtcore-rtp-topologies-update].  The translator's
   main purpose is to make the peer look to the other peer like
   something it is compatible with.  There can also be other reasons
   than compatibility to insert a translator in the form of a middlebox
   or gateway, for example a need to monitor the media streams.  If the
   stream transport characteristics are changed by the translator,
   appropriate media handling can require thorough understanding of the
   application logic, specifically any congestion control or media
   adaptation.

5.3.  Point to Multipoint Using Multicast

   The Point to Multi-point topology is using Multicast to interconnect
   the session participants.  This includes both Topo-ASM and Topo-SSM
   in [I-D.westerlund-avtcore-rtp-topologies-update].

   Special considerations need to be made as multicast is a one to many
   distribution system.  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



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   control the bit-rate sent on the path to itself.  It is also common
   that streams that improve transport robustness are sent in their own
   multicast group to allow for interworking with legacy or to support
   different levels of protection.

   The result of this is some common behaviours for RTP multicast:

   1.  Multicast applications use a group of RTP sessions, not one.
       Each endpoint will need to be a member of a number of RTP
       sessions in order to perform well.

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

   3.  Multicast applications need signalling functions to identify the
       relationships between RTP sessions.

   4.  Multicast applications need signalling functions to identify the
       relationships between SSRCs in different RTP sessions.

   All multicast configurations share a signalling requirement; all of
   the 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 is to be
   noted that SDP offer/answer [RFC3264] is not appropriate for ensuring
   this property.  The signalling aspects of multicast are not explored
   further in this memo.

   Security solutions for this type of group communications are also
   challenging.  First of all the key-management and the security
   protocol needs to support group communication.  Source authentication
   requires special solutions.  For more discussion on this please
   review Options for Securing RTP Sessions
   [I-D.ietf-avtcore-rtp-security-options].

5.4.  Point to Multipoint Using an RTP Transport Translator

   This mode is described as Topo-Translator in
   [I-D.westerlund-avtcore-rtp-topologies-update].

   Transport Translators (Relays) result in an RTP session situation
   that is very similar to how an ASM group RTP session would behave.

   One of the most important aspects with the simple relay is that it is
   only rewriting transport headers, no RTP modifications nor media
   transcoding occur.  The most obvious downside of this basic relaying
   is that the translator has no control over how many streams need to
   be delivered to a receiver.  Nor can it simply select to deliver only



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   certain streams, as this creates session inconsistencies: If the
   translator temporarily stops a stream, this prevents some receivers
   from reporting on it.  From the sender's perspective it will look
   like a transport failure.  Applications needing to stop or switch
   streams in the central node ought to consider using an RTP mixer to
   avoid this issue.

   The Transport Translator has the same signalling requirement as
   multicast: All participants need to have the same payload type
   configuration.  Most of the ASM security issues also arise here.
   Some alternatives when it comes to solution do exist, as there exists
   a central node to communicate with, one that also can enforce some
   security policies depending on the level of trust placed in the node.

5.5.  Point to Multipoint Using an RTP Mixer

   A mixer, described by Topo-Mixer in
   [I-D.westerlund-avtcore-rtp-topologies-update], is a centralised node
   that selects or mixes content in a conference to optimise the RTP
   session so that each endpoint only needs connect to one entity, the
   mixer.  The media sent from the mixer to the endpoint can be
   optimised in different ways.  These optimisations include methods
   like only choosing media from the currently most active speaker or
   mixing together audio so that only one audio stream is needed.

   Mixers have some downsides, the first is that the mixer has to be a
   trusted node as they repacketize the media, and can perform media
   transformation operations.  When using SRTP, both media operations
   and repacketization requires that the mixer verifies integrity,
   decrypts the content, performs the operation and forms new RTP
   packets, encrypts and integrity-protects them.  This applies to all
   types of mixers.  The second downside is that all these operations
   and optimisations of the session requires processing.  How much
   depends on the implementation, as will become evident below.

   A mixer, unlike a pure transport translator, is always application
   specific: the application logic for stream mixing or stream selection
   has to be embedded within the mixer, and controlled using application
   specific signalling.  The implementation of a mixer can take several
   different forms, as discussed below.

   A Mixer can also contain translator functionalities, like a media
   transcoder to adjust the media bit-rate or codec used for a
   particular RTP media stream.

6.  RTP Multiplexing: When to Use Multiple RTP Sessions





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   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.

6.1.  RTP and RTCP Protocol Considerations

   This section discusses RTP and RTCP aspects worth considering when
   selecting between using an additional SSRC and Multiple RTP sessions.

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

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

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

   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.




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   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 might also be
   appropriate to multiplex streams of the same medium using different
   SSRC values in other scenarios where the last two problems do not
   apply."

   Let's consider one argument at a time.  The first is an argument for
   using different SSRC for each individual media stream, which 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.  As we saw in the
   discussion of RTP mixers, the RTP mixer has to embed application
   logic in order to handle streams anyway; the separation of streams
   according to stream type is just another piece of application logic,
   which might or might not be appropriate for a particular application.
   A type of application that can mix different media sources "blindly"
   is the audio only "telephone" bridge; most other type of application
   needs application-specific logic to perform the mix correctly.

   The fifth argument discusses network aspects that we will discuss
   more below in Section 6.3.  It also goes into aspects of
   implementation, like decomposed endpoints where different processes



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   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 to use
   different RTP sessions for media streams that don't share a media
   type.  This document supports the first point; it is very valid.  The
   later is one thing which is further discussed in this document as
   something the application developer needs to make a conscious choice
   for, but where imposing a single solution on all usages of RTP is
   inappropriate.

6.1.1.1.  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 was 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 has resulted in the creation of a specification
   Multiple Media Types in an RTP Session specification
   [I-D.ietf-avtcore-multi-media-rtp-session] which intends to update
   this recommendation.  That document has a detailed analysis of the
   potential issues in having multiple media types in the same RTP
   session.  This document tries to provide an moreover arching
   consideration regarding the usage of RTP session and considers
   multiple media types in one RTP session as possible choice for the
   RTP application designer.

6.1.2.  Multiple SSRCs in a Session

   Using multiple SSRCs in an RTP session at one endpoint requires
   resolving some unclear aspects of the RTP specification.  These could
   potentially lead to some interoperability issues as well as some
   potential significant inefficencies.  These are further discussed in
   "RTP Considerations for Endpoints Sending Multiple Media Streams"
   [I-D.lennox-avtcore-rtp-multi-stream].  A application designer needs
   to consider these issues and the impact availability or lack of the
   optimization in the endpoints has on their application.

   If an application will become affected by the issues described, using
   Multiple RTP sessions can mitigate these issues.





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6.1.3.  Handling Varying Sets of Senders

   In some applications, the set of simultaneously active sources varies
   within a larger set of session members.  A receiver can then possibly
   try to use a set of decoding chains that is smaller than the number
   of senders, switching the decoding chains between different senders.
   As each media decoding chain can contain state, either the receiver
   needs to either be able to save the state of swapped-out senders, or
   the sender needs to be able to send data that permits the receiver to
   reinitialise when it resumes activity.

   This behaviour will cause similar issues independent of Additional
   SSRC or Multiple RTP session.

6.1.4.  Cross Session RTCP Requests

   There currently exists no functionality to make truly synchronised
   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 streams in the same RTP session
   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.

   When using multiple RTP sessions, 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.

6.1.5.  Binding Related Sources

   A common problem in a number of various RTP extensions has been how
   to bind related RTP sources and their media streams together.  This
   issue is common to both using additional SSRCs and Multiple RTP
   sessions.

   The solutions can be divided into some groups, RTP/RTCP based,
   Signalling based (SDP), grouping related RTP sessions, and grouping
   SSRCs within an RTP session.  Most solutions are explicit, but some
   implicit methods have also been applied to the problem.

   The SDP-based signalling solutions are:

   SDP Media Description Grouping:  The SDP Grouping Framework [RFC5888]
      uses various semantics to group any number of media descriptions.
      These has previously been considered primarily as grouping RTP
      sessions, but this might change.



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   SDP SSRC grouping:  Source-Specific Media Attributes in SDP [RFC5576]
      includes a solution for grouping SSRCs the same way as the
      Grouping framework groupes Media Descriptions.

   SDP MSID grouping:  Media Stream Identifiers [I-D.ietf-mmusic-msid]
      includes a solution for grouping SSRCs that is independent of
      their allocation to RTP sessions.

   This supports a lot of use cases.  All these solutions have
   shortcomings 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.

   Within RTP/RTCP based solutions when binding to a endpoint or
   synchronization context, i.e.  the CNAME has not be sufficient and
   one has multiple RTP sessions has been to using the same SSRC value
   across all the RTP sessions.  RTP Retransmission [RFC4588] is
   multiple RTP session mode, Generic FEC [RFC5109], as well as the RTP
   payload format for Scalable Video Coding [RFC6190] in Multi Session
   Transmission (MST) mode uses this method.  This method clearly works
   but might have some downside in RTP sessions with many participating
   SSRCs.  The birthday paradox ensures that if you populate a single
   session with 9292 SSRCs at random, the chances are approximately 1%
   that at least one collision will occur.  When a collision occur this
   will force one to change SSRC in all RTP sessions and thus
   resynchronizing all of them instead of only the single media stream
   having the collision.

   It can be noted that Section 8.3 of the RTP Specification [RFC3550]
   recommends using a single SSRC space across all RTP sessions for
   layered coding.

   Another solution that has been applied to binding SSRCs has been an
   implicit method used by RTP Retransmission [RFC4588] when doing
   retransmissions in the same RTP session as the source RTP media
   stream.  This issues an RTP retransmission request, and then await a
   new SSRC carrying the RTP retransmission payload and where that SSRC
   is from the same CNAME.  This limits a requestor to having only one
   outstanding request on any new source SSRCs per endpoint.

   There exists no RTP/RTCP based mechanism capable of supporting
   explicit association accross multiple RTP sessions as well within an
   RTP session.  A proposed solution for handling this issue is
   [I-D.westerlund-avtext-rtcp-sdes-srcname].  If accepted, this can
   potentially also be part of an SDP based solution also by reusing the
   same identifiers and name space.





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6.1.6.  Forward Error Correction

   There exist a number of Forward Error Correction (FEC) based schemes
   for how to reduce the packet loss of the original streams.  Most of
   the FEC schemes will protect a single source flow.  The protection is
   achieved by transmitting a certain amount of redundant information
   that is encoded such that it can repair one or more packet losses
   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 related flows as discussed
   above.  Looking at the history of these schemes, there are schemes
   using multiple SSRCs and schemes using multiple RTP sessions, and
   some schemes that support both modes of operation.

   Using multiple RTP sessions supports the case where some set of
   receivers might not be able to utilise 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 therefore in its own RTP session, allows for
   flexibility.  This is especially useful when receivers see very
   heterogeneous packet loss rates.  Those receivers that are not seeing
   packet loss don't need to join the multicast group with the FEC data,
   and so avoid the overhead of receiving unnecessary FEC packets, for
   example.

6.1.7.  Transport Translator Sessions

   A basic Transport Translator relays any incoming RTP and RTCP packets
   to the other participants.  The main difference between Additional
   SSRCs and Multiple RTP Sessions resulting from this use case is that
   with Additional SSRCs 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.

6.2.  Interworking Considerations

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





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6.2.1.  Types of Interworking

   It is not uncommon that applications or services of similar usage,
   especially the ones intended for interactive communication, encounter
   a situation where one want to interconnect two or more of these
   applications.

   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.

   There are two fundamental approaches to gatewaying: RTP Translator
   interworking (RTP bridging), where the gateway acts as an RTP
   Translator, and the two applications are members of the same RTP
   session, and Gateway Interworking (with RTP termination), where there
   are independent RTP sessions running from each interconnected
   application to the gateway.

6.2.2.  RTP Translator Interworking

   From an RTP perspective the RTP Translator approach could work if all
   the applications are using the same codecs with the same payload
   types, 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 might not always be true.

   When one is gatewaying via an RTP Translator, a natural requirement
   is that the two applications being interconnected need to use the
   same approach to multiplexing.  Furthermore, if one of the
   applications is capable of working in several modes (such as being
   able to use Additional SSRCs or Multiple RTP sessions at will), and
   the other one is not, successful interconnection depends on locking
   the more flexible application into the operating mode where
   interconnection can be successful, even if no participants using the
   less flexible application are present when the RTP sessions are being
   created.

6.2.3.  Gateway Interworking

   When one terminates RTP sessions at the gateway, there are certain
   tasks that the gateway has to carry out:

   o  Generating appropriate RTCP reports for all media streams
      (possibly based on incoming RTCP reports), originating from SSRCs
      controlled by the gateway.




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   o  Handling SSRC collision resolution in each application's RTP
      sessions.

   o  Signalling, choosing and policing appropriate bit-rates for each
      session.

   If either of the applications has any security applied, e.g.  in the
   form of SRTP, the gateway needs to be able to decrypt incoming
   packets and re-encrypt them in the other application's security
   context.  This is necessary even if all that's needed is a simple
   remapping of SSRC numbers.  If this is done, the gateway also needs
   to be a member of the security contexts of both sides, of course.

   Other tasks a gateway might need to apply include transcoding (for
   incompatible codec types), rescaling (for incompatible video size
   requirements), suppression of content that is known not to be handled
   in the destination application, or the addition or removal of
   redundancy coding or scalability layers to fit the need of the
   destination domain.

   From the above, we can see that the gateway needs to have an intimate
   knowledge of the application requirements; a gateway is by its nature
   application specific, not a commodity product.

   This fact reveals 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 endpoints, 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 requires that each RTP session needs
   different master keys, as use of the same key in two RTP sessions can
   result in two-time pads that completely breaks the confidentiality of
   the packets.

6.2.4.  Multiple SSRC Legacy Considerations












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   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 endpoint 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 endpoint and the
   RTP mixer, meaning that the mixer functions as an RTP-terminating
   gateway.

   When establishing RTP sessions that can contain endpoints 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.

   This indicates that gateways attempting to interconnect to this class
   of devices has to make sure that only one media stream of each type
   gets delivered to the endpoint if it's expecting only one, and that
   the multiplexing format is what the device expects.  It is highly
   unlikely that RTP translator-based interworking can be made to
   function successfully in such a context.

6.3.  Network Considerations

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

6.3.1.  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.  Additional SSRC 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 when



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   using multiple RTP sessions, where each media stream for which a
   separate QoS handling is desired can be in a different RTP session
   that can be sent over different 5-tuples.

   It also needs to be noted that packet marking based QoS mechanisms
   can have limitations.  A general observation is that different DSCP
   can be assigned to different packets within in a flow as well as
   within an RTP Media Stream.  However, care needs to be taken when
   considering which forwarding behaviours that are applied on path due
   to these DSCPs.  In some cases the forwarding behaviour can result in
   packet reordering.  For more discussion of this see
   [I-D.ietf-dart-dscp-rtp].

   More specific to the choice between using one or more RTP session can
   be the method for assigning marking to packets.  If this is done
   using a network ingress function, it can have issues discriminating
   the different RTP media streams.  The network API on the endpoint
   also needs to be capable of setting the marking on a per packet basis
   to reach the full functionality.

6.3.2.  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 endpoints 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 endpoint 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 endpoint can use.  In practice, implementation will
      probably benefit from using more server ports to simplify
      implementation or avoid performance bottlenecks.

   NAT State:  If an endpoint 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



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      Office/Home Office (SOHO) NATs, memory or processing are usually
      the most limited resources.  For large scale NATs serving many
      internal endpoints, available external ports are likely the scarce
      resource.  Port limitations is primarily a problem for larger
      centralised NATs where endpoint 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 takes after finding the
      first valid candidate pair following the specified ICE procedures
      are: 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the pacing
      timer, which ICE specifies to be no smaller than 20 ms.  That
      assumes a message in one direction, and then an immediate
      triggered check back.  The reason it isn't more, is that ICE first
      finds one candidate pair that works prior to attempting 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.  However, packet loss causes extra
      delays, at least 100 ms, which is the minimal retransmission timer
      for ICE.

   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
      ought to 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, ought to 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.

   Additional SSRC keeps the additional media streams within one RTP
   Session and transport flow and does not introduce any additional NAT
   traversal complexities per media stream.  This can be compared with
   normally one or two additional transport flows per RTP session when



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   using multiple RTP sessions.  Additional lower layer transport flows
   will be needed, 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-transport-multiplexing].

6.3.3.  Multicast

   Multicast groups provides a powerful semantics for a number of real-
   time applications, especially the ones that desire broadcast-like
   behaviours with one endpoint 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
   RTP session per multicast group is used.

   In addition, the transport flow considerations in multicast are a bit
   different from unicast; NATs are not useful in the multicast
   environment, meaning that the entire port range of each multicast
   address is available for distinguishing between RTP sessions.

   Thus it appears easiest and most straightforward to use multiple RTP
   sessions for sending different media flows used for adapting to
   network conditions.

6.3.4.  Multiplexing multiple RTP Session on a Single Transport

   For applications that don'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-transport-multiplexing].
   Using such a solution would allow Multiple RTP session without most
   of the perceived downsides of Multiple RTP sessions creating a need
   for additional transport flows, but this solution would require
   support from all functions that handle RTP packets, including
   firewalls.





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6.4.  Security and Key Management Considerations

   When dealing with point-to-point, 2-member RTP sessions only, there
   are few security issues that are relevant to the choice of having one
   RTP session or multiple RTP sessions.  However, there are a few
   aspects of multiparty sessions that might warrant consideration.  For
   general information of possible methods of securing RTP, please
   review RTP Security Options [I-D.ietf-avtcore-rtp-security-options].

6.4.1.  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) are 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 issues.

   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.

   SRTP [RFC3711] has no special functions for dealing with different
   sets of master keys for different SSRCs.  The key-management
   functions have different capabilities to establish different set of
   keys, normally on a per endpoint basis.  For example, DTLS-SRTP
   [RFC5764] and Security Descriptions [RFC4568] establish different
   keys for outgoing and incoming traffic from an endpoint.  This key
   usage has to be written into the cryptographic context, possibly
   associated with different SSRCs.

6.4.2.  Key Management for Multi-party session

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





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   Multi-party sessions, such as transport translator based sessions and
   multicast sessions, cannot use Security Description [RFC4568] nor
   DTLS-SRTP [RFC5764] without an extension as each endpoint provides
   its set of keys.  In centralised conferences, the signalling
   counterpart is a conference server and the media plane unicast
   counterpart (to which DTLS messages would be sent) is the transport
   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.

6.4.3.  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
   an RTP translator or mixer to rewrite an SSRC value in the RTP packet
   of an unencrypted session, 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 tag needs to 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.

7.  Archetypes

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

7.1.  Single SSRC per Session

   In this archetype each endpoint 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
   endpoints have one media stream each.  If the application needs
   additional media flows between the endpoints, they will have to
   establish additional RTP sessions.

   The Pros:

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



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   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 media
       stream with current key-management, since each media stream is
       directly related to an RTP session, and the keying operates on a
       per-session basis.

   The Cons:

   a.  The number of RTP sessions grows directly in proportion with the
       number of media streams, 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.

       *  Higher potential for a single media stream to fail during
          transport between the endpoints.

   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 might become a problem for centralised
       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 might be needed, and the fact that
       they're impossible can cause issues.





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   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
       require SSRC translation.

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

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

   RTP applications that need to inter-work with legacy RTP
   applications, like most deployed VoIP and video conferencing
   solutions, 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, needs its own RTP session.  In addition, the
   dynamic behaviour that can arise in multi-party applications can tax
   the signalling system and make timely media establishment more
   difficult.

7.2.  Multiple SSRCs of the Same Media Type

   In this archetype, each RTP session serves only a single media type.
   The RTP session can contain multiple media streams, either from a
   single endpoint or from multiple endpoints.  This commonly creates a
   low number of RTP sessions, typically only one for audio and one for
   video, with a corresponding need for two listening ports when using
   RTP/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.



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   3.  Works well with media type de-composite endpoints.

   4.  Enables Flow-based QoS with different prioritisation 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 endpoint.

   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, unless these are split into different RTP
       sessions.

   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 fate 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 prioritisation between media types.  It
   handles multi-party session well, independently of multicast or
   centralised transport distribution, as additional sources can
   dynamically enter and leave the session.

7.3.  Multiple Sessions for one Media type

   In this archetype one goes one step further than in the above
   (Section 7.2) by using multiple RTP sessions also for a single media
   type, but still not as far as having a single SSRC per RTP session.
   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 archetype are scalability over
   multicast, simulcast, need for extended QoS prioritisation of media
   streams due to their usage in the application, or the need for fine-
   grained signalling using today's tools.



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   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.  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 prioritisation for flow based mechanisms.

   5.  Works well with de-composite endpoints.

   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 endpoint 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.

   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 synchronised 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
       need to 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 endpoint.

   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



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       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 archetype 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
   endpoint will never use the SSRC for more than a single media type.

   The Pros:

   1.  Single RTP session which implies:

       *  Minimal NAT/FW state.

       *  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 endpoint 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.  Less suitable for interworking with other applications that uses
       individual RTP sessions per media type or multiple sessions for a
       single media type, due to need of SSRC translation.

   b.  Negotiation of bandwidth for the different media types is
       currently not possible in SDP.  This requires SDP extensions to



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       enable payload or source specific bandwidth.  Likely to be a
       problem due to media type asymmetry in needed bandwidth.

   c.  Not suitable for de-composite endpoints.

   d.  Flow based QoS cannot provide separate treatment to some media
       streams compared to others 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 do not
       support the RTP specification fully when it comes to handling
       multiple SSRC per endpoint, 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
       associations, most key-management will have difficulties
       establishing such a session.

7.5.  Summary

   There are some clear relations between these archetypes.  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 needed.














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   The two other archetypes "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.  Summary considerations and guidelines

8.1.  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 6.1.5 there exist drawbacks in using the same SSRC in
      multiple RTP sessions as a mechanism to bind related media streams
      together.  It is instead suggested 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).

   Use additional SSRCs additional Media Sources:  In the cases where an
      RTP endpoint needs to transmit additional media streams of the
      same media type in the application, with the same processing
      requirements at the network and RTP layers, it is suggested to
      send them as additional SSRCs in the same RTP session.  For
      example a telepresence room where there are three cameras, and
      each camera captures 2 persons sitting at the table, sending each
      camera as its own SSRC within a single RTP session is suggested.

   Use additional RTP sessions for streams with different requirements:
      When media streams have different processing requirements from the
      network or the RTP layer at the endpoints, it is suggested that
      the different types of streams are put in different RTP sessions.
      This includes the case where different participants want different
      subsets of the set of RTP streams.

   When using multiple RTP Sessions use grouping:  When using Multiple
      RTP session solutions, it is suggested to explicitly group the
      involved RTP sessions when needed using the signalling mechanism,
      for example The Session Description Protocol (SDP) Grouping
      Framework.  [RFC5888], using some appropriate grouping semantics.




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   RTP/RTCP Extensions May Support Additional SSRCs as well as Multiple RTP sessions:
      When defining an RTP or RTCP extension, the creator needs to
      consider if this extension is applicable to usage with additional
      SSRCs and Multiple RTP sessions.  Any extension intended to be
      generic is suggested 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 expected to include support for both
      additional SSRCs and multiple RTP sessions 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.  IANA Considerations

   This document makes no request of IANA.

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

10.  Security Considerations

   There is discussion of the security implications of choosing SSRC vs
   Multiple RTP session in Section 6.4.

11.  References

11.1.  Normative References

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

11.2.  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]





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              Alvestrand, H., "Why RTP Sessions Should Be Content
              Neutral", draft-alvestrand-rtp-sess-neutral-01 (work in
              progress), June 2012.

   [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-avtcore-6222bis]
              Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
              (work in progress), July 2013.

   [I-D.ietf-avtcore-multi-media-rtp-session]
              Westerlund, M., Perkins, C., and J. Lennox, "Sending
              Multiple Types of Media in a Single RTP Session", draft-
              ietf-avtcore-multi-media-rtp-session-05 (work in
              progress), February 2014.

   [I-D.ietf-avtcore-rtp-security-options]
              Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", draft-ietf-avtcore-rtp-security-options-10
              (work in progress), January 2014.

   [I-D.ietf-avtext-multiple-clock-rates]
              Petit-Huguenin, M. and G. Zorn, "Support for Multiple
              Clock Rates in an RTP Session", draft-ietf-avtext-
              multiple-clock-rates-11 (work in progress), November 2013.

   [I-D.ietf-dart-dscp-rtp]
              Black, D. and P. Jones, "Differentiated Services
              (DiffServ) and Real-time Communication", draft-ietf-dart-
              dscp-rtp-07 (work in progress), September 2014.

   [I-D.ietf-mmusic-msid]
              Alvestrand, H., "WebRTC MediaStream Identification in the
              Session Description Protocol", draft-ietf-mmusic-msid-06
              (work in progress), June 2014.

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-11 (work in progress), September 2014.

   [I-D.ietf-payload-rtp-howto]



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              Westerlund, M., "How to Write an RTP Payload Format",
              draft-ietf-payload-rtp-howto-13 (work in progress),
              January 2014.

   [I-D.lennox-avtcore-rtp-multi-stream]
              Lennox, J., Westerlund, M., Wu, W., and C. Perkins, "RTP
              Considerations for Endpoints Sending Multiple Media
              Streams", draft-lennox-avtcore-rtp-multi-stream-02 (work
              in progress), February 2013.

   [I-D.lennox-mmusic-sdp-source-selection]
              Lennox, J. and H. Schulzrinne, "Mechanisms for Media
              Source Selection in the Session Description Protocol
              (SDP)", draft-lennox-mmusic-sdp-source-selection-05 (work
              in progress), October 2012.

   [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-02 (work in progress),
              July 2012.

   [I-D.westerlund-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              westerlund-avtcore-rtp-topologies-update-02 (work in
              progress), February 2013.

   [I-D.westerlund-avtcore-transport-multiplexing]
              Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
              Sessions onto a Single Lower-Layer Transport", draft-
              westerlund-avtcore-transport-multiplexing-07 (work in
              progress), October 2013.

   [I-D.westerlund-avtext-rtcp-sdes-srcname]
              Westerlund, M., "RTCP Source Description Item SRCNAME to
              Label Individual Media Sources", draft-westerlund-avtext-
              rtcp-sdes-srcname-03 (work in progress), October 2013.

   [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.L. 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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   [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.

   [RFC5109]  Li, A., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, December 2007.

   [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.

   [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.-K., Schierl, T., and A.
              Eleftheriadis, "RTP Payload Format for Scalable Video
              Coding", RFC 6190, May 2011.

   [RFC6222]  Begen, A., Perkins, C., and D. Wing, "Guidelines for
              Choosing RTP Control Protocol (RTCP) Canonical Names
              (CNAMEs)", RFC 6222, April 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.

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

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



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   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
   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 can 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 need 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



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        almost never a field to indicate which Payload Type is reported,
        so sending feedback for a specific media stream is difficult
        without extending existing RTCP reporting.

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

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

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

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

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

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

Appendix B.  Proposals 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 Multiple RTP session
   or Additional SSRC.  These extensions are:

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






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   MSID:  A Media Stream identification scheme that can be used to
      signal relationships between SSRCs that can be in the same or in
      different RTP sessions.  Described in [I-D.ietf-mmusic-msid]

   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 endpoint can handle within a given RTP Session.  That
      ensures that usage of Additional SSRC occurs when supported and
      without overloading an endpoint.  This extension is proposed in
      [I-D.westerlund-avtcore-max-ssrc].

Appendix C.  Signalling considerations

   Signalling is not an architectural consideration for RTP itself, so
   this discussion has been moved to an appendix.  However, it is hugely
   important for anyone building complete applications, so it is
   deserving of discussion.

   The issues raised here need to be addressed in the WGs that deal with
   signalling; they cannot be addressed by tweaking, extending or
   profiling RTP.

C.1.  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.

C.1.1.  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:

   o  Bitrate/Bandwidth exist today only at aggregate or a common any
      media stream limit, unless either codec-specific bandwidth
      limiting or RTCP signalling using TMMBR is used.



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   o  Which SSRC that will use which RTP Payload Types (this will be
      visible from the first media packet, but is sometimes useful to
      know before packet arrival).

   Some of these issues are clearly SDP's problem rather than RTP
   limitations.  However, if the aim is to deploy an solution using
   additional SSRCs 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
   additional SSRC only, a number of signalling extensions are needed to
   clarify that there are multiple sets of media streams with different
   properties and that they need in fact be kept different, since a
   single set will not satisfy the application's requirements.

   For some parameters, such as resolution and framerate, a SSRC-linked
   mechanism has been proposed:
   [I-D.lennox-mmusic-sdp-source-selection].

C.1.2.  SDP Prevents Multiple Media Types

   SDP chose to use the m= line both to delineate an RTP session and to
   specify the top level of the MIME 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.  This binding has to be
   loosened in order to use SDP to describe RTP sessions containing
   multiple MIME top level types.

   There is an accepted WG item in the MMUSIC WG to define how multiple
   media lines describe a single underlying transport
   [I-D.ietf-mmusic-sdp-bundle-negotiation] and thus it becomes possible
   in SDP to define one RTP session with media types having different
   MIME top level types.

C.1.3.  Signalling 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, an
   application might choose to take all incoming audio RTP streams, mix
   them and play them 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.  The signalling
   used will have to identify the media streams affected by their RTP-
   level identifiers, which means that they have to be identified either
   by their session or by their SSRC + session.



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   In some applications, the receiver cannot utilise the media stream at
   all before it has received the signalling message describing the
   media stream and its usage.  In other applications, there exists a
   default handling that is appropriate.

   If all media streams in an RTP session are to be treated in the same
   way, identifying the session is enough.  If SSRCs in a session are to
   be treated differently, signalling needs to identify both the session
   and the SSRC.

   If this signalling affects how any RTP central node, like an RTP
   mixer or translator that selects, mixes or processes streams, treats
   the streams, the node will also need to receive the same signalling
   to know how to treat media streams with different usage in the right
   fashion.

Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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


   Bo Burman
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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


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

   Email: csp@csperkins.org






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   Harald Tveit Alvestrand
   Google
   Kungsbron 2
   Stockholm  11122
   Sweden

   Email: harald@alvestrand.no











































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