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Versions: 00 01 02 03 04                                                
Internet Engineering Task Force                              Basso-AT&T
Internet Draft                                            Civanlar-AT&T
                                                        Gentric-Philips
                                                         Herpel-Thomson
                                                      Lifshitz-Optibase
                                                            Lim-mp4cast
                                                            Perkins-ISI
                                                   Van Der Meer-Philips
                                                          February 2002
                                                    Expires August 2002
Document: draft-ietf-avt-mpeg4-multisl-04.txt


                 RTP Payload Format for MPEG-4 Streams


Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that
   other groups may also distribute working documents as Internet-
   Drafts. 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 specification is a product of the Audio/Video Transport working
   group within the Internet Engineering Task Force and ISO/IEC MPEG-4
   ad hoc group on MPEG-4 over Internet. Comments are solicited and
   should be addressed to the working group's mailing list at
   avt@ietf.org and/or the authors.

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt
   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.

   <<
   Note for the RFC editor:
   XXXX should be replaced with this RFC number and YYYY replaced by
   the number given to the companion RFC which draft is: draft-ietf-
   avt-mpeg4-simple-**.txt.
   This document also contains a MIME type registration form that is
   intended to be taken as-is and therefore makes reference to this
   document, using the temporary placeholder: XXXX.
   >>





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                RTP Payload Format for MPEG-4 Streams   February 2002


Abstract

   This document describes a payload format for transporting MPEG-4
   encoded data using RTP. MPEG-4 is a recent standard from ISO/IEC for
   the coding of natural and synthetic audio-visual data. Several
   services provided by RTP are beneficial for MPEG-4 encoded data
   transport over the Internet. Additionally, the use of RTP makes it
   possible to synchronize MPEG-4 data with other real-time data types.

Table of Contents

   1. Introduction....................................................3
   1.1 Overview of MPEG-4 End-System Architecture.....................3
   1.2 The simplified MPEG-4 terminal model...........................4
   1.3 The complete MPEG-4 terminal model.............................4
   1.3.1 The Sync Layer and DMIF......................................6
   2. Analysis of the carriage of MPEG-4 over IP......................8
   2.1 The Sync Layer point of view...................................8
   2.2 The Elementary Stream point of view............................9
   2.3 How the two views reconcile...................................10
   2.4 Rationale for features........................................11
   2.5 Relation with RFC 3016........................................11
   3. Payload format.................................................13
   3.1 RTP Header Fields Usage.......................................14
   3.2 RTP payload structure.........................................16
   3.3 Payload Header Section structure..............................17
   3.3.1 Payload Header structure....................................18
   3.3.2 Fields of a Payload Header..................................19
   3.4 RSLHSection structure.........................................21
   3.4.1 RSLH structure..............................................22
   3.4.2 Removal of fields...........................................22
   3.4.3 Mapping of OCR..............................................23
   3.4.4 Degradation Priority........................................23
   3.5 Payload Section structure.....................................23
   3.6 Interleaving..................................................24
   3.6.1 Time stamp based interleaving (TSBI)........................25
   3.6.2 Index based interleaving (IBI)..............................26
   3.6.3 SL streams that should not be interleaved...................26
   3.7 Fragmentation Rules...........................................26
   4. Types and names................................................28
   4.1 MIME type registration........................................28
   4.2 Concatenation of parameters...................................33
   4.3 Usage of SDP..................................................33
   4.3.1 The a=fmtp keyword..........................................33
   4.3.2 SDP example.................................................33
   5. IANA considerations............................................34
   6. Other issues...................................................34
   6.1 SL-packetized stream reconstruction...........................34
   6.2 Handling of scene description streams.........................38
   6.3 Overlap with RFC 3016.........................................39
   6.4 Multiplexing..................................................40
   7. Security considerations........................................41
   8. Acknowledgements...............................................42

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   9. References.....................................................42
   10. Authors's addresses...........................................43
   APPENDIX: Examples of usage.......................................44
   Appendix.1 RFC 3016 compatible MPEG-4 Video (no SL)...............44
   Appendix.2 MPEG-4 Video with SL...................................46
   Appendix.3 Low delay MPEG-4 Audio (no SL).........................48
   Appendix.4 Media delivery MPEG-4 Audio (no SL)....................50
   Appendix.5 AAC with interleaving (no SL)..........................51
   Appendix.6 AAC with Index-based interleaving and SL...............53


1. Introduction

   MPEG-4 is a recent standard from ISO/IEC for the coding of natural
   and synthetic audio-visual data in the form of audiovisual objects
   that are arranged into an audiovisual scene by means of a scene
   description [1][2][3][4]. This draft specifies an RTP [5] payload
   format for transporting MPEG-4 encoded data streams. It supplements
   RFC 3016 in the respect that it can transport all MPEG-4 stream
   types while being compatible with RFC 3016 for the transport of
   MPEG-4 video.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in
   this document are to be interpreted as described in RFC 2119 [6].

   The benefits of using RTP for MPEG-4 data stream transport include:

   i. Ability to synchronize MPEG-4 streams with other RTP payloads,
   one example is the transport and synchronization of MPEG-4 video
   associated with AMR audio in mobile networks.

   ii. Monitoring MPEG-4 delivery performance through RTCP.

   iii. Combining MPEG-4 and other real-time data streams received from
   multiple end-systems into a set of consolidated streams through RTP
   mixers.

   iv. Converting data types, etc. through the use of RTP translators.

1.1 Overview of MPEG-4 End-System Architecture

   Two types of terminals can use this specification. One case is a
   complete MPEG-4 terminal i.e. a terminal implementing the MPEG-4
   system [1] specification and possibly also MPEG-4 video [2] and
   audio [3]. Another possibility is a terminal implementing only a
   part of this set of MPEG-4 specification; one example is a terminal
   using MPEG-4 video [2] but not MPEG-4 systems as in RFC3016.

   This document is structured so as to be understandable from both
   points of view (with or without MPEG-4 systems). The target is also
   that services deployed for one type of terminal can be adapted for
   the other type with only a minor change in the session description

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   because the media formats are the same. Another key assumption is
   that the properties of streams of various types (video, audio, scene
   description) can be described with the same Elementary Stream model
   so that this same payload format can transport any MPEG-4 stream.

1.2 The simplified MPEG-4 terminal model

   In the simplified MPEG-4 model MPEG-4 systems [1] is not used.
   However the concept of Elementary Stream remains, by MPEG
   definition:  "A consecutive flow of mono-media data from a single
   source entity to a single destination entity on the compression
   layer". Indeed both MPEG-4 video [2] and MPEG-4 audio [3] documents
   describe how respectively audio and video bit streams are fragmented
   into pieces that are called Access Units, again by MPEG definition:
   "An individually accessible portion of data within an Elementary
   Stream. An access unit is the smallest data entity to which timing
   information can be attributed". Each Access Unit has by this
   definition a number of media independent basic properties:
   . Composition time stamp (CTS)
   . Framing
   . Possibly decoding time stamp (DTS)

   Furthermore both the video [2] and audio [3] specification also
   define how Access Units (AU) shall be themselves fragmented since in
   the spirit of Application Level Framing AUs should be fragmented in
   such a way that decoders can process the packets arriving
   immediately after a packet loss. In this case the signaling of
   Access Unit fragment boundaries is also required.

   In order to be understandable from this point of view this payload
   format is described in terms of Access Units (AU) and Access Units
   fragments. This specification does not make reference to media
   specific properties (but for a few exceptions). Indeed it is the
   purpose of this specification to provide RTP transport for all media
   types in MPEG-4 in a generic fashion.

   In this mode of operation the RTP framework is used for transport of
   timing and synchronization and protocols such as H.323, SIP, RTSP,
   etc, can be used for control.

1.3 The complete MPEG-4 terminal model

   Fig. 1 below shows the layered architecture of a terminal, which
   implements the complete MPEG-4 systems model. The Compression Layer
   processes individual audio-visual media streams. The MPEG-4
   compression schemes are defined in the ISO/IEC specifications 14496-
   2 [2] and 14496-3 [3]. The compression schemes in MPEG-4 achieve
   efficient encoding over a bandwidth ranging from a few kbps to many
   Mbps. The audio-visual content compressed by this layer is organized
   into Elementary Streams (ESs).

   The MPEG-4 standard specifies MPEG-4 compliant streams. Within the
   constraint of this compliance the compression layer is unaware of a

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   specific delivery technology, but it can be made to react to the
   characteristics of a particular delivery layer such as the path-MTU
   or loss characteristics. Also, some compressors can be designed to
   be delivery specific for implementation efficiency. In such cases
   the compressor may work in a non-optimal fashion with delivery
   technologies that are different than the one it is specifically
   designed to operate with.

   The hierarchical relations, location and properties of ESs in a
   presentation are described by a dynamic set of Object Descriptors
   (ODs). Each OD groups one or more ES Descriptors referring to a
   single content item (audio-visual object). Hence, multiple
   alternative or hierarchical representations of each content item are
   possible.

   ODs are themselves conveyed through one or more ESs. A complete set
   of ODs can be seen as an MPEG-4 resource or session description at a
   stream level. The resource description may itself be hierarchical,
   i.e. an ES conveying an OD may describe other ESs conveying other
   ODs.

   The session description is accompanied by a dynamic scene
   description, Binary Format for Scene (BIFS), again conveyed through
   one or more ESs. At this level, content is identified in terms of
   audio-visual objects. The spatio-temporal location of each object is
   defined by BIFS. The audio-visual content of those objects that are
   synthetic and static are described by BIFS also. Natural and
   animated synthetic objects may refer to an OD that points to one or
   more ESs that carries the coded representation of the object or its
   animation data.
























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   media aware        +-----------------------------------------+
   delivery unaware   |           COMPRESSION LAYER             |
   14496-2 Visual     |streams from as low as Kbps to multi-Mbps|
   14496-3 Audio      +-----------------------------------------+

                                                      Elementary
                                                      Stream
   ===================================================Interface

   (ESI)
                     +-------------------------------------------+
   media and         |              SYNC LAYER                   |
   delivery unaware  | manages elementary streams, their synch-  |
   14496-1 Systems   | ronization and hierarchical relations     |
                     +-------------------------------------------+

                                                       DMIF
                                                       Application
   ====================================================Interface

   (DAI)
                     +-------------------------------------------+
   delivery aware    |               DELIVERY LAYER              |
   media  unaware    |provides transparent access to and delivery|
   14496-6 DMIF      | of content irrespective of delivery       |
                     |                technology                 |
                     +-------------------------------------------+

   Figure 1: Conceptual MPEG-4 terminal architecture

   By conveying the session (or resource) description as well as the
   scene (or content composition) description through their own ESs, it
   is made possible to change portions of the content composition and
   the number and properties of media streams that carry the audio-
   visual content separately and dynamically at well known instants in
   time.

   One or more initial Scene Description streams and the corresponding
   OD stream are pointed to by an initial object descriptor (IOD). In
   this context the IOD needs to be made available to the receivers
   through some out-of-band means that are out of scope of this payload
   specification. However in the context of transport on IP networks it
   is defined in a separate document [9].

   The Compression Layer organizes the ESs in Access Units (AU), the
   smallest elements that can be attributed individual timestamps. The
   Access Units concept defines the boundary between media specific
   processing and delivery specific processing. That is to say
   transport should not depend on the nature of the media data but only
   on AU properties.


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1.3.1 The Sync Layer and DMIF

   The Sync Layer (SL) that primarily provides the synchronization
   between streams defines a homogeneous encapsulation of ESs carrying
   media or control data (ODs, BIFS). Integer or fractional AUs are
   then encapsulated in SL packets.

   All consecutive data from one stream is called an SL-packetized
   stream. The interface between the compression layer and the SL is
   called the Elementary Stream Interface (ESI). The ESI is informative
   i.e. it is extremely useful in order to define concepts and
   mechanisms but does not have to be implemented.

   The Delivery Layer in MPEG-4 consists of the Delivery Multimedia
   Integration Framework defined in ISO/IEC 14496-6 [4]. This layer is
   media unaware but delivery technology aware. It provides transparent
   access to and delivery of content irrespective of the technologies
   used.  The interface between the SL and DMIF is called the DMIF
   Application Interface (DAI). It offers content location independent
   procedures for establishing MPEG-4 sessions and access to transport
   channels. This payload format can be used as an instance of the
   MPEG-4 Delivery Layer but is otherwise not tied to DMIF.

   The ESs from the encoders are fed into the SL with indications of AU
   boundaries, random access points, desired composition time and the
   current time. The Sync Layer fragments the ESs into SL packets, each
   containing a header that encodes information conveyed through the
   ESI. If the AU is larger than a SL packet, subsequent packets
   containing remaining parts of the AU are generated with subset
   headers until the complete AU is packetized. One SL packet describes
   an Access Units or fragments thereof, the SL packet header contains
   extended timing and framing information; the SL packet payload
   contains the bit stream frame (AU) or fragment. For the complete
   list of features of the Sync Layer refer to the MPEG-4 systems
   specification [1]. The syntax of the Sync Layer is configurable and
   can be adapted to the needs of the stream to be transported. This
   includes the possibility to select the presence or absence of
   individual syntax elements as well as configuration of their length
   in bits. The configuration for each individual stream is conveyed in
   a SLConfigDescriptor, which is an integral part of the ES Descriptor
   for this stream. The MPEG-4 SLConfigDescriptor, being configuration
   information, is not carried by the media stream itself but is rather
   transported via an ObjectDescriptor Stream encoded using the MPEG-4
   Object Description framework. This can be done in a separate stream
   using this payload format (see section 6.2 for details). The
   SLConfigDescriptor MAY also be transported by other means (for
   example as a MIME parameter, see section 4.1).

   An important point is to note that this draft could just as well
   have been entirely written in terms of SL packets instead of Access
   Units and Access Unit fragments. However this could have created
   confusion for implementers who only need basic properties and do not
   want to cope with the additional complexity of the Sync Layer.

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   Instead this specification refers to the Sync Layer only when
   needed.


2. Analysis of the carriage of MPEG-4 over IP

   As explained above when transporting MPEG-4 audio and video,
   applications may or may not require the use of MPEG-4 systems. To
   achieve the highest level of interoperability between all MPEG-4
   applications, it is desirable that (a) in both cases the same MPEG-4
   transport format can be used and that (b) receivers that have no
   MPEG-4 system knowledge can easily skip the MPEG-4 system specific
   information, if any.

   An example of application not requiring MPEG-4 system is audio/video
   streaming from a single source. Examples of applications that would
   benefit from MPEG-4 system features are:
   . Audio/video streaming mixing RTP and non-RTP sources (e.g. local
   storage in the .mp4 interchange format)
   . Rich multimedia applications including 2D, 2.5D or 3D interactive
   scenes with multiple graphical/audio/video objects and/or a
   composition variable in time and/or according to a server-push
   and/or server-pull model.
   . Applications involving Digital Right Management for some or all
   parts/streams in the content
   . Applications involving the use of advanced meta-data and the
   associated content management features as provided by the MPEG suite
   of relevant standards (MPEG-7 and MPEG-11).

2.1 The Sync Layer point of view

   RTP is perfectly suitable to transport MPEG-4 audio and MPEG-4
   video, but when using MPEG-4 systems a problem arises from the fact
   that both RTP and MPEG-4 systems contain a synchronization layer.
   In particular, the RTP header duplicates some of the information
   provided in SL packet headers such as the composition timestamps
   (CTS) and Access Unit boundaries.

   To avoid unnecessary overhead and potential interoperability risks
   when transporting MPEG-4 systems, it is desirable to remove the
   redundancy between the SL packet header and the RTP packet header.
   To be independent on the use of MPEG-4 systems, synchronization can
   rely on the parameters provided in the RTP header. Another desired
   property is to have compatibility with RFC3016 for MPEG-4 video
   transport.

   This is achieved in the following fashion (also depicted in figure
   5): In case SL headers are used, the redundant fields are removed
   from the SL header. The remaining information from the SL header, if
   any, is contained inside the RTP packet payload, together with the
   SL packet payload. Some of this information is also useful for
   transport over RTP when an MPEG-4 system is not used. For that
   reason this information is split into "general useful information"

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   and "MPEG-4 systems only information". The "general useful
   information" hereinafter called Payload Header is carried by a
   number of fields configurable using parameters defined in section
   4.1; all receivers MUST parse these fields. The "MPEG-4 systems only
   information", if any, is contained in an auxiliary header,
   hereinafter called Remaining SL Packet Header (RSLH), also
   configured using parameters (see section 4.1) and preceded by a
   length field, so that non-MPEG-4-system devices MAY skip this
   information.

                                           +------------+
                extended framing and       | AU or AU   |
                  timing information       | fragment   |
                                           +------------+
                                   |              |
                                   |              |
                                   |              |
                                   |              |
                                   V              V

                            <----------SL Packet-------->

                            +---------------------------+
                            |   SL Packet   | SL Packet |
                            |    Header     | Payload   |
                            +---------------------------+
                                  |                |
                                  |                |
         +-------------+----------+---+            |
         |             |              |            |
         V             V              V            V
   +-----------+ +-----------+ +-------------+ +-----------+
   |RTP Packet | |  Payload  | | Remaining SL| | SL Packet |
   |  Header   | |  Header   | |    Header   | | Payload   |
   +-----------+ +-----------+ +-------------+ +-----------+

                 <----RTP Packet Payload------------------->

   Figure 5: Mapping of ES into SL, then SL Packet into RTP packet

2.2 The Elementary Stream point of view

   Another way to see the mapping of Elementary Streams (i.e. Access
   Units or AU fragments) into RTP packets is depicted in Figure 6. In
   this view the "basic" timing and fragmentation information listed in
   section 1.2 is obtained directly at the codec interfaces and mapped
   into the RTP header or the RTP Payload Header.

   For example this RTP payload format has been designed so that it is
   by default configured to be identical to RFC 3016 for the
   recommended MPEG-4 video configurations, specifically in this case
   the Payload Header is empty. Hence receivers that comply with this
   payload specification can decode such RTP payload without knowledge

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   about the Sync Layer (see the relevant examples in Appendix). In a
   similar fashion but with non-empty Payload Headers, MPEG-4 audio
   (see Appendix 3 and 4 for examples) can be transported without
   explicit use of the Sync Layer.


                               +------------+
        basic framing and      | AU or AU   |
        timing information     | fragment   |
                               +------------+
                |                    |
                |                    |
         +-------------+             |
         |             |             |
         V             V             V
   +-----------+ +-----------+ +-----------+
   |RTP Packet | |  Payload  | |           |
   |  Header   | |  Header   | | Payload   |
   +-----------+ +-----------+ +-----------+

                 <----RTP Packet Payload--->


   Figure 6: Direct mapping of Elementary Streams into RTP packet

2.3 How the two views reconcile

   A simple concept enables to unify these apparently antagonistic
   points of view: a terminal that does not implement the Sync Layer
   can skip (ignore) the Remaining SL Header, if present.

   There are also cases when an Elementary Stream is such that SL
   packets are reduced to the media (compressed) data (empty headers)
   and in that case implementations do not actually need to be aware of
   the Sync Layer at all. In these cases it is logically equivalent to
   say that the Sync Layer is not implemented or to say that the SL
   packet headers are completely empty (or fully map into the RTP
   headers). The Sync Layer can then be seen as a purely conceptual
   construction that does not have to be implemented at all. Examples
   are video transported as in RFC3016 (see below) and some audio modes
   (see Annex).

   The above described MPEG-4 system model also deals with session
   setup through Object Descriptors. In cases where the complete MPEG-4
   system framework is not used a replacement for this key functionally
   is required. In fact for simple (audio/video) systems only the
   knowledge of the decoder configuration is needed; we will see how
   this specification defines options so that decoder configuration can
   also be signaled without MPEG-4 system.

   In conclusion this payload format is intended to be capable of
   transporting data formatted according to the Sync Layer


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   specification but is also useful without the Sync Layer, or when the
   Sync Layer is invisible, which is equivalent to not using it.


2.4 Rationale for features

   This payload format has a number of uncommon features that are best
   understood by first considering their rationale:

   . Genericity: The payload structure does not depend on the nature of
   the stream (audio, video, scene, etc). In this respect the apparent
   complexity of this specification should be compared to the
   complexity of the only alternative solution, which would have been
   the specification and implementation of many different RTP payload
   formats.
   . Variable geometry: this payload format is highly configurable i.e.
   the structure of the RTP payload depends on MIME parameters;
   actually all the Payload Header components are optional and most of
   them have a configurable size. This is aligned with the Sync Layer
   definition and allows optimal efficiency in terms of payload size
   per packet.
   . Two packing style (single and multiple): the rationale for
   transporting a single AU or AU fragment per RTP packet is
   simplicity, it is also the packing style for backward compatibility
   with RFC3016. The rationale for transporting multiple AU per RTP
   packet is efficiency, at the cost of sensitivity to losses.
   . Two interleaving methods: the rationale for interleaving is to
   enable various error concealment strategies in case of packet losses
   when packing several AU or AU fragments per RTP packets. The need
   for two interleaving methods arises from the fact that the default
   one, based on time stamps, is the most efficient but does not work
   for all configurations. Another method, based on indexes, is
   therefore required.
   . The rationale for transporting multiple interleaved AU fragments
   per RTP packet is to benefit from advanced error resiliency
   properties of bit streams (such as MPEG-4 audio version 2).

2.5 Relation with RFC 3016

   The following set of figures displays the relationship between the
   MPEG-4 RTP payload formats; there are 4 MPEG-4-related RTP payload
   formats. The FlexMux is a really separate issue [11] and need not be
   discussed here apart from the fact that is shares with this work the
   MPEG-4 Sync Layer as the interface into the MPEG-4 domain. RFC 3016
   describes transport of MPEG-4 video and LATM (for speech and audio
   codecs). This specification defines transport of any MPEG-4 type of
   data, with or without the Sync Layer. RFC YYYY describes a subset of
   the configurations that this specification can handle.

   Figure 2 displays the situation for video; note that this
   specification is compatible with RFC 3016. Figure 3 displays the
   situation for audio, note the presence of the LATM multiplex, which
   makes RFC 3016 audio transport incompatible with this specification.

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   Figure 4 displays the situation for other MPEG-4 streams, including
   BIFS, ODS, IPMP, etc.

   +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
   |                                                               |
   |                                   MPEG-4 Video                |
   |                                                               | I
   |+++++++++++++++++++++++|                                       | S
   |                       |                                       | O
   |     Sync Layer        |                                       | /
   |                       |                                       | M
   |+++++++++++++++++++++++|                                       | P
   |            |          |                                       | E
   | FlexMux    |          |                                       | G
   |            |          |   <- same RTP packet structure ->     |
   |++++++++++++|          +++++++++++++++++++++++++++|++++++++++++|***
   |            |                         |           |            |
   | FlexMux    |     RFC XXXX            | RFC YYYY  |  RFC 3016  | I
   |   RTP      |  MPEG-4 generic RTP     |           |    for     | E
   | payload    |     payload             +++++++++++++   Video    | T
   |            |                                     |            | F
   +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

   Figure 2: Relationship of MPEG-4 RTP payload formats for the
   transport of video

   +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
   |                                                               |
   |                                   MPEG-4 Audio                |
   |                                                               | I
   |+++++++++++++++++++++++|                                       | S
   |                       |                                       | O
   |     Sync Layer        |                                       | /
   |                       |                                       | M
   |+++++++++++++++++++++++|                          +++++++++++++| P
   |            |          |                          |            | E
   | FlexMux    |          |                          |   LATM     | G
   |            |          |                          |            |
   |++++++++++++|          +++++++++++++++++++++++++++|++++++++++++|***
   |            |                         |           |            |
   | FlexMux    |     RFC XXXX            | RFC YYYY  |  RFC 3016  | I
   |   RTP      |  MPEG-4 generic RTP     |           |    for     | E
   | payload    |     payload             +++++++++++++   Audio    | T
   |            |                                     |            | F
   +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

   Figure 3: Relationship of MPEG-4 RTP payload formats for the
   transport of audio






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   ++++++++++++++++++++++++++++++++++++++++++++++++++++
   |                                                  |
   |                                MPEG-4 system     |
   |                                                  | I
   |+++++++++++++++++++++++|                          | S
   |                       |                          | O
   |     Sync Layer        |                          | /
   |                       |                          | M
   |+++++++++++++++++++++++|                          | P
   |            |          |                          | E
   | FlexMux    |          |                          | G
   |            |          |                          |
   |++++++++++++|          +++++++++++++++++++++++++++|***
   |            |                         |           |
   | FlexMux    |     RFC XXXX            | RFC YYYY  | I
   |   RTP      |  MPEG-4 generic RTP     |           | E
   | payload    |     payload             ++++++++++++| T
   |            |                                     | F
   ++++++++++++++++++++++++++++++++++++++++++++++++++++

   Figure 4: Relationship of MPEG-4 RTP payload formats for the
   transport of MPEG-4 system streams (including BIFS, ODS, IPMP).

3. Payload Format

   One or more Access Units or Access Unit fragments (see section 3.9
   for fragmentation rules) are mapped into each RTP packet. Some
   information attached to these AU or AU Fragment is mapped onto the
   RTP header (see section 3.1), some form an additional payload
   header. The resulting RTP payload is described in section 3.2, it is
   composed of 3 parts (see figure 5):
   . a Payload Header section (optional)
   . a RSLH (Remaining SL Header) section (optional)
   . a Payload Section.
   These are described respectively in section 3.3, 3.4 and 3.5 of this
   memo.

   When transporting SL streams, SL Packet Headers are transformed into
   Remaining SL Header (RSLH) with some fields extracted to be mapped
   in the RTP header and others extracted to be mapped in the
   corresponding Payload Header. The AU or AU fragment data (SL packet
   payload) i.e. Elementary Stream codec data is unchanged.

   When transporting Elementary Streams there is no RSLH section.

   This payload format has two packing styles. The "Single" packing
   style is a packing style where a single AU or AU fragment is
   transported per RTP packet. The "Multiple" packing style is a
   packing style where possibly more than one AU or AU fragment are
   transported per RTP packet. The default packing style is the
   "Single" packing style.


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   In the "Multiple" packing style, AU or AU fragments MUST be in
   decoding order inside one RTP packet. Decoding order is defined by
   the relevant codec specification. Note that decoding order and
   presentation order may be different, typically for video streams
   containing B frames (see [2]). According to the MPEG-4 system model
   the decoding order may be quantified using decoding time stamps
   (DTS).

   RTP Packets SHOULD be sent in the decoding order. In case of
   interleaving the first AU or AU fragment of each RTP packet is used
   as reference as in the following examples of RTP packets containing
   interleaved SL packets.
   This sequence is correct: [0,2,4][1,3,5]
   This sequence is correct: [0,3,6][1,2][4,5]
   This sequence is correct: [0,3,6][1,4][2,5]
   This sequence is prohibited: [0,4,2][1,5,3]
   This sequence is prohibited: [1,3,5][0,2,4]
   This sequence is prohibited: [0,3,6][2,5][1,4]

   In the "Multiple" packing style the Payload Header and RSLH contains
   fields with relative values, they MUST have sufficient bits to
   encode the difference i.e. senders MUST make sure that no fields
   undergo roll over inside one RTP packet. This may limit the number
   of SL packets inside one RTP packet and, when interleaving, may
   limit the interleaving period as detailed in section 3.6.

   The size and/or number of the payload(s) SHOULD be adjusted such
   that the resulting RTP packet is not larger than the path-MTU. To
   handle larger packets, this payload format relies on lower layers
   for fragmentation, which may not be desirable.

3.1 RTP Header Fields Usage

   Payload Type (PT):
        The assignment of an RTP payload type for this new packet
        format is outside the scope of this document, and will not be
        specified here. It is expected that the RTP profile for a
        particular class of applications will assign a payload type for
        this encoding, or if that is not done then a payload type in
        the dynamic range shall be chosen.

   Marker (M) bit:
        The M bit is set to 1 when all AU fragments in the RTP packet
        are Access Units ends.

        Specifically the M bit is set to 0 when the RTP packet contains
        one or more AU fragments that are not Access Unit ends, and the
        M bit is set to 1 for RTP packets that contain either:
        . A single complete Access Unit
        . The last fragment of an Access Unit
        . Several complete Access Units
        . Several last fragments of Access Units


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        . A mix of complete Access Units and last fragments of Access
        Units

        Therefore for streams where all SL packets are complete Access
        Units the M bit is 1 for all RTP packets. Note also that in
        terms of Sync Layer this means that the M bit is related to the
        accessUnitEndFlag.

   Extension (X) bit:
        Defined by the RTP profile used.

   Sequence Number:
        The RTP sequence number should be generated by the sender with
        a constant random offset.

   Timestamp:
        Set to a value corresponding to the compositionTimeStamp (CTS)
        of the first AU or AU fragment in the RTP packet. This mapping
        is established as follows:

        If CTS has less than 32 bits length, the RTP timestamp is
        generated to extend it out to 32 bits using the number of
        wraparounds. If CTS has more than 32 bits length, the RTP
        timestamp uses the 32 LSB of it. When using the Sync Layer the
        resolution of the timestamp (timeStampLength) is available from
        the SL configuration data and shall be used by receivers to
        reconstruct CTS with the original bit length. It is RECOMMENDED
        to use timeStampLength=32.

        When an RTP packet starts with a non-initial AU fragment, the
        timestamp of the initial fragment SHALL be used.

        For SL streams where CTS is never present the RTP packetizer
        SHOULD convey a reading of a local clock at the time the RTP
        packet is created.

        Note that since, according to RFC1889 [5, Section 5.1],
        timestamps are recommended to start at a random value, a
        receiver is not in the general case able to reconstruct the
        original MPEG-4 Time Stamps (CTS, DTS, OCR). This is not an
        issue for synchronization of multiple RTP streams. However,
        applications where streams from multiple sources are to be
        synchronized (for example one stream from local storage,
        another from a RTP streaming server) may have to transport out
        of band the random offset used to map CTS into RTP timestamp,
        which is not in the scope of this specification.
        Note also that since RTP devices may re-stamp the stream, all
        time stamps inside of the RTP payload (CTS and DTS in the
        Payload Header, OCR in RSLH) MUST be expressed as difference to
        the RTP time stamp. Since this subtraction may lead to negative
        values, the offset MUST be encoded as a two's complement signed
        integer in network octet order. Note these offsets (delta)
        typically require much fewer bits to be encoded than the

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        original length. Nevertheless senders MUST make sure that these
        fields have enough bits to encode these differences.

        When startCompositionTimeStamp is signaled in the
        SLConfigDescriptor the RTP time stamps MUST start with this
        value.

   SSRC, CC and CSRC fields are used as described in RFC 1889 [5].

   RTCP SHOULD be used as defined in RFC 1889 [5].

3.2 RTP payload structure

   The packet payload structure consists of 3 octet-aligned sections.

   The first section is the Payload Header Section and contains Payload
   Headers. Each Payload Header contains basic fragmentation and timing
   information (relative to the RTP timestamp) for one AU or AU
   fragment. The Payload Header structure is described in 3.3. In the
   "Single" packing style this section is empty by default.

   The second section is the RSLH Section and contains Remaining SL
   Headers (RSLH). The RSLH structure is described in 3.4. By default
   this section is empty.

   The last section (Payload Section) contains the AU or AU fragment
   codec bit stream fragments and is described in section 3.5. This
   section is never empty.

   The Nth Payload Header in the Payload Header Section, the Nth RSLH
   in the RSLH Section and the Nth AU or AU fragment payload in the
   Payload Section correspond to the Nth AU or AU fragment transported
   by the RTP packet.





















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   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|X|  CC   |M|     PT      |       sequence number         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :            contributing source (CSRC) identifiers             :
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |                                                               |
   |              Payload Header Section (octet aligned)           |
   |                                                               |
   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               |                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
   |                                                               |
   |              RSLH Section (octet aligned)                     |
   |                                                               |
   |               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               |                                               |
   +-+-+-+-+-+-+-+-+                                               |
   |                                                               |
   |              Payload Section (octet aligned)                  |
   |                                                               |
   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               :...OPTIONAL RTP padding        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   Figure 5: RTP packet for MPEG-4

3.3 Payload Header Section structure

   If the Payload Header Section consumes a non-integer number of
   octets, up to 7 zero-valued padding bits MUST be inserted at the end
   in order to achieve octet-alignment.

   In the "Single" packing style the Payload Header Section consists of
   a single Payload Header.

   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | Payload Header (x bits )        : padding bits|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Figure 6: Payload Header Section structure in "Single" packing style

   In the "Multiple" packing style the Payload Header section consist
   of a 2 octets field giving the size in bits (in network octet order)


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   of the following block of bit-wise concatenated PayloadHeaders. This
   size excludes the padding bits, if any.

   This size field is absent in the "Single" packing style not because
   it is not needed (which would be a minor gain) but for compatibility
   with RFC 3016.

   This size field is also absent when the value would always be zero
   because the Payload Header is always empty, which happens when a
   constant payload size in signaled using ConstantSize (see below).

   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | Payload Header section size   |                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
   | as many bit-wise concatenated Payload Headers                 |
   | as AU or AU fragments in this RTP packet                      |
   |                                 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                 : padding bits|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Figure 7: Payload Header Section structure in "Multiple" packing
   style

3.3.1 Payload Header structure

   The Payload Header content depends on parameters (as described in
   section 4.1); by default it is empty for the "Single" packing style
   and, in the "Multiple" packing style, contains at least the
   PayloadSize field, except when ConstantSize is signaled.

   When all options are used the Payload Header structure and the
   relationship with the related parameter is given in table 1.

   +===========================+=================================+
   | Fields of Payload Header  | Number of bits (parameters)     |
   +===========================+=================================+
   | PayloadSize               | SizeLength                      |
   +---------------------------+---------------------------------+
   | Index                     | IndexLength                     |
   +---------------------------+---------------------------------+
   | IndexDelta                | IndexDeltaLength                |
   +---------------------------+---------------------------------+
   | CTSFlag                   | 1      If (CTSDeltaLength > 0)  |
   +---------------------------+---------------------------------+
   | CTSDelta                  | CTSDeltaLength If (CTSFlag==1)  |
   +---------------------------+---------------------------------+
   | DTSFlag                   | 1      If (DTSDeltaLength > 0)  |
   +---------------------------+---------------------------------+
   | DTSDelta                  | DTSDeltaLength If (DTSFlag==1)  |
   +---------------------------+---------------------------------+


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   Table 1: Payload Header fields and parameters giving the sizes

   In the general case a receiver can only discover the size of a
   Payload Header by parsing it since for example the presence of
   CTSDelta is signaled by the value of CTSFlag.

3.3.2 Fields of a Payload Header

   PayloadSize:
        Indicates the size in octets of the associated Payload, which
        can be found in the Payload Section of the RTP packet. The
        length in bits of this field is signaled by the SizeLength
        parameter (see section 4.1).

        There is an exception to that. In the "Multiple" packing style
        when a RTP packet contains only one AU or AU fragment, the
        PayloadSize field SHALL contain the size of the entire
        corresponding AU. There are two reasons, firstly the size of
        the fragment is not needed when there is only one fragment in
        the RTP packet, secondly this is useful in order to detect if a
        full Access Unit has been received after the loss of a packet
        carrying a M bit set to 1.

   Index, IndexDelta:
        Encodes the serial number of the associated AU or AU fragment.
        IndexDelta is useful for interleaving (see section 3.6). When
        transporting a SL stream, Index and IndexDelta SHALL be used to
        encode the packetSequenceNumber field of the SL Packet Header,
        if present.

        Index is optional and -if present- appears in the first Payload
        Header of a RTP packet.

        The length in bits of the Index field is defined by the
        IndexLength parameter (see section 4.1).

        IndexDelta is optional and -if present- appears for subsequent
        (non-first) Payload Headers of a RTP packet.

        The length in bits of the IndexDelta field is defined by the
        IndexDeltaLength parameter (see section 4.1).

        Both Index and IndexDelta MUST be incremented so that 2
        consecutive AU or AU fragments SHALL be distinguishable. One
        exception for Index is described in 3.6.1.

        If the parameter IndexDeltaLength is defined, non-first AU or
        AU fragments inside a RTP packet have their serial number
        encoded as a difference (thus the name IndexDelta). IndexDelta
        MUST have sufficient bits to encode this difference. This
        difference is relative to the previous AU or AU fragment in the
        RTP packet according to (with i>=0):
        Serial number(0) = Index(0)

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        Serial number (i+1) = Serial number (i) + IndexDelta(i+1) + 1

        If the parameter IndexDeltaLength is not defined the default
        value is zero and then the IndexDelta field is not present for
        non-first AU or AU fragments. Nevertheless receivers SHALL then
        apply the above formula with IndexDelta equal to zero. In other
        words by default the serial number is incremented by 1 for each
        AU or AU fragment in the RTP packet.


   CTSFlag (1 bit):
        Indicates whether the CTSDelta field is present.
        A value of 1 indicates that the CTSDelta field is present, a
        value of 0 that it is not present.

        If CTSDeltaLength is not zero, CTSFlag is present in all
        Payload Headers regardless of whether the AU fragment is an
        Access Unit start or not.

   CTSDelta (CTSDeltaLength bits):
        Specifies the value of the CTS as a 2-complement offset (delta)
        from the timestamp in the RTP header of the RTP packet. The
        length in bits of each CTSDelta field is specified by the
        CTSDeltaLength parameter (see section 4.1). CTSDelta MUST have
        sufficient bits to encode this difference.

        The CTSDelta field is present if CTSFlag is 1.

        For the first Payload Header of each RTP packet CTSFlag is
        always 0, since the composition time stamp of the first AU or
        AU fragment in the RTP packet is mapped to the RTP time stamp.
        When using the Sync Layer the sender MUST remove the
        compositionTimeStamp from the RSLH.

        Senders MUST finish assembling a RTP packet for which CTSDelta
        would roll over since this would prevent the receiver from
        reconstructing the correct CTS. This can result in sub optimal
        RTP packets (smaller than the MTU) depending on the MTU, the AU
        or AU fragment sizes and CTSDeltaLength.

   DTSFlag (1 bit):
        Indicates whether the DTSDelta field is present. A value of 1
        indicates that DTSDelta is present, a value of 0 that it is not
        present.

        If DTSDeltaLength is not zero, DTSFlag is present in all
        Payload Headers regardless of whether the AU fragment is an
        Access Unit start or not. When transporting SL streams the
        receiver needs this flag in order to reconstruct the
        decodingTimeStampFlag of SL Packet Headers.

   DTSDelta (DTSDeltaLength  bits):


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        Encodes (compositionTimeStamp - decodingTimeStamp) for the same
        AU or AU fragment(always positive). The length in bits of each
        DTSDelta field is specified by the DTSDeltaLength parameter
        (see section 4.1).

        Senders MUST make sure that DTSDeltaLength is large enough to
        encode the difference between CTS and DTS (otherwise the DTS
        computed by the receiver would be incorrect).

        The DTSDelta field appears when DTSFlag is 1. The sender MUST
        always remove the decodingTimeStamp from the RSLH.

        If DTSDelta is zero i.e. if decodingTimeStamp equals
        compositionTimeStamp then DTSFlag MUST be set to 0 and no
        DTSDelta field SHALL be present.


3.4 RSLHSection structure

   This section is present only when using the Sync Layer, and then,
   when the rules in the previous section have left remaining fields.

   This section first consists of a field (RSLHSectionSize) giving the
   size in bits of the following block of bit-wise concatenated RSLHs
   (this size does not include padding bits).

   If the section consumes a non-integer number of octets, up to 7 zero
   padding bits MUST be inserted at the end in order to achieve octet-
   alignment.

   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | RSLHSectionSize (RSLHSectionSizeLength bits)| RSLH (variable|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               |
   | number of bits)                                             |
   |                                                             |
   |         +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |         | RSLH (variable number of bits)                    |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | etc                                                         |
   | as many bit-wise concatenated RSLHs                         |
   | as SL Packets in this RTP packet                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | RSLH (variable number of bits)                              |
   |                                               +-+-+-+-+-+-+-+
   |                                               : padding bits|
   |-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Figure 8: RSLHSection structure

   The length in bits of the RSLHSectionSize field is
   RSLHSectionSizeLength and is specified with a default value of zero

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   indicating that the whole RSLHSection is absent. Note that for
   compatibility with RFC 3016 we need to be able to make the
   RSLHSection disappear completely, including the RSLHSectionSize
   field. This is the reason why there is such a variable length with a
   zero default value indicating the absence of the RSLHSectionSize
   field.

   +=================================+===============================+
   | Fields of RSLHSection           |         Number of bits        |
   +=================================+===============================+
   | RSLHSectionSize                 |       RSLHSectionSizeLength   |
   +---------------------------------+-------------------------------+
   | all bit-wise concatenated RSLHs |       RSLHSectionSize         |
   +---------------------------------+-------------------------------+

   Table 2: Sizes in bits inside RSLHSection

   Parsing of the bit-wise concatenated RSLHs requires MPEG-4 system
   awareness, specifically it requires to understand the MPEG-4
   Sync Layer (SL) syntax and the modifications to this syntax
   described in the next section.

   However thanks to the RSLHSectionSize field non-MPEG-4-system
   receivers can skip this part by rounding up RSLPHSize/8 to the next
   integer number of octets. This means that receivers not implementing
   the Sync Layer can process streams containing Sync Layer specific
   items by simply ignoring the parts they would not be able to parse.

3.4.1 RSLH structure

   RSLH is present only when using the Sync Layer, and then, when the
   rules in the previous section have left remaining fields.

   A Remaining SL Packet Header (RSLH) is what remains of an SL header
   after modifications for mapping into this payload format.

   The following modifications of the SL Packet Header MUST be applied.
   The other fields of the SL Packet Header MUST remain unchanged but
   are bit-shifted to fill in the gaps left by the operations specified
   below.

3.4.2 Removal of fields

   The following SL Packet Header fields -if present- are removed since
   they are mapped either in the RTP header or in the corresponding
   Payload Header:
   . compositionTimeStampFlag
   . compositionTimeStamp
   . decodingTimeStampFlag
   . decodingTimeStamp
   . packetSequenceNumber
   . AccessUnitEndFlag (in "Single" packing style only)


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   The AccessUnitEndFlag, when present for a given stream, MUST be
   removed from every RSLH when using the "Single" packing style since
   it has the same meaning as the Marker bit (and for compatibility
   with RFC 3016). However when using the "Multiple" packing style,
   AccessUnitEndFlag MUST NOT be removed since it is useful to signal
   individual AU ends.

3.4.3 Mapping of OCR

   Furthermore if the SL Packet header contains an OCR, then this field
   is encoded in the RSLH as a 2-complement difference (delta) exactly
   like a compositionTimeStamp or a decodingTimeStamp in the
   PayloadHeader. The length in bit of this difference is indicated by
   the OCRDeltaLength parameter (see section 4.1).

   With this payload format OCRs MUST have the same clock frequency as
   Time Stamps.

   If compositionTimeStamp is not present for a SL packet that has OCR
   then the OCR SHALL be encoded as a difference to the RTP time stamp.

3.4.4 Degradation Priority

   For streams that use the optional degradationPriority field in the
   SL Packet Headers, only SL packets with the same degradation
   priority SHALL be transported by one RTP packet so that components
   may dispatch the RTP packets according to appropriate QoS or
   protection schemes. Furthermore only the first RSLH of one RTP
   packet SHALL contain the degradationPriority field since it would be
   otherwise redundant.

3.5 Payload Section structure

   The Payload Section contains the concatenated AU or AU fragment
   Payloads. By definition AU or AU fragment Payloads are octet
   aligned.

   For efficiency SL packets do not carry their own payload size. This
   is not an issue for RTP packets that contain a single SL Packet.
   However in the "Multiple" packing style the size of each AU or AU
   fragment payload MUST be available to the receiver.

   If the AU or AU fragment payload size is constant for a stream, the
   size information SHOULD NOT be transported in the RTP packet.
   However in that case it MUST be signaled using the ConstantSize
   parameter (see section 4.1).

   If the AU or AU fragment payload size is variable then the size of
   each AU or AU fragment payload MUST be indicated in the
   corresponding Payload Header. In order to do so the Payload Header
   MUST contain a PayloadSize field. The number of bits on which this
   PayloadSize field is encoded MUST be indicated using the SizeLength
   parameter (see section 4.1).

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   The absence of either ConstantSize or SizeLength indicates the
   "Single" packing style i.e. that a single AU or AU fragment is
   transported in each RTP packet for that stream.


   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | AU or AU fragment (variable number of octets)               |
   |                                                             |
   |                                                             |
   |                             +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                             | AU or AU fragment             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
   |                                                             |
   |         (variable number of octets)                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | etc                                                         |
   | as many octet-wise concatenated AU or AU fragment           |
   | as required to finish RTP packet                            |
   |-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Figure 9: Payload Section structure

3.6 Interleaving

   SL Packets MAY be interleaved. Senders MAY perform interleaving.
   Receivers MUST support interleaving. Additional specifications MAY
   restrict this support by explicit signaling (see for example
   RFCYYYY).

   Note for Sync Layer implementers: the AUSequenceNumber field of the
   SL Header MUST NOT be used for interleaving since firstly it may
   collide with the Scene Description Carousel usage described in
   section 6.2 and secondly it is not visible to receivers that do not
   implement the Sync Layer and would skip the RSLH section
   transporting AUSequenceNumber.

   When interleaving of AU or AU fragments is used it SHALL be
   implemented using the IndexDelta fields of the Payload Header.
   Senders MUST NOT make RTP packets for which IndexDelta rolls over.
   Therefore depending on the interleaving scheme (if any), the MTU and
   the AU or AU fragment sizes, senders wishing to make optimally sized
   RTP packets (i.e. close to the MTU) will need to set
   IndexDeltaLength to a properly large value.

   Senders SHOULD use non zero values of IndexDeltaLength only for
   streams that exhibit interleaving, so that this can be interpreted
   by receivers as an indication that interleaving maybe present.

   There are, based on this, two ways for a receiver to implement de-
   interleaving:

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   . Time-Stamp-Based-Interleaving (TSBI see section 3.6.1) uses
   IndexDelta and timestamps.
   . Index-Based-Interleaving (see section 3.6.2) uses IndexDelta and
   Index.

   This is signaled using mime parameters as in the following table.
   Note that the need for two methods arises from two facts: firstly
   the time stamp based method is more economical and in basic cases
   (no multiple AU fragments, CTS always defined) simpler to implement.
   Secondly, unfortunately this method does not always work as
   explained below.


   ==================================================================
   |                | IndexDeltaLength = 0 | IndexDeltaLength !=  0 |
   ------------------------------------------------------------------
   | IndexLength=0  |   no interleaving    |          TSBI          |
   ------------------------------------------------------------------
   | IndexLength!=0 |   no interleaving,   |   Index=0  |  Index!=0 |
   |                |   SL.packetSeqNum    |-------------------------
   |                |    transport         |    TSBI    |    IBI    |
   ==================================================================


3.6.1 Time stamp based interleaving (TSBI)

   The conjunction of RTP time stamp, IndexDelta and CTS may allow a
   receiver to un-ambiguously re-order AU or AU fragments based on
   their time stamps (CTS).

   This is possible and efficient for streams where only complete
   Access Units are transported and receivers can always compute the
   time stamp of each Access Unit.

   In case of Access Units of constant duration (e.g. audio streams)
   the explicit presence of CTS in the Payload Header is not even
   required; Indeed then we have (i being the index of one AU in one
   RTP packet):
   CTS(0) = RTP-TS
   for (i >= 1): CTS(i) = CTS(i-1) + (IndexDelta(i)+1)*AU_duration

   AU_duration, when constant, can be either signaled in SLConfig or be
   deduced from the decoder configuration (see the "Config" MIME
   parameter).

   Senders MUST use either IndexLength=0 or set all Index values in all
   packets to zero so that receivers can detect this as an indication
   that de-interleaving SHOULD be performed using time stamps.

   When using the Sync Layer and when interleaving senders MUST use for
   SL.timeStampLength values large enough to prevent the CTS from
   rolling over more often than a packet loss burst length. Pre-
   existing SL streams that do not comply with this requirement cannot

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                RTP Payload Format for MPEG-4 Streams   February 2002


   be interleaved using this payload format (or by using IBI as in
   3.6.2)

3.6.2 Index based interleaving (IBI)

   The timestamp-based interleaving algorithm described in the previous
   section does not work when a CTS cannot always be computed for all
   AU or AU fragments (for example after a packet loss); this happens:
        . If the AU duration is not constant (SL durationFlag = 0) and
        CTS is not signaled  (SL useTimeStampsFlag= 0).
        . When interleaving AU fragments.

   When interleaving, senders of such streams MUST use the index-based
   technique described in this section.

   The conjunction of RTP sequence number, Index and IndexDelta can
   produce a quasi-unique identifier for each AU or AU fragment so that
   a receiver can unambiguously reconstruct the original order even in
   case of out-of-order packets, packet loss or duplication (see the
   pseudo code in 3.3.2 and 6.1). Specifically the RTP sequence number
   is used to re-order packets and inside one RTP packet we have:
   Serial number(0) = Index(0)
   Serial number(i+1) = Serial number(i) + IndexDelta(i+1) + 1   (i>=0)

   This requires, however, that IndexLength is not too small. For that
   reason senders when interleaving in this fashion MUST use for
   IndexLength values large enough to prevent Index from rolling over
   more often than a typical loss burst length. Pre-existing SL streams
   that do not comply with this requirement (specifically if
   SL.packetSeqNumLength is too small) cannot be interleaved using this
   payload format (or should use TSBI).

   Receivers SHOULD interpret non-zero values in the Index field as an
   indication that de-interleaving can be performed using Index and
   IndexDelta but cannot be performed using timestamps.

3.6.3 SL streams that should not be interleaved

   SL streams for which both SL.timeStampLength and
   SL.packetSeqNumLength are too small SHOULD NOT be interleaved with
   this payload format, the reason being that small values would cause
   a receiver to drop a large part of the stream in case of packet
   loss. The actual minimal length depends on network loss properties
   and on the expected quality of service.

3.7 Fragmentation Rules

   MPEG-4 Access Units are the default fragments for MPEG-4 bitstreams
   and SHOULD be mapped directly into RTP packets of this format with
   two exceptions:
   - Access Units larger than the MTU
   - When using interleaving for better packet loss resilience.


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   This section gives rules to apply when performing Access Unit
   fragmentation. Let us first explain the context before describing
   the rules.

   For error resilience purposes some MPEG-4 codecs define optional
   syntax of Access Units fragments that are independently decodable.
   Examples are Video Packets for video and Error Sensitivity
   Categories (ESC) for audio. This always corresponds to specific
   bitstream syntax, which is signaled in the DecoderSpecificInfo
   inside the DecoderConfig in SLConfig, and/or using the corresponding
   parameters as described in section 4.1.
   Thanks to that, decoders are aware whether encoders are operating in
   such a mode or not (however since this codec configuration is an
   opaque data block this is not explicitly signaled by this payload
   format).

   If not operating in such a mode it is obvious that the decoder has
   to skip packets after a loss until an Access Unit start is received.
   Similarly decoder implementations that do not implement robust
   decoding of Access Units fragments have to discard all packets after
   a packet loss until an Access Unit start is received. In the same
   way decoder implementations that do not implement re-synchronization
   at any Access Units start have to discard all packets after a packet
   loss until a Random Access Point Access Unit is received. These are
   all obvious things that a good implementation would do.

   However serious problems would arise for decoder implementations
   that try to restart decoding after a packet loss if independently
   decodable fragments are signaled (in the decoder configuration) but
   the fragments actually received are not independently decodable
   because the RTP sender has made RTP packets on different boundaries
   than the fragments provided by the encoder (so this issue applies to
   the interface between the encoder and the RTP sender and to the RTP
   sender component itself). Indeed the decoder has in general no way
   to detect such a faulty fragment (except for MPEG-4 video).

   For this reason the following rules must be applied:

   In the spirit of ALF this payload format should transport either
   complete Access Units or fragments of Access Units that are
   independently decodable. Specifically when a given codec has an
   independently decodable Access Unit fragments optional syntax this
   option SHOULD be used.

   Independently decodable Access Units fragments SHOULD NOT be split
   across several RTP packets.

   An MPEG-4 audio stream encoded using the ESC syntax MUST NOT split
   one ESC across 2 RTP packets.

   When using MPEG-4 Video Packets since all Video Packets start with a
   specific resynchronization marker that can be unambiguously detected
   this rule is not needed. However it is strongly RECOMMENDED to

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   always adapt the Video Packet size to fit the MTU. In any case a
   video AU or AU fragment start MUST always be aligned with either:
        . a VOP start.
        . a Video Packet start.
        . or a GOV followed by the first (or only) Video Packet of the
        following VOP.

4. Types and Names

   This section describes the MIME types and names associated with this
   payload format. Section 4.1 registers the MIME types, as per RFC
   2048.

   This format may require additional information about the mapping to
   be made available to the receiver. This is done using parameters
   described in the next section. The absence of any of these fields is
   equivalent to a field set to the default value, which is always zero
   for numerical parameters. The absence of any such parameters
   resolves into a default "basic" configuration compatible with
   RFC3016 for MPEG-4 video.

   In the MPEG-4 framework the SL stream configuration information is
   carried using the Object Descriptor. For compatibility with
   receivers that do not implement the full MPEG-4 system specification
   this information MAY also be signaled using parameters described
   here. When such information is present both in an Object Descriptor
   and as a parameter of this payload format it MUST be exactly the
   same.

   For transport of MPEG-4 audio and video without the use of MPEG-4
   systems, as well as to support non-MPEG-4 system receivers, it is
   also possible to transport information on the profile and level of
   the stream and on the decoder configuration. This is also described
   in the next section.

   Finally this MIME type also defines a mode parameter and a profile
   parameter that are intended for derivations of this payload format.
   One such derivation is described in the companion RFC YYYY.

4.1 MIME type registration

   MIME media type name:  "video" or "audio" or "application"

   "video" MUST be used for MPEG-4 Visual streams (i.e. video as
   defined in ISO/IEC 14496-2 (Streamtype = 4) and/or graphics as
   defined in ISO/IEC 14496-1 (Streamtype = 3)) or MPEG-4 Systems
   streams that convey information needed for an audio/visual
   presentation.

   "audio" MUST be used for MPEG-4 Audio streams (ISO/IEC 14496-3)
   (Streamtype = 5)) or MPEG-4 Systems streams that convey information
   needed for an audio only presentation.


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   "application" MUST be used for MPEG-4 Systems streams (ISO/IEC14496-
   1 (all other StreamType values)) that serve other purposes than
   audio/visual presentation, e.g. in some cases when MPEG-J streams
   are transmitted.

   MIME subtype name: mpeg4-generic

   Required parameters: none

   Optional parameters:

   mode:
        The mode in which this specification is used. This
        specification itself defines only the default mode
        (Mode=default). When the mode parameter is not present the
        default mode SHALL be assumed. In the default mode all
        parameters are OPTIONAL and as defined here. Other modes may be
        defined as needed in other RFCs. A mode MUST be a subset of
        this specification. Specifically when defining a mode care MUST
        be taken that an implementation of this specification can
        decode the payload format corresponding to this new mode. For
        this reason a mode MUST NOT specify new default values for MIME
        parameters and MIME parameters MUST be present (unless they
        have the default value) even if it is redundant in case the
        mode assigns fixed values. A mode may define additionally that
        some MIME parameters are required instead of optional, that
        some MIME parameters have fixed values (or ranges), and that
        there are rules restricting the usage (for example RFCYYYY
        forbids the carriage of multiple AU fragments in the same RTP
        packet and -logically- uses only TSBI interleaving).

   profile:
        The meaning of this parameter may be defined by a mode. This is
        meant to be used in order to define sub-configurations of a
        given mode, for example the maximum delay (and therefore the
        size of buffers) induced by the usage of interleaving.
        Implementations of this specification can ignore this
        parameter.

   DTSDeltaLength:
        The number of bits on which the DTSDelta field is encoded in
        each Payload Header. The default value is zero and indicates
        the absence of DTSFlag and DTSDelta in the Payload Header (the
        stream does not transport decodingTimeStamps). A value larger
        than zero indicates that there is a DTSFlag in each Payload
        Header. Since decodingTimeStamp, if present, must be encoded as
        a difference to the RTP time stamp, the DTSDeltaLength
        parameter MUST be present in order to transport
        decodingTimeStamps with this payload format.

   CTSDeltaLength:
        The number of bits on which the CTSDelta field is encoded. The
        default value is zero and indicates the absence of the CTSFlag

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        and CTSDelta fields in Payload Header. Non-zero values MUST NOT
        be signaled in the "Single" packing style. Since
        compositionTimeStamps, if present, must be encoded as a
        difference to the RTP time stamp, the CTSDeltaLength parameter
        MUST be present in order to transport compositionTimeStamps
        using this payload format (in the "Multiple" packing style).
        However CTSDeltaLength SHOULD be set to zero (or not signaled)
        for streams that have a constant Access Unit duration (which
        can be explicitly signaled using the DurationFlag and
        AccessUnitDuration field of SLConfigDescriptor).

   OCRDeltaLength:
        The number of bits on which the OCRDelta field is encoded in
        RSLH. The default value is zero and indicates the absence of
        OCR for this stream. Since objectClockReference -if present-
        must be encoded as a difference to the RTP time stamp, the
        OCRDeltaLength parameter MUST be present in order to transport
        objectClockReferences with this payload format.

   SizeLength:
        The number of bits on which the PayloadSize field of a Payload
        Header is encoded. The default value is zero and indicates the
        "Single" packing style (unless ConstantSize is present).
        Simultaneous presence of this parameter and ConstantSize is
        illegal. Either the SizeLength or ConstantSize parameter MUST
        be present in order to signal the "Multiple" packing style of
        this payload format.

   ConstantSize:
        The constant size in octets of each AU or AU fragment Payload
        for this stream. The default value is zero and indicates
        variable AU or AU fragment Payload size (or the "Single"
        packing style if SizeLength is absent). Simultaneous presence
        of this parameter and SizeLength is illegal. Either the
        SizeLength or ConstantSize parameter MUST be present in order
        to signal the "Multiple" packing style of this payload format.
        When ConstantSize is present the PayloadSize field of the
        Payload Header in the RTP packets MUST NOT be present.

   IndexLength:
        The number of bits on which the Index is encoded in the first
        Payload Header of a RTP packet. The default value is zero and
        indicates the absence of Index and IndexDelta for all Payload
        Headers. Since SL.packetSequenceNumber -if present- must be
        mapped in the Payload Header, the IndexLength parameter MUST be
        present in order to transport SL.packetSequenceNumber with this
        payload format.

   IndexDeltaLength:
        The number of bits on which the IndexDelta are encoded in any
        non-first Payload Header. The default value is zero and
        indicates that the serial number MUST be incremented by one for
        each AU or AU fragment in the RTP packet (see section 3.5). A

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        non-zero IndexDeltaLength parameter MUST be present when using
        interleaving with this payload format.

   RSLHSectionSizeLength:
        The number of bits that is used to encode the RSLHSectionSize
        field. The default value is zero and indicates the absence of
        the whole RSLHSection for all RTP packets of this stream.

   SLConfigDescriptor:
        A base-64 encoding of the SLConfigDescriptor. This SHALL be the
        original SLConfigDescriptor and it SHALL be the same as the one
        transported by the OD framework, if any.

   profile-level-id:
        A decimal representation of the MPEG-4 Profile Level indication
        value. For audio this parameter indicates which MPEG-4 Audio
        tool subsets are applied to encode the audio stream and is
        defined in ISO/IEC 14496-1 [1]. For video this parameter
        indicates which MPEG-4 Visual tool subsets are applied to
        encode the video stream and is defined in Table G-1 of ISO/IEC
        14496-2 [2]. This parameter MAY be used in the capability
        exchange or session setup procedure to indicate MPEG-4 Profile
        and Level combination of which the relevant MPEG-4 media codec
        is capable. If this parameter is not specified its default
        value is 1 (Simple Profile/Level 1) for video (for
        compatibility with RFC 3016) and otherwise 254 (0xFE being
        defined in ISO/IEC 14496-1 [1] as being the generic default
        value).

   config:
        A hexadecimal representation of an octet string that expresses
        the media payload configuration. Configuration data is mapped
        onto the octet string in an MSB-first basis. The first bit of
        the configuration data SHALL be located at the MSB of the first
        octet. In the last octet, zero-valued padding bits, if
        necessary, shall follow the configuration data. For audio
        streams, config is the audio object type specific decoder
        configuration data AudioSpecificConfig() as defined in ISO/IEC
        14496-3 [3]. For video this expresses the MPEG-4 Visual
        configuration information, as defined in subclause 6.2.1 Start
        codes of ISO/IEC14496-2 [2] and the configuration information
        indicated by this parameter SHALL be the same as the
        configuration information in the corresponding MPEG-4 Visual
        stream, except for first-half-vbv-occupancy and latter-half-
        vbv-occupancy, if it exists, which may vary in the repeated
        configuration information inside an MPEG-4 Visual stream (See
        6.2.1 Start codes of ISO/IEC14496-2).

   StreamType:
        The integer value that indicates the type of MPEG-4 stream that
        is carried; its coding corresponds to the values of the
        streamType as defined for the DecoderConfigDescriptor in
        ISO/IEC 14496-1.

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   Encoding considerations:
        System bitstreams MUST be generated according to MPEG-4 System
        specifications (ISO/IEC 14496-1). Video bitstreams MUST be
        generated according to MPEG-4 Visual specifications (ISO/IEC
        14496-2). Audio bitstreams MUST be generated according to MPEG-
        4 Audio specifications (ISO/IEC 14496-3). If the Sync Layer is
        used SL streams MUST be generated according to MPEG-4 Sync
        Layer specifications (ISO/IEC 14496-1 section 10), then in
        order to read the RSLH parts of this format the
        SLConfigDescriptor is required. These bitstreams are binary
        data and MUST be encoded for non-binary transport (for Email,
        the Base64 encoding is sufficient).  This type is also defined
        for transfer via RTP.  The RTP packets MUST be packetized
        according to the RTP payload format defined in RFC XXXX.

   Security considerations:
        As in RFC XXXX.

   Interoperability considerations:
        MPEG-4 provides a large and rich set of tools for the coding of
        visual objects.  For effective implementation of the standard,
        subsets of the MPEG-4 tool sets have been provided for use in
        specific applications. These subsets, called "Profiles", limit
        the size of the tool set a decoder is required to implement. In
        order to restrict computational complexity, one or more
        "Levels" are set for each Profile. A Profile@Level combination
        allows:
        . A codec builder to implement only the subset of the standard
        he needs, while maintaining interoperability with other MPEG-4
        devices included in the same combination, and
        . Checking whether MPEG-4 devices comply with the standard
        ('conformance testing').

        A stream SHALL be compliant with the MPEG-4 Profile@Level
        specified by the parameter "profile-level-id". Interoperability
        between a sender and a receiver may be achieved by specifying
        the parameter "profile-level-id" in MIME content, or by
        arranging in the capability exchange/announcement procedure to
        set this parameter mutually to the same value.

   Published specification:
        The specifications for MPEG-4 streams are presented in ISO/IEC
        14469-1, 14469-2, and 14469-3.  The RTP payload format is
        described in RFC XXXX.

   Applications that use this media type:
        Multimedia streaming and conferencing tools.

   Additional information: none

   Magic number(s): none


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   File extension(s):
        None. A file format with the extension .mp4 has been defined
        for MPEG-4 content but is not directly correlated with this
        MIME type which sole purpose is RTP transport.

   Macintosh File Type Code(s): none

   Person & email address to contact for further information:
        Authors of RFC XXXX.

   Intended usage: COMMON

   Author/Change controller:
        Authors of RFC XXXX, IETF Audio/Video Transport working group.

4.2 Concatenation of parameters

   Multiple parameters SHOULD be expressed as a MIME media type string,
   in the form of a semicolon-separated list of parameter=value pairs
   (see examples below).

4.3 Usage of SDP

4.3.1 The a=fmtp keyword

   It is assumed that one typical way to transport the above-described
   parameters associated with this payload format is via an SDP [10]
   message for example transported to the client in reply to a RTSP
   [13] DESCRIBE message or via SAP [14]. In that case the (a=fmtp)
   keyword MUST be used as described in RFC 2327 [10, section 6]. The
   syntax being then:

   a=fmtp:<format> <parameter name>=<value>

4.3.2 SDP example

   The following is an example of SDP syntax for the description of a
   session containing one MPEG-4 video, one MPEG-4 audio stream and
   three MPEG-4 system streams, the first one being BIFS, the second
   one OD stream and the third one IPMP. All are transported using this
   format and the AVP profile [12]. Note the usage of some MIME
   parameters: all stream display their StreamType; the video stream
   uses DTS with DTSDelta encoded on 4 bits; the audio stream uses the
   "Multiple" packing style with 12 bits to describe the size of each
   AU or AU fragment payload. See the Appendix for more examples.

   o= ....
   I= ....
   c=IN IP4 123.234.71.112
   m=video 1034 RTP/AVP 97
   a=rtpmap:97 mpeg4-generic
   a=fmtp:97 StreamType=4;DTSDeltaLength=4
   m=audio 1810  RTP/AVP 98

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   a=rtpmap:98 mpeg4-generic
   a=fmtp:98 StreamType=5;SizeLength=12;
   m=application 1234  RTP/AVP 99
   a=rtpmap:99 mpeg4-generic
   a=fmtp:99 StreamType=3
   m=application 1236  RTP/AVP 100
   a=rtpmap:100 mpeg4-generic
   a=fmtp:100 StreamType=1
   m=application 1238  RTP/AVP 101
   a=rtpmap:101 mpeg4-generic
   a=fmtp:101 StreamType=7

5. IANA Considerations

   One new MIME subtype is to be registered, see Section 4.1.

6. Other issues

6.1 SL-packetized stream reconstruction

   The purpose of this section is to document how a receiver can
   reconstruct a valid SL-packetized stream. This reconstruction is
   performed by reversing the payload structure rules (section 3). We
   explicitly describe here the most complex transformations.

   In the following let (i) be the index of SL packets inside one RTP
   packet (starting at zero for each RTP packet), let SLPacketHeader.x
   denote field x of the reconstructed SL packet header, let
   PayloadHeader.x denote field x of the received PayloadHeader, etc.

   SLPacketHeader.packetSequenceNumber is restored from
   PayloadHeader.Index and PayloadHeader.IndexDelta using:

   If ( IndexLength == 0) { // or is absent
      if ( SLConfig.packetSeqNumLength == 0 ) {
          // this stream does not have SL packet sequence number
      }
      else {
          // illegal, normally the sender MUST map
          // SLPacketHeader.packetSequenceNumber in PayloadHeader
          // and set a relevant IndexLength value;
          // otherwise it is unfortunately impossible for the receiver
          // to reconstruct the correct sequence
      }
   }
   else { // IndexLength is not zero
      if ( SLConfig.packetSeqNumLength == 0 ) {
          // the original SL stream does not have SL packet
          // sequence numbers, typically the sender inserted them
          // in order to implement interleaving at the RTP level;
          // they must be ignored for SL stream reconstruction
      }
      else {

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         if (i == 0){ // first SL packet in RTP packet
           SLPacketHeader.packetSequenceNumber(0) =
   PayloadHeader.Index(0);
         }
         else { // remaining SL packets
           SLPacketHeader.packetSequenceNumber(i+1)=
              SLPacketHeader.packetSequenceNumber(i)
              + PayloadHeader.IndexDelta(i+1)
              +1;
         }
   }

   All time stamps (CTS, DTS, OCR), when present, are restored from the
   delta values. Time stamps flags (CTSFlag, DTSFlag) in PayloadHeader
   are used to reconstruct respectively the compositionTimeStampFlag
   and decodingTimeStampFlag of SLPacketHeader. The function
   corrected(x) for the RTP time stamp transformation is the mapping
   from 32 bits to SLConfig.timeStampLength, which may be smaller or
   larger than 32 bits:

   If (timeStampLength < 32 ) { // short SL time stamps
      corrected(x) = LSB(x); // only the timeStampLength LSBits of x
   }
   else If (timeStampLength > 32 ) { // long SL time stamps
      corrected(x) = x + m; // start with m=0
      if ( x(i) < x(i-1) ) { // 32 bits RTPTS roll over has occurred
      {
          m += 2^32;
      }
   }
   else If (timeStampLength = 32 ) { // recommended value
      corrected(x) = x; // direct mapping
   }


   if ( CTSDeltaLength == 0) { // or CTSDeltaLength is absent
      // CTS is not transported for this RTP stream
      if (i == 0){ // first SL packet in RTP packet
         if ( SLConfig.useTimeStamps == 1 ) {
            if ( SLPacketHeader.accessUnitStartFlag == 1 ) {
               SLPacketHeader.compositionTimeStampFlag(0) = 1;
               SLPacketHeader.compositionTimeStamp(0) =
                   corrected(RTP TimeStamp);
            }
            else {
               // ignore
            }
         }
         else {
             // empty
         }
      }
      else { // non-first SL packets in RTP packet

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         if ( SLConfig.useTimeStamps == 1 ) {
             if ( SLPacketHeader.accessUnitStartFlag == 1 ) {
                SLPacketHeader.compositionTimeStampFlag(i) = 0;
             }
             else {
                // ignore
             }
         }
         else {
             // empty
         }
      }
   }
   else { // CTSDeltaLength is not zero
      // CTS is transported for this stream
      if ( SLConfig.useTimeStamps == 1 ) {
         if ( SLPacketHeader.accessUnitStartFlag == 1 ) {
             SLPacketHeader.compositionTimeStampFlag(i) =
                      PayloadHeader.CTSFlag(i);
             SLPacketHeader.compositionTimeStamp(i) =
                    corrected(RTP TimeStamp) +
   PayloadHeader.CTSDelta(i);
         }
         else {
            // ignore CTSFlag (which must be zero)
         }
      else {
         // this is strange and sub-optimal at best
         // a receiver should ignore this
      }
   }

   if ( DTSDeltaLength == 0) { // or DTSDeltaLength is absent
      // DTS is not transported for this stream
      if ( SLConfig.useTimeStamps == 1 ) {
         if ( SLPacketHeader.accessUnitStartFlag == 1 ) {
             SLPacketHeader.decodingTimeStampFlag(i) = 0;
         }
         else {
             // ignore
         }
      }
      else {
          // empty
      }
   }
   else {
      // DTS is transported for this stream
      if ( SLConfig.useTimeStamps == 1 ) {
         if ( SLPacketHeader.accessUnitStartFlag == 1 ) {
              SLPacketHeader.decodingTimeStampFlag(i) =
                  PayloadHeader.DTSFlag(i);
              SLPacketHeader.decodingTimeStamp(i)=

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                  SLPacketHeader.compositionTimeStamp(i)
                  - PayloadHeader.DTSDelta(i); // DTS <= CTS always
         }
         else {
             // ignore DTSFlag (which must be zero)
         }
      }
      else {
         // this is strange and sub-optimal at best
         // a receiver should ignore this
      }
   }

   if ( OCRDeltaLength == 0) { // or OCRDeltaLength is absent
      // the RTP stream does not transport any OCR
      if ( SLConfig.OCRLenght == 0 ) {
          // this stream does not have any OCR
      }
      else {
          // illegal, normally the sender MUST detect
          // OCRs, replace them with OCRDelta and set
          // a relevant OCRDeltaLength value
      }
   }
   else {
      if ( SLConfig.OCRLenght == 0 ) {
         // this is strange and sub-optimal at best
         // a receiver should ignore this
      }
      else {
          SLPacketHeader.OCRflag(i) = RSLH.OCRFlag(i);
          if ( SLPacketHeader.OCRflag(i) == 1) {
               SLPacketHeader.objectClockReference(i) =
                    corrected(RTP TimeStamp) + RSLH.OCRDelta(i);
          }
      }
   }


   In the "Single" packing style the AccessUnitEndFlag, if needed, is
   restored from the M bit, as follows:

   if ( SLConfig.useAccessUnitEndFlag == 0 ) {
       // this SL stream does not signal access unit ends
   else {
       SLPacketHeader.AccessUnitEndFlag = M bit;
   }

   In the "Multiple" packing style the AccessUnitEndFlag is untouched
   in RSLH.

   The other SL packet header fields SHALL remain as found in RSLH.


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   It is obvious that in the general case the reconstruction of the
   original SL packetized stream requires SL-awareness. However this
   payload format allows in all cases a receiver that does not know
   about the SL syntax to reconstruct the semantic of Elementary
   Streams for the following very useful features:
   - Packet order (decoding order)
   - Access Unit boundaries (using the M bit)
   - Access Unit fragments (fragment boundaries using PayloadSize)
   - Composition Time Stamps, according to:
      compositionTimeStamp(i) = RTP TimeStamp + CTSDelta(i);
   - Decoding Time Stamps, according to:
      decodingTimeStamp(i) = compositionTimeStamp(i) - DTSDelta(i);
   - Packet serial number, according to:
      if (i == 0){ // first SL packet in RTP packet
           packet serial number(0) = Index(0);
         }
         else { // remaining SL packets
           packet serial number (i+1) = packet serial number (i)
               + IndexDelta(i+1) + 1;
         }

6.2 Handling of scene description streams

   MPEG-4 introduces new stream types as described in section 1 namely
   Object Descriptors and BIFS. In the following both OD and BIFS are
   discussed on the same basis i.e. as "scene description".

   Considering scene description as a "stream-able" type of content is
   a rather new concept and for that reasons some specific comments are
   needed.

   Typically scene descriptions are encoded in such a way that
   information loss would in the general case cripple the presentation
   beyond any hope of repair by the receiver. This is acceptable for a
   number of multimedia applications were the scene is first made
   available via reliable channels to the client and then played. This
   payload format is not primarily intended for this type of
   applications for which download of MPEG-4 interchange (.mp4) files
   would be typical. However this payload format can also be used. It
   is then RECOMMENDED however that the RTP packets should be
   transported using TCP (for example inside RTSP as described in [13,
   section 10.12]) or any other reliable protocol.

   On the other hand MPEG-4 has introduced the possibility to
   dynamically change the scene description by sending animation
   information (changes in parameters) and structural change
   information (updates). Since this information has to be sent in a
   timely fashion MPEG-4 has defined a number of techniques in order to
   encode the scene description in a manner that makes it behave
   similarly to other temporal encoding schemes such as audio and
   video. This payload format is intended for this usage.



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   Note that in many cases the application will consist of first the
   reliable transmission of a static initial scene followed by the
   streaming of animations and updates. For this reason the usage of
   this payload format is attractive since it offers a unique solution.

   Senders must be aware that suitable schemes should be used when
   scene description streams transport sensitive configuration
   information. For example in case the RTP packet transporting an OD-
   update command would be lost, the corresponding media stream would
   not be accessible by the receiver.

   Redundancy is a possibility and may either be added by tools
   hierarchically higher than this payload format, e.g. by packet based
   FEC, re-transmission, or similar tools. In such a case, the general
   congestion control principles have to be observed.

   Since BIFS and OD streams may be modified during the session with
   update commands, there is a need to send both update commands and
   full BIFS/OD refresh. For that reason MPEG-4 defines Random Access
   Points (RAP) for scene description streams (OD and BIFS) where by
   definition a decoder can restart decoding i.e. receives a "full
   update" of the scene. This mechanism is called Scene and Object
   Description Carousel. The AU Sequence Number field of SL Packet
   Header is used to support this behavior at the Sync Layer. When two
   access units are sent consecutively with the same AU Sequence
   Number, the second one is assumed to be a semantic repetition of the
   first. If a receiver starts to listen in the middle of a session or
   has detected losses, it can ignore all received AUs until such a
   RAP. The periodicity of transmission of these RAPs should be
   chosen/adjusted depending on the application and the network it is
   deployed on; i.e. exactly like Intra-coded frames for video, it is
   the responsibility of the sender to make sure the periodicity of
   RAPs is suitable.

6.3 Overlap with RFC 3016

   This payload format has been designed to have a (large) overlap with
   RFC 3016 [7]. The conditions for this overlap are:

   Conditions for RFC 3016:
   C1. MPEG-4 video elementary streams only
   C2. There MUST be a single VOP or Video Packet per RTP packet (which
   is only recommended in RFC 3016)
   C3. The decoder configuration MUST be signaled out-of-band either
   using the Config mime parameter or using the OD framework

   Conditions for this payload format:
   C4. No MIME parameters defined (or all set to zero), i.e. "Single"
   packing style with empty Payload Header and empty RSLH.
   C5. Receivers MUST be ready to accept (and ignore) video
   configuration headers (e.g. VOSH, VO and VOL) and visual-object-
   sequence-end-code transported in-band.


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   Under conditions C2 and C4 the MPEG-4 video RTP packet structures
   are identical. Since C4 and C5 MUST be supported by implementations
   of this specification the conditions for RTP streams backward
   compatibility of this specification with RFC3016 are established
   when RFC3016 is used with condition C1, C2 and C3. Technically the
   most stringent condition is C2 but it is also a condition that makes
   a lot of sense for many reasons, whatever the application.

   Furthermore the MIME parameters have been aligned, specifically the
   parameters "config" and "profile-level-id" have the same name and
   signification in RFC3016 and in this memo.

   The remaining difference is therefore the MIME subtype name. It
   would be desirable then that specifications built upon this memo and
   enforcing the above minor usage restrictions of RFC3016 in order to
   provide a backward compatible solution would then specify that
   receivers can interpret the MIME subtype name "MP4V-ES" as being
   equivalent to MIME type "video" with subtype name "mpeg4-generic"
   and vice versa.

   In short this payload format is backward compatible with RFC3016 for
   video used in the recommended fashion.

6.4 Multiplexing

   An advanced MPEG-4 session may involve a large number of objects
   that may be as many as a few hundred, transporting each ES as an
   individual RTP stream may not always be practical. Allocating and
   controlling hundreds of destination addresses for each MPEG-4
   session may pose insurmountable session administration problems.
   The input/output processing overhead at the end-points will be
   extremely high also. Additionally, low delay transmission of low
   bitrate data streams, e.g. facial animation parameters, results in
   extremely high header overheads.

   To solve these problems, MPEG-4 data transport requires a
   multiplexing scheme that allows selective bundling of several ESs.
   This is beyond the scope of the payload format defined here.

   The MPEG-4's Flexmux multiplexing scheme may be used for this
   purpose and a specific RTP payload format is being developed [11].

   Another approach may be to develop a generic RTP multiplexing scheme
   usable for MPEG-4 data. The multiplexing scheme reported in [8] may
   be a candidate for this approach.

   For MPEG-4 applications, the multiplexing technique needs to address
   the following requirements:

   i. The ESs multiplexed in one stream can change frequently during a
   session. Consequently, the coding type, individual packet size and
   temporal relationships between the multiplexed data units must be
   handled dynamically.

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   ii. The multiplexing scheme should have a mechanism to determine the
   ES identifier (ES_ID) for each of the multiplexed packets. ES_ID is
   not a part of the SL header.

   iii. In general, an SL packet does not contain information about its
   size. The multiplexing scheme should be able to delineate the
   multiplexed packets whose lengths may vary from a few octets to
   close to the path-MTU.


7. Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [5]. This implies that confidentiality of the media
   streams is achieved by encryption. Because the data compression used
   with this payload format is applied end-to-end, encryption may be
   performed on the compressed data so there is no conflict between the
   two operations. The packet processing complexity of this payload
   type (i.e. excluding media data processing) does not exhibit any
   significant non-uniformity in the receiver side to cause a denial-
   of-service threat.

   However, it is possible to inject non-compliant MPEG streams (Audio,
   Video, and Systems) to overload the receiver/decoder's buffers,
   which might compromise the functionality of the receiver or even
   crash it. This is especially true for end-to-end systems like MPEG
   where the buffer models are precisely defined.

   MPEG-4 Systems supports stream types including commands that are
   executed on the terminal like OD commands, BIFS commands, etc. and
   programmatic content like MPEG-J (Java(TM) Byte Code) and
   ECMAScript. It is possible to use one or more of the above in a
   manner non-compliant to MPEG to crash or temporarily make the
   receiver unavailable.

   Authentication mechanisms can be used to validate of the sender and
   the data to prevent security problems due to non-compliant malignant
   MPEG-4 streams.

   A security model is defined in MPEG-4 Systems streams carrying MPEG-
   J access units which comprises Java(TM) classes and objects. MPEG-J
   defines a set of Java APIs and a secure execution model. MPEG-J
   content can call this set of APIs and Java(TM) methods from a set of
   Java packages supported in the receiver within the defined security
   model. According to this security model, downloaded byte code is
   forbidden to load libraries, define native methods, start programs,
   read or write files, or read system properties.

   Receivers can implement intelligent filters to validate the buffer
   requirements or parametric (OD, BIFS, etc.) or programmatic (MPEG-J,


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   ECMAScript) commands in the streams. However, this can increase the
   complexity significantly.

8. Acknowledgements

   This document evolved across several years through many revisions
   thanks to contributions from a large number of people since it is
   based on work within the IETF AVT working group and various ISO MPEG
   working groups, especially the 4-on-IP ad-hoc group. The authors
   wish to thank Olivier Avaro, Stephen Casner, Guido Fransceschini,
   Art Howarth, Dave Mackie, Dave Singer, and Stephan Wenger for their
   valuable comments and support. Attentive readers and early
   implementers also found flaws and bugs, thank you all.

9. References

   [1] ISO/IEC 14496-1:2001 MPEG-4 Systems

   [2] ISO/IEC 14496-2:2001 MPEG-4 Visual

   [3] ISO/IEC 14496-3:2001 MPEG-4 Audio

   [4] ISO/IEC 14496-6:2001 Delivery Multimedia Integration Framework.

   [5] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, RTP: A
   Transport Protocol for Real Time Applications, RFC 1889, Internet
   Engineering Task Force, January 1996.

   [6] S. Bradner, Key words for use in RFCs to Indicate Requirement
   Levels, RFC 2119, Internet Engineering Task Force, March 1997.

   [7] Y. Kikuchi, T. Nomura, S. Fukunaga, Y. Matsui, H. Kimata, RTP
   payload format for MPEG-4 Audio/Visual streams, Internet Engineering
   Task Force, RFC 3016.

   [8] B. Thompson, T. Koren, D. Wing, Tunneling multiplexed Compressed
   RTP ("TCRTP"), work in progress, draft-ietf-avt-tcrtp-04.txt, July
   2001.

   [9] D. Singer, Y Lim, A Framework for the delivery of MPEG-4 over
   IP-based Protocols, work in progress, draft-singer-mpeg4-ip-02.txt,
   May 2001.

   [10] M. Handley, V. Jacobson, SDP: Session Description Protocol, RFC
   2327, Internet Engineering Task Force, April 1998.

   [11] C.Roux & al, RTP Payload Format for MPEG-4 FlexMultiplexed
   Streams, work in progress, draft-curet-avt-rtp-mpeg4-flexmux-00.txt,
   February 2001.

   [12] H. Schulzrinne, RTP Profile for Audio and Video Conferences
   with Minimal Control, RFC 1890, Internet Engineering Task Force,
   January 1996.

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   [13] H. Schulzrinne, A. Rao, R. Lanphier, Real Time Streaming
   Protocol, RFC 2326, Internet Engineering Task Force, April 1998.

   [14] M. Handley, C. Perkins, E. Whelan, Session Announcement
   Protocol, RFC 2974, Internet Engineering Task Force, October 2000.


10. Authors' Addresses

   Andrea Basso
   AT&T Labs Research
   200 Laurel Avenue
   Middletown, NJ 07748
   USA
   e-mail: basso@research.att.com

   M. Reha Civanlar
   AT&T Labs - Research
   200 Laurel Ave. South, A5 4D04
   Middletown, NJ 07748
   USA
   e-mail: civanlar@research.att.com

   Philippe Gentric
   Philips Digital Networks, MP4Net
   51 rue Carnot
   92156 Suresnes
   France
   e-mail: philippe.gentric@philips.com

   Carsten Herpel
   THOMSON multimedia
   Karl-Wiechert-Allee 74
   30625 Hannover
   Germany
   e-mail: herpelc@thmulti.com

   Zvi Lifshitz
   Optibase Ltd.
   7 Shenkar St.
   Herzliya 46120
   Israel
   e-mail: zvil@optibase.com

   Young-Kwon Lim
   net&tv Co., Ltd.
   5th Floor Himart Building
   1007-46 Sadang-Dong Dongjak-Gu,
   Seoul, 156-090,
   Korea
   e-mail : young@netntv.co.kr


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   Colin Perkins
   USC Information Sciences Institute
   3811 N. Fairfax Drive suite 200
   Arlington, VA 22203
   USA
   e-mail : csp@isi.edu

   Jan van der Meer
   Philips Digital Networks
   Building WDB-1
   Prof Holstlaan 4
   5656 AA Eindhoven
   Netherlands
   e-mail : jan.vandermeer@philips.com

APPENDIX: Examples of usage

   This section describes a number of examples of how this payload
   format can be used either with or without the Sync Layer. In all
   examples the Sync Layer syntax is given (which shows how it may
   become invisible in cases 1,3,4 and 5).

   A C++-like syntax called SDL (Syntactic Description Language)
   defined in [1, section 14] is used to economically describe MPEG-4
   system data structures.

   These examples assume that the (a=fmtp) SDP syntax is used to convey
   the MIME parameters of the payload format.

Appendix.1 RFC 3016 compatible MPEG-4 Video (no SL)

   This is an example of a video stream compatible with RFC 3016.

SLConfigDescriptor

   In this example the SLConfigDescriptor is:

   class SLConfigDescriptor extends BaseDescriptor : bit(8)
   tag=SLConfigDescrTag {
    bit(8) predefined;
    if (predefined==0) {
     bit(1) useAccessUnitStartFlag; = 0
     bit(1) useAccessUnitEndFlag; = 1
     bit(1) useRandomAccessPointFlag; = 0
     bit(1) hasRandomAccessUnitsOnlyFlag; = 0
     bit(1) usePaddingFlag; = 0
     bit(1) useTimeStampsFlag; = 0
     bit(1) useIdleFlag; = 0
     bit(1) durationFlag; = 0
     bit(32) timeStampResolution; = 0
     bit(32) OCRResolution; = 0
     bit(8) timeStampLength; = 0
     bit(8) OCRLength; = 0

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     bit(8) AU_Length; = 0
     bit(8) instantBitrateLength; = 0
     bit(4) degradationPriorityLength; = 0
     bit(5) AU_seqNumLength; = 0
     bit(5) packetSeqNumLength; = 0
     bit(2) reserved=0b11;
    }
    if (durationFlag) {
     bit(32) timeScale; // NOT USED
     bit(16) accessUnitDuration;  // NOT USED
     bit(16) compositionUnitDuration;  // NOT USED
    }
    if (!useTimeStampsFlag) {
     bit(timeStampLength) startDecodingTimeStamp; = 0
     bit(timeStampLength) startCompositionTimeStamp; = 0
    }
   }

SL Packet Header structure

   With this configuration we have the following SL packet header
   structure:

   aligned(8) class SL_PacketHeader (SLConfigDescriptor SL) {
    if (SL.useAccessUnitEndFlag) {
     bit(1) accessUnitEndFlag; // 1 bit
    }
   }

   In this case this payload produces RTP packets that are exactly
   conformant to RFC 3016 and the SL is reduced to a purely logical
   construction that neither sender nor receiver need to implement.

Parameters

   This configuration is the default one; no parameters are required.

RTP packet structure

   Note that accessUnitEndFlag is mapped to the RTP header M bit.

   +=========================================+=============+
   | Field                                   |  size       |
   +=========================================+=============+
   | RTP header                              |    -        |
   +-----------------------------------------+-------------+
   | Access Unit or AU fragment              | 1400 octets |
   +-----------------------------------------+-------------+

Overhead

   In this example we have an RTP overhead of 40 octets for 1400 octets
   of payload i.e. 3 % overhead.

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Appendix.2 MPEG-4 Video with SL

   Let us consider the case of a 30 frames per second MPEG-4 video
   stream which bit rate is high enough that Access Units have to be
   split in several SL packets (typically above 300 kb/s).

   Let us assume also that the video codec generates in that case Video
   Packets suitable to fit in one SL packet i.e that the video codec is
   MTU aware and the MTU is 1500 octets. We assume furthermore that
   this stream contains B frames and that decodingTimeStamps are
   present.

SLConfigDescriptor

   In this example the SLConfigDescriptor is:

   class SLConfigDescriptor extends BaseDescriptor : bit(8)
   tag=SLConfigDescrTag {
    bit(8) predefined;
    if (predefined==0) {
     bit(1) useAccessUnitStartFlag; = 1
     bit(1) useAccessUnitEndFlag; = 0
     bit(1) useRandomAccessPointFlag; = 1
     bit(1) hasRandomAccessUnitsOnlyFlag; = 0
     bit(1) usePaddingFlag; = 0
     bit(1) useTimeStampsFlag; = 1
     bit(1) useIdleFlag; = 0
     bit(1) durationFlag; = 0
     bit(32) timeStampResolution; = 30
     bit(32) OCRResolution; = 0
     bit(8) timeStampLength; = 32
     bit(8) OCRLength; = 0
     bit(8) AU_Length; = 0
     bit(8) instantBitrateLength; = 0
     bit(4) degradationPriorityLength; = 0
     bit(5) AU_seqNumLength; = 0
     bit(5) packetSeqNumLength; = 0
     bit(2) reserved=0b11;
    }
    if (durationFlag) {
     bit(32) timeScale; // NOT USED
     bit(16) accessUnitDuration;  // NOT USED
     bit(16) compositionUnitDuration;  // NOT USED
    }
    if (!useTimeStampsFlag) {
     bit(timeStampLength) startDecodingTimeStamp; // NOT USED
     bit(timeStampLength) startCompositionTimeStamp; // NOT USED
    }
   }
   The useRandomAccessPointFlag is set so that the
   randomAccessPointFlag can indicate that the corresponding SL packet
   contains a GOV and the first Video Packet of an Intra coded frame.

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SL Packet Header structure
   With this configuration we have the following SL packet header
   structure:

   aligned(8) class SL_PacketHeader (SLConfigDescriptor SL) {
    bit(1) accessUnitStartFlag; // 1 bit
    if (accessUnitStartFlag) {
      bit(1) randomAccessPointFlag; // 1 bit
      bit(1) decodingTimeStampFlag; // 1 bit
      bit(1) compositionTimeStampFlag; // 1 bit
      if (decodingTimeStampFlag) {
         bit(SL.timeStampLength) decodingTimeStamp;
      }
      if (compositionTimeStampFlag) {
         bit(SL.timeStampLength) compositionTimeStamp;
      }
   }

Parameters
   decodingTimeStamps are encoded on 32 bits, which is much more than
   needed for delta. Therefore the sender will use DTSDeltaLength to
   signal that only 7 bits are used for the coding of relative DTS in
   the RTP packet.

   The RSLHSectionSize cannot exceed 4 (bits), which is encoded on 3
   bits and signaled by RSLHSectionSizeLength. The resulting
   concatenated fmtp line is:

   a=fmtp:<format> DTSDeltaLength=7;RSLHSectionSizeLength=3

RTP packet structure
   Two cases can occur; for packets that transport first fragments of
   Access Units we have:

   +=========================================+=============+
   | Field                                   |  size       |
   +=========================================+=============+
   | RTP header                              |    -        |
   +-----------------------------------------+-------------+
   | DTSFlag = (1)                           |  1 bit      |
   +-----------------------------------------+-------------+
   | DTSDelta                                |  7 bits     |
   +-----------------------------------------+-------------+
   | bits to octet alignment                 |  0 bits     |
   +-----------------------------------------+-------------+
   | RSLHSectionSize = (100)                 |  3 bits     |
   +-----------------------------------------+-------------+
   | accessUnitStartFlag = (1)               |  1 bit      |
   +-----------------------------------------+-------------+
   | randomAccessPointFlag                   |  1 bit      |
   +-----------------------------------------+-------------+
   | decodingTimeStampFlag                   |  1 bit      |

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   +-----------------------------------------+-------------+
   | compositionTimeStampFlag                |  1 bit      |
   +-----------------------------------------+-------------+
   | bits to octet alignment =(0)            |  1 bit      |
   +-----------------------------------------+-------------+
   | SL packet payload                       |  N octets   |
   +-----------------------------------------+-------------+


   For packets that transport non-first fragments of Access Units we
   have:

   +=========================================+=============+
   | Field                                   |  size       |
   +=========================================+=============+
   | RTP header                              |    -        |
   +-----------------------------------------+-------------+
   | DTSFlag = 0                             |  1 bit      |
   +-----------------------------------------+-------------+
   | bits to octet alignment = (0000000)     |  7 bits     |
   +-----------------------------------------+-------------+
   | RSLHSectionSize = (001)                 |  3 bits     |
   +-----------------------------------------+-------------+
   | accessUnitStartFlag = (0)               |  1 bit      |
   +-----------------------------------------+-------------+
   | bits to octet alignment = (0000)        |  4 bits     |
   +-----------------------------------------+-------------+
   | SL packet payload                       |  N octets   |
   +-----------------------------------------+-------------+

Overhead estimation

   In this example we have a RTP overhead of 40 + 2 octets for 1400
   octets of payload i.e. 3 % overhead.

Appendix.3 Low delay MPEG-4 Audio (no SL)

   This example is for a low delay audio service. For this reason a
   single Access Unit is transported in each RTP packet (in terms of
   Sync Layer each SL packet contains a complete Access Unit).

SLConfigDescriptor

   Since CTS=DTS and Access Unit duration is constant, signaling of
   MPEG-4 time stamps is not needed (the durationFlag of SLConfig is
   set).

   We also assume here an audio Object Type for which all Access Units
   are Random Access Points, which is signaled using the
   hasRandomAccessUnitsOnlyFlag in the SLConfigDescriptor.

   We assume furthermore a mode where the Access Unit size is constant
   and equal to 5 octets (which is signaled with AU_Length).

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   In this example the SLConfigDescriptor is:

   class SLConfigDescriptor extends BaseDescriptor : bit(8)
   tag=SLConfigDescrTag {
    bit(8) predefined;
    if (predefined==0) {
     bit(1) useAccessUnitStartFlag; = 0
     bit(1) useAccessUnitEndFlag; = 0
     bit(1) useRandomAccessPointFlag; = 0
     bit(1) hasRandomAccessUnitsOnlyFlag; = 1
     bit(1) usePaddingFlag; = 0
     bit(1) useTimeStampsFlag; = 0
     bit(1) useIdleFlag; = 0
     bit(1) durationFlag; = 1 // signals constant AU duration
     bit(32) timeStampResolution; = 0
     bit(32) OCRResolution; = 0
     bit(8) timeStampLength; = 0
     bit(8) OCRLength; = 0
     bit(8) AU_Length; = 5
     bit(8) instantBitrateLength; = 0
     bit(4) degradationPriorityLength; = 0
     bit(5) AU_seqNumLength; = 0
     bit(5) packetSeqNumLength; = 0
     bit(2) reserved=0b11;
    }
    if (durationFlag) {
     bit(32) timeScale; = 1000 // for milliseconds
     bit(16) accessUnitDuration; = 10 // ms
     bit(16) compositionUnitDuration; = 10 // ms
    }
    if (!useTimeStampsFlag) {
     bit(timeStampLength) startDecodingTimeStamp; = 0
     bit(timeStampLength) startCompositionTimeStamp; = 0
    }
   }

SL packet header

   With this configuration the SL packet header is empty. The Sync
   Layer is reduced to a purely logical construction that neither
   sender nor receiver need to implement.

Parameters

   No parameters are required.

RTP packet structure

   Note that the RTP header M bit must be set to 1.

   +=========================================+=============+
   | Field                                   |  size       |

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   +=========================================+=============+
   | RTP header                              |    -        |
   +-----------------------------------------+-------------+
   | Access Unit                             |  5 octets   |
   +-----------------------------------------+-------------+


Overhead estimation

   The overhead is extremely large i.e. more than 800 %, since 40
   octets of headers are required to transport 5 octets of data. Note
   however that RTP header compression would work well since time
   stamps increments are constant.

Appendix.4 Media delivery MPEG-4 Audio (no SL)

   This example is for a media delivery service where delay is not an
   issue but efficiency is. In this case several Access Units are
   transported in each RTP packet.

SLConfigDescriptor

   Similar to previous example.

SL packet header

   With this configuration the SL packet header is empty. The Sync
   Layer is reduced to a purely logical construction that neither
   sender nor receiver need to implement.

Parameters

   The absence of RSLHSectionSizeLength indicates that the RSLHSection
   is empty.

   The size of SL Packets (which are all complete Access Units in this
   case) is constant and is indicated  with:
   a=fmtp:<format> ConstantSize=5

   This also indicates to the receiver that the "Multiple" packing
   style will be used, the 2 octets field that would give the size of
   the Payload Header Section is ommited since in this case this field
   always contains zero (the Payload Header Section is always empty due
   to the absence of any other MIME parameter).

RTP packet structure
   Note that the RTP header M bit is always set to 1, which indicates
   to the receiver that only complete Access Units are transported.

   +=========================================+=============+
   | Field                                   |  size       |
   +=========================================+=============+
   | RTP header                              |    -        |

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   +-----------------------------------------+-------------+
   | Access Unit data                        |  5 octets   |
   +-----------------------------------------+-------------+
   | Access Unit data                        |  5 octets   |
   +-----------------------------------------+-------------+
   | etc, until MTU is reached                             |
   +-----------------------------------------+-------------+
   | Access Unit data                        |  5 octets   |
   +-----------------------------------------+-------------+

Overhead estimation

   The overhead is 3% i.e. minimal.


Appendix.5 AAC with interleaving (no SL)

   Let us consider AAC at 128 kb/s where each Access Unit is in the
   average 320 octets. Interleaving is applied using a continuous
   interleaving scheme (see table below) where 4 Access Units are used
   to construct each RTP packet in order to match a MTU of 1500 octets.

   IndexDelta is constant and equal to 2 (since +1 is automatically
   added); it is encoded on 2 bits.

   As explained in section 3.8 this is a time stamp based interleaving
   (TSBI) scheme (IndexLength=0); indeed receivers know that each
   payload is a complete Access Unit because all RTP packets have the M
   bit set to 1 and therefore, since Access Unit duration is constant,
   Access Unit timestamps can be computed from RTP timestamps and
   IndexDelta values; this can be used for de-interleaving even in case
   of losses.
   Note that it is  also be possible to use IndexLength=2 so as to
   maintain a octet alignement in the Payload Header portions; in this
   case however the value of these two bits MUST be zero as stated in
   3.8.1. This solution is used in the companion RFC YYYY.

   +-----------------------------------------------------------------+
   | RTP packet | RTP Timestamp |    Aus          |    IndexDelta    |
   +-----------------------------------------------------------------+
   |    1       |   CTS(AU1)    |             1   |  -               |
   +-----------------------------------------------------------------+
   |    2       |   CTS(AU2)    |          2, 5   |  -,2             |
   +-----------------------------------------------------------------+
   |    3       |   CTS(AU3)    |       3, 6, 9   |  -,2,2           |
   +-----------------------------------------------------------------+
   |    4       |   CTS(AU4)    |    4, 7,10,13   |  -,2,2,2         |
   +-----------------------------------------------------------------+
   |    5       |   CTS(AU8)    |    8,11,14,17   |  -,2,2,2         |
   +-----------------------------------------------------------------+
   |    6       |   CTS(AU12)   |   12,15,18,21   |  -,2,2,2         |
   +-----------------------------------------------------------------+
   |    7       |   CTS(AU16)   |   16,19,22,25   |  -,2,2,2         |

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   +----------------------------------------------------------------+
   |    8       |   CTS(AU20)   |   20,23,26,29   |  -,2,2,2         |
   +-----------------------------------------------------------------+
   |    9       |   CTS(AU24)   |   24,27,30,33   |  -,2,2,2         |
   +-----------------------------------------------------------------+
   |    10      |   CTS(AU28)   |   28,31,34,37   |  -,2,2,2         |
   +-----------------------------------------------------------------+
   |                              etc                                |
   +-----------------------------------------------------------------+

SLConfigDescriptor

   Similar to previous example.

SL Packet Header

   Similar to previous example (empty).

Parameters

   The resulting concatenated fmtp line is:

   a=fmtp:<format> SizeLength=9; IndexDeltaLength=2;

RTP packet structure

   +=========================================+=============+
   | Field                                   |  size       |
   +=========================================+=============+
   | RTP header                              |    -        |
   +-----------------------------------------+-------------+
                      Payload Header Section
   +=========================================+=============+
   | PayloadHeaderSection size = (42)        |  2 octets   |
   +-----------------------------------------+-------------+
   | PayloadSize                             |  9 bits     |
   +-----------------------------------------+-------------+
   | PayloadSize                             |  9 bits     |
   +-----------------------------------------+-------------+
   | IndexDelta                              |  2 bits     |
   +-----------------------------------------+-------------+
   | PayloadSize                             |  9 bits     |
   +-----------------------------------------+-------------+
   | IndexDelta                              |  2 bits     |
   +-----------------------------------------+-------------+
   | PayloadSize                             |  9 bits     |
   +-----------------------------------------+-------------+
   | IndexDelta                              |  2 bits     |
   +-----------------------------------------+-------------+
   | bits to octet alignment = (000000)      |  6 bits     |
   +-----------------------------------------+-------------+
                         Payload Section
   +=========================================+=============+

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   | AAC Access Unit                         |   x octets  |
   +-----------------------------------------+-------------+
   | AAC Access Unit                         |   x octets  |
   +-----------------------------------------+-------------+
   | AAC Access Unit                         |   x octets  |
   +-----------------------------------------+-------------+
   | AAC Access Unit                         |   x octets  |
   +-----------------------------------------+-------------+


Overhead estimation

   The PayloadHeaderSection is 8 octets; in this example we have
   therefore a RTP overhead of 40 + 8 octets for 1400 octets (approx)
   of payload i.e. around 4 % overhead.



Appendix.6 AAC with Index-based interleaving and SL

   Let us consider AAC around 130 kb/s where each Access Unit is split
   in 4 SL packets corresponding to Error Sensitivity Categories (ESC)
   of maximum 90 octets for which interleaving is very useful in terms
   of error resilience. We thus use an interleaving scheme where 15 SL
   Packets (extracted from 15 consecutive Access Units) are used to
   construct each RTP packet in order to match a MTU of 1500 octets.
   Note that since ESC fragments are not octet aligned we also use the
   paddingFlag and paddingBits features of the Sync Layer. The
   interleaving sequence is 4 RTP packets and 350 ms long, which is too
   long for conferencing but perfectly OK for Internet radio.

   Since the sequence contains 60 SL packets, IndexLength is set to 16
   bits so as to provide a safe margin in case of long loss bursts.
   This will also indicate to the receiver that this is a Index-Based-
   Interleaving scheme (and indeed CTS cannot be computed for SL
   packets that are not AU starts so TSBI would not work).

   2 bits are enough for IndexDelta, which is constant and equal to 3
   (since +1 is automatically added).

   Note that the 4th RTP packet in each sequence has its M bit set to 1
   since it contains 15 SL packets transporting the end of 15
   consecutive Access Units.

   With this scheme a sender (for example upon reception of RTCP
   reports indicating high loss rates) can (for example) choose to
   duplicate for each interleaving sequence the first RTP packet that
   contains the most useful data in terms of ESC or apply other error
   protection techniques, with due care to congestion issues.

   In this example we will also show several other SL features (OCR, AU
   boundary flags, padding, as detailed below).


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   One feature demonstrated by this example is the degradation
   priority. We assume degradation priority can take 4 different
   values, mapped to Error Sensitivity Categories, and is encoded on 2
   bits. This interleaving scheme makes sure that only SL packets of
   identical degradation priorities are grouped in the same RTP packet
   (3.6.3) and that only the first RSLH of each RTP packet transports
   the degradation priority. We also assume that for each last SL
   packet of each RTP packet the server inserts an OCR.

SLConfigDescriptor
   In this example the SLConfigDescriptor is:
   class SLConfigDescriptor extends BaseDescriptor : bit(8)
   tag=SLConfigDescrTag {
    bit(8) predefined;
    if (predefined==0) {
     bit(1) useAccessUnitStartFlag; = 1
     bit(1) useAccessUnitEndFlag; = 1
     bit(1) useRandomAccessPointFlag; = 0
     bit(1) hasRandomAccessUnitsOnlyFlag; = 1
     bit(1) usePaddingFlag; = 1 // we need to signal padding bits
     bit(1) useTimeStampsFlag; = 0
     bit(1) useIdleFlag; = 0
     bit(1) durationFlag; = 1
     bit(32) timeStampResolution; = 0
     bit(32) OCRResolution; = 30
     bit(8) timeStampLength; = 0
     bit(8) OCRLength; = 32
     bit(8) AU_Length; = 0
     bit(8) instantBitrateLength; = 0
     bit(4) degradationPriorityLength; = 2
     bit(5) AU_seqNumLength; = 0
     bit(5) packetSeqNumLength; = 6
     bit(2) reserved=0b11;
    }
    if (durationFlag) {
     bit(32) timeScale; = 1000// milliseconds
     bit(16) accessUnitDuration; = 23.22 // ms
     bit(16) compositionUnitDuration; = 23.22 // ms
    }
    if (!useTimeStampsFlag) {
     bit(timeStampLength) startDecodingTimeStamp; = 0
     bit(timeStampLength) startCompositionTimeStamp; = 0
    }
   }

SL Packet Header structure
   With this configuration we have the following SL packet header
   structure:
   aligned(8) class SL_PacketHeader (SLConfigDescriptor SL) {
    bit(1) accessUnitStartFlag;
    bit(1) accessUnitEndFlag;
    bit(1) OCRflag;
    bit(1) paddingFlag;

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    if (paddingFlag) bit(3) paddingBits;
    bit(SL.packetSeqNumLength) packetSequenceNumber;
    bit(1) DegPrioflag;
    if (DegPrioflag) {
     bit(SL.degradationPriorityLength) degradationPriority;}
    if (OCRflag) {
     bit(SL.OCRLength) objectClockReference;}
    }
   }

Parameters
   The resulting concatenated fmtp line is:
   a=fmtp:<format> SizeLength=7; RSLHSectionSizeLength=8;
   IndexLength=16; IndexDeltaLength=2; OCRDeltaLength=16

RTP packet structure
   +=========================================+=============+
   | Field                                   |  size       |
   +=========================================+=============+
   | RTP header                              |    -        |
   +-----------------------------------------+-------------+
                       Payload Header Section
   +=========================================+=============+
   | Payload Header Section size = 149 bits  |  2 octets   |
   +-----------------------------------------+-------------+
   | PayloadSize                             |  7 bits     |
   +-----------------------------------------+-------------+
   | Index                                   |  16 bits    |
   +-----------------------------------------+-------------+
   | PayloadSize                             |  7 bits     |
   +-----------------------------------------+-------------+
   | IndexDelta = (11)                       |  2 bits     |
   +-----------------------------------------+-------------+
   |            etc + 12 times 9 bits                      |
   +-----------------------------------------+-------------+
   | PayloadSize                             |  7 bits     |
   +-----------------------------------------+-------------+
   | IndexDelta = (11)                       |  2 bits     |
   +-----------------------------------------+-------------+
   | bits to octet alignment = (000)         |  3 bits     |
   +-----------------------------------------+-------------+
                         RSLHSection
   +=========================================+=============+
   | RSLHSectionSize =  (10000111)           |  8 bits     |
   +-----------------------------------------+-------------+
   | accessUnitStartFlag                     |  1 bit      |
   +-----------------------------------------+-------------+
   | accessUnitEndFlag                       |  1 bit      |
   +-----------------------------------------+-------------+
   | OCRFlag = (0)                           |  1 bit      |
   +-----------------------------------------+-------------+
   | paddingFlag = (1)                       |  1 bit      |
   +-----------------------------------------+-------------+

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   | paddingBits                             |  3 bits     |
   +-----------------------------------------+-------------+
   | DegPrioflag = (1)                       |  1 bit      |
   +-----------------------------------------+-------------+
   | degradationPriority                     |  2 bits     |
   +-----------------------------------------+-------------+
   | accessUnitStartFlag                     |  1 bit      |
   +-----------------------------------------+-------------+
   | accessUnitEndFlag                       |  1 bit      |
   +-----------------------------------------+-------------+
   | OCRFlag = (0)                           |  1 bit      |
   +-----------------------------------------+-------------+
   | paddingFlag = (1)                       |  1 bit      |
   +-----------------------------------------+-------------+
   | paddingBits                             |  3 bits     |
   +-----------------------------------------+-------------+
   | DegPrioflag = (0)                       |  1 bit      |
   +-----------------------------------------+-------------+
   |              etc + 12 times 8 bits                    |
   +-----------------------------------------+-------------+
   | accessUnitStartFlag                     |  1 bit      |
   +-----------------------------------------+-------------+
   | accessUnitEndFlag                       |  1 bit      |
   +-----------------------------------------+-------------+
   | OCRFlag = (1)                           |  1 bit      |
   +-----------------------------------------+-------------+
   | OCRDelta                                |  16 bits    |
   +-----------------------------------------+-------------+
   | paddingFlag = (0)                       |  1 bit      |
   +-----------------------------------------+-------------+
   | DegPrioflag = (0)                       |  1 bit      |
   +-----------------------------------------+-------------+
   | bits to octet alignment = (000)         |  3 bits     |
   +-----------------------------------------+-------------+
                         Payload Section
   +=========================================+=============+
   | SL packet payload                       |max 90 octets|
   +-----------------------------------------+-------------+
   |             etc + 13  SL packets                      |
   +-----------------------------------------+-------------+
   | SL packet payload                       |max 90 octets|
   +-----------------------------------------+-------------+

   Note that in the above table the last SL packet in the RTP packet
   has a payload that is octet-aligned (at the end). When this happens
   paddingFlag is set to zero and the paddingBits field is omitted.

Overhead estimation

   The PayloadHeaderSection is 19 octets, the RSLHSection is 16 octets;
   in this example we have therefore a RTP overhead of 40 + 35 octets
   for 1350 octets of payload i.e. around 6 % overhead.


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