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Versions: (draft-westerlund-avtcore-multiplex-architecture) 00 01 02 03 04 05 06

Network Working Group                                      M. Westerlund
Internet-Draft                                                 B. Burman
Intended status: Informational                                  Ericsson
Expires: January 3, 2019                                      C. Perkins
                                                   University of Glasgow
                                                           H. Alvestrand
                                                                  Google
                                                                 R. Even
                                                                  Huawei
                                                            July 2, 2018


    Guidelines for using the Multiplexing Features of RTP to Support
                         Multiple Media Streams
               draft-ietf-avtcore-multiplex-guidelines-06

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 https://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 January 3, 2019.








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Copyright Notice

   Copyright (c) 2018 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (https://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
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   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.1.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   4
     2.2.  Subjects Out of Scope . . . . . . . . . . . . . . . . . .   5
   3.  RTP Multiplexing Overview . . . . . . . . . . . . . . . . . .   5
     3.1.  Reasons for Multiplexing and Grouping RTP Streams . . . .   5
     3.2.  RTP Multiplexing Points . . . . . . . . . . . . . . . . .   6
       3.2.1.  RTP Session . . . . . . . . . . . . . . . . . . . . .   7
       3.2.2.  Synchronisation Source (SSRC) . . . . . . . . . . . .   8
       3.2.3.  Contributing Source (CSRC)  . . . . . . . . . . . . .  10
       3.2.4.  RTP Payload Type  . . . . . . . . . . . . . . . . . .  10
     3.3.  Issues Related to RTP Topologies  . . . . . . . . . . . .  11
     3.4.  Issues Related to RTP and RTCP Protocol . . . . . . . . .  12
       3.4.1.  The RTP Specification . . . . . . . . . . . . . . . .  13
       3.4.2.  Multiple SSRCs in a Session . . . . . . . . . . . . .  15
       3.4.3.  Binding Related Sources . . . . . . . . . . . . . . .  15
       3.4.4.  Forward Error Correction  . . . . . . . . . . . . . .  17
   4.  Considerations for RTP Multiplexing . . . . . . . . . . . . .  17
     4.1.  Interworking Considerations . . . . . . . . . . . . . . .  17
       4.1.1.  Application Interworking  . . . . . . . . . . . . . .  18
       4.1.2.  RTP Translator Interworking . . . . . . . . . . . . .  18
       4.1.3.  Gateway Interworking  . . . . . . . . . . . . . . . .  19
       4.1.4.  Multiple SSRC Legacy Considerations . . . . . . . . .  20
     4.2.  Network Considerations  . . . . . . . . . . . . . . . . .  20
       4.2.1.  Quality of Service  . . . . . . . . . . . . . . . . .  20
       4.2.2.  NAT and Firewall Traversal  . . . . . . . . . . . . .  21
       4.2.3.  Multicast . . . . . . . . . . . . . . . . . . . . . .  23
     4.3.  Security and Key Management Considerations  . . . . . . .  24
       4.3.1.  Security Context Scope  . . . . . . . . . . . . . . .  24
       4.3.2.  Key Management for Multi-party sessions . . . . . . .  25
       4.3.3.  Complexity Implications . . . . . . . . . . . . . . .  25



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   5.  RTP Multiplexing Design Choices . . . . . . . . . . . . . . .  26
     5.1.  Single SSRC per Endpoint  . . . . . . . . . . . . . . . .  26
     5.2.  Multiple SSRCs of the Same Media Type . . . . . . . . . .  27
     5.3.  Multiple Sessions for one Media type  . . . . . . . . . .  28
     5.4.  Multiple Media Types in one Session . . . . . . . . . . .  30
     5.5.  Summary . . . . . . . . . . . . . . . . . . . . . . . . .  31
   6.  Guidelines  . . . . . . . . . . . . . . . . . . . . . . . . .  31
   7.  Open Issues . . . . . . . . . . . . . . . . . . . . . . . . .  33
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  33
   9.  Security Considerations . . . . . . . . . . . . . . . . . . .  33
   10. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  33
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  33
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  33
     11.2.  Informative References . . . . . . . . . . . . . . . . .  34
   Appendix A.  Dismissing Payload Type Multiplexing . . . . . . . .  38
   Appendix B.  Signalling Considerations  . . . . . . . . . . . . .  40
     B.1.  Session Oriented Properties . . . . . . . . . . . . . . .  40
     B.2.  SDP Prevents Multiple Media Types . . . . . . . . . . . .  41
     B.3.  Signalling RTP stream Usage . . . . . . . . . . . . . . .  41
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  42

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 was from the beginning designed for multiple
   participants in a communication session.  It supports many topology
   paradigms and usages, as defined in [RFC7667].  RTP has several
   multiplexing points designed for different purposes.  These enable
   support of multiple RTP streams and switching between different
   encoding or packetization of the media.  By using multiple RTP
   sessions, sets of RTP 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 RTP
   stream identifier (SSRC), and the RTP payload type to meet the
   application's needs.

   There have been increased interest in more advanced usage of RTP.
   For example, multiple RTP streams can be used when a single endpoint
   has multiple media sources (like multiple cameras or microphones)
   that need to be sent simultaneously.  Consequently, questions are
   raised regarding the most appropriate RTP usage.  The limitations in
   some implementations, RTP/RTCP extensions, and signalling has also
   been exposed.  The authors also hope that clarification on the
   usefulness of some functionalities in RTP will result in more
   complete implementations in the future.




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   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.

   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 designs of RTP usage are discussed.
   Finally, some guidelines and examples are provided.

2.  Definitions

2.1.  Terminology

   The definitions in Section 3 of [RFC3550] are referenced normatively.

   The taxonomy defined in [RFC7656] is referenced normatively.

   The following terms and abbreviations are used in this document:

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

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

   RTP Sink:  An endpoint that receives RTP streams.  The RTP Sink is
      identified using one or more SSRCs.  The SSRCs used by an RTP sink
      can be both RTP source ones, as well as used soley to represent
      the RTP sink.  There can be more than one RTP Sink for one RTP
      source.

   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 Group:  One or more RTP sessions that are used together



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      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
      sessions, and designate synchronisation contexts that can cross
      RTP sessions; i.e. SSRCs that map to a common CNAME can be assumed
      to have RTCP SR timing information derived from a common clock
      such that they can be synchronised for playout.

   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 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.  RTP Multiplexing Overview

3.1.  Reasons for Multiplexing and Grouping RTP Streams

   There are several reasons why an endpoint might choose to send
   multiple media streams.  In the below discussion, please keep in mind
   that the reasons for having multiple RTP streams vary and include but
   are not limited to the following:

   o  Multiple media sources

   o  Multiple RTP streams might be needed to represent one media source
      (for instance when using layered encodings)

   o  A retransmission stream might repeat some parts of the content of
      another RTP stream

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

   o  Alternative encodings, for instance using different codecs for the
      same audio stream



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

   For each of these reasons, it is necessary to decide if each
   additional RTP stream is sent within the same RTP session as the
   other RTP streams, or if it is necessary to use additional RTP
   sessions to group the RTP streams.  The choice suitable for one
   reason, might not be the choice suitable for another reason.  The
   clearest understanding is associated with multiplexing multiple media
   sources of the same media type.  However, all reasons 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 SSRC or RTP session multiplexing 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 additional RTP sessions
   and when additional RTP streams in the same RTP session is the right
   choice.

3.2.  RTP Multiplexing Points

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



























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                           |
                           | packets
           +--             v
           |        +------------+
           |        |   Socket   |   Transport Protocol Demultiplexing
           |        +------------+
           |            ||  ||
      RTP  |       RTP/ ||  |+-----> SCTP ( ...and any other protocols)
   Session |       RTCP ||  +------> STUN (multiplexed using same port)
           +--          ||
           +--          ||
           |      (split by SSRC)
           |      ||    ||    ||
           |      ||    ||    ||
     RTP   |     +--+  +--+  +--+
   Streams |     |PB|  |PB|  |PB| Jitter buffer, process RTCP, etc.
           |     +--+  +--+  +--+
           +--      |   |      |
           +--      |  /       |
           |        +----+     |
           |         /   |     |
   Payload |     +---+ +---+ +---+
   Formats |     |Dec| |Dec| |Dec| Decoders
           |     +---+ +---+ +---+
           +--

                   Figure 1: RTP Demultiplexing Process

3.2.1.  RTP Session

   An RTP Session is the highest semantic layer in the RTP protocol, and
   represents an association between a group of communicating endpoints.
   RTP does not contain a session identifier, yet RTP sessions must be
   possible to separate both across different endpoints and within a
   single endpoint.

   For RTP session separation across 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 Session Description Protocol (SDP) [RFC4566] "a=ssrc:" attribute
   [RFC5576]).  Thus, the scope of an RTP session is determined by the
   participants' network interconnection topology, in combination with



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   RTP and RTCP forwarding strategies deployed by the endpoints and any
   middleboxes, and by the signalling.

   For RTP session separation within a single endpoint, RTP relies on
   the underlying transport layer, and on the signalling to identify RTP
   sessions in a manner that is meaningful to the application.  A single
   endpoint can have one or more transport flows for the same RTP
   session.  The signalling layer might give RTP sessions an explicit
   identifier, or the 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 by receiving the multiple UDP flows used as identified
   based on their UDP source and destination IP addresses and UDP port
   numbers.  Another example is SDP media descriptions (the "m=" line
   and the following associated lines) signals the transport flow and
   RTP session configuration for the endpoints part of the RTP session.
   SDP grouping framework [RFC5888] allows labeling of the media
   descriptions, for example used so that RTP Session Groups can be
   created.  With Negotiating Media Multiplexing Using the Session
   Description Protocol (SDP)[I-D.ietf-mmusic-sdp-bundle-negotiation],
   multiple media descriptions where each represents the RTP streams
   sent or received for a media source are part of a common RTP session.

   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.

3.2.2.  Synchronisation Source (SSRC)

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

   The SSRC is a 32-bit identifier.  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, e.g. using
   the SDP "a=ssrc:" attribute [RFC5576], the SSRC is chosen at random.
   It is not dependent on the network address of the endpoint, and is



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   intended to be unique within an RTP session.  SSRC collisions can
   occur, and are handled as specified in [RFC3550] and [RFC5576],
   resulting in the SSRC of the colliding RTP sources and/or sinks
   changing.  An RTP source that changes its network transport address
   during a session have to choose a new SSRC identifier to avoid being
   interpreted as looped source, unless the transport layer mechansism,
   e.g ICE [RFC5245], handle such changes

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

   In some cases, the same SSRC identifier value is used to relate
   streams in two different RTP sessions, such as in RTP retransmission
   [RFC4588].  This is to be avoided since there is no guarantee that
   SSRC values are unique across RTP sessions.  For the RTP
   retransmission [RFC4588] case it is recommended to use explicit
   binding of the source RTP stream and the redundancy stream, e.g.
   using the RepairedRtpStreamId RTCP SDES item [I-D.ietf-avtext-rid].

   Note that RTP sequence number and RTP timestamp are scoped by the
   SSRC.  Each RTP source will have a different SSRC, and the
   corresponding RTP 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.

   Conceptual 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 stream in
      itself (e.g. an endpoint or middlebox only receiving in an RTP
      session).  It still needs an SSRC for use as source in RTCP
      reports.

   Note that an endpoint that generates more than one media type, e.g. a
   conference participant sending both audio and video, need not (and
   should not) use the same SSRC value across RTP sessions.  RTCP
   Compound packets containing the CNAME SDES item is the designated
   method to bind an SSRC to a CNAME, effectively cross-correlating



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   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.

3.2.3.  Contributing Source (CSRC)

   The Contributing Source (CSRC) is not a separate identifier.  Rather
   an SSRC 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.

   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
   an RTP stream, possibly modified.  The RTP header extension for
   Mixer-to-Client Audio Level Indication [RFC6465] expands on the
   receiver's 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.

3.2.4.  RTP Payload Type

   Each RTP 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 (PT) field in the RTP packet header.
   The combination of SSRC and PT therefore identifies a specific RTP
   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].




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   The RTP payload type is scoped by the sending endpoint within an RTP
   session.  PT has the same meaning across all RTP streams in an RTP
   session.  All SSRCs sent from a single endpoint share the same
   payload type 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.  If it is acceptable to send multiple formats of the
   same media source as separate RTP streams (with separate SSRC),
   simulcast [I-D.ietf-mmusic-sdp-simulcast] can be used.

   Other aspects of RTP payload format use are described in How to Write
   an RTP Payload Format [RFC8088].

   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
   an RTP multiplexing point does not work).  The RTP payload type is,
   however, used to determine how to consume and decode an RTP stream.
   The RTP payload type number is sometimes used to associate an RTP
   stream with the signalling; this is not recommended since a specific
   payload type value can be used in multiple bundled "m=" sections
   [I-D.ietf-mmusic-sdp-bundle-negotiation].  This association is only
   possible if unique RTP payload type numbers are used in each context.

3.3.  Issues Related to RTP Topologies

   The impact of how RTP multiplexing is performed will in general vary
   with how the RTP session participants are interconnected, described
   by RTP Topology [RFC7667].

   Even the most basic use case, denoted Topo-Point-to-Point in
   [RFC7667], raises a number of considerations that are discussed in
   detail in following sections.  They range over such aspects as:

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

   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?



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

   o  etc.

   For some point to multi-point topologies (e.g.  Topo-ASM and Topo-SSM
   in [RFC7667]), multicast is used to interconnect the session
   participants.  Special considerations (documented in Section 4.2.3)
   are then needed as multicast is a one-to-many distribution system.

   Sometimes an RTP communication can end up in a situation when the
   communicating peers are not compatible 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
   [RFC7667].  The translator's main purpose is to make the peers look
   compatible to each other.  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 RTP 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.

   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.

3.4.  Issues Related to RTP and RTCP Protocol

   Using multiple RTP streams is a well-supported feature of RTP.
   However, for most implementers or people writing RTP/RTCP
   applications or extensions attempting to apply multiple streams, it
   can be unclear when it is most appropriate to add an additional RTP
   stream in an existing RTP session and when it is better to use
   multiple RTP sessions.  This section discusses the various
   considerations needed.



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3.4.1.  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.

   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.






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   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 argument is for
   using different SSRC for each individual RTP stream, which is very
   basic.

   The second argument is advocating against demultiplexing RTP streams
   within a session based on their RTP payload type numbers, 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 argument is against multiplexing RTP packets that require
   different handling into the same session.  As we saw in the
   discussion of RTP mixers, the RTP mixer must embed application logic
   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.  One type
   of application that can mix different media sources "blindly" is the
   audio-only "telephone" bridge; most other types of applications need
   application-specific logic to perform the mix correctly.

   The fifth argument discusses network aspects that we will discuss
   more below in Section 4.2.  It also goes into aspects of
   implementation, like Split Component Terminal (see Section 3.10 of
   [RFC7667]) endpoints where different processes or inter-connected
   devices handle different aspects of the whole multi-media session.

   A summary of RFC 3550's view on multiplexing is to use unique SSRCs
   for anything that is its own media/packet stream, and 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
   latter needs further discussion, as imposing a single solution on all
   usages of RTP is inappropriate.  Multiple Media Types in an RTP
   Session specification [I-D.ietf-avtcore-multi-media-rtp-session]



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   provides a detailed analysis of the potential issues in having
   multiple media types in the same RTP session.  This document provides
   a wider scope for an RTP session and considers multiple media types
   in one RTP session as a possible choice for the RTP application
   designer.

3.4.2.  Multiple SSRCs in a Session

   Using multiple SSRCs at one endpoint in an RTP session requires
   resolving some unclear aspects of the RTP specification.  These could
   potentially lead to some interoperability issues as well as some
   potential significant inefficiencies, as further discussed in "RTP
   Considerations for Endpoints Sending Multiple Media Streams"
   [RFC8108].  An RTP application designer should consider these issues
   and the possible applicaiton impact from lack of appropriate RTP
   handling or optimization in the peer endpoints.

   Using multiple RTP sessions can potentially mitigate application
   issues caused by multiple SSRCs in an RTP session.

3.4.3.  Binding Related Sources

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

   The solutions can be divided into a few groups:

   o  RTP/RTCP based

   o  Signalling based (SDP)

   o  grouping related RTP sessions

   o  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, [I-D.ietf-mmusic-sdp-bundle-negotiation] groups multiple
      media descriptions as a single RTP session.




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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 groups Media Descriptions.

   SDP MSID grouping:  Media Stream Identifiers [I-D.ietf-mmusic-msid]
      specifies a Session Description Protocol (SDP) Grouping mechanism
      for RTP streams that can be used to specify relations between RTP
      streams.  This mechanism is used to signal the association between
      the SDP concept of "media description" and the WebRTC concept of
      "MediaStream" / "MediaStreamTrack" (Corresponds to the [RFC7656]
      term "Source Stream") using SDP signalling.

   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.

   An RTP/RTCP-based solution is to use the RTCP SDES CNAME to bind the
   RTP streams to an endpoint or synchronization context.  For
   applications with a single RTP stream per type (Media, Source or
   Redundancy) this is sufficient independent if one or more RTP
   sessions are used.  However, some applications choose not to use it
   because of perceived complexity or a desire not to implement RTCP and
   instead use the same SSRC value to bind related RTP streams across
   multiple RTP sessions.  RTP Retransmission [RFC4588] in multiple RTP
   session mode and Generic FEC [RFC5109] both use this method.  This
   method may work but might have some downsides in RTP sessions with
   many participating SSRCs.  When an SSRC collision occurs, this will
   force one to change SSRC in all RTP sessions and thus resynchronize
   all of them instead of only the single media stream having the
   collision.  Therefore, it is not recommended to use identical SSRC
   values to relate RTP streams.

   Another solution to bind SSRCs is an implicit method used by RTP
   Retransmission [RFC4588] when doing retransmissions in the same RTP
   session as the source RTP stream.  The receiver missing a packet
   issues an RTP retransmission request, and then awaits 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.

   RTP Payload Format Restrictions [I-D.ietf-mmusic-rid] provides an
   RTP/RTCP based mechanism to unambiguously identify the RTP streams
   within an RTP session and restrict the streams' payload format
   parameters in a codec-agnostic way beyond what is provided with the
   regular Payload Types.  The mapping is done by specifying an "a=rid"
   value in the SDP offer/answer signalling and having the corresponding
   "rtp-stream-id" value as an SDES item and an RTP header extension.



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   The RID solution also includes a solution for binding redundancy RTP
   streams to their original source RTP streams, given that those use
   RID identifiers.

   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.  Based on the experience so far however, we recommend
   to use a solution doing explicit binding between the RTP streams so
   what the used SSRC values are do not matter.  That way solutions
   using multiple RTP streams in a single RTP session and multiple RTP
   sessions uses the same solution.

3.4.4.  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 the redundant information protects.  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 and if separating RTP sessions
   on transport level, FEC can easily be ignored already on transport
   level.

   In usages involving multicast, having the FEC information on its own
   multicast group allows for similar 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.

4.  Considerations for RTP Multiplexing

4.1.  Interworking Considerations

   There are several different kinds of interworking, and this section
   discusses two; interworking between different applications including
   the implications of potentially different RTP multiplexing point



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   choices and limitations that have to be considered when working with
   some legacy applications.

4.1.1.  Application Interworking

   It is not uncommon that applications or services of similar but not
   identical 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 must then either
   change the multiplexing structure or adhere to the respective
   limitations in each application.

   There are two fundamental approaches to gatewaying: RTP Translator
   interworking (RTP bridging), where the gateway acts as an RTP
   Translator, with the two applications being 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.

4.1.2.  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, and have the same
   capabilities in number of simultaneous RTP 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, an important
   consideration is if the two applications being interconnected need to
   use the same approach to multiplexing.  If one side is using RTP
   session multiplexing and the other is using SSRC multiplexing with
   bundle, the mapping of SDP "m=" lines between both sides requires
   that the order in bundled and not bundled sides will be the same to
   allow routing without mapping, it is possible for the RTP translator
   to map the RTP streams between both sides.  There are also challenges
   with SSRC collision handling since there may be a collision on the
   SSRC multiplexing side but the RTP session multiplexing side will not
   be aware of any collision unless SSRC translation is applied on the
   RTP translator.  Furthermore, if one of the applications is capable
   of working in several modes (such as being able to use additional RTP
   streams in one RTP session 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 are using



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   the less flexible application when the RTP sessions are being
   created.

4.1.3.  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 RTP streams (possibly
      based on incoming RTCP reports), originating from SSRCs controlled
      by the gateway.

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

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

   For applications that uses any security mechanism, 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), media-level adaptations that cannot be
   solved through media negotiation (such as 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 needs 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
   application evolution by blocking RTP and RTCP extensions that the
   applications have been extended with but that are unknown to the
   gateway.

   If one uses security functions, like SRTP, and as can be seen from
   above, they incur both additional risk due to the requirement to have
   the gateway in the 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



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   RTP session needs different master keys, as use of the same key in
   two RTP sessions can for some ciphers result in two-time pads that
   completely breaks the confidentiality of the packets.

4.1.4.  Multiple SSRC Legacy Considerations

   Historically, the most common RTP use cases have been point to point
   Voice over IP (VoIP) or streaming applications, commonly with no more
   than one media source per 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 RTP 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.

4.2.  Network Considerations

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

4.2.1.  Quality of Service

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




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   However, for a flow based scheme there is a clear difference between
   the multiplexing methods.  Additional SSRC will result in all RTP
   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.

   It must also be noted that packet marking based QoS mechanisms can
   have limitations.  A general observation is that different
   Differentiated Services Code Points (DSCP) can be assigned to
   different packets within a flow as well as within an RTP stream.
   However, care must 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 [RFC7657].

   The method for assigning marking to packets can impact what number of
   RTP sessions to choose.  If this marking is done using a network
   ingress function, it can have issues discriminating the different RTP
   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.

4.2.2.  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 analyse 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



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      to an external address will result in a state that has to be kept
      in the NAT.  That state is a limited resource.  In home or Small
      Office/Home Office (SOHO) NATs, memory or processing are usually
      the most limited resources.  For large scale NATs serving many
      internal 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 Extra Delay:  Performing 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.  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.

   Using additional RTP streams in the same RTP session and transport
   flow does not introduce any additional NAT traversal complexities per
   RTP stream.  This can be compared with normally one or two additional



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   transport flows per RTP session when 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.  At time of writing no such mechanism was defined.

4.2.3.  Multicast

   Multicast groups provides a powerful tool 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.  There are also the RTP/RTCP extension to
   better support Source Specific Multicast (SSM) [RFC5760].  Another
   application is the Many to Many communication, which RTP [RFC3550]
   was originally built to support.  But the multicast 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.  For broadcast type of applications 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 with port translation 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, when using broadcast applications it appears easiest and most
   straightforward to use multiple RTP sessions for sending different
   media flows used for adapting to network conditions.  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.

   For many to many applications there is different needs.  Here it will
   depend on how the actual application is realized what is the most
   appropriate choice.  With sender side congestion control there might
   not exist any benefit with using multiple RTP session.

   The properties of a broadcast application using RTP multicast:





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   1.  Uses 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.  The applications need signalling functions to identify the
       relationships between RTP sessions.

   4.  The applications need signalling or RTP/RTCP functions to
       identify the relationships between SSRCs in different RTP
       sessions when needs beyond CNAME exists.

   Both broadcast and many to many multicast applications do 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 in broadcast/
   multicast context.  The signalling aspects of broadcast/multicast are
   not explored further in this memo.

   Security solutions for this type of group communications are also
   challenging.  First, the key-management and the security protocol
   needs to support group communication.  Second, source authentication
   requires special solutions.  For more discussion on this please
   review Options for Securing RTP Sessions [RFC7201].

4.3.  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 [RFC7201].

4.3.1.  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 are (so far) 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 and not by another party having
   access to the key.  At least unless TESLA source authentication
   [RFC4383], which adds delay to achieve source authentication.  In



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   most cases symmetric ciphers provide sufficient security properties,
   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 RTP
   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 that 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 sets 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.

4.3.2.  Key Management for Multi-party sessions

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

   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-perc-srtp-ekt-diet] or a MIKEY [RFC3830] based solution
   that allows for keying all session participants with the same master
   key is needed.

4.3.3.  Complexity Implications

   The usage of security functions can surface complexity implications
   from 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



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   value in the RTP packet of an unencrypted session, the cost is higher
   when using cryptographic security functions.  For example, if using
   SRTP [RFC3711], the actual security context and exact crypto key are
   determined by the SSRC field value.  If one changes SSRC, the
   encryption and authentication must use 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.

5.  RTP Multiplexing Design Choices

   This section discusses how some RTP multiplexing design choices can
   be used in applications to achieve certain goals, and a summary of
   the implications of such choices.  For each design there is
   discussion of benefits and downsides.

5.1.  Single SSRC per Endpoint

   In this design 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 RTP stream or bi-directional, i.e. both endpoints have
   one RTP stream each.  If the application needs additional media flows
   between the endpoints, they will have to establish additional RTP
   sessions.

   The Pros:

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

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

   3.  It is possible to control security association per RTP stream
       with current key-management, since each RTP stream is directly
       related to an RTP session, and the most used keying mechanisms
       operates on a per-session basis.

   The Cons:

   a.  The number of RTP sessions grows directly in proportion with the
       number of RTP streams, which has the implications:

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





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       *  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 RTP stream to fail during
          transport between the endpoints.

   b.  When the number of RTP sessions grows, the amount of explicit
       state for relating RTP stream also grows, linearly, depending on
       how the application needs to relate RTP 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 RTP stream usage is highly dynamic,
       i.e. entering and leaving, the amount of signalling can grow
       high.  Issues can also arise from the timely establishment of
       additional RTP sessions.

   e.  If, against the recommendation, the same SSRC value is reused in
       multiple RTP sessions rather than being randomly chosen,
       interworking with applications that use a different multiplexing
       structure will require SSRC translation.

   RTP applications that need to interwork with legacy RTP applications
   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
   RTP 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.

5.2.  Multiple SSRCs of the Same Media Type

   In this design, each RTP session serves only a single media type.
   The RTP session can contain multiple RTP 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:



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   1.  Low number of RTP sessions needed compared to Single SSRC per
       Endpoint case.  This implies:

       *  Reduced NAT/FW state

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

   2.  Works well with Split Component Terminal (see Section 3.10 of
       [RFC7667]) where the split is per media type.

   3.  Enables Flow-based QoS with different prioritisation between
       media types.

   4.  For applications with dynamic usage of RTP streams, i.e.
       frequently added and removed, having much of the state associated
       with the RTP session rather than per individual SSRC can avoid
       the need for in-session signalling of meta-information about each
       SSRC.

   5.  Low overhead for security association establishment.

   The Cons:

   a.  Some potential for concern with legacy implementations that don't
       support the RTP specification fully when it comes to handling
       multiple SSRC per endpoint.

   b.  Not possible to control security association for sets of RTP
       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 RTP 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
   enables 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.

5.3.  Multiple Sessions for one Media type

   This design goes one step further than above (Section 5.2) by using
   multiple RTP sessions also for a single media type.  The main reason
   for going in this direction is that the RTP application needs
   separation of the RTP streams due to their usage.  Some typical



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   reasons for going to this design are scalability over multicast,
   simulcast, need for extended QoS prioritisation of RTP streams due to
   their usage in the application, or the need for fine- grained
   signalling using today's tools.

   The Pros:

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

   2.  The application can indicate its usage of the RTP streams on RTP
       session level, in case 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 Split Component Terminal (see Section 3.10 of
       [RFC7667]).

   6.  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.  For RTP streams that are part of scalability, simulcast or
       transport robustness, a method to bind sources across multiple
       RTP sessions is needed.

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

   e.  Higher overhead for security association establishment due to the
       increased number of RTP sessions.

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



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   For more complex RTP applications that have several different usages
   for RTP 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.

5.4.  Multiple Media Types in one Session

   This design uses a single RTP session for multiple different media
   types, like audio and video, and possibly also transport robustness
   mechanisms like FEC or Retransmission.  An endpoint can have zero,
   one or more media sources per media type.  Resulting in a number of
   RTP streams of various media types and both source and redundancy
   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.  Can handle dynamic allocations of RTP 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.

   3.  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 the potential need of SSRC translation.

   b.  Negotiation of bandwidth for the different media types is
       currently only possible using RID [I-D.ietf-mmusic-rid] in SDP.

   c.  Not suitable for Split Component Terminal (see Section 3.10 of
       [RFC7667]).

   d.  Flow-based QoS cannot provide separate treatment of RTP streams
       compared to others in the single RTP session.



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

   f.  Not suitable for applications where some receivers like to
       receive only a subset of the RTP 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.

5.5.  Summary

   There are some clear similarities between these designs.  Both the
   "Single SSRC per Endpoint" and the "Multiple Media Types in one
   Session" are cases that 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 SSRC level binding.  From another perspective, the two
   solutions are the two extreme points when it comes to number of RTP
   sessions needed.

   The two other designs "Multiple SSRCs of the Same Media Type" and
   "Multiple Sessions for one Media Type" are two examples that
   primarily allows for some implicit mapping of the role or usage of
   the RTP streams based on which RTP session they appear in.  It thus
   potentially allows for less signalling and in particular reduces the
   need for real-time signalling in dynamic sessions.  They also
   represent points in between the first two designs when it comes to
   amount of RTP sessions established, i.e. representing an attempt to
   balance the amount of RTP sessions with the functionality the
   communication session provides both on network level and on
   signalling level.

6.  Guidelines

   This section contains a number of multi-stream guidelines for
   implementers or specification writers.





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   Do not use the same SSRC across RTP sessions:  As discussed in
      Section 3.4.3 there exist drawbacks in using the same SSRC in
      multiple RTP sessions as a mechanism to bind related RTP streams
      together.  It is instead recommended to use a mechanism to
      explicitly signal the relation, either in RTP/RTCP or in the
      signalling mechanism used to establish the RTP session(s).

   Use additional RTP streams for additional media sources:  In the
      cases where an RTP endpoint needs to transmit additional RTP
      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 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 RTP stream within a single RTP session is suggested.

   Use additional RTP sessions for streams with different requirements:

      When RTP 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 a signalling mechanism,
      for example The Session Description Protocol (SDP) Grouping
      Framework [RFC5888], using some appropriate grouping semantics.

   RTP/RTCP Extensions Support Multiple RTP Streams as well as Multiple
   RTP sessions:
      When defining an RTP or RTCP extension, the creator needs to
      consider if this extension is applicable to use with additional
      SSRCs and multiple RTP sessions.  Any extension intended to be
      generic must support both.  Extensions 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 must include support for both multiple RTP
      streams in the same RTP sessions and multiple RTP sessions, such
      that application developers can choose freely from the set of
      mechanisms without concerning themselves with which of the
      multiplexing choices a particular solution supports.





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7.  Open Issues

   There are currently some issues that needs to be resolved before this
   document is ready to be published:

   1.  Does the MSID text need to be updated and clarified based on the
       evolution of MSID since previous version.  Section 3.4.3.

   2.  Changed definitions needs review and consideration.

8.  IANA Considerations

   This document makes no request of IANA.

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

9.  Security Considerations

   The security considerations of the RTP specification [RFC3550] and
   any applicable RTP profile [RFC3551],[RFC4585],[RFC3711], the
   extensions for sending multiple media types in a single RTP session
   [I-D.ietf-avtcore-multi-media-rtp-session], MSID
   [I-D.ietf-mmusic-msid], RID [I-D.ietf-mmusic-rid], BUNDLE
   [I-D.ietf-mmusic-sdp-bundle-negotiation], [RFC5760], [RFC5761], apply
   if selected and thus needs to be considered in the evaluation.

   There is discussion of the security implications of choosing multiple
   SSRC vs multiple RTP sessions in Section 4.3.

10.  Contributors

   Hui Zheng (Marvin) from Huawei contributed to WG draft versions -04
   and -05 of the document.

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, DOI 10.17487/RFC3550,
              July 2003, <https://www.rfc-editor.org/info/rfc3550>.








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   [RFC7656]  Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
              B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
              for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
              DOI 10.17487/RFC7656, November 2015,
              <https://www.rfc-editor.org/info/rfc7656>.

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.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-13 (work in
              progress), December 2015.

   [I-D.ietf-avtext-rid]
              Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
              Identifier Source Description (SDES)", draft-ietf-avtext-
              rid-09 (work in progress), October 2016.

   [I-D.ietf-mmusic-msid]
              Alvestrand, H., "WebRTC MediaStream Identification in the
              Session Description Protocol", draft-ietf-mmusic-msid-16
              (work in progress), February 2017.

   [I-D.ietf-mmusic-rid]
              Roach, A., "RTP Payload Format Restrictions", draft-ietf-
              mmusic-rid-15 (work in progress), May 2018.

   [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-52 (work in progress), May 2018.

   [I-D.ietf-mmusic-sdp-simulcast]
              Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
              "Using Simulcast in SDP and RTP Sessions", draft-ietf-
              mmusic-sdp-simulcast-13 (work in progress), June 2018.






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   [I-D.ietf-perc-srtp-ekt-diet]
              Jennings, C., Mattsson, J., McGrew, D., Wing, D., and F.
              Andreasen, "Encrypted Key Transport for DTLS and Secure
              RTP", draft-ietf-perc-srtp-ekt-diet-07 (work in progress),
              March 2018.

   [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,
              DOI 10.17487/RFC2198, September 1997,
              <https://www.rfc-editor.org/info/rfc2198>.

   [RFC2205]  Braden, R., Ed., Zhang, L., Berson, S., Herzog, S., and S.
              Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1
              Functional Specification", RFC 2205, DOI 10.17487/RFC2205,
              September 1997, <https://www.rfc-editor.org/info/rfc2205>.

   [RFC2474]  Nichols, K., Blake, S., Baker, F., and D. Black,
              "Definition of the Differentiated Services Field (DS
              Field) in the IPv4 and IPv6 Headers", RFC 2474,
              DOI 10.17487/RFC2474, December 1998,
              <https://www.rfc-editor.org/info/rfc2474>.

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, DOI 10.17487/RFC2974,
              October 2000, <https://www.rfc-editor.org/info/rfc2974>.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,
              <https://www.rfc-editor.org/info/rfc3261>.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,
              <https://www.rfc-editor.org/info/rfc3264>.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
              September 2002, <https://www.rfc-editor.org/info/rfc3389>.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              DOI 10.17487/RFC3551, July 2003,
              <https://www.rfc-editor.org/info/rfc3551>.





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   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,
              <https://www.rfc-editor.org/info/rfc3711>.

   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              DOI 10.17487/RFC3830, August 2004,
              <https://www.rfc-editor.org/info/rfc3830>.

   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
              <https://www.rfc-editor.org/info/rfc4103>.

   [RFC4383]  Baugher, M. and E. Carrara, "The Use of Timed Efficient
              Stream Loss-Tolerant Authentication (TESLA) in the Secure
              Real-time Transport Protocol (SRTP)", RFC 4383,
              DOI 10.17487/RFC4383, February 2006,
              <https://www.rfc-editor.org/info/rfc4383>.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <https://www.rfc-editor.org/info/rfc4566>.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
              <https://www.rfc-editor.org/info/rfc4568>.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              DOI 10.17487/RFC4585, July 2006,
              <https://www.rfc-editor.org/info/rfc4585>.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,
              <https://www.rfc-editor.org/info/rfc4588>.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
              February 2008, <https://www.rfc-editor.org/info/rfc5104>.

   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December
              2007, <https://www.rfc-editor.org/info/rfc5109>.



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   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,
              <https://www.rfc-editor.org/info/rfc5245>.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
              <https://www.rfc-editor.org/info/rfc5576>.

   [RFC5760]  Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
              Protocol (RTCP) Extensions for Single-Source Multicast
              Sessions with Unicast Feedback", RFC 5760,
              DOI 10.17487/RFC5760, February 2010,
              <https://www.rfc-editor.org/info/rfc5760>.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761,
              DOI 10.17487/RFC5761, April 2010,
              <https://www.rfc-editor.org/info/rfc5761>.

   [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,
              DOI 10.17487/RFC5764, May 2010,
              <https://www.rfc-editor.org/info/rfc5764>.

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888,
              DOI 10.17487/RFC5888, June 2010,
              <https://www.rfc-editor.org/info/rfc5888>.

   [RFC6465]  Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
              time Transport Protocol (RTP) Header Extension for Mixer-
              to-Client Audio Level Indication", RFC 6465,
              DOI 10.17487/RFC6465, December 2011,
              <https://www.rfc-editor.org/info/rfc6465>.

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
              <https://www.rfc-editor.org/info/rfc7201>.

   [RFC7657]  Black, D., Ed. and P. Jones, "Differentiated Services
              (Diffserv) and Real-Time Communication", RFC 7657,
              DOI 10.17487/RFC7657, November 2015,
              <https://www.rfc-editor.org/info/rfc7657>.




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   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,
              <https://www.rfc-editor.org/info/rfc7667>.

   [RFC7826]  Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
              and M. Stiemerling, Ed., "Real-Time Streaming Protocol
              Version 2.0", RFC 7826, DOI 10.17487/RFC7826, December
              2016, <https://www.rfc-editor.org/info/rfc7826>.

   [RFC8088]  Westerlund, M., "How to Write an RTP Payload Format",
              RFC 8088, DOI 10.17487/RFC8088, May 2017,
              <https://www.rfc-editor.org/info/rfc8088>.

   [RFC8108]  Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session",
              RFC 8108, DOI 10.17487/RFC8108, March 2017,
              <https://www.rfc-editor.org/info/rfc8108>.

Appendix A.  Dismissing Payload Type Multiplexing

   This section documents a number of reasons why using the payload type
   as a multiplexing point is unsuitable for most things related to
   multiple RTP streams.  If one attempts to use Payload type
   multiplexing beyond its 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 RTP 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, RTP 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 RTP timestamp rate.  When
        this is not possible, payload type multiplexing cannot be used.

   2.   Many RTP payload formats can fragment a media object over
        multiple RTP 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 RTP stream are sent in
        sequence.

   3.   Some media formats require uninterrupted sequence number space
        between media parts.  These are media formats where any missing



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        RTP sequence number will result in decoding failure or invoking
        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 RTP 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, which stream a
        packet loss relates to, and thus to which stream to potentially
        apply packet loss concealment or other stream-specific loss
        mitigation mechanisms.

   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 RTP streams based on stream ID (SSRC), packet
        (sequence numbers) and time interval (RTP Timestamps).  There is
        almost never a field to indicate which payload type is reported,
        so sending feedback for a specific RTP payload type 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 RTP stream when using payload type multiplexing.





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   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 does not understand the indication that
        different RTP payload types are different RTP streams might be
        slightly confused by the large amount of possibly overlapping or
        identically defined RTP payload types.

Appendix B.  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.

   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.  RTSP
   [RFC7826] 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.

B.1.  Session Oriented Properties

   One aspect of the existing signalling is that it is focused around
   RTP sessions, or at least in the case of SDP the media description.
   There are a number of things that are signalled on media description
   level but those are not necessarily strictly bound to an RTP session
   and could be of interest to signal specifically for a particular RTP
   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 as a common
      "any RTP stream" limit, unless either codec-specific bandwidth
      limiting or RTCP signalling using TMMBR is used.

   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).




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   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 RTP 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 RTP 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 RTP payload type, resolution and
   framerate, a SSRC-linked mechanism has been proposed in
   [I-D.ietf-mmusic-rid]

B.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 and is 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.

   [I-D.ietf-mmusic-sdp-bundle-negotiation] describes how to let
   multiple SDP media descriptions use a single underlying transport in
   SDP, which allows to define one RTP session with media types having
   different MIME top level types.

B.3.  Signalling RTP stream Usage

   RTP streams being transported in RTP has some particular usage in an
   RTP application.  This usage of the RTP 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 RTP streams there will be more than a single usage or
   purpose among the set of RTP streams being sent or received.  RTP
   applications will need to signal this usage somehow.  The signalling
   used will have to identify the RTP streams affected by their RTP-
   level identifiers, which means that they have to be identified either
   by their session or by their SSRC + session.

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



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   If all RTP 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 RTP streams with different usage in the right
   fashion.

Authors' Addresses

   Magnus Westerlund
   Ericsson
   Torshamsgatan 23
   SE-164 80 Kista
   Sweden

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


   Bo Burman
   Ericsson
   Gronlandsgatan 31
   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


   Roni Even
   Huawei

   Email: roni.even@huawei.com






































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