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Versions: (draft-lennox-raiarea-rtp-grouping-taxonomy) 00 01 02 03 04 05 06 07 08 RFC 7656

Network Working Group                                          J. Lennox
Internet-Draft                                                     Vidyo
Intended status: Informational                                  K. Gross
Expires: January 21, 2016                                            AVA
                                                           S. Nandakumar
                                                            G. Salgueiro
                                                           Cisco Systems
                                                          B. Burman, Ed.
                                                                Ericsson
                                                           July 20, 2015


A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol
                             (RTP) Sources
               draft-ietf-avtext-rtp-grouping-taxonomy-08

Abstract

   The terminology about, and associations among, Real-Time Transport
   Protocol (RTP) sources can be complex and somewhat opaque.  This
   document describes a number of existing and proposed properties and
   relationships among RTP sources, and defines common terminology for
   discussing protocol entities and their relationships.

Status of This Memo

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

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

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

   This Internet-Draft will expire on January 21, 2016.

Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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



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

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
   2.  Concepts  . . . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.1.  Media Chain . . . . . . . . . . . . . . . . . . . . . . .   5
       2.1.1.  Physical Stimulus . . . . . . . . . . . . . . . . . .   9
       2.1.2.  Media Capture . . . . . . . . . . . . . . . . . . . .   9
       2.1.3.  Raw Stream  . . . . . . . . . . . . . . . . . . . . .   9
       2.1.4.  Media Source  . . . . . . . . . . . . . . . . . . . .  10
       2.1.5.  Source Stream . . . . . . . . . . . . . . . . . . . .  10
       2.1.6.  Media Encoder . . . . . . . . . . . . . . . . . . . .  11
       2.1.7.  Encoded Stream  . . . . . . . . . . . . . . . . . . .  12
       2.1.8.  Dependent Stream  . . . . . . . . . . . . . . . . . .  12
       2.1.9.  Media Packetizer  . . . . . . . . . . . . . . . . . .  12
       2.1.10. RTP Stream  . . . . . . . . . . . . . . . . . . . . .  13
       2.1.11. RTP-based Redundancy  . . . . . . . . . . . . . . . .  13
       2.1.12. Redundancy RTP Stream . . . . . . . . . . . . . . . .  14
       2.1.13. RTP-based Security  . . . . . . . . . . . . . . . . .  14
       2.1.14. Secured RTP Stream  . . . . . . . . . . . . . . . . .  15
       2.1.15. Media Transport . . . . . . . . . . . . . . . . . . .  15
       2.1.16. Media Transport Sender  . . . . . . . . . . . . . . .  16
       2.1.17. Sent RTP Stream . . . . . . . . . . . . . . . . . . .  17
       2.1.18. Network Transport . . . . . . . . . . . . . . . . . .  17
       2.1.19. Transported RTP Stream  . . . . . . . . . . . . . . .  17
       2.1.20. Media Transport Receiver  . . . . . . . . . . . . . .  17
       2.1.21. Received Secured RTP Stream . . . . . . . . . . . . .  18
       2.1.22. RTP-based Validation  . . . . . . . . . . . . . . . .  18
       2.1.23. Received RTP Stream . . . . . . . . . . . . . . . . .  18
       2.1.24. Received Redundancy RTP Stream  . . . . . . . . . . .  18
       2.1.25. RTP-based Repair  . . . . . . . . . . . . . . . . . .  18
       2.1.26. Repaired RTP Stream . . . . . . . . . . . . . . . . .  18
       2.1.27. Media Depacketizer  . . . . . . . . . . . . . . . . .  19
       2.1.28. Received Encoded Stream . . . . . . . . . . . . . . .  19
       2.1.29. Media Decoder . . . . . . . . . . . . . . . . . . . .  19
       2.1.30. Received Source Stream  . . . . . . . . . . . . . . .  19
       2.1.31. Media Sink  . . . . . . . . . . . . . . . . . . . . .  19
       2.1.32. Received Raw Stream . . . . . . . . . . . . . . . . .  20
       2.1.33. Media Render  . . . . . . . . . . . . . . . . . . . .  20
     2.2.  Communication Entities  . . . . . . . . . . . . . . . . .  20
       2.2.1.  Endpoint  . . . . . . . . . . . . . . . . . . . . . .  22



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       2.2.2.  RTP Session . . . . . . . . . . . . . . . . . . . . .  22
       2.2.3.  Participant . . . . . . . . . . . . . . . . . . . . .  23
       2.2.4.  Multimedia Session  . . . . . . . . . . . . . . . . .  23
       2.2.5.  Communication Session . . . . . . . . . . . . . . . .  24
   3.  Concepts of Inter-Relations . . . . . . . . . . . . . . . . .  24
     3.1.  Synchronization Context . . . . . . . . . . . . . . . . .  24
       3.1.1.  RTCP CNAME  . . . . . . . . . . . . . . . . . . . . .  25
       3.1.2.  Clock Source Signaling  . . . . . . . . . . . . . . .  25
       3.1.3.  Implicitly via RtcMediaStream . . . . . . . . . . . .  25
       3.1.4.  Explicitly via SDP Mechanisms . . . . . . . . . . . .  25
     3.2.  Endpoint  . . . . . . . . . . . . . . . . . . . . . . . .  25
     3.3.  Participant . . . . . . . . . . . . . . . . . . . . . . .  26
     3.4.  RtcMediaStream  . . . . . . . . . . . . . . . . . . . . .  26
     3.5.  Multi-Channel Audio . . . . . . . . . . . . . . . . . . .  26
     3.6.  Simulcast . . . . . . . . . . . . . . . . . . . . . . . .  27
     3.7.  Layered Multi-Stream  . . . . . . . . . . . . . . . . . .  28
     3.8.  RTP Stream Duplication  . . . . . . . . . . . . . . . . .  29
     3.9.  Redundancy Format . . . . . . . . . . . . . . . . . . . .  30
     3.10. RTP Retransmission  . . . . . . . . . . . . . . . . . . .  31
     3.11. Forward Error Correction  . . . . . . . . . . . . . . . .  33
     3.12. RTP Stream Separation . . . . . . . . . . . . . . . . . .  34
     3.13. Multiple RTP Sessions over one Media Transport  . . . . .  35
   4.  Mapping from Existing Terms . . . . . . . . . . . . . . . . .  35
     4.1.  Telepresence Terms  . . . . . . . . . . . . . . . . . . .  35
       4.1.1.  Audio Capture . . . . . . . . . . . . . . . . . . . .  35
       4.1.2.  Capture Device  . . . . . . . . . . . . . . . . . . .  35
       4.1.3.  Capture Encoding  . . . . . . . . . . . . . . . . . .  36
       4.1.4.  Capture Scene . . . . . . . . . . . . . . . . . . . .  36
       4.1.5.  Endpoint  . . . . . . . . . . . . . . . . . . . . . .  36
       4.1.6.  Individual Encoding . . . . . . . . . . . . . . . . .  36
       4.1.7.  Media Capture . . . . . . . . . . . . . . . . . . . .  36
       4.1.8.  Media Consumer  . . . . . . . . . . . . . . . . . . .  36
       4.1.9.  Media Provider  . . . . . . . . . . . . . . . . . . .  37
       4.1.10. Stream  . . . . . . . . . . . . . . . . . . . . . . .  37
       4.1.11. Video Capture . . . . . . . . . . . . . . . . . . . .  37
     4.2.  Media Description . . . . . . . . . . . . . . . . . . . .  37
     4.3.  Media Stream  . . . . . . . . . . . . . . . . . . . . . .  37
     4.4.  Multimedia Conference . . . . . . . . . . . . . . . . . .  37
     4.5.  Multimedia Session  . . . . . . . . . . . . . . . . . . .  38
     4.6.  Multipoint Control Unit (MCU) . . . . . . . . . . . . . .  38
     4.7.  Multi-Session Transmission (MST)  . . . . . . . . . . . .  38
     4.8.  Recording Device  . . . . . . . . . . . . . . . . . . . .  39
     4.9.  RtcMediaStream  . . . . . . . . . . . . . . . . . . . . .  39
     4.10. RtcMediaStreamTrack . . . . . . . . . . . . . . . . . . .  39
     4.11. RTP Sender  . . . . . . . . . . . . . . . . . . . . . . .  39
     4.12. RTP Session . . . . . . . . . . . . . . . . . . . . . . .  39
     4.13. Single Session Transmission (SST) . . . . . . . . . . . .  39
     4.14. SSRC  . . . . . . . . . . . . . . . . . . . . . . . . . .  39



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   5.  Security Considerations . . . . . . . . . . . . . . . . . . .  40
   6.  Acknowledgement . . . . . . . . . . . . . . . . . . . . . . .  40
   7.  Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  40
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  41
   9.  Informative References  . . . . . . . . . . . . . . . . . . .  41
   Appendix A.  Changes From Earlier Versions  . . . . . . . . . . .  44
     A.1.  Modifications Between WG Version -07 and -08  . . . . . .  44
     A.2.  Modifications Between WG Version -06 and -07  . . . . . .  45
     A.3.  Modifications Between WG Version -05 and -06  . . . . . .  45
     A.4.  Modifications Between WG Version -04 and -05  . . . . . .  46
     A.5.  Modifications Between WG Version -03 and -04  . . . . . .  46
     A.6.  Modifications Between WG Version -02 and -03  . . . . . .  47
     A.7.  Modifications Between WG Version -01 and -02  . . . . . .  47
     A.8.  Modifications Between WG Version -00 and -01  . . . . . .  48
     A.9.  Modifications Between Version -02 and -03 . . . . . . . .  48
     A.10. Modifications Between Version -01 and -02 . . . . . . . .  48
     A.11. Modifications Between Version -00 and -01 . . . . . . . .  48
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  49

1.  Introduction

   The existing taxonomy of sources in the Real-Time Transport Protocol
   (RTP) [RFC3550] has previously been regarded as confusing and
   inconsistent.  Consequently, a deep understanding of how the
   different terms relate to each other becomes a real challenge.
   Frequently cited examples of this confusion are (1) how different
   protocols that make use of RTP use the same terms to signify
   different things and (2) how the complexities addressed at one layer
   are often glossed over or ignored at another.

   This document improves clarity by reviewing the semantics of various
   aspects of sources in RTP.  As an organizing mechanism, it approaches
   this by describing various ways that RTP sources are transformed on
   their way between sender and receiver, and how they can be grouped
   and associated together.

   All non-specific references to ControLling mUltiple streams for
   tElepresence (CLUE) in this document map to [I-D.ietf-clue-framework]
   and all references to Web Real-Time Communications (WebRTC) map to
   [I-D.ietf-rtcweb-overview].

2.  Concepts

   This section defines concepts that serve to identify and name various
   transformations and streams in a given RTP usage.  For each concept,
   alternate definitions and usages that co-exist today are listed along
   with various characteristics that further describes the concept.
   These concepts are divided into two categories, one related to the



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   chain of streams and transformations that media can be subject to,
   the other for entities involved in the communication.

2.1.  Media Chain

   In the context of this document, Media is a sequence of synthetic or
   Physical Stimuli (Section 2.1.1) (sound waves, photons, key-strokes),
   represented in digital form.  Synthesized Media is typically
   generated directly in the digital domain.

   This section contains the concepts that can be involved in taking
   Media at a sender side and transporting it to a receiver, which may
   recover a sequence of physical stimuli.  This chain of concepts is of
   two main types, streams and transformations.  Streams are time-based
   sequences of samples of the physical stimulus in various
   representations, while transformations changes the representation of
   the streams in some way.

   The below examples are basic ones and it is important to keep in mind
   that this conceptual model enables more complex usages.  Some will be
   further discussed in later sections of this document.  In general the
   following applies to this model:

   o  A transformation may have zero or more inputs and one or more
      outputs.

   o  A stream is of some type, such as audio, video, real-time text,
      etc.

   o  A stream has one source transformation and one or more sink
      transformations (with the exception of Physical Stimulus
      (Section 2.1.1) that may lack source or sink transformation).

   o  Streams can be forwarded from a transformation output to any
      number of inputs on other transformations that support that type.

   o  If the output of a transformation is sent to multiple
      transformations, those streams will be identical; it takes a
      transformation to make them different.

   o  There are no formal limitations on how streams are connected to
      transformations.

   It is also important to remember that this is a conceptual model.
   Thus real-world implementations may look different and have different
   structure.





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   To provide a basic understanding of the relationships in the chain we
   first introduce the concepts for the sender side (Figure 1).  This
   covers physical stimuli until media packets are emitted onto the
   network.

               Physical Stimulus
                      |
                      V
           +----------------------+
           |     Media Capture    |
           +----------------------+
                      |
                 Raw Stream
                      V
           +----------------------+
           |     Media Source     |<- Synchronization Timing
           +----------------------+
                      |
                Source Stream
                      V
           +----------------------+
           |    Media Encoder     |
           +----------------------+
                      |
                Encoded Stream      +------------+
                      V             |            V
           +----------------------+ | +----------------------+
           |   Media Packetizer   | | | RTP-based Redundancy |
           +----------------------+ | +----------------------+
                      |             |            |
                      +-------------+  Redundancy RTP Stream
               Source RTP Stream                 |
                      V                          V
           +----------------------+   +----------------------+
           |  RTP-based Security  |   |  RTP-based Security  |
           +----------------------+   +----------------------+
                      |                          |
              Secured RTP Stream   Secured Redundancy RTP Stream
                      V                          V
           +----------------------+   +----------------------+
           |   Media Transport    |   |   Media Transport    |
           +----------------------+   +----------------------+

             Figure 1: Sender Side Concepts in the Media Chain

   In Figure 1 we have included a branched chain to cover the concepts
   for using redundancy to improve the reliability of the transport.




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   The Media Transport concept is an aggregate that is decomposed in
   Section 2.1.15.

   In Figure 2 we review a receiver media chain matching the sender
   side, to look at the inverse transformations and their attempts to
   recover identical streams as in the sender chain, subject to what may
   be lossy compression and imperfect Media Transport.  Note that the
   streams out of a reverse transformation, like the Source Stream out
   the Media Decoder are in many cases not the same as the corresponding
   ones on the sender side, thus they are prefixed with a "Received" to
   denote a potentially modified version.  The reason for not being the
   same lies in the transformations that can be of irreversible type.
   For example, lossy source coding in the Media Encoder prevents the
   Source Stream out of the Media Decoder to be the same as the one fed
   into the Media Encoder.  Other reasons include packet loss or late
   loss in the Media Transport transformation that even RTP-based
   Repair, if used, fails to repair.  However, some transformations are
   not always present, like RTP-based Repair that cannot operate without
   Redundancy RTP Streams.
































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          +----------------------+   +----------------------+
          |   Media Transport    |   |   Media Transport    |
          +----------------------+   +----------------------+
            Received |                 Received | Secured
            Secured RTP Stream       Redundancy RTP Stream
                     V                          V
          +----------------------+   +----------------------+
          | RTP-based Validation |   | RTP-based Validation |
          +----------------------+   +----------------------+
                     |                          |
            Received RTP Stream   Received Redundancy RTP Stream
                     |                          |
                     |     +--------------------+
                     V     V
          +----------------------+
          |   RTP-based Repair   |
          +----------------------+
                     |
            Repaired RTP Stream
                     V
          +----------------------+
          |  Media Depacketizer  |
          +----------------------+
                     |
           Received Encoded Stream
                     V
          +----------------------+
          |    Media Decoder     |
          +----------------------+
                     |
           Received Source Stream
                     V
          +----------------------+
          |      Media Sink      |--> Synchronization Information
          +----------------------+
                     |
            Received Raw Stream
                     V
          +----------------------+
          |    Media Renderer    |
          +----------------------+
                     |
                     V
             Physical Stimulus

            Figure 2: Receiver Side Concepts of the Media Chain





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2.1.1.  Physical Stimulus

   The Physical Stimulus is a physical event in the analog domain that
   can be sampled and converted to digital form by an appropriate sensor
   or transducer.  This include sound waves making up audio, photons in
   a light field, or other excitations or interactions with sensors,
   like keystrokes on a keyboard.

2.1.2.  Media Capture

   Media Capture is the process of transforming the analog Physical
   Stimulus (Section 2.1.1) into digital Media using an appropriate
   sensor or transducer.  The Media Capture performs a digital sampling
   of the physical stimulus, usually periodically, and outputs this in
   some representation as a Raw Stream (Section 2.1.3).  This data is
   considered "Media", because it includes data that is periodically
   sampled, or made up of a set of timed asynchronous events.  The Media
   Capture is normally instantiated in some type of device, i.e. media
   capture device.  Examples of different types of media capturing
   devices are digital cameras, microphones connected to A/D converters,
   or keyboards.

   Characteristics:

   o  A Media Capture is identified either by hardware/manufacturer ID
      or via a session-scoped device identifier as mandated by the
      application usage.

   o  A Media Capture can generate an Encoded Stream (Section 2.1.7) if
      the capture device supports such a configuration.

   o  The nature of the Media Capture may impose constraints on the
      clock handling in some of the subsequent steps.  For example, many
      audio or video capture devices are not completely free in
      selecting the sample rate.

2.1.3.  Raw Stream

   A Raw Stream is the time progressing stream of digitally sampled
   information, usually periodically sampled and provided by a Media
   Capture (Section 2.1.2).  A Raw Stream can also contain synthesized
   Media that may not require any explicit Media Capture, since it is
   already in an appropriate digital form.








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2.1.4.  Media Source

   A Media Source is the logical source of a time progressing digital
   media stream synchronized to a reference clock.  This stream is
   called a Source Stream (Section 2.1.5).  This transformation takes
   one or more Raw Streams (Section 2.1.3) and provides a Source Stream
   as output.  The output is synchronized with a reference clock
   (Section 3.1), which can be as simple as a system local wall clock or
   as complex as an NTP synchronized clock.

   The output can be of different types.  One type is directly
   associated with a particular Media Capture's Raw Stream.  Others are
   more conceptual sources, like an audio mix of multiple Source Streams
   (Figure 3).  Mixing multiple streams typically requires that the
   input streams are possible to relate in time, meaning that they have
   to be Source Streams (Section 2.1.5) rather than Raw Streams.  In
   Figure 3, the generated Source Stream is a mix of the three input
   Source Streams.

                Source    Source    Source
                Stream    Stream    Stream
                  |         |         |
                  V         V         V
              +--------------------------+
              |        Media Source      |<-- Reference Clock
              |           Mixer          |
              +--------------------------+
                            |
                            V
                      Source Stream

         Figure 3: Conceptual Media Source in form of Audio Mixer

   Another possible example of a conceptual Media Source is a video
   surveillance switch, where the input is multiple Source Streams from
   different cameras, and the output is one of those Source Streams
   based on some selection criteria, like a round-robin or based on some
   video activity measure.

2.1.5.  Source Stream

   A Source Stream is a stream of digital samples that has been
   synchronized with a reference clock and comes from particular Media
   Source (Section 2.1.4).







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2.1.6.  Media Encoder

   A Media Encoder is a transform that is responsible for encoding the
   media data from a Source Stream (Section 2.1.5) into another
   representation, usually more compact, that is output as an Encoded
   Stream (Section 2.1.7).

   The Media Encoder step commonly includes pre-encoding
   transformations, such as scaling, resampling etc.  The Media Encoder
   can have a significant number of configuration options that affects
   the properties of the Encoded Stream.  This include properties such
   as codec, bit-rate, start points for decoding, resolution, bandwidth
   or other fidelity affecting properties.

   Scalable Media Encoders need special attention as they produce
   multiple outputs that are potentially of different types.  As shown
   in Figure 4, a scalable Media Encoder takes one input Source Stream
   and encodes it into multiple output streams of two different types;
   at least one Encoded Stream that is independently decodable and one
   or more Dependent Streams (Section 2.1.8).  Decoding requires at
   least one Encoded Stream and zero or more Dependent Streams.  A
   Dependent Stream's dependency is one of the grouping relations this
   document discusses further in Section 3.7.

                              Source Stream
                                    |
                                    V
                       +--------------------------+
                       |  Scalable Media Encoder  |
                       +--------------------------+
                          |         |   ...    |
                          V         V          V
                       Encoded  Dependent  Dependent
                       Stream    Stream     Stream

            Figure 4: Scalable Media Encoder Input and Outputs

   There are also other variants of encoders, like so-called Multiple
   Description Coding (MDC).  Such Media Encoders produce multiple
   independent and thus individually decodable Encoded Streams.
   However, (logically) combining multiple of these Encoded Streams into
   a single Received Source Stream during decoding leads to an
   improvement in perceptual reproduced quality when compared to
   decoding a single Encoded Stream.

   Creating multiple Encoded Streams from the same Source Stream, where
   the Encoded Streams are neither in a scalable nor in an MDC




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   relationship is commonly utilized in Simulcast
   [I-D.ietf-mmusic-sdp-simulcast] environments.

2.1.7.  Encoded Stream

   A stream of time synchronized encoded media that can be independently
   decoded.

   Due to temporal dependencies, an Encoded Stream may have limitations
   in where decoding can be started.  These entry points, for example
   Intra frames from a video encoder, may require identification and
   their generation may be event based or configured to occur
   periodically.

2.1.8.  Dependent Stream

   A stream of time synchronized encoded media fragments that are
   dependent on one or more Encoded Streams (Section 2.1.7) and zero or
   more Dependent Streams to be possible to decode.

   Each Dependent Stream has a set of dependencies.  These dependencies
   must be understood by the parties in a Multimedia Session that intend
   to use a Dependent Stream.

2.1.9.  Media Packetizer

   The transformation of taking one or more Encoded (Section 2.1.7) or
   Dependent Streams (Section 2.1.8) and putting their content into one
   or more sequences of packets, normally RTP packets, and output Source
   RTP Streams (Section 2.1.10).  This step includes both generating RTP
   payloads as well as RTP packets.  The Media Packetizer then selects
   which Synchronization source(s) (SSRC) [RFC3550] and RTP Sessions to
   use.

   The Media Packetizer can combine multiple Encoded or Dependent
   Streams into one or more RTP Streams:

   o  The Media Packetizer can use multiple inputs when producing a
      single RTP Stream.  One such example is SRST packetization when
      using Scalable Video Coding (SVC) (Section 3.7).

   o  The Media Packetizer can also produce multiple RTP Streams, for
      example when Encoded and/or Dependent Streams are distributed over
      multiple RTP Streams.  One example of this is MRMT packetization
      when using SVC (Section 3.7).






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2.1.10.  RTP Stream

   An RTP Stream is a stream of RTP packets containing media data,
   source or redundant.  The RTP Stream is identified by an SSRC
   belonging to a particular RTP Session.  The RTP Session is identified
   as discussed in Section 2.2.2.

   A Source RTP Stream is an RTP Stream directly related to an Encoded
   Stream (Section 2.1.7), targeted for transport over RTP without any
   additional RTP-based Redundancy (Section 2.1.11) applied.

   Characteristics:

   o  Each RTP Stream is identified by a Synchronization source (SSRC)
      [RFC3550] that is carried in every RTP and RTP Control Protocol
      (RTCP) packet header.  The SSRC is unique in a specific RTP
      Session context.

   o  At any given point in time, a RTP Stream can have one and only one
      SSRC, but SSRCs for a given RTP Stream can change over time.  SSRC
      collision and clock rate change [RFC7160] are examples of valid
      reasons to change SSRC for an RTP Stream.  In those cases, the RTP
      Stream itself is not changed in any significant way, only the
      identifying SSRC number.

   o  Each SSRC defines a unique RTP sequence numbering and timing
      space.

   o  Several RTP Streams, each with their own SSRC, may represent a
      single Media Source.

   o  Several RTP Streams, each with their own SSRC, can be carried in a
      single RTP Session.

2.1.11.  RTP-based Redundancy

   RTP-based Redundancy is defined here as a transformation that
   generates redundant or repair packets sent out as a Redundancy RTP
   Stream (Section 2.1.12) to mitigate network transport impairments,
   like packet loss and delay.  Note that this excludes the type of
   redundancy that most suitable Media Encoders (Section 2.1.6) may add
   to the media format of the Encoded Stream (Section 2.1.7) that makes
   it cope better with inevitable RTP packet losses.

   The RTP-based Redundancy exists in many flavors; they may be
   generating independent Repair Streams that are used in addition to
   the Source Stream (like RTP Retransmission (Section 3.10) and some
   special types of Forward Error Correction, like RTP stream



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   duplication (Section 3.8)), they may generate a new Source Stream by
   combining redundancy information with source information (Using XOR
   FEC (Section 3.11) as a redundancy payload (Section 3.9)), or
   completely replace the source information with only redundancy
   packets.

2.1.12.  Redundancy RTP Stream

   A Redundancy RTP Stream is an RTP Stream (Section 2.1.10) that
   contains no original source data, only redundant data, which may
   either be used standalone or be combined with one or more Received
   RTP Streams (Section 2.1.23) to produce Repaired RTP Streams
   (Section 2.1.26).

2.1.13.  RTP-based Security

   The optional RTP-based Security transformation applies security
   services such as authentication, integrity protection and
   confidentiality to an input RTP Stream, like what is specified in The
   Secure Real-time Transport Protocol (SRTP) [RFC3711], producing a
   Secured RTP Stream (Section 2.1.14).  Either an RTP Stream
   (Section 2.1.10) or a Redundancy RTP Stream (Section 2.1.12) can be
   used as input to this transformation.

   In SRTP and the related Secure RTCP (SRTCP), all of the above
   mentioned security services are optional, except for integrity
   protection of SRTCP, which is mandatory.  Also confidentiality
   (encryption) is effectively optional in SRTP, since it is possible to
   use a NULL encryption algorithm.  As described in [RFC7201], the
   strength of SRTP data origin authentication depends on the
   cryptographic transform and key management used, for example in group
   communication where it is sometimes possible to authenticate group
   membership but not the actual RTP Stream sender.

   RTP-based Security and RTP-based Redundancy can be combined in a few
   different ways.  One way is depicted in Figure 1, where an RTP Stream
   and its corresponding Redundancy RTP Stream are protected by separate
   RTP-based Security transforms.  In other cases, like when a Media
   Translator is adding FEC in Section 3.2.1.3 of
   [I-D.ietf-avtcore-rtp-topologies-update], a middlebox can apply RTP-
   based Redundancy to an already Secured RTP Stream instead of a Source
   RTP Stream.  One example of that is depicted in Figure 5 below.









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               Source RTP Stream    +------------+
                      V             |            V
           +----------------------+ | +----------------------+
           |  RTP-based Security  | | | RTP-based Redundancy |
           +----------------------+ | +----------------------+
                      |             |            |
                      |             |  Redundancy RTP Stream
                      +-------------+            |
                      |                          V
                      |               +----------------------+
              Secured RTP Stream      |  RTP-based Security  |
                      |               +----------------------+
                      |                          |
                      |            Secured Redundancy RTP Stream
                      V                          V
           +----------------------+   +----------------------+
           |   Media Transport    |   |   Media Transport    |
           +----------------------+   +----------------------+

            Figure 5: Adding Redundancy to a Secured RTP Stream

   In this case, the Redundancy RTP Stream may already have been secured
   for confidentiality (encrypted) by the first RTP-based Security, and
   it may therefore not be necessary to apply additional confidentiality
   protection in the second RTP-based Security.  To avoid attacks and
   negative impact on RTP-based Repair (Section 2.1.25) and the
   resulting Repaired RTP Stream (Section 2.1.26), it is however still
   necessary to have this second RTP-based Security apply both
   authentication and integrity protection to the Redundancy RTP Stream.

2.1.14.  Secured RTP Stream

   A Secured RTP Stream is a Source or Redundancy RTP Stream that is
   protected through RTP-based Security (Section 2.1.13) by one or more
   of the confidentiality, integrity, or authentication security
   services.

2.1.15.  Media Transport

   A Media Transport defines the transformation that the RTP Streams
   (Section 2.1.10) are subjected to by the end-to-end transport from
   one RTP sender to one specific RTP receiver (an RTP Session
   (Section 2.2.2) may contain multiple RTP receivers per sender).  Each
   Media Transport is defined by a transport association that is
   normally identified by a 5-tuple (source address, source port,
   destination address, destination port, transport protocol), but a
   proposal exists for sending multiple transport associations on a
   single 5-tuple [I-D.westerlund-avtcore-transport-multiplexing].



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   Characteristics:

   o  Media Transport transmits RTP Streams of RTP Packets from a source
      transport address to a destination transport address.

   o  Each Media Transport contains only a single RTP Session.

   o  A single RTP Session can span multiple Media Transports.

   The Media Transport concept sometimes needs to be decomposed into
   more steps to enable discussion of what a sender emits that gets
   transformed by the network before it is received by the receiver.
   Thus we provide also this Media Transport decomposition (Figure 6).

                               RTP Stream
                                    |
                                    V
                       +--------------------------+
                       |  Media Transport Sender  |
                       +--------------------------+
                                    |
                             Sent RTP Stream
                                    V
                       +--------------------------+
                       |    Network Transport     |
                       +--------------------------+
                                    |
                         Transported RTP Stream
                                    V
                       +--------------------------+
                       | Media Transport Receiver |
                       +--------------------------+
                                    |
                                    V
                           Received RTP Stream

                Figure 6: Decomposition of Media Transport

2.1.16.  Media Transport Sender

   The first transformation within the Media Transport (Section 2.1.15)
   is the Media Transport Sender.  The sending Endpoint (Section 2.2.1)
   takes an RTP Stream and emits the packets onto the network using the
   transport association established for this Media Transport, thereby
   creating a Sent RTP Stream (Section 2.1.17).  In the process, it
   transforms the RTP Stream in several ways.  First, it generates the
   necessary protocol headers for the transport association, for example
   IP and UDP headers, thus forming IP/UDP/RTP packets.  In addition,



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   the Media Transport Sender may queue, intentionally pace or otherwise
   affect how the packets are emitted onto the network, thereby
   potentially introducing delay and delay variations [RFC5481] that
   characterize the Sent RTP Stream.

2.1.17.  Sent RTP Stream

   The Sent RTP Stream is the RTP Stream as entering the first hop of
   the network path to its destination.  The Sent RTP Stream is
   identified using network transport addresses, like for IP/UDP the
   5-tuple (source IP address, source port, destination IP address,
   destination port, and protocol (UDP)).

2.1.18.  Network Transport

   Network Transport is the transformation that subjects the Sent RTP
   Stream (Section 2.1.17) to traveling from the source to the
   destination through the network.  This transformation can result in
   loss of some packets, delay and delay variation on a per packet
   basis, packet duplication, and packet header or data corruption.
   This transformation produces a Transported RTP Stream
   (Section 2.1.19) at the exit of the network path.

2.1.19.  Transported RTP Stream

   The Transported RTP Stream is the RTP Stream that is emitted out of
   the network path at the destination, subjected to the Network
   Transport's transformation (Section 2.1.18).

2.1.20.  Media Transport Receiver

   The Media Transport Receiver is the receiver Endpoint's
   (Section 2.2.1) transformation of the Transported RTP Stream
   (Section 2.1.19) by its reception process, which results in the
   Received RTP Stream (Section 2.1.23).  This transformation includes
   transport checksums being verified.  Sensible system designs
   typically either discard packets with mis-matching checksums, or pass
   them on while somehow marking them in the resulting Received RTP
   Stream so to alert subsequent transformations about the possible
   corrupt state.  In this context it is worth noting that there is
   typically some probability for corrupt packets to pass through
   undetected (with a seemingly correct checksum).  Other
   transformations can compensate for delay variations in receiving a
   packet on the network interface and providing it to the application
   (de-jitter buffer).






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2.1.21.  Received Secured RTP Stream

   This is the Secured RTP Stream (Section 2.1.14) resulting from the
   Media Transport (Section 2.1.15) aggregate transformation.

2.1.22.  RTP-based Validation

   RTP-based Validation is the reverse transformation of RTP-based
   Security (Section 2.1.13).  If this transformation fails, the result
   is either not usable and must be discarded, or may be usable but
   cannot be trusted.  If the transformation succeeds, the result can be
   a Received RTP Stream (Section 2.1.23) or a Received Redundancy RTP
   Stream (Section 2.1.24), depending on what was input to the
   corresponding RTP-based Security transformation, but can also be a
   Received Secured RTP Stream (Section 2.1.21) in case several RTP-
   based Security transformations were applied.

2.1.23.  Received RTP Stream

   The Received RTP Stream is the RTP Stream (Section 2.1.10) resulting
   from the Media Transport's aggregate transformation (Section 2.1.15),
   i.e. subjected to packet loss, packet corruption, packet duplication,
   delay, and delay variation from sender to receiver.

2.1.24.  Received Redundancy RTP Stream

   The Received Redundancy RTP Stream is the Redundancy RTP Stream
   (Section 2.1.12) resulting from the Media Transport transformation,
   i.e. subjected to packet loss, packet corruption, delay, and delay
   variation from sender to receiver.

2.1.25.  RTP-based Repair

   RTP-based Repair is a Transformation that takes as input zero or more
   Received RTP Streams (Section 2.1.23) and one or more Received
   Redundancy RTP Streams (Section 2.1.24), and produces one or more
   Repaired RTP Streams (Section 2.1.26) that are as close to the
   corresponding sent Source RTP Streams (Section 2.1.10) as possible,
   using different RTP-based repair methods, for example the ones
   referred in RTP-based Redundancy (Section 2.1.11).

2.1.26.  Repaired RTP Stream

   A Repaired RTP Stream is a Received RTP Stream (Section 2.1.23) for
   which Received Redundancy RTP Stream (Section 2.1.24) information has
   been used to try to recover the Source RTP Stream (Section 2.1.10) as
   it was before Media Transport (Section 2.1.15).




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2.1.27.  Media Depacketizer

   A Media Depacketizer takes one or more RTP Streams (Section 2.1.10),
   depacketizes them, and attempts to reconstitute the Encoded Streams
   (Section 2.1.7) or Dependent Streams (Section 2.1.8) present in those
   RTP Streams.

   In practical implementations, the Media Depacketizer and the Media
   Decoder may be tightly coupled and share information to improve or
   optimize the overall decoding and error concealment process.  It is,
   however, not expected that there would be any benefit in defining a
   taxonomy for those detailed (and likely very implementation-
   dependent) steps.

2.1.28.  Received Encoded Stream

   The Received Encoded Stream is the received version of an Encoded
   Stream (Section 2.1.7).

2.1.29.  Media Decoder

   A Media Decoder is a transformation that is responsible for decoding
   Encoded Streams (Section 2.1.7) and any Dependent Streams
   (Section 2.1.8) into a Source Stream (Section 2.1.5).

   In practical implementations, the Media Decoder and the Media
   Depacketizer may be tightly coupled and share information to improve
   or optimize the overall decoding process in various ways.  It is
   however not expected that there would be any benefit in defining a
   taxonomy for those detailed (and likely very implementation-
   dependent) steps.

   A Media Decoder has to deal with any errors in the Encoded Streams
   that resulted from corruption or failure to repair packet losses.
   Therefore, it commonly is robust to error and losses, and includes
   concealment methods.

2.1.30.  Received Source Stream

   The Received Source Stream is the received version of a Source Stream
   (Section 2.1.5).

2.1.31.  Media Sink

   The Media Sink receives a Source Stream (Section 2.1.5) that
   contains, usually periodically, sampled media data together with
   associated synchronization information.  Depending on application,
   this Source Stream then needs to be transformed into a Raw Stream



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   (Section 2.1.3) that is conveyed to the Media Render
   (Section 2.1.33), synchronized with the output from other Media
   Sinks.  The Media Sink may also be connected with a Media Source
   (Section 2.1.4) and be used as part of a conceptual Media Source.

   The Media Sink can further transform the Source Stream into a
   representation that is suitable for rendering on the Media Render as
   defined by the application or system-wide configuration.  This
   include sample scaling, level adjustments etc.

2.1.32.  Received Raw Stream

   The Received Raw Stream is the received version of a Raw Stream
   (Section 2.1.3).

2.1.33.  Media Render

   A Media Render takes a Raw Stream (Section 2.1.3) and converts it
   into Physical Stimulus (Section 2.1.1) that a human user can
   perceive.  Examples of such devices are screens, and D/A converters
   connected to amplifiers and loudspeakers.

   An Endpoint can potentially have multiple Media Renders for each
   media type.

2.2.  Communication Entities

   This section contains concepts for entities involved in the
   communication.






















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      +------------------------------------------------------------+
      | Communication Session                                      |
      |                                                            |
      | +----------------+                      +----------------+ |
      | | Participant A  |    +------------+    | Participant B  | |
      | |                |    | Multimedia |    |                | |
      | | +------------+ |<==>| Session    |<==>| +------------+ | |
      | | | Endpoint A | |    |            |    | | Endpoint B | | |
      | | |            | |    +------------+    | |            | | |
      | | | +----------+-+----------------------+-+----------+ | | |
      | | | | RTP      | |                      | |          | | | |
      | | | | Session  |-+---Media Transport----+>|          | | | |
      | | | | Audio    |<+---Media Transport----+-|          | | | |
      | | | |          | |          ^           | |          | | | |
      | | | +----------+-+----------|-----------+-+----------+ | | |
      | | |            | |          v           | |            | | |
      | | |            | | +-----------------+  | |            | | |
      | | |            | | | Synchronization |  | |            | | |
      | | |            | | |     Context     |  | |            | | |
      | | |            | | +-----------------+  | |            | | |
      | | |            | |          ^           | |            | | |
      | | | +----------+-+----------|-----------+-+----------+ | | |
      | | | | RTP      | |          v           | |          | | | |
      | | | | Session  |<+---Media Transport----+-|          | | | |
      | | | | Video    |-+---Media Transport----+>|          | | | |
      | | | |          | |                      | |          | | | |
      | | | +----------+-+----------------------+-+----------+ | | |
      | | +------------+ |                      | +------------+ | |
      | +----------------+                      +----------------+ |
      +------------------------------------------------------------+

    Figure 7: Example Point to Point Communication Session with two RTP
                                 Sessions

   Figure 7 shows a high-level example representation of a very basic
   point-to-point Communication Session between Participants A and B.
   It uses two different audio and video RTP Sessions between A's and
   B's Endpoints, where each RTP Session is a group communications
   channel that can potentially carry a number of RTP Streams.  It is
   using separate Media Transports for those RTP Sessions.  The
   Multimedia Session shared by the Participants can, for example, be
   established using SIP (i.e., there is a SIP Dialog between A and B).
   The terms used in Figure 7 are further elaborated in the sub-sections
   below.







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2.2.1.  Endpoint

   An Endpoint is a single addressable entity sending or receiving RTP
   packets.  It may be decomposed into several functional blocks, but as
   long as it behaves as a single RTP stack entity it is classified as a
   single "Endpoint".

   Characteristics:

   o  Endpoints can be identified in several different ways.  While RTCP
      Canonical Names (CNAMEs) [RFC3550] provide a globally unique and
      stable identification mechanism for the duration of the
      Communication Session (see Section 2.2.5), their validity applies
      exclusively within a Synchronization Context (Section 3.1).  Thus
      one Endpoint can handle multiple CNAMEs, each of which can be
      shared among a set of Endpoints belonging to the same Participant
      (Section 2.2.3).  Therefore, mechanisms outside the scope of RTP,
      such as application defined mechanisms, must be used to provide
      Endpoint identification when outside this Synchronization Context.

   o  An Endpoint can be associated with at most one Participant
      (Section 2.2.3) at any single point in time.

   o  In some contexts, an Endpoint would typically correspond to a
      single "host", for example a computer using a single network
      interface and being used by a single human user.  In other
      contexts, a single "host" can serve multiple Participants, in
      which case each Participant's Endpoint may share properties, for
      example the IP address part of a transport address.

2.2.2.  RTP Session

   An RTP Session is an association among a group of Participants
   communicating with RTP.  It is a group communications channel which
   can potentially carry a number of RTP Streams.  Within an RTP
   Session, every Participant can find meta-data and control information
   (over RTCP) about all the RTP Streams in the RTP Session.  The
   bandwidth of the RTCP control channel is shared between all
   Participants within an RTP Session.

   Characteristics:

   o  An RTP Session can carry one ore more RTP Streams.

   o  An RTP Session shares a single SSRC space as defined in RFC3550
      [RFC3550].  That is, the Endpoints participating in an RTP Session
      can see an SSRC identifier transmitted by any of the other
      Endpoints.  An Endpoint can receive an SSRC either as SSRC or as a



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      Contributing source (CSRC) in RTP and RTCP packets, as defined by
      the Endpoints' network interconnection topology.

   o  An RTP Session uses at least two Media Transports
      (Section 2.1.15), one for sending and one for receiving.
      Commonly, the receiving Media Transport is the reverse direction
      of the Media Transport used for sending.  An RTP Session may use
      many Media Transports and these define the session's network
      interconnection topology.

   o  A single Media Transport always carries a single RTP Session.

   o  Multiple RTP Sessions can be conceptually related, for example
      originating from or targeted for the same Participant
      (Section 2.2.3) or Endpoint (Section 2.2.1), or by containing RTP
      Streams that are somehow related (Section 3).

2.2.3.  Participant

   A Participant is an entity reachable by a single signaling address,
   and is thus related more to the signaling context than to the media
   context.

   Characteristics:

   o  A single signaling-addressable entity, using an application-
      specific signaling address space, for example a SIP URI.

   o  A Participant can participate in several Multimedia Sessions
      (Section 2.2.4).

   o  A Participant can be comprised of several associated Endpoints
      (Section 2.2.1).

2.2.4.  Multimedia Session

   A Multimedia Session is an association among a group of Participants
   (Section 2.2.3) engaged in the communication via one or more RTP
   Sessions (Section 2.2.2).  It defines logical relationships among
   Media Sources (Section 2.1.4) that appear in multiple RTP Sessions.

   Characteristics:

   o  A Multimedia Session can be composed of several RTP Sessions with
      potentially multiple RTP Streams per RTP Session.

   o  Each Participant in a Multimedia Session can have a multitude of
      Media Captures and Media Rendering devices.



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   o  A single Multimedia Session can contain media from one or more
      Synchronization Contexts (Section 3.1).  An example of that is a
      Multimedia Session containing one set of audio and video for
      communication purposes belonging to one Synchronization Context,
      and another set of audio and video for presentation purposes (like
      playing a video file) with a separate Synchronization Context that
      has no strong timing relationship and need not be strictly
      synchronized with the audio and video used for communication.

2.2.5.  Communication Session

   A Communication Session is an association among two or more
   Participants (Section 2.2.3) communicating with each other via one or
   more Multimedia Sessions (Section 2.2.4).

   Characteristics:

   o  Each Participant in a Communication Session is identified via an
      application-specific signaling address.

   o  A Communication Session is composed of Participants that share at
      least one Multimedia Session, involving one or more parallel RTP
      Sessions with potentially multiple RTP Streams per RTP Session.

   For example, in a full mesh communication, the Communication Session
   consists of a set of separate Multimedia Sessions between each pair
   of Participants.  Another example is a centralized conference, where
   the Communication Session consists of a set of Multimedia Sessions
   between each Participant and the conference handler.

3.  Concepts of Inter-Relations

   This section uses the concepts from previous sections, and looks at
   different types of relationships among them.  These relationships
   occur at different abstraction levels and for different purposes, but
   the reason for the needed relationship at a certain step in the media
   handling chain may exist at another step.  For example, the use of
   Simulcast (Section 3.6)) implies a need to determine relations at RTP
   Stream level, but the underlying reason is that multiple Media
   Encoders use the same Media Source, i.e. to be able to identify a
   common Media Source.

3.1.  Synchronization Context

   A Synchronization Context defines a requirement on a strong timing
   relationship between the Media Sources, typically requiring alignment
   of clock sources.  Such a relationship can be identified in multiple
   ways as listed below.  A single Media Source can only belong to a



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   single Synchronization Context, since it is assumed that a single
   Media Source can only have a single media clock and requiring
   alignment to several Synchronization Contexts (and thus reference
   clocks) will effectively merge those into a single Synchronization
   Context.

3.1.1.  RTCP CNAME

   RFC3550 [RFC3550] describes Inter-media synchronization between RTP
   Sessions based on RTCP CNAME, RTP and Network Time Protocol (NTP)
   [RFC5905] formatted timestamps of a reference clock.  As indicated in
   [RFC7273], despite using NTP format timestamps, it is not required
   that the clock be synchronized to an NTP source.

3.1.2.  Clock Source Signaling

   [RFC7273] provides a mechanism to signal the clock source in Session
   Description Protocol (SDP) [RFC4566] both for the reference clock as
   well as the media clock, thus allowing a Synchronization Context to
   be defined beyond the one defined by the usage of CNAME source
   descriptions.

3.1.3.  Implicitly via RtcMediaStream

   WebRTC defines "RtcMediaStream" with one or more
   "RtcMediaStreamTracks".  All tracks in a "RtcMediaStream" are
   intended to be synchronized when rendered, implying that they must be
   generated such that synchronization is possible.

3.1.4.  Explicitly via SDP Mechanisms

   The SDP Grouping Framework [RFC5888] defines an m= line (Section 4.2)
   grouping mechanism called "Lip Synchronization" (with LS
   identification-tag) for establishing the synchronization requirement
   across m= lines when they map to individual sources.

   Source-Specific Media Attributes in SDP [RFC5576] extends the above
   mechanism when multiple Media Sources are described by a single m=
   line.

3.2.  Endpoint

   Some applications requires knowledge of what Media Sources originate
   from a particular Endpoint (Section 2.2.1).  This can include such
   decisions as packet routing between parts of the topology, knowing
   the Endpoint origin of the RTP Streams.





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   In RTP, this identification has been overloaded with the
   Synchronization Context (Section 3.1) through the usage of the RTCP
   source description CNAME (Section 3.1.1).  This works for some
   usages, but in others it breaks down.  For example, if an Endpoint
   has two sets of Media Sources that have different Synchronization
   Contexts, like the audio and video of the human Participant as well
   as a set of Media Sources of audio and video for a shared movie,
   CNAME would not be an appropriate identification for that Endpoint.
   Therefore, an Endpoint may have multiple CNAMEs.  The CNAMEs or the
   Media Sources themselves can be related to the Endpoint.

3.3.  Participant

   In communication scenarios, it is commonly needed to know which Media
   Sources originate from which Participant (Section 2.2.3).  One reason
   is, for example, to enable the application to display Participant
   Identity information correctly associated with the Media Sources.
   This association is handled through the signaling solution to point
   at a specific Multimedia Session where the Media Sources may be
   explicitly or implicitly tied to a particular Endpoint.

   Participant information becomes more problematic due to Media Sources
   that are generated through mixing or other conceptual processing of
   Raw Streams or Source Streams that originate from different
   Participants.  This type of Media Sources can thus have a dynamically
   varying set of origins and Participants.  RTP contains the concept of
   CSRC that carry information about the previous step origin of the
   included media content on RTP level.

3.4.  RtcMediaStream

   An RtcMediaStream in WebRTC is an explicit grouping of a set of Media
   Sources (RtcMediaStreamTracks) that share a common identifier and a
   single Synchronization Context (Section 3.1).

3.5.  Multi-Channel Audio

   There exist a number of RTP payload formats that can carry multi-
   channel audio, despite the codec being a single-channel (mono)
   encoder.  Multi-channel audio can be viewed as multiple Media Sources
   sharing a common Synchronization Context.  These are independently
   encoded by a Media Encoder and the different Encoded Streams are
   packetized together in a time synchronized way into a single Source
   RTP Stream, using the used codec's RTP Payload format.  Examples of
   codecs that support multi-channel audio are PCMA and PCMU [RFC3551],
   AMR [RFC4867], and G.719 [RFC5404].





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3.6.  Simulcast

   A Media Source represented as multiple independent Encoded Streams
   constitutes a Simulcast [I-D.ietf-mmusic-sdp-simulcast] or MDC of
   that Media Source.  Figure 8 shows an example of a Media Source that
   is encoded into three separate Simulcast streams, that are in turn
   sent on the same Media Transport flow.  When using Simulcast, the RTP
   Streams may be sharing RTP Session and Media Transport, or be
   separated on different RTP Sessions and Media Transports, or any
   combination of these two.  One major reason to use separate Media
   Transports is to make use of different Quality of Service for the
   different Source RTP Streams.  Some considerations on separating
   related RTP Streams are discussed in Section 3.12.

                            +----------------+
                            |  Media Source  |
                            +----------------+
                     Source Stream  |
             +----------------------+----------------------+
             |                      |                      |
             V                      V                      V
    +------------------+   +------------------+   +------------------+
    |  Media Encoder   |   |  Media Encoder   |   |  Media Encoder   |
    +------------------+   +------------------+   +------------------+
             | Encoded              | Encoded              | Encoded
             | Stream               | Stream               | Stream
             V                      V                      V
    +------------------+   +------------------+   +------------------+
    | Media Packetizer |   | Media Packetizer |   | Media Packetizer |
    +------------------+   +------------------+   +------------------+
             | Source               | Source               | Source
             | RTP                  | RTP                  | RTP
             | Stream               | Stream               | Stream
             +-----------------+    |    +-----------------+
                               |    |    |
                               V    V    V
                          +-------------------+
                          |  Media Transport  |
                          +-------------------+

                Figure 8: Example of Media Source Simulcast

   The Simulcast relation between the RTP Streams is the common Media
   Source.  In addition, to be able to identify the common Media Source,
   a receiver of the RTP Stream may need to know which configuration or
   encoding goals that lay behind the produced Encoded Stream and its
   properties.  This enables selection of the stream that is most useful
   in the application at that moment.



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3.7.  Layered Multi-Stream

   Layered Multi-Stream (LMS) is a mechanism by which different portions
   of a layered or scalable encoding of a Source Stream are sent using
   separate RTP Streams (sometimes in separate RTP Sessions).  LMSs are
   useful for receiver control of layered media.

   A Media Source represented as an Encoded Stream and multiple
   Dependent Streams constitutes a Media Source that has layered
   dependencies.  Figure 9 represents an example of a Media Source that
   is encoded into three dependent layers, where two layers are sent on
   the same Media Transport using different RTP Streams, i.e. SSRCs, and
   the third layer is sent on a separate Media Transport.

                            +----------------+
                            |  Media Source  |
                            +----------------+
                                    |
                                    |
                                    V
       +---------------------------------------------------------+
       |                      Media Encoder                      |
       +---------------------------------------------------------+
               |                    |                     |
        Encoded Stream       Dependent Stream     Dependent Stream
               |                    |                     |
               V                    V                     V
       +----------------+   +----------------+   +----------------+
       |Media Packetizer|   |Media Packetizer|   |Media Packetizer|
       +----------------+   +----------------+   +----------------+
               |                    |                     |
          RTP Stream           RTP Stream            RTP Stream
               |                    |                     |
               +------+      +------+                     |
                      |      |                            |
                      V      V                            V
                +-----------------+              +-----------------+
                | Media Transport |              | Media Transport |
                +-----------------+              +-----------------+

           Figure 9: Example of Media Source Layered Dependency

   It is sometimes useful to make a distinction between using a single
   Media Transport or multiple separate Media Transports when (in both
   cases) using multiple RTP Streams to carry Encoded Streams and
   Dependent Streams for a Media Source.  Therefore, the following new
   terminology is defined here:




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   SRST:  Single RTP Stream on a Single Media Transport

   MRST:  Multiple RTP Streams on a Single Media Transport

   MRMT:  Multiple RTP Streams on Multiple Media Transports

   MRST and MRMT relations needs to identify the common Media Encoder
   origin for the Encoded and Dependent Streams.  When using different
   RTP Sessions (MRMT), a single RTP Stream per Media Encoder, and a
   single Media Source in each RTP Session, common SSRC and CNAMEs can
   be used to identify the common Media Source.  When multiple RTP
   Streams are sent from one Media Encoder in the same RTP Session
   (MRST), then CNAME is the only currently specified RTP identifier
   that can be used.  In cases where multiple Media Encoders use
   multiple Media Sources sharing Synchronization Context, and thus
   having a common CNAME, additional heuristics or identification need
   to be applied to create the MRST or MRMT relationships between the
   RTP Streams.

3.8.  RTP Stream Duplication

   RTP Stream Duplication [RFC7198], using the same or different Media
   Transports, and optionally also delaying the duplicate [RFC7197],
   offers a simple way to protect media flows from packet loss in some
   cases (see Figure 10).  This is a specific type of redundancy.  All
   but one Source RTP Stream (Section 2.1.10) are effectively Redundancy
   RTP Streams (Section 2.1.12), but since both Source and Redundant RTP
   Streams are the same, it does not matter which one is which.  This
   can also be seen as a specific type of Simulcast (Section 3.6) that
   transmits the same Encoded Stream (Section 2.1.7) multiple times.





















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                            +----------------+
                            |  Media Source  |
                            +----------------+
                     Source Stream  |
                                    V
                            +----------------+
                            | Media Encoder  |
                            +----------------+
                    Encoded Stream  |
                        +-----------+-----------+
                        |                       |
                        V                       V
               +------------------+    +------------------+
               | Media Packetizer |    | Media Packetizer |
               +------------------+    +------------------+
                 Source | RTP Stream     Source | RTP Stream
                        |                       V
                        |                +-------------+
                        |                | Delay (opt) |
                        |                +-------------+
                        |                       |
                        +-----------+-----------+
                                    |
                                    V
                          +-------------------+
                          |  Media Transport  |
                          +-------------------+

               Figure 10: Example of RTP Stream Duplication

3.9.  Redundancy Format

   The RTP Payload for Redundant Audio Data [RFC2198] defines a
   transport for redundant audio data together with primary data in the
   same RTP payload.  The redundant data can be a time delayed version
   of the primary or another time delayed Encoded Stream using a
   different Media Encoder to encode the same Media Source as the
   primary, as depicted in Figure 11.













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              +--------------------+
              |    Media Source    |
              +--------------------+
                        |
                   Source Stream
                        |
                        +------------------------+
                        |                        |
                        V                        V
              +--------------------+   +--------------------+
              |   Media Encoder    |   |   Media Encoder    |
              +--------------------+   +--------------------+
                        |                        |
                        |                 +------------+
                  Encoded Stream          | Time Delay |
                        |                 +------------+
                        |                        |
                        |     +------------------+
                        V     V
              +--------------------+
              |  Media Packetizer  |
              +--------------------+
                        |
                        V
                   RTP Stream

   Figure 11: Concept for usage of Audio Redundancy with different Media
                                 Encoders

   The Redundancy format is thus providing the necessary meta
   information to correctly relate different parts of the same Encoded
   Stream.  The case depicted above (Figure 11) relates the Received
   Source Stream fragments coming out of different Media Decoders, to be
   able to combine them together into a less erroneous Source Stream.

3.10.  RTP Retransmission

   Figure 12 shows an example where a Media Source's Source RTP Stream
   is protected by a retransmission (RTX) flow [RFC4588].  In this
   example the Source RTP Stream and the Redundancy RTP Stream share the
   same Media Transport.










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          +--------------------+
          |    Media Source    |
          +--------------------+
                    |
                    V
          +--------------------+
          |   Media Encoder    |
          +--------------------+
                    |                              Retransmission
              Encoded Stream     +--------+     +---- Request
                    V            |        V     V
          +--------------------+ | +--------------------+
          |  Media Packetizer  | | | RTP Retransmission |
          +--------------------+ | +--------------------+
                    |            |           |
                    +------------+  Redundancy RTP Stream
             Source RTP Stream               |
                    |                        |
                    +---------+    +---------+
                              |    |
                              V    V
                       +-----------------+
                       | Media Transport |
                       +-----------------+

          Figure 12: Example of Media Source Retransmission Flows

   The RTP Retransmission example (Figure 12) illustrates that this
   mechanism works purely on the Source RTP Stream.  The RTP
   Retransmission transform buffers the sent Source RTP Stream and, upon
   request, emits a retransmitted packet with an extra payload header as
   a Redundancy RTP Stream.  The RTP Retransmission mechanism [RFC4588]
   is specified such that there is a one to one relation between the
   Source RTP Stream and the Redundancy RTP Stream.  Therefore, a
   Redundancy RTP Stream needs to be associated with its Source RTP
   Stream.  This is done based on CNAME selectors and heuristics to
   match requested packets for a given Source RTP Stream with the
   original sequence number in the payload of any new Redundancy RTP
   Stream using the RTX payload format.  In cases where the Redundancy
   RTP Stream is sent in a different RTP Session than the Source RTP
   Stream, the RTP Session relation is signaled by using the SDP Media
   Grouping's [RFC5888] Flow Identification (FID identification-tag)
   semantics.








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3.11.  Forward Error Correction

   Figure 13 shows an example where two Media Sources' Source RTP
   Streams are protected by Forward Error Correction (FEC).  Source RTP
   Stream A has a RTP-based Redundancy transformation in FEC Encoder 1.
   This produces a Redundancy RTP Stream 1, that is only related to
   Source RTP Stream A.  The FEC Encoder 2, however, takes two Source
   RTP Streams (A and B) and produces a Redundancy RTP Stream 2 that
   protects them jointly, i.e. Redundancy RTP Stream 2 relates to two
   Source RTP Streams (a FEC group).  FEC decoding, when needed due to
   packet loss or packet corruption at the receiver, requires knowledge
   about which Source RTP Streams that the FEC encoding was based on.

   In Figure 13 all RTP Streams are sent on the same Media Transport.
   This is however not the only possible choice.  Numerous combinations
   exist for spreading these RTP Streams over different Media Transports
   to achieve the communication application's goal.

       +--------------------+                +--------------------+
       |   Media Source A   |                |   Media Source B   |
       +--------------------+                +--------------------+
                 |                                     |
                 V                                     V
       +--------------------+                +--------------------+
       |   Media Encoder A  |                |   Media Encoder B  |
       +--------------------+                +--------------------+
                 |                                     |
           Encoded Stream                        Encoded Stream
                 V                                     V
       +--------------------+                +--------------------+
       | Media Packetizer A |                | Media Packetizer B |
       +--------------------+                +--------------------+
                 |                                     |
        Source RTP Stream A                   Source RTP Stream B
                 |                                     |
           +-----+---------+-------------+         +---+---+
           |               V             V         V       |
           |       +---------------+  +---------------+    |
           |       | FEC Encoder 1 |  | FEC Encoder 2 |    |
           |       +---------------+  +---------------+    |
           |  Redundancy   |     Redundancy   |            |
           |  RTP Stream 1 |     RTP Stream 2 |            |
           V               V                  V            V
       +----------------------------------------------------------+
       |                    Media Transport                       |
       +----------------------------------------------------------+

             Figure 13: Example of FEC Redundancy RTP Streams



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   As FEC Encoding exists in various forms, the methods for relating FEC
   Redundancy RTP Streams with its source information in Source RTP
   Streams are many.  The XOR based RTP FEC Payload format [RFC5109] is
   defined in such a way that a Redundancy RTP Stream has a one to one
   relation with a Source RTP Stream.  In fact, the RFC requires the
   Redundancy RTP Stream to use the same SSRC as the Source RTP Stream.
   This requires the use of either a separate RTP Session, or the
   Redundancy RTP Payload format [RFC2198].  The underlying relation
   requirement for this FEC format and a particular Redundancy RTP
   Stream is to know the related Source RTP Stream, including its SSRC.

3.12.  RTP Stream Separation

   RTP Streams can be separated exclusively based on their SSRCs, at the
   RTP Session level, or at the Multi-Media Session level.

   When the RTP Streams that have a relationship are all sent in the
   same RTP Session and are uniquely identified based on their SSRC
   only, it is termed an SSRC-Only Based Separation.  Such streams can
   be related via RTCP CNAME to identify that the streams belong to the
   same Endpoint.  SSRC-based approaches [RFC5576], when used, can
   explicitly relate various such RTP Streams.

   On the other hand, when RTP Streams that are related are sent in the
   context of different RTP Sessions to achieve separation, it is known
   as RTP Session-based separation.  This is commonly used when the
   different RTP Streams are intended for different Media Transports.

   Several mechanisms that use RTP Session-based separation rely on it
   to enable an implicit grouping mechanism expressing the relationship.
   The solutions have been based on using the same SSRC value in the
   different RTP Sessions to implicitly indicate their relation.  That
   way, no explicit RTP level mechanism has been needed, only signaling
   level relations have been established using semantics from Grouping
   of Media lines framework [RFC5888].  Examples of this are RTP
   Retransmission [RFC4588], SVC Multi-Session Transmission [RFC6190]
   and XOR Based FEC [RFC5109].  RTCP CNAME explicitly relates RTP
   Streams across different RTP Sessions, as explained in the previous
   section.  Such a relationship can be used to perform inter-media
   synchronization.

   RTP Streams that are related and need to be associated can be part of
   different Multimedia Sessions, rather than just different RTP
   Sessions within the same Multimedia Session context.  This puts
   further demand on the scope of the mechanism(s) and its handling of
   identifiers used for expressing the relationships.





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3.13.  Multiple RTP Sessions over one Media Transport

   [I-D.westerlund-avtcore-transport-multiplexing] describes a mechanism
   that allows several RTP Sessions to be carried over a single
   underlying Media Transport.  The main reasons for doing this are
   related to the impact of using one or more Media Transports (using a
   common network path or potentially have different ones).  The fewer
   Media Transports used, the less need for NAT/FW traversal resources
   and smaller number of flow based Quality of Service (QoS).

   However, Multiple RTP Sessions over one Media Transport imply that a
   single Media Transport 5-tuple is not sufficient to express in which
   RTP Session context a particular RTP Stream exists.  Complexities in
   the relationship between Media Transports and RTP Session already
   exist as one RTP Session contains multiple Media Transports, e.g.
   even a Peer-to-Peer RTP Session with RTP/RTCP Multiplexing requires
   two Media Transports, one in each direction.  The relationship
   between Media Transports and RTP Sessions as well as additional
   levels of identifiers need to be considered in both signaling design
   and when defining terminology.

4.  Mapping from Existing Terms

   This section describes a selected set of terms from some relevant
   IETF RFC and Internet Drafts (at the time of writing), using the
   concepts from previous sections.

4.1.  Telepresence Terms

   The terms in this sub-section are used in the context of CLUE
   [I-D.ietf-clue-framework].  Note that some terms listed in this sub-
   section use the same names as terms defined elsewhere in this
   document.  Unless explicitly stated (as "RTP Taxonomy") and in this
   sub-section, they are to be read as references to the CLUE-specific
   term within this sub-section.

4.1.1.  Audio Capture

   Defined in CLUE as a Media Capture (Section 4.1.7) for audio.
   Describes an audio Media Source (Section 2.1.4).

4.1.2.  Capture Device

   Defined in CLUE as a device that converts physical input into an
   electrical signal.  Identifies a physical entity performing an RTP
   Taxonomy Media Capture (Section 2.1.2) transformation.





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4.1.3.  Capture Encoding

   Defined in CLUE as a specific encoding (Section 4.1.6) of a Media
   Capture (Section 4.1.7).  Describes an Encoded Stream (Section 2.1.7)
   related to CLUE specific semantic information.

4.1.4.  Capture Scene

   Defined in CLUE as a structure representing a spatial region captured
   by one or more Capture Devices (Section 4.1.2), each capturing media
   representing a portion of the region.  Describes a set of spatially
   related Media Sources (Section 2.1.4).

4.1.5.  Endpoint

   Defined in CLUE as a CLUE-capable device which is the logical point
   of final termination through receiving, decoding and rendering and/or
   initiation through capturing, encoding, and sending of media streams
   (Section 4.1.10).  CLUE further defines it to consist of one or more
   physical devices with source and sink media streams, and exactly one
   [RFC4353] Participant.  Describes exactly one Participant
   (Section 2.2.3) and one or more RTP Taxonomy Endpoints
   (Section 2.2.1).

4.1.6.  Individual Encoding

   Defined in CLUE as a set of parameters representing a way to encode a
   Media Capture (Section 4.1.7) to become a Capture Encoding
   (Section 4.1.3).  Describes the configuration information needed to
   perform a Media Encoder (Section 2.1.6) transformation.

4.1.7.  Media Capture

   Defined in CLUE as a source of media, such as from one or more
   Capture Devices (Section 4.1.2) or constructed from other media
   streams (Section 4.1.10).  Describes either an RTP Taxonomy Media
   Capture (Section 2.1.2) or a Media Source (Section 2.1.4), depending
   on in which context the term is used.

4.1.8.  Media Consumer

   Defined in CLUE as a CLUE-capable device that intends to receive
   Capture Encodings (Section 4.1.3).  Describes the media receiving
   part of an RTP Taxonomy Endpoint (Section 2.2.1).







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4.1.9.  Media Provider

   Defined in CLUE as a CLUE-capable device that intends to send Capture
   Encodings (Section 4.1.3).  Describes the media sending part of an
   RTP Taxonomy Endpoint (Section 2.2.1).

4.1.10.  Stream

   Defined in CLUE as a Capture Encoding (Section 4.1.3) sent from a
   Media Provider (Section 4.1.9) to a Media Consumer (Section 4.1.8)
   via RTP.  Describes an RTP Stream (Section 2.1.10).

4.1.11.  Video Capture

   Defined in CLUE as a Media Capture (Section 4.1.7) for video.
   Describes a video Media Source (Section 2.1.4).

4.2.  Media Description

   A single Session Description Protocol (SDP) [RFC4566] media
   description (or media block; an m-line and all subsequent lines until
   the next m-line or the end of the SDP) describes part of the
   necessary configuration and identification information needed for a
   Media Encoder transformation, as well as the necessary configuration
   and identification information for the Media Decoder to be able to
   correctly interpret a received RTP Stream.

   A Media Description typically relates to a single Media Source.  This
   is for example an explicit restriction in WebRTC.  However, nothing
   prevents that the same Media Description (and same RTP Session) is
   re-used for multiple Media Sources
   [I-D.ietf-avtcore-rtp-multi-stream].  It can thus describe properties
   of one or more RTP Streams, and can also describe properties valid
   for an entire RTP Session (via [RFC5576] mechanisms, for example).

4.3.  Media Stream

   RTP [RFC3550] uses media stream, audio stream, video stream, and
   stream of (RTP) packets interchangeably, which are all RTP Streams.

4.4.  Multimedia Conference

   A Multimedia Conference is a Communication Session (Section 2.2.5)
   between two or more Participants (Section 2.2.3), along with the
   software they are using to communicate.






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4.5.  Multimedia Session

   SDP [RFC4566] defines a Multimedia Session as a set of multimedia
   senders and receivers and the data streams flowing from senders to
   receivers, which would correspond to a set of Endpoints and the RTP
   Streams that flow between them.  In this document, Multimedia Session
   (Section 2.2.4) also assumes those Endpoints belong to a set of
   Participants that are engaged in communication via a set of related
   RTP Streams.

   RTP [RFC3550] defines a Multimedia Session as a set of concurrent RTP
   Sessions among a common group of Participants.  For example, a video
   conference may contain an audio RTP Session and a video RTP Session.
   This would correspond to a group of Participants (each using one or
   more Endpoints) sharing a set of concurrent RTP Sessions.  In this
   document, Multimedia Session also defines those RTP Sessions to have
   some relation and be part of a communication among the Participants.

4.6.  Multipoint Control Unit (MCU)

   This term is commonly used to describe the central node in any type
   of star topology [I-D.ietf-avtcore-rtp-topologies-update] conference.
   It describes a device that includes one Participant (Section 2.2.3)
   (usually corresponding to a so-called conference focus) and one or
   more related Endpoints (Section 2.2.1) (sometimes one or more per
   conference Participant).

4.7.  Multi-Session Transmission (MST)

   One of two transmission modes defined in H.264 based SVC [RFC6190],
   the other mode being SST (Section 4.13).  In Multi-Session
   Transmission (MST), the SVC Media Encoder sends Encoded Streams and
   Dependent Streams distributed across two or more RTP Streams in one
   or more RTP Sessions.  The term "MST" is ambiguous in RFC 6190,
   especially since the name indicates the use of multiple "sessions",
   while MST type packetization is in fact required whenever two or more
   RTP Streams are used for the Encoded and Dependent Streams,
   regardless if those are sent in one or more RTP Sessions.
   Corresponds either to MRST or MRMT (Section 3.7) stream relations
   defined in this document.  The SVC RTP Payload RFC [RFC6190] is not
   particularly explicit about how the common Media Encoder
   (Section 2.1.6) relation between Encoded Streams (Section 2.1.7) and
   Dependent Streams (Section 2.1.8) is to be implemented.








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4.8.  Recording Device

   WebRTC specifications use this term to refer to locally available
   entities performing a Media Capture (Section 2.1.2) transformation.

4.9.  RtcMediaStream

   A WebRTC RtcMediaStream is a set of Media Sources (Section 2.1.4)
   sharing the same Synchronization Context (Section 3.1).

4.10.  RtcMediaStreamTrack

   A WebRTC RtcMediaStreamTrack is a Media Source (Section 2.1.4).

4.11.  RTP Sender

   RTP [RFC3550] uses this term, which can be seen as the RTP protocol
   part of a Media Packetizer (Section 2.1.9).

4.12.  RTP Session

   Within the context of SDP, a singe m= line can map to a single RTP
   Session (Section 2.2.2) or multiple m= lines can map to a single RTP
   Session.  The latter is enabled via multiplexing schemes such as
   BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation], for example, which
   allows mapping of multiple m= lines to a single RTP Session.

4.13.  Single Session Transmission (SST)

   One of two transmission modes defined in H.264 based SVC [RFC6190],
   the other mode being MST (Section 4.7).  In Single Session
   Transmission (SST), the SVC Media Encoder sends Encoded Streams
   (Section 2.1.7) and Dependent Streams (Section 2.1.8) combined into a
   single RTP Stream (Section 2.1.10) in a single RTP Session
   (Section 2.2.2), using the SVC RTP Payload format.  The term "SST" is
   ambiguous in RFC 6190, in that it sometimes refers to the use of a
   single RTP Stream, like in sections relating to packetization, and
   sometimes appears to refer to use of a single RTP Session, like in
   the context of discussing SDP.  Closely corresponds to SRST
   (Section 3.7) defined in this document.

4.14.  SSRC

   RTP [RFC3550] defines this as "the source of a stream of RTP
   packets", which indicates that an SSRC is not only a unique
   identifier for the Encoded Stream (Section 2.1.7) carried in those
   packets, but is also effectively used as a term to denote a Media
   Packetizer (Section 2.1.9).  In [RFC3550], it is stated that "a



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   synchronization source may change its data format, e.g., audio
   encoding, over time".  The related Encoded Stream data format in an
   RTP Stream (Section 2.1.10) is identified by the RTP Payload Type.
   Changing data format for an Encoded Stream effectively also changes
   what Media Encoder (Section 2.1.6) that is used for the Encoded
   Stream.  No ambiguity is introduced to SSRC as Encoded Stream
   identifier by allowing RTP Payload Type changes, as long as only a
   single RTP Payload Type is valid for any given RTP Time Stamp.  This
   is aligned with and further described by Section 5.2 of [RFC3550].

5.  Security Considerations

   The purpose of this document is to make clarifications and reduce the
   confusion prevalent in RTP taxonomy because of inconsistent usage by
   multiple technologies and protocols making use of the RTP protocol.
   It does not introduce any new security considerations beyond those
   already well documented in the RTP protocol [RFC3550] and each of the
   many respective specifications of the various protocols making use of
   it.

   Having a well-defined common terminology and understanding of the
   complexities of the RTP architecture will help lead us to better
   standards, avoiding security problems.

6.  Acknowledgement

   This document has many concepts borrowed from several documents such
   as WebRTC [I-D.ietf-rtcweb-overview], CLUE [I-D.ietf-clue-framework],
   and Multiplexing Architecture
   [I-D.westerlund-avtcore-transport-multiplexing].  The authors would
   like to thank all the authors of each of those documents.

   The authors would also like to acknowledge the insights, guidance and
   contributions of Magnus Westerlund, Roni Even, Paul Kyzivat, Colin
   Perkins, Keith Drage, Harald Alvestrand, Alex Eleftheriadis, Mo
   Zanaty, Stephan Wenger, and Bernard Aboba.

7.  Contributors

   Magnus Westerlund has contributed the concept model for the media
   chain using transformations and streams model, including rewriting
   pre-existing concepts into this model and adding missing concepts.
   The first proposal for updating the relationships and the topologies
   based on this concept was also performed by Magnus.







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8.  IANA Considerations

   This document makes no request of IANA.

9.  Informative References

   [I-D.ietf-avtcore-rtp-multi-stream]
              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-08 (work in progress),
              July 2015.

   [I-D.ietf-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              ietf-avtcore-rtp-topologies-update-10 (work in progress),
              July 2015.

   [I-D.ietf-clue-framework]
              Duckworth, M., Pepperell, A., and S. Wenger, "Framework
              for Telepresence Multi-Streams", draft-ietf-clue-
              framework-22 (work in progress), April 2015.

   [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-23 (work in progress), July 2015.

   [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-00 (work in progress), January 2015.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-14
              (work in progress), June 2015.

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








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   [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,
              <http://www.rfc-editor.org/info/rfc2198>.

   [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, <http://www.rfc-editor.org/info/rfc3550>.

   [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,
              <http://www.rfc-editor.org/info/rfc3551>.

   [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,
              <http://www.rfc-editor.org/info/rfc3711>.

   [RFC4353]  Rosenberg, J., "A Framework for Conferencing with the
              Session Initiation Protocol (SIP)", RFC 4353,
              DOI 10.17487/RFC4353, February 2006,
              <http://www.rfc-editor.org/info/rfc4353>.

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

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

   [RFC4867]  Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
              "RTP Payload Format and File Storage Format for the
              Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
              (AMR-WB) Audio Codecs", RFC 4867, DOI 10.17487/RFC4867,
              April 2007, <http://www.rfc-editor.org/info/rfc4867>.

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

   [RFC5404]  Westerlund, M. and I. Johansson, "RTP Payload Format for
              G.719", RFC 5404, DOI 10.17487/RFC5404, January 2009,
              <http://www.rfc-editor.org/info/rfc5404>.



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   [RFC5481]  Morton, A. and B. Claise, "Packet Delay Variation
              Applicability Statement", RFC 5481, DOI 10.17487/RFC5481,
              March 2009, <http://www.rfc-editor.org/info/rfc5481>.

   [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,
              <http://www.rfc-editor.org/info/rfc5576>.

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

   [RFC5905]  Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch,
              "Network Time Protocol Version 4: Protocol and Algorithms
              Specification", RFC 5905, DOI 10.17487/RFC5905, June 2010,
              <http://www.rfc-editor.org/info/rfc5905>.

   [RFC6190]  Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              "RTP Payload Format for Scalable Video Coding", RFC 6190,
              DOI 10.17487/RFC6190, May 2011,
              <http://www.rfc-editor.org/info/rfc6190>.

   [RFC7160]  Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
              Clock Rates in an RTP Session", RFC 7160,
              DOI 10.17487/RFC7160, April 2014,
              <http://www.rfc-editor.org/info/rfc7160>.

   [RFC7197]  Begen, A., Cai, Y., and H. Ou, "Duplication Delay
              Attribute in the Session Description Protocol", RFC 7197,
              DOI 10.17487/RFC7197, April 2014,
              <http://www.rfc-editor.org/info/rfc7197>.

   [RFC7198]  Begen, A. and C. Perkins, "Duplicating RTP Streams",
              RFC 7198, DOI 10.17487/RFC7198, April 2014,
              <http://www.rfc-editor.org/info/rfc7198>.

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

   [RFC7273]  Williams, A., Gross, K., van Brandenburg, R., and H.
              Stokking, "RTP Clock Source Signalling", RFC 7273,
              DOI 10.17487/RFC7273, June 2014,
              <http://www.rfc-editor.org/info/rfc7273>.





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Appendix A.  Changes From Earlier Versions

   NOTE TO RFC EDITOR: Please remove this section prior to publication.

A.1.  Modifications Between WG Version -07 and -08

   Addresses comments from IESG evaluation.

   o  Made text more firm around what improvements this document
      introduces.

   o  Clarified the distinction between analog and digital in sections
      2.1.1 and 2.1.2.

   o  Removed the explicit requirement that a Source RTP Stream must
      send at least some data from an Encoded Stream, replacing it with
      a statement that it is directly related to the Encoded Stream.

   o  Moved the clarification that RTP-based Redundancy excludes Media
      Encoder redundancy data in an Encoded Stream from Section 2.1.10
      (RTP Stream) to 2.1.11 (RTP-based Redundancy), since that
      statement applies to RTP-based Redundancy rather than to RTP
      Stream.

   o  Added clarification that a Media Transport Sender can
      intentionally pace packet transmission.

   o  Aligned text around delay variation to use this term throughout,
      and added a reference to RFC 5481.

   o  Added that RTP Session is a group communications channel that can
      potentially carry a number of RTP Streams, as an additional
      clarification below Figure 7.

   o  Added a clarification in Section 4.1 around Telepresence Terms on
      which references are to CLUE terms and which are to other sections
      of this document, for terms that have the same name in CLUE as in
      this document.

   o  Clarified in Section 4.14 what SSRC data format changes means,
      since the RFC 3550 SSRC definition mentions this possibility.

   o  Editorial improvements.








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A.2.  Modifications Between WG Version -06 and -07

   Addresses comments from AD review and GenArt review.

   o  Added RTP-based Security and RTP-based Validation transform
      sections, as well as Secured RTP Stream and Received Secured RTP
      Stream sections.

   o  Improved wording in Abstract and Introduction sections.

   o  Clarified what is considered "media" in section 2.1.2 Media
      Capture.

   o  Changed a number of "Characteristics" lists to more suitable prose
      text.

   o  Re-worded text around use of Encoded and Dependent RTP Streams in
      section 2.1.9 Media Packetizer.

   o  Clarified description of Source RTP Stream in section 2.1.10.

   o  Clarified motivation to use separate Media Transports for
      Simulcast in section 3.6.

   o  Added local descriptions of terms imported from CLUE framework.

   o  Editorial improvements.

A.3.  Modifications Between WG Version -05 and -06

   o  Clarified that a Redundancy RTP Stream can be used standalone to
      generate Repaired RTP Streams.

   o  Clarified that (in accordance with above) RTP-based Repair takes
      zero or more Received RTP Streams and one or more Received
      Redundancy RTP Streams as input.

   o  Changed Figure 6 to more clearly show that Media Transport is
      terminated in the Endpoint, not in the Participant.

   o  Added a sentence to Endpoint section that clarifies there may be
      contexts where a single "host" can serve multiple Participants,
      making those Endpoints share some properties.

   o  Merged previous section 3.5 on SST/MST with previous section 3.8
      on Layered Multi-Stream into a common section discussing the
      scalable/layered stream relation, and moved improved, descriptive




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      text on SST and MST to new sub-sections 4.7 and 4.13, describing
      them as existing terms.

   o  Editorial improvements.

A.4.  Modifications Between WG Version -04 and -05

   o  Editorial improvements.

A.5.  Modifications Between WG Version -03 and -04

   o  Changed "Media Redundancy" and "Media Repair" to "RTP-based
      Redundancy" and "RTP-based Repair", since those terms are more
      specific and correct.

   o  Changed "End Point" to "Endpoint" and removed Editor's Note on
      this.

   o  Clarified that a Media Capture may impose constraints on clock
      handling.

   o  Clarified that mixing multiple Raw Streams into a Source Stream is
      not possible, since that requires mixed streams to have a timing
      relation, requiring them to be Source Streams, and added an
      example.

   o  Clarified that RTP-based Redundancy excludes the type of encoding
      redundancy found within the encoded media format in an Encoded
      Stream.

   o  Clarified that a Media Transport contains only a single RTP
      Session, but a single RTP Session can span multiple Media
      Transports.

   o  Clarified that packets with seemingly correct checksum that are
      received by a Media Transport Receiver may still be corrupt.

   o  Clarified that a corrupt packet in a Media Transport Receiver is
      typically either discarded or somehow marked and passed on in the
      Received RTP Stream.

   o  Added Synchronization Context to Figure 6.

   o  Editorial improvements and clarifications.







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A.6.  Modifications Between WG Version -02 and -03

   o  Changed section 3.5, removing SST-SS/MS and MST-SS/MS, replacing
      them with SRST, MRST, and MRMT.

   o  Updated section 3.8 to align with terminology changes in section
      3.5.

   o  Added a new section 4.12, describing the term Multimedia
      Conference.

   o  Changed reference from I-D to now published RFC 7273.

   o  Editorial improvements and clarifications.

A.7.  Modifications Between WG Version -01 and -02

   o  Major re-structure

   o  Moved media chain Media Transport detailing up one section level

   o  Collapsed level 2 sub-sections of section 3 and thus moved level 3
      sub-sections up one level, gathering some introductory text into
      the beginning of section 3

   o  Added that not only SSRC collision, but also a clock rate change
      [RFC7160] is a valid reason to change SSRC value for an RTP stream

   o  Added a sub-section on clock source signaling

   o  Added a sub-section on RTP stream duplication

   o  Elaborated a bit in section 2.2.1 on the relation between End
      Points, Participants and CNAMEs

   o  Elaborated a bit in section 2.2.4 on Multimedia Session and
      synchronization contexts

   o  Removed the section on CLUE scenes defining an implicit
      synchronization context, since it was incorrect

   o  Clarified text on SVC SST and MST according to list discussions

   o  Removed the entire topology section to avoid possible
      inconsistencies or duplications with draft-ietf-avtcore-rtp-
      topologies-update, but saved one example overview figure of
      Communication Entities into that section




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   o  Added a section 4 on mapping from existing terms with one sub-
      section per term, mainly by moving text from sections 2 and 3

   o  Changed all occurrences of Packet Stream to RTP Stream

   o  Moved all normative references to informative, since this is an
      informative document

   o  Added references to RFC 7160, RFC 7197 and RFC 7198, and removed
      unused references

A.8.  Modifications Between WG Version -00 and -01

   o  WG version -00 text is identical to individual draft -03

   o  Amended description of SVC SST and MST encodings with respect to
      concepts defined in this text

   o  Removed UML as normative reference, since the text no longer uses
      any UML notation

   o  Removed a number of level 4 sections and moved out text to the
      level above

A.9.  Modifications Between Version -02 and -03

   o  Section 4 rewritten (and new communication topologies added) to
      reflect the major updates to Sections 1-3

   o  Section 8 removed (carryover from initial -00 draft)

   o  General clean up of text, grammar and nits

A.10.  Modifications Between Version -01 and -02

   o  Section 2 rewritten to add both streams and transformations in the
      media chain.

   o  Section 3 rewritten to focus on exposing relationships.

A.11.  Modifications Between Version -00 and -01

   o  Too many to list

   o  Added new authors

   o  Updated content organization and presentation




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Authors' Addresses

   Jonathan Lennox
   Vidyo, Inc.
   433 Hackensack Avenue
   Seventh Floor
   Hackensack, NJ  07601
   US

   Email: jonathan@vidyo.com


   Kevin Gross
   AVA Networks, LLC
   Boulder, CO
   US

   Email: kevin.gross@avanw.com


   Suhas Nandakumar
   Cisco Systems
   170 West Tasman Drive
   San Jose, CA  95134
   US

   Email: snandaku@cisco.com


   Gonzalo Salgueiro
   Cisco Systems
   7200-12 Kit Creek Road
   Research Triangle Park, NC  27709
   US

   Email: gsalguei@cisco.com


   Bo Burman (editor)
   Ericsson
   Kistavagen 25
   SE-16480 Stockholm
   Sweden

   Email: bo.burman@ericsson.com






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