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RFC 7656

A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources

Pages: 46
Part 2 of 2 – Pages 25 to 46
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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 the
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   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 for 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
   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

3.1.1.  RTCP CNAME

   [RFC3550] describes inter-media synchronization between RTP sessions
   based on RTCP CNAME, RTP, and timestamps of a reference clock
   formatted using the Network Time Protocol (NTP) [RFC5905].  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 the
   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.
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   Source-Specific Media Attributes in SDP [RFC5576] extends the above
   mechanism when multiple media sources are described by a single "m="

3.2.  Endpoint

   Some applications require 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.

   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, information about which media sources
   originate from which participant (Section 2.2.3) is commonly needed.
   One reason is, for example, to enable the application to correctly
   display participant identity information associated with the media
   sources.  This association is handled through signaling 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 when there are media
   sources that are generated through mixing or other conceptual
   processing of raw streams or source streams that originate from
   different participants.  These types of media sources can thus have a
   dynamically varying set of origins and participants.  RTP contains
   the concept of CSRC that carries information about the previous step
   origin of the included media content on the 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).
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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],
   Adaptive Multi Rate (AMR) [RFC4867], and G.719 [RFC5404].

3.6.  Simulcast

   A media source represented as multiple independent encoded streams
   constitutes a simulcast [SDP-SIMULCAST] or Modification Detection
   Code (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 an RTP session and media
   transport, or be separated on different RTP sessions and media
   transports, or be any combination of these two.  One major reason to
   use separate media transports is to make use of different quality of
   service (QoS) for the different source RTP streams.  Some
   considerations on separating related RTP streams are discussed in
   Section 3.12.
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                            |  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 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  |
       |                      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 need 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 SSRCs 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 have
   a common CNAME, additional heuristics or identification need to be
   applied to create the MRST or MRMT relationships between the RTP
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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.

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

               Figure 10: Example of RTP Stream Duplication
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3.9.  Redundancy Format

   "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.

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

   Figure 11: Concept for Usage of Audio Redundancy with Different Media

   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
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   example, the source RTP stream and the redundancy RTP stream share
   the same media transport.

          |    Media Source    |
          |   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 transforms buffers from 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 FEC.  Source RTP stream A has an 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 multimedia 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
   as a 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 the media-line
   grouping 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

   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

   [TRANSPORT-MULTIPLEX] 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
   having different ones).  The fewer media transports used, the less
   need for NAT/firewall traversal resources and smaller number of flow-
   based 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 sessions 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 needs 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
   RFCs and Internet-Drafts (at the time of writing), using the concepts
   from previous sections.

4.1.  Telepresence Terms

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

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 that 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
   participant [RFC4353].  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
   reused for multiple media sources [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 a
   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 [RTP-TOPOLOGIES] 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 a Single-Session Transmission (SST)
   (Section 4.14).  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
Top   ToC   Page 41
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 Receiver

   RTP [RFC3550] uses this term, which can be seen as the RTP protocol
   part of a media depacketizer (Section 2.1.27).

4.12.  RTP Sender

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

4.13.  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 [SDP-BUNDLE], for example, which allows mapping of multiple
   "m=" lines to a single RTP session.

4.14.  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 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.
Top   ToC   Page 42
4.15.  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
   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 the data format for an encoded stream effectively also
   changes what media encoder (Section 2.1.6) is used for the encoded
   stream.  No ambiguity is introduced to SSRC as an encoded stream
   identifier by allowing RTP payload type changes, as long as only a
   single RTP payload type is valid for any given RTP Timestamp.  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

   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.  Informative References

              Duckworth, M., Pepperell, A., and S. Wenger, "Framework
              for Telepresence Multi-Streams", Work in Progress,
              draft-ietf-clue-framework-22, April 2015.

   [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,

   [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, <>.
Top   ToC   Page 43
   [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,

   [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,

   [RFC4353]  Rosenberg, J., "A Framework for Conferencing with the
              Session Initiation Protocol (SIP)", RFC 4353,
              DOI 10.17487/RFC4353, February 2006,

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <>.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,

   [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, <>.

   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December
              2007, <>.

   [RFC5404]  Westerlund, M. and I. Johansson, "RTP Payload Format for
              G.719", RFC 5404, DOI 10.17487/RFC5404, January 2009,

   [RFC5481]  Morton, A. and B. Claise, "Packet Delay Variation
              Applicability Statement", RFC 5481, DOI 10.17487/RFC5481,
              March 2009, <>.

   [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,
Top   ToC   Page 44
   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888,
              DOI 10.17487/RFC5888, June 2010,

   [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,

   [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,

   [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,

   [RFC7197]  Begen, A., Cai, Y., and H. Ou, "Duplication Delay
              Attribute in the Session Description Protocol", RFC 7197,
              DOI 10.17487/RFC7197, April 2014,

   [RFC7198]  Begen, A. and C. Perkins, "Duplicating RTP Streams",
              RFC 7198, DOI 10.17487/RFC7198, April 2014,

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,

   [RFC7273]  Williams, A., Gross, K., van Brandenburg, R., and H.
              Stokking, "RTP Clock Source Signalling", RFC 7273,
              DOI 10.17487/RFC7273, June 2014,

              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              Work in Progress, draft-ietf-avtcore-rtp-multi-stream-08,
              July 2015.

              Westerlund, M. and S. Wenger, "RTP Topologies", Work in
              Progress, draft-ietf-avtcore-rtp-topologies-update-10,
              July 2015.
Top   ToC   Page 45
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", Work in Progress,
              draft-ietf-mmusic-sdp-bundle-negotiation-23, July 2015.

              Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
              "Using Simulcast in SDP and RTP Sessions", Work in
              Progress, draft-ietf-mmusic-sdp-simulcast-01, July 2015.

              Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
              Sessions onto a Single Lower-Layer Transport", Work in
              Progress, draft-westerlund-avtcore-transport-multiplexing-
              07, October 2013.

              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", Work in Progress,
              draft-ietf-rtcweb-overview-14, June 2015.


   This document has many concepts borrowed from several documents such
   as WebRTC [WEBRTC-OVERVIEW], CLUE [CLUE-FRAME], and Multiplexing
   Architecture [TRANSPORT-MULTIPLEX].  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.


   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.
Top   ToC   Page 46
Authors' Addresses

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


   Kevin Gross
   AVA Networks, LLC
   Boulder, CO
   United States


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


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


   Bo Burman (editor)
   Kistavagen 25
   SE-16480 Stockholm