Subject: [AVT] draft-ietf-avt-mpeg4-simple-00.txt for review

From: [email-masked]
Date: Fri, 21 Sep 2001 16:04:42 +0200

Dear all,

Please find attached for your review draft-ietf-avt-mpeg4-simple-00.txt. The
document is an update of draft-vandermeer-mpeg-4-simple-00.txt and of
draft-vandermeer-mpeg-4-simple-01.txt; the latter was distributed on the
reflector, but not uploaded. The new name of the document is
draft-ietf-avt-mpeg4-simple-00.txt, as agreed in the London IETF meeting.
The document incorporates comments from IETF, MPEG and ISMA.

As in the previous draft there are three configurations with four use cases.
The main difference with the previous draft is that the mode parameter
signals use cases instead of configurations. The four modes are :
- mode=CELP-cbr, used for constant bit rate CELP
- mode=CELP-vbr, used for variable bit rate CELP
- mode=AAC-lbr, used for AAC at low bitrates
- mode=AAC-hbr, used for AAC at higher bitrates
The payload configuration in the variable bit rate CELP mode is the same as
in the low bitrates AAC mode.

Another important difference is that there is now a statement that "For
interoperability reasons, applications that transport MPEG-4 video over RTP
SHOULD use the payload format and associated names and parameters defined in
RFC 3016 if the functionality provided by RFC 3016 can meet the requirements
of that application." In addition there are several more or less editorial
improvements. The document is submitted for uploaded.

Your review is highly appreciated; any comment you may have is very welcome.

Kind regards,

Jan van der Meer

************************
Jan van der Meer
Philips - Digital Networks - MP4Net
Building WDB-140
P.O. Box 80021
5600 JZ Eindhoven
The Netherlands
Phone +31 402 735 774
Fax +31 402 735 545
Email [email-masked]
************************
Internet Engineering Task Force                        J. van der Meer
Internet Draft                                     Philips Electronics
                                                             D. Mackie
                                                    Cisco Systems Inc.
                                                        V. Swaminathan
                                                 Sun Microsystems Inc.
                                                             D. Singer
                                                        Apple Computer

                                                         September 2001
                                                     Expires March 2002

   Document: draft-ietf-avt-mpeg4-simple-00.txt

   Use of "RFC-generic" for MPEG-4 Elementary Streams with no SL layer

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."

   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.

0 Abstract

   The MPEG Committee (ISO/IEC JTC1/SC29 WG11) is a working group in ISO 
   that recently produced the MPEG-4 [1] standard. MPEG defines tools to 
   compress content such as audio-visual information into elementary 
   streams. In [6] a generic RTP payload format is defined for transport 
   of any non-multiplexed MPEG-4 elementary stream. To achieve the generic 
   MPEG-4 functionality, [6] addresses detailed issues related to the 
   MPEG-4 SL layer. However, many initial applications will not use the SL 
   Layer. To facilitate usage of [6] by such applications, this document 
   describes how to use [6] when no SL layer is used. 

   This specification is a product of the Audio/Video Transport working
   group within the Internet Engineering Task Force. Comments are 
   solicited and should be addressed to the working group's mailing 
   list at [email-masked] and/or the authors.

1. Introduction

   The MPEG Committee is Working Group 11 (WG11) in ISO/IEC JTC1 SC29 
   that specified the MPEG-1, MPEG-2 and, more recently, the MPEG-4 
   standards [1]. The MPEG-4 standard specifies compression of 
   audio-visual data into for example an audio or video elementary 
   stream. In the MPEG-4 standard, these streams take the form of 
   audiovisual objects that may be arranged into an audio-visual scene 
   by means of a scene description. Each MPEG-4 elementary stream 
   consists of a sequence of Access Units; in case of audio an Access 
   Unit (AU) is an audio frame and in case of video a picture. 

   The MPEG-4 system specification is a rather abstract specification in 
   the sense that no transport format for MPEG-4 elementary streams is 
   defined. Instead, a conceptual SL layer has been specified to store 
   transport specific information such as time stamps and random access 
   point information. When transporting an MPEG-4 elementary stream, 
   transport information from the SL layer is typically mapped to the 
   actual transport layer. Note however that the SL layer is conceptual 
   and may not exist in practice.

   In [6], a general payload format is defined for transport of a single 
   MPEG-4 elementary stream over RTP. The RTP payload format specified 
   in [6] allows for carriage of any information that may be contained in 
   the MPEG-4 SL layer, either by mapping to the RTP header fields or by 
   carriage in specific fields defined in the RTP payload. Consequently, 
   the format defined in [6] is very generic and complete; for example, 
   transcoding issues from and to the SL layer are described in detail. 
   
   However, in many initial MPEG-4 applications the SL layer does not 
   exist in practice. Such applications do not require any knowledge of 
   the SL layer. While the use of [6] is highly desirable for all MPEG-4 
   applications, to understand [6] may be difficult without knowledge of 
   the MPEG-4 SL layer. Therefore in this document the use of [6] is 
   described without requiring knowledge of the SL layer to understand 
   its functionality.

   Sophisticated features on interleaving of fragmented Access Units are 
   defined in [6]. Because initial applications do not require these 
   complicated features, these features are not supported in this 
   document. Hence, only a functional subset of [6] is supported. 

   In [6], a general and configurable payload structure is defined for 
   transport of MPEG-4 streams. This allows for the design of receivers 
   that can be configured to receive any MPEG-4 stream. Configuration of 
   the payload is provided to accommodate transport of any MPEG-4 stream, 
   but for a specific MPEG-4 elementary stream typically only very few 
   configurations are needed. So as to allow for the design of simplified, 
   but dedicated receivers, this specifications requires that specific 
   modes are defined for transport of MPEG-4 streams. In this document 
   only modes are defined for transport of MPEG-4 CELP and AAC streams, 
   but in future new RFCs are expected to specify additional modes for 
   transport of other MPEG-4 streams. 

   In summary, this document:
   - is intended for applications that do not apply the SL layer;
   - describes how to use [6] without requiring knowledge of the SL layer;
   - defines a functional but true subset of [6]; 
   - defines modes how to use this specification for transport of MPEG-4
     CELP and AAC streams.
   
   The use of [6] defined in this document is simple to implement and 
   reasonably efficient. It allows for optional interleaving of Access 
   Units (such as audio frames) to increase error resiliency in packet 
   loss.

   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 [3].

2. Carriage of MPEG-4 elementary streams over RTP 

2.1 MPEG-4 stream type identification

   Information on the type of MPEG-4 stream that is carried in the 
   payload is conveyed by format parameters in an SDP message or by other 
   means.

2.2 MPEG Access Units

   For carriage of compressed audio-visual data MPEG defines Access 
   Units. An MPEG Access Unit (AU) is the smallest data entity to which 
   timing information can be attributed. In case of audio an Access 
   Unit represents an audio frame and in case of video a picture. MPEG 
   Access Units are by definition byte aligned. If for example an audio 
   frame is not byte aligned, up to 7 zero-padding bits MUST be inserted 
   at the end of the frame to achieve a byte-aligned Access Unit. 
   Decoders MUST be able to decode AUs in which such padding is applied.

   Consistent with the MPEG-4 specification, this document requires that 
   each MPEG-4 video Access Unit includes all the coded data of a 
   picture, any video stream headers that may precede the coded picture 
   data, and any video stream stuffing that may follow it, up to, but not 
   including the startcode indicating the start of a new video stream or 
   the next Access Unit.

2.3 Concatenation of Access Units

   Frequently it is possible to carry multiple Access Units in one RTP 
   packet. This is particularly useful for audio; for example, when AAC 
   is used for encoding of a stereo signal at 64 kbits/sec, AAC frames 
   contain on average approximately 200 bytes. On a LAN with a 1500 byte 
   MTU this would allow on average 7 complete AAC frames to be carried 
   per AAC packet.

   Access Units may have a fixed size in bytes, but a variable size is 
   also possible. To facilitate parsing in case of multiple concatenated 
   AUs in one RTP packet, the size of each AU is made known to the 
   receiver. When concatenating in case of a constant AU size, this size 
   is communicated through a format parameter. When concatenating in case 
   of variable size AUs, the RTP payload carries an AU size field for 
   each contained AU. In combination with the RTP payload length the 
   size information allows the RTP payload to be split by the receiver 
   back into the individual AUs.

   To simplify the implementation of [6] defined in this document, it 
   is required that when multiple AUs are carried in an RTP packet, that 
   each AU MUST be complete, i.e. the number of AUs in an RTP packet 
   MUST be integral.

2.4 Fragmentation of Access Units

   MPEG allows for very large Access Units. Since most IP networks have 
   significantly smaller MTU's, this payload format allows to fragment 
   the AUs over multiple RTP packets so as to avoid IP layer 
   fragmentation. To simplify the implementation of [6] defined in this 
   document, an RTP packet SHALL either carry one or more complete 
   Access Units or a single fragment of one Access Unit. 

2.5 Interleaving

   When an RTP packet carries a contiguous sequence of Access Units, 
   the loss of such packet can result in "decoding gaps" for the user. 
   One method to alleviate this problem is to allow for the Access 
   Units to be interleaved in the RTP packets. For a modest cost in 
   latency and implementation complexity, significant error resiliency 
   to packet loss can be achieved. 

   To support optional interleaving of Access Units, this payload 
   format allows for index information to be sent for each Access Unit. 
   The RTP sender is free to choose the interleaving pattern without 
   propagating this information to the receiver(s). Indeed the sender 
   could dynamically adjust the interleaving pattern based on the 
   Access Unit size, error rates, etc. The RTP receiver does not need 
   to know the interleaving pattern used, it only need extract the
   index information of the Access Unit and insert the Access Unit into 
   the appropriate sequence in the rendering queue. An example of 
   interleaving is given below. 

   Assume that an RTP packet contains 3 AUs, and that the AUs are 
   numbered 1, 2, 3, 4, etc. If an interleaving group length of 9 is 
   chosen, then RTP packet(i) contain the following AU(n):
   RTP packet(1):  AU(1),  AU(4),  AU(7)
   RTP packet(2):  AU(2),  AU(5),  AU(8)
   RTP packet(3):  AU(3),  AU(6),  AU(9)
   RTP packet(4):  AU(10), AU(13), AU(16)
   RTP packet(5):  AU(11), AU(14), AU(17)
   Etc.

2.6 Time stamp information

   MPEG-4 defines two type of time stamps, the decoding time stamp DTS 
   and the composition time stamp CTS. The RTP timestamp is equivalent 
   to the composition time stamp.

   The RTP time stamp MUST carry the sampling instance of the first AU 
   (fragment) in the RTP packet. When multiple AUs are carried within 
   an RTP packet, the time stamps of subsequent AUs can be calculated 
   if the frame period of each AU is known. For audio and video this 
   is possible if the frame rate is constant. However, in some cases it
   is not possible to make such calculation, for example for variable 
   frame rate video and for MPEG-4 BIFS streams carrying composition 
   information. To support such cases, this payload format can be 
   configured to carry a CTS in the RTP payload for each contained 
   Access Unit. A CTS time stamp MAY be conveyed in the RTP payload 
   only for non-first AUs in the RTP packet, and SHALL NOT be conveyed 
   for the first AU (fragment), as the time stamp for the latter is 
   carried by the RTP time stamp. 

   The DTS timestamp is applied only in MPEG video streams that use 
   bi-directional coding, i.e. when pictures may be predicted in both 
   forward and backward direction by using either a reference picture in 
   the past, or a reference picture in the future. The DTS cannot be 
   carried in the RTP header. In some cases the DTS can be derived from 
   the RTP time stamp using frame rate information; this requires deep 
   parsing in the video stream, which may be considered objectionable. 
   But if the video frame rate is variable, the required information 
   is not even present in the video stream. For both reasons, the 
   capability has been defined to optionally carry a DTS in the RTP 
   payload for each contained Access Unit.

   Since RTP time stamps may be re-stamped by RTP devices, each CTS 
   and DTS contained in the RTP payload is coded differentially from the 
   RTP time stamp, so as to avoid extensive parsing by re-stamping 
   devices. 

2.7 Carriage of auxiliary information.

   This payload format defines a specific field to carry auxiliary data 
   on the contained MPEG-4 stream, representing MPEG-4 system information. 
   The auxiliary data corresponds to the RSLH field defined in [6]. 
   Receivers MAY use the auxiliary data to decode the contained stream, 
   but receivers that have no interest in such data MAY skip the 
   auxiliary data field. To facilitate skipping of the data, and to avoid 
   the need for parsing it, the auxiliary data field is preceded by a 
   field that specifies the length of the auxiliary data.

2.8 Format parameters and the conditional presence and length of fields

   To support the features described in the previous sections several 
   fields are defined for carriage in the RTP payload. However, their use 
   strongly depends on the type of MPEG-4 elementary stream that is 
   carried. Sometimes a specific field is needed with a certain length, 
   while in other cases such field is not needed at all. To be efficient 
   in either case, the fields needed for these features are configurable 
   by means of format parameters. In general, a format parameter defines 
   the presence and length of associated fields. A length of zero 
   indicates absence of the field. As a consequence, parsing of the 
   payload requires knowledge of format parameters. The format 
   parameters are conveyed to the receiver via SDP messages or through 
   other means.

2.9 Global structure of payload format

   The payload structure in [6] is described in terms derived from the 
   SL layer. In this document exactly the same structure is described 
   in more general terms, so as to improve the readability for people 
   with no knowledge of the SL layer. So the payload structure described 
   below corresponds on bit level exactly to the payload structure 
   defined in [6].

   The RTP payload following the RTP header, contains three byte aligned 
   data sections, of which the first two MAY be empty. See figure 1.

          +---------+-----------+-----------+---------------+
          | RTP     | AU Header | Auxiliary | Access Unit   |
          | Header  | Section   | Section   | Data Section  |
          +---------+-----------+-----------+---------------+

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

   Figure 1: Data sections within an RTP packet

   The first data section is the AU (Access Unit) Header Section, that 
   contains one or more AU-headers; however, each AU-header MAY be empty, 
   in which case the entire AU Header Section is empty. The second 
   section is the Auxiliary Section, containing auxiliary data; also 
   this section MAY be configured empty. The third section is the Access 
   Unit Data Section, containing either a single fragment of one Access 
   Unit or one or more complete Access Units. The Access Unit Data 
   Section is never empty.

   When compared to the terms used in [6], the AU Header Section exactly 
   corresponds to the MSLHSection, the Auxiliary Section to the 
   RSLHSection, and the Access Unit Data Section to the SLPPSection. 

2.10 Modes to transport MPEG-4 streams

   While it is possible to build fully configurable receivers capable of 
   receiving any MPEG-4 stream, this specification also allows for the 
   design of simplified, but dedicated receivers, that are capable for 
   example to receive only one type of MPEG-4 stream. This is achieved by 
   requiring that specific modes be defined for using this specification. 
   Each mode defines how to transport specific MPEG-4 streams, for example 
   by defining suitable constraints or payload configurations. Modes can 
   be defined as deemed appropriate. However, each mode MUST be in full 
   compliance with this specification.

   The applied mode MUST be signalled. Signalling the mode is particularly 
   important for receivers that are only capable of decoding a particular 
   mode. Such receivers need to determine whether that particular mode is 
   applied, so as to avoid problems with processing of payloads that are 
   beyond the capabilities of the receiver. 

   In this internet draft only modes are defined for transport of MPEG-4 
   CELP and AAC streams. However, in future new RFCs are expected to 
   specify additional modes of using this specification for transport of 
   other MPEG-4 streams. 

2.11 Alignment with "RFC-generic" and RFC 3016

   This document defines a subset of the "RTP payload format for MPEG-4 
   streams" [6]. The main characteristic of this subset is that each RTP 
   payload is only allowed to contain either a single fragment of one 
   Access Unit or one or more complete Access Units. Obviously, RTP 
   payloads that apply this subset in conformance with this document 
   conform also to [6]. Receivers that comply with [6] are able to decode 
   MPEG-4 streams carried in compliance with this document. 

   Receivers designed to only comply to this document may not be able to 
   decode a RTP payload that conforms to [6] but not to this document.
   Such receivers may also not be capable of exploiting some of features 
   of the SL layer supported in [6], such as knowledge of AU-start, random 
   access information and other information carried in the SL header, but 
   not described in this document. 

   Furthermore, this payload can be configured to be identical to the 
   payload format defined in RFC 3016 for the MPEG-4 video configurations 
   recommended in RFC 3016. Hence, receivers that comply with RFC 3016 
   can decode such RTP payload. Vice versa, receivers that comply with the 
   specification in this document SHOULD be able to decode payloads, names 
   and parameters defined for MPEG-4 video in RFC 3016.

   For interoperability reasons, applications that transport MPEG-4 video 
   over RTP SHOULD use the payload format and associated names and 
   parameters defined in RFC 3016 if the functionality provided by RFC 3016 
   can meet the requirements of that application.

3 Payload Format

3.1 RTP Header Fields Usage

   Payload Type (PT): The assignment of an RTP payload type for this
   RTP 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 to indicate that the RTP packet 
   payload includes the end of each Access Unit of which data is 
   contained in this RTP packet. As the payload either carries one or 
   more complete Access Units or a single fragment of an Access Unit, 
   the M is always set to set to 1, except when the packet carries a 
   single fragment of an Access Unit that is not the last one.  

   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: Indicates the sampling instance of the first AU contained 
   in the RTP payload. This sampling instance is equivalent to the CTS 
   in the MPEG-4 time domain. The clock rate of the RTP time stamp MAY 
   be expressed as part of the RTPMAP. If an audio or video stream with 
   a fixed frame rate is transported, the rate SHOULD be set to the same 
   value as the sampling frequency of the audio or video frames (number 
   of samples per second). 
   In all cases, the sender SHALL make sure that RTP time stamps
   are identical only if the RTP time stamp refers to fragments of the
   same Access Unit.
   According to RFC 1889 [2] (section 5.1), RTP timestamps are 
   recommended to start at a random value for security reasons. However, 
   then a receiver is, in the general case, not able to reconstruct the 
   original MPEG Time Stamps, which creates problems for applications 
   where streams from multiple sources are to be synchronized. Therefore 
   the usage of a random offset SHOULD be avoided.

   SSRC: set as described in RFC1889 [2]. 

   CC and CSRC fields are used as described in RFC 1889 [2].

   RTCP SHOULD be used as defined in RFC 1889 [2].

3.2 RTP Payload Structure

   As already noted in section 2.9 of this document, this document uses 
   more general names to describe exactly the same payload structure as 
   defined in [6]. For mapping between section names in [6] and in this 
   document see section 2.9. 

3.2.1 The AU Header Section

   When present, the AU Header Section consists of the AU-header-length 
   field, followed by a number of AU-headers. See figure 2.

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
   |AU-headers-length|AU-header|AU-header|      |AU-header|padding|
   |                 |   (1)   |   (2)   |      |   (n)   | bits  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+

   Figure 2: The AU Header Section 

   The AU-headers are configured using format parameters and MAY be empty. 
   If the AU-header is configured empty, the AU-headers-length field 
   SHALL not be present and consequently the AU Header Section is empty. 
   If the AU-header is not configured empty, then the AU-headers-length 
   is a two byte field that specifies the length in bits of the 
   immediately following AU-headers.

   Each AU-header is associated with a single Access Unit (fragment) 
   contained in the Access Unit Data Section in the same RTP packet. For 
   each contained Access Unit (fragment) there is exactly one AU-header. 
   Within the AU Header Section, the AU-headers are bit-wise concatenated 
   in the order in which the Access Units are contained in the Access 
   Unit Data Section. Hence, the n-th AU-header refers to the n-th AU 
   (fragment). If the concatenated AU-headers consume a non-integer 
   number of bytes, up to 7 zero-padding bits MUST be inserted at the end 
   in order to achieve byte-alignment of the AU Header Section.

3.2.1.1 The AU-header

   The AU-header contains the fields given in figure 3. The length in 
   bits of the above fields with the exception of the CTS-flag and 
   the DTS-flag fields is defined by format parameters; see section 4.1. 
   If a format parameter has the default value of zero, then the 
   associated field is not present. 

   +---------------------------------------+
   |     AU-size                           |
   +---------------------------------------+
   |     AU-Index / AU-Index-delta         |
   +---------------------------------------+
   |     CTS-flag                          |
   +---------------------------------------+
   |     CTS-delta                         |
   +---------------------------------------+
   |     DTS-flag                          |
   +---------------------------------------+
   |     DTS-delta                         |
   +---------------------------------------+

   Figure 3: The fields in the AU-header. If used, the AU-Index field 
             only occurs in the first AU-header within an AU Header 
             Section; in any other AU-header the AU-Index-delta field 
             occurs instead.

   AU-size: indicates the size in bytes of the associated Access Unit 
         in the Access Unit Data Section in the same RTP packet. When the 
         AU-size is associated to an AU fragment, the AU size indicates 
         the size of the entire AU and not the size of the fragment. This 
         can be exploited to determine whether a packet contains an entire 
         AU or a fragment, which is particularly useful after losing a 
         packet carrying the last fragment of an AU. 

   AU-Index: indicates the serial number of the associated Access Unit 
         (fragment). For each (in time) consecutive AU or AU fragment, 
         the serial number is incremented with 1. When present, the 
         AU-Index field occurs in the first AU-header in the AU Header 
         Section, but MUST not occur in any subsequent (non-first) 
         AU-header in that Section. To encode the serial number in any 
         such non-first AU-header, the AU-Index-delta field is used. 
         When each AU-Index field is coded with the value 0, the serial 
         number of the AU (fragment) is not specified and in that case 
         receivers MAY ignore the AU-Index field. 

   AU-Index-delta: The AU-Index-delta field is an unsigned integer 
         that specifies the serial number of the associated AU as the 
         difference with respect to the serial number of the previous 
         Access Unit. Hence, for the n-th (n>1) AU the serial number is 
         found from: 
         AU-Index(n) = AU-Index(n-1) + AU-Index-delta(n) + 1
         If the AU-Index field is present in the first AU-header in 
         the AU Header Section, then the AU-Index-delta field MUST be 
         present in any subsequent (non-first) AU-header. When the 
         AU-Index-delta is coded with the value 0, it indicates that 
         the Access Units are consecutive in time. An AU-Index-delta  
         value larger than 0 signals that interleaving is applied.

   CTS-flag: Indicates whether the CTS-delta field is present. 
         A value of 1 indicates that the field is present, a value of 0 
         that it is not present. 
         The CTS-flag field MUST be present in each AU-header if the 
         length of the CTS-delta field is signalled to be larger than 
         zero. In that case, the CTS-flag field MUST have the value 0 
         in the first AU-header and MAY have the value 1 in all non-first 
         AU-headers. The CTS-flag field SHOULD be 0 for any non-first 
         fragment of an Access Unit. 

   CTS-delta: Encodes the CTS by specifying the value of CTS as a 2's
         complement offset (delta) from the timestamp in the RTP header 
         of this RTP packet. The CTS MUST use the same clock rate as the
         time stamp in the RTP header. 

   DTS-flag: Indicates whether the DTS-delta field is present. A value 
         of 1 indicates that DTS-delta is present, a value of 0 that it 
         is not present. 
         The DTS-flag field MUST be present in each AU-header if the 
         length of the DTS-delta field is signalled to be larger than 
         zero. The DTS-flag field SHOULD be 0 for any non-first 
         fragment of an Access Unit. 

   DTS-delta: specifies the value of the DTS as a 2's complement offset 
         (delta) from the CTS timestamp. The DTS MUST use the same clock 
         rate as the time stamp in the RTP header.  

   If present, the fields MUST occur in the mutual order given in 
   figure 3. In the general case a receiver can only discover the size 
   of an AU-header by parsing it since the presence of the CTS-delta 
   and DTS-delta fields is signalled by the value of the CTS-flag and 
   DTS-flag, respectively.

3.2.2 The Auxiliary Section

   The Auxiliary Section consists of the auxiliary-data-size field 
   followed by the auxiliary-data field. Receivers MAY (but are not 
   required to) parse the auxiliary-data field; to facilitate skipping 
   of the auxiliary-data field by receivers, the auxiliary-data-size 
   field indicates the length in bits of the auxiliary-data. If the  
   concatenation of the auxiliary-data-size and the auxiliary-data 
   fields consume a non-integer number of bytes, up to 7 zero padding 
   bits MUST be inserted immediately after the auxiliary data in order 
   to achieve byte-alignment. See figure 4.    

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+
   | auxiliary-data-size   | auxiliary-data       |padding bits |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+

   Figure 4: The fields in the Auxiliary Section 

   The length in bits of the auxiliary-data-size field is configurable 
   by a format parameter; see section 4.1. The default length of zero 
   indicates that the entire Auxiliary Section is absent.

   auxiliary-data-size; specifies the length in bits of the immediately 
         following auxiliary-data field;

   auxiliary-data; the auxiliary-data field contains the Remaining SL 
         headers (RSLHs) as defined in [6].

3.2.3 The Access Unit Data Section

   The Access Unit Data Section contains an integer number of complete 
   Access Units or a single fragment of one AU. The Access Unit Data 
   Section is never empty. If data of more than one Access Units is 
   contained, then the AUs are concatenated into a contiguous string of 
   bytes. See figure 5. The AUs inside the Access Unit Data Section 
   MUST be in decoding order.

   The size and number of Access Units 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.

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 
   |AU(1)                                                              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-                                |
   |                                                                   |
   |     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               |AU(2)                                              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                                   |
   |                                                                   |
   |                            -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               | AU(n)                             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               |
   |-+-+-+-+-+-+-+-+

   Figure 5: Access Unit Data Section; each AU is byte aligned. 

   When multiple Access Units are carried, the size of each AU MUST be 
   made available to the receiver. If the AU size is variable then the 
   size of each AU MUST be indicated in the AU-size field of the 
   corresponding AU-header. However, if the AU size is constant for a 
   stream, this mechanism SHOULD NOT be used, but instead the fixed size 
   SHOULD be signalled by the format parameter "ConstantSize", see 
   section 4.1. 
 
   The absence of both AU-size in the AU-header and the ConstantSize 
   format parameter indicates carriage of a single AU (fragment), i.e. 
   that a single Access Unit (fragment) is transported in each RTP 
   packet for that stream.

3.2.3.1 Fragmentation

   A packet SHALL carry either one or more Access Units, or a single
   fragment of an Access Unit.  Fragments of the same Access Unit have
   the same time stamp but differing RTP sequence numbers. The marker
   bit in the RTP header is 1 on the last fragment of an Access Unit, 
   and 0 on all other fragments.

3.2.3.2 Interleaving

   Access Units MAY be interleaved. Senders MAY perform interleaving.
   Receivers MUST support interleaving.

   When interleaving of Access Units is used it SHALL be implemented 
   using the AU-Index and AU-Index-delta fields in the AU-header.

   Based on the RTP sequence number, the RTP time stamp, the AU-Index and 
   the AU-Index-delta, a receiver can unambiguously reconstruct the 
   original order even in case of out-of-order packets, packet loss or 
   duplication. Note that for this purpose the AU-Index is redundant when 
   the RTP time stamp and the AU-Index-delta values are sufficient for 
   placing the AUs correctly in time. In such cases receivers MAY ignore 
   the AU-Index value and senders MAY code the AU-Index field with the 
   value 0, but only if they code each AU-Index field with that value. 

   When interleaving is applied, a de-interleave buffer is needed in 
   receivers to put the Access Units in their correct logical consecutive 
   order in time. This requires the computation of the time stamp for 
   each Access Unit. In case of a fixed time duration per Access Unit, 
   the time-stamp of each access unit i in an RTP packet with RTP 
   time-stamp T is calculated as follows:

   Timestamp[0] = T
   Timestamp[i, i > 0] = T +(Sum(for k=1 to i of (AU-Index-delta[k] 
                         + 1))) * access-unit-duration

   When AU-Index-delta is always 0, this reduces to T + I * (access-unit- 
   duration). This is the non-interleaved case, the frames are consecutive 
   in time. Note that the AU-Index field (present for the first Access 
   Unit) is not needed in this calculation. Hence in cases where the 
   Access-unit-duration has a fixed and known value, the AU-Index does not 
   need to provide index information and can be coded with the value 0. 
   See also the semantics of the AU-Index field in 3.2.1.1.

   When an RTP packet arrives (after any re-ordering has been done),
   receivers may 'flush' all Access Units from the interleave buffer 
   which have a time-stamp strictly less than the time-stamp of the 
   arriving packet. Similarly the first Access Unit of every arriving 
   packet can always be flushed (as no following packet can provide an 
   earlier Access Unit), and any Access Units which are consecutive with 
   it which have already been received. Access Units should also be 
   flushed in time to be played; this can be important if there is loss 
   before end-of-stream, before a silence interval, or before a large 
   drop-out.

3.2.3.2.1 Constraints for interleaving 

   The size of the packets should be suitably chosen to be appropriate
   to both the path MTU and the duration and capacity of the receiver's
   de-interleave buffer. The maximum packet size for a session should be 
   chosen not to exceed the path MTU.

   In order to control receiver latency and mitigate the effects of loss, 
   there are profile-based limits on the size of the packet. This is 
   expressed as a duration: it is calculated from the duration of the 
   Access Units contained within a packet. It is NOT the difference in 
   time-stamp between the first and last Access Unit in a packet.

   No matter what interleaving scheme is used, the scheme must be 
   analyzed to calculate the minimum number of frames a receiver has to
   buffer in order to de-interleave. 

   The maximum packet duration in milliseconds, and the maximum 
   de-interleave buffer required at the receiver, for the two profiles,
   shall not exceed:

   RTP transport profile 0 -- 200 milliseconds 
   RTP transport profile 1 -- 500 milliseconds 

   When interleaving is applied, the applied RTP transport profile MUST 
   be signalled by the profile parameter; see section 4.1.

   Note that for low bit-rate material, the duration limit may make
   packets shorter than the MTU size.

3.3 Usage of this specification

3.3.1 General

   Usage of this specification requires definition of a mode. A mode 
   defines how use this specification for transport of one or more types 
   of MPEG-4 streams. Each mode may specify constraints and payload 
   configurations as deemed appropriate. 

   Senders MUST signal the mode that they use by the format parameter 
   Mode. In this document only modes are defined for transport of MPEG-4 
   CELP and AAC streams, but more modes are expected to be defined in 
   future RFCs. 

3.3.2 Modes for MPEG-4 CELP and AAC streams

   Four modes are defined for transport of MPEG-4 CELP and AAC streams. 
   In each of these modes, the same requirements apply for the rtpmap 
   attributes. The general form of an rtpmap attribute is:
   a=rtpmap:<payload type><encoding name>/<clock rate>[/<encoding 
             parameters>] 
   For audio streams, <encoding parameters> specifies the number of
   audio channels. This parameter may be omitted if the number of 
   channels is one, provided no additional parameters are needed.
   In all four modes, the following attributes are REQUIRED:
   a) The encoding name
   b) The RTP clock rate MUST be expressed. It is recommended that this 
      be the sampling rate of the audio, to give sample-accurate timing. 
      However, other rates MAY be used (e.g. 90 kHz).
   c) The number of audio channels MUST be specified, for example as 2 
      for  stereo material (see RFC 2327) and MAY be specified as 1 for 
      mono material; 1 is the default.

3.3.3 Constant bit-rate CELP.

   This mode is signalled by mode=CELP-cbr. In this mode one or more 
   fixed size CELP frames can be transported in one RTP packet; there is 
   no support for interleaving. The RTP payload consist of one or more 
   concatenated CELP frames, each of the same size. Both the AU Header 
   Section and the Auxiliary Section are empty. 

   The format parameter ConstantSize MUST be provided to specify the 
   length of each CELP frame.

   For an example see below.

   m=audio 49230 RTP/AVP 96
   a=rtpmap:96 mpeg-generic/44100/2
   a=fmtp:96 streamtype=5; profile-level-id=15; mode=CELP-cbr; config=
   AudioSpecificConfig(); ConstantSize=xxx;

   The AudioSpecificConfig() specifies that the audio stream type is CELP.

3.3.4 Variable bit-rate CELP

   This mode is signalled by mode=CELP-vbr. With this mode in one RTP 
   packet one or more variable size CELP frames can be transported with 
   optional interleaving. As the largest possible frame size in this mode 
   is greater than the maximum CELP frames size, there is no support for 
   fragmentation on the CELP frames.

   In this mode the RTP payload consists of the AU Header Section, 
   followed by one or more concatenated CELP frames. The Auxiliary Section 
   is empty. For each CELP frame contained in the payload there is a one 
   byte AU-header in the AU Header Section to provide : 
   (a) the size of each CELP frame in the payload and 
   (b) index information for computing the sequence (and hence timing) of 
       each CELP frame.
   Transport of CELP frames requires that the AU-size field is coded with 
   6 bits. In this mode therefore 6 bits are allocated to the AU-size 
   field, and 2 bits to the AU-Index(-delta) field. Each AU-Index field 
   MUST be coded with the value 0. In the AU Header Section, the 
   concatenated AU-headers are preceded by the 16-bit AU-headers-length 
   field, as specified in 3.2.1. 

   Next to the required format parameters, the following parameters MUST 
   be present: 
   SizeLength, IndexLength, and IndexDeltaLength. 
   When interleaving is applied (AU-Index-delta coded with a value larger 
   than 0), also the parameter Profile MUST be present. 

   Example :

   m=audio 49230 RTP/AVP 96
   a=rtpmap:96 mpeg4-generic/44100/2
   a=fmtp:96 streamtype=5; profile-level-id=15; mode=CELP-vbr; config=
   AudioSpecificConfig(); SizeLength=6; IndexLength=2; IndexDeltaLength=2; 
   Profile=1

   The AudioSpecificConfig() specifies that audio stream type is CELP.

3.3.5 Low bit-rate AAC

   This mode is signalled by AAC-lbr. This mode supports transport of one 
   or more variable size AAC frames with optional support for interleaving 
   and fragmenting. The maximum size of an AAC frame (fragment) in this 
   mode is 63 bytes. 

   The payload configuration in this mode is the same as in the variable 
   bit-rate CELP mode as defined in 3.3.4. The RTP payload consists of the 
   AU Header Section, followed by concatenated AAC frames. The Auxiliary 
   Section is empty. For each AAC frame contained in the payload the one 
   byte AU-header provides : 
   (a) the size of each AAC frame in the payload and 
   (b) index information for computing the sequence (and hence timing) of 
       each AAC frame.
   In the AU-header, the AU-size is coded with 6 and the AU-Index(-delta) 
   with 2 bits; the AU-Index field MUST have the value 0 in each AU-header. 
   In the AU-header Section, the concatenated AU-headers are preceded by 
   the 16-bit AU-headers-length field, as specified in 3.2.1. 

   Next to the required format parameters, the following parameters MUST 
   be present: 
   SizeLength, IndexLength, and IndexDeltaLength. 
   When interleaving is applied (AU-Index-delta coded with a value larger 
   than 0), also the parameter Profile MUST be present. 

   Example :

   m=audio 49230 RTP/AVP 96
   a=rtpmap:96 mpeg4-generic/44100/2
   a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-lbr; config=
   AudioSpecificConfig(); SizeLength=6; IndexLength=2; IndexDeltaLength=2; 
   Profile=1

   The AudioSpecificConfig() specifies that audio stream type is AAC.

3.3.6 High bit-rate AAC

   This mode is signalled by mode=AAC-hbr. This mode supports transport 
   of one or more large variable size AAC frames in one RTP packet with 
   optional support for interleaving and fragmenting. The maximum size of 
   an AAC frame (fragment) in this mode is 8191 bytes. 

   In this mode the RTP payload consists of the AU Header Section, 
   followed by one or more concatenated AAC frames. The Auxiliary Section 
   is empty. For each AAC frame contained in the payload there is an 
   AU-header in the AU Header Section to provide :
   (a) the size of each AAC frame in the payload and 
   (b) index information for computing the sequence (and hence timing) of 
       each AAC frame.
   To code the maximum size of an AAC frame requires 13 bits. Therefore in 
   this configuration 13 bits are allocated to the AU-size, and 3 bits 
   to the AU-Index(-delta) field. Thus each AU-header has a size of 2 
   bytes. Each AU-Index field MUST be coded with the value 0. In the 
   AU Header Section, the concatenated AU-headers are preceded by the 
   16-bit AU-headers-length field, as specified in 3.2.1. 

   Next to the required format parameters, the following parameters MUST 
   be present: 
   SizeLength, IndexLength, and IndexDeltaLength. 
   When interleaving is applied (AU-Index-delta coded with a value larger 
   than 0), also the parameter Profile MUST be present.

   Example :
   m=audio 49230 RTP/AVP 96
   a=rtpmap:96 mpeg4-generic/44100/2
   a=fmtp:96 streamtype=5; profile-level-id=15; mode= AAC-hbr; config=
   AudioSpecificConfig(); SizeLength=13; IndexLength=3; IndexDeltaLength=3; 
   Profile=1

   The AudioSpecificConfig() specifies that the audio stream type is AAC. 

4. Types and names 

   This section describes the MIME types and names associated with this 
   payload format.
    
   Depending on the required payload configuration, format parameters may 
   need to be available to the receiver. This is done using the parameters 
   described in the next section. The absence of any of these parameters 
   is equivalent to the associated field set to its default value, which 
   is always zero. The absence of any such parameters resolves into a 
   default "basic" configuration. 
   
   In the MPEG-4 framework the SL stream configuration information is 
   carried using the Object Descriptor. When such information is present 
   both in an Object Descriptor and as a parameter of this payload format 
   it MUST be exactly the same. 

4.1 MIME types 
 
   This specification uses exactly the same MIME types as [6], and hence 
   no further MIME type registration is required. In [6] uses the MIME 
   media type names: "video" or "audio" or "application".

      "video" SHOULD be used for any MPEG Video stream or any MPEG-4 
      System (ISO/IEC 14496-1) stream that conveys information needed for 
      an audio-visual presentation.

      "audio" SHOULD be used for any MPEG Audio streams and any MPEG-4 
      System (ISO/IEC 14496-1) stream that conveys information needed for 
      an audio-only presentation.

      "application" SHOULD be used for MPEG-4 Systems streams 
      (ISO/IEC14496-1) 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:

      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.

      Profile-level-id: 
      A decimal representation of the MPEG-4 Profile Level indication. 
      This parameter MUST be used in the capability exchange or session 
      set-up procedure to indicate the MPEG-4 Profile and Level 
      combination of which the relevant MPEG-4 media codec is capable 
      of. 
      For audio streams, this parameter is the decimal value from Table 5 
      (audioProfileLevelIndicationValues) in ISO/IEC 14496-1, indicating 
      which MPEG-4 Audio tool subsets are applied to encode the audio 
      stream. 
      For visual streams, this parameter is the decimal value from Table 
      G-1 (FLC table for profile and level indication of ISO/IEC 14496-2, 
      indicating which MPEG-4 Visual tool subsets are applied to encode 
      the visual stream.
    
      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, if necessary to achieve byte alignment, up to 
      7 zero-valued padding bits 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.
      For visual streams, config is the MPEG-4 Visual configuration 
      information, as defined in subclause 6.2.1 Start codes of 
      ISO/IEC14496-2. 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). 

      Mode: 
      The mode in which this specification is used. The following modes 
      can be signalled :
      mode=CELP-cbr,
      mode=CELP-vbr,
      mode=AAC-lbr and
      mode=AAC-hbr.
      Other modes are expected to be defined in future RFCs. When defining 
      a new mode care MUST be taken that an implementation of all features 
      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; in particular, 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.  

   Optional parameters:

      ConstantSize:
      The constant size in bytes of each Access Unit for this stream. 
      Simultaneous presence of ConstantSize and the SizeLength 
      parameters is not permitted.

      SizeLength: 
      The number of bits on which the AU-size field is encoded in the 
      AU-header. Simultaneous presence of SizeLength and the ConstantSize 
      parameter is not permitted.

      IndexLength:
      The number of bits on which the AU-Index is encoded in the first 
      AU-header. The default value of zero indicates the absence of the 
      AU-Index and AU-Index-delta fields in each AU-header.

      IndexDeltaLength:
      The number of bits on which the AU-Index-delta field is encoded in 
      any non-first AU-header. 

      CTSDeltaLength:
      The number of bits on which the CTS-delta field is encoded in the 
      AU-header. 

      DTSDeltaLength:
      The number of bits on which the DTS-delta field is encoded in the 
      AU-header. 

      AuxiliaryDataSizeLength:
      The number of bits that is used to encode the auxiliary-data-size 
      field. 

      Profile:
      The decimal representation of the RTP transport profile.
 
   Applications MAY use more parameters, in addition to those defined 
   above. Receivers MUST tolerate the presence of such additional 
   parameters, but these parameters SHALL not impact the decoding of 
   receivers that comply to this specification. 

   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 Visual 
   specifications (ISO/IEC 14496-3). The RTP packets MUST be packetized 
   according to the RTP payload format defined in RFC <self-reference-to-
   this>. 
    
   Security considerations: 
   As in RFC <self-reference-to-this>. 
    
   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 interworking 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 <self-reference-to-this>. 
    
   Applications which use this media type: 
   Multimedia streaming and conferencing tools, Internet messaging and 
   Email applications.
    
   Additional information: none 
    
   Magic number(s): none 
    
   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 <self-reference-to-this>. 
    
   Intended usage: COMMON 
    
   Author/Change controller: 
   Authors of RFC <self-reference-to-this>. 
    
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 
   (for parameter usage examples see Appendix A). 
    
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 a SDP message 
   for example transported to the client in reply to a RTSP DESCRIBE of 
   via SAP. In that case the (a=fmtp) keyword MUST be used as described 
   in RFC 2327 [8, section 6]. The syntax being then: 
    
   a=fmtp:<format> <parameter name>=<value>[; <parameter name>=<value>] 

5. Security Considerations

   RTP packets using the payload format defined in this specification 
   are subject to the security considerations discussed in the RTP 
   specification [2]. 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, 
   ECMAScript) commands in the streams. However, this can increase the 
   complexity significantly. 

6. References

   [1] ISO/IEC International Standard 14496 (MPEG-4); "Information 
   technology - Coding of audio-visual objects", January 2000
  
   [2] Schulzrinne, Casner, Frederick, Jacobson RTP: A Transport
   Protocol for Real Time Applications  RFC 1889, Internet Engineering
   Task Force, January 1996.

   [3] S. Bradner, Key words for use in RFCs to Indicate Requirement
   Levels, RFC 2119, March 1997.

   [4] D. Hoffman, G. Fernando, V. Goyal, M. Civanlar, RTP payload 
   format for MPEG1/MPEG2 Video, RFC 2250, January 1998.

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

   [6] Avaro, Basso, Casner, Civanlar, Gentric, Herpel, Lim, Perkins, 
   van der Meer, RTP payload format for MPEG-4 streams, work in progress, 
   draft-gentric-avt-mpeg4-multiSL-01.txt, January 2001.

   [7] D. Singer, Y Lim, A Framework for the delivery of MPEG-4 over
   IP-based Protocols, work in progress, draft-singer-mpeg4-ip-
   01.txt,October 2000.

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

7. Author Adresses

   Jan van der Meer
   Philips Digital Networks
   Cederlaan 4
   5600 JB Eindhoven
   Netherlands
   Email : [email-masked]

   David Mackie
   Cisco Systems Inc.
   170 West Tasman Dr.
   San Jose, CA 95034
   Email: [email-masked]

   Viswanathan Swaminathan
   Sun Microsystems Inc.
   901 San Antonio Road, M/S UMPK15-214
   Palo Alto, CA 94303
   Email: [email-masked]

   David Singer
   Apple Computer, Inc.
   One Infinite Loop, MS:302-3MT
   Cupertino  CA 95014
   Email: [email-masked]

   Full Copyright Statement

   "Copyright (C) The Internet Society (date). All Rights Reserved. This 
   document and translations of it may be copied and furnished to others, 
   and derivative works that comment on or otherwise explain it or assist 
   in its implementation may be prepared, copied, published and 
   distributed, in whole or in part, without restriction of any kind, 
   provided that the above copyright notice and this paragraph are 
   included on all such copies and derivative works. However, this 
   document itself may not be modified in any way, such as by removing 
   the copyright notice or references to the Internet Society or other 
   Internet organizations, except as needed for the purpose of developing 
   Internet standards in which case the procedures for copyrights defined 
   in the Internet Standards process MUST be followed, or as required to 
   translate it into.

APPENDIX: Usage of this payload format

Appendix A. Examples

A.1 Examples of delay analysis with interleave

A.1.1 Group interleave

   An example of regular interleave is when packets are formed into 
   groups.  If the number of packets in a group is N, packet 0 contains 
   frame 0, frame N, frame 2N, and so on;  packet 1 contains frame 1, 
   frame 1+N, 1+2N, and so on.  The AU-Index field is used to document 
   the sequence of the packet within the group (or the first frame in the 
   packet, which is the same thing in this scheme), and all the
   AU-Index-delta fields contain N-1.

   Receivers can tell when a new interleave group is starting, by noting
   that the computed time-stamp of the first frame in a packet is later
   than any previously computed time-stamp.  This is because no
   following packet can contain an earlier RTP timestamp (RTP rules),
   and the second and subsequent frames in a packet have larger
   time-stamps (the frames in a packet are also in time-order).

   If the group size is 3, then packets are formed as follows:

   Packet   Time-stamp   Frame Numbers       AU-Index, AU-Index-delta 
   0        T[0]         0, 3, 6             0, 2, 2
   1        T[1]         1, 4, 7             0, 2, 2
   2        T[2]         2, 5, 8             0, 2, 2
   3        T[9]         9,12,15             0, 2, 2

   In this case, the receiver would have to buffer 4 frames at least
   from packets 0 and 1, and can flush all frames when packet 2 arrives.
   (Frame 0 can be flushed as packet 0 arrives, since it is the earliest
   frame we hold, and likewise frame 1 from packet 1; we are therefore
   holding 3,4,6,7 until packet 2 arrives).

   If there is loss, then the receiver may wait longer than is strictly
   necessary before it emits frames.  For example, say packet 1 is lost
   from the above example.  Packet 0 allows frame 0 to be emitted, and
   then packet 2 arrives, allowing us to notice the loss of frame 1, and
   emit frame 2 and 3.  Then it is not until the arrival of packet 3
   (which has a time-stamp beyond the times of all the frames seen so
   far), that we can finish dealing with the loss, even though the first
   group has, in fact, ended.  (This is in contrast to schemes which
   signal the group size explicitly;  if the receiver knows that this is
   packet 3 of 3, then even if 2 of 3 is missing, it can de-interleave
   this group without waiting for the next one to start).

   In the above example the AU-Index is coded with the value 0, as 
   required for the modes defined in this document. To reconstruct the 
   original order, the RTP time stamp and the AU-Index-delta are used. 
   See also 3.2.3.2.

A.1.2 Continuous interleave

   In continuous interleave, once the scheme is 'primed', the number of
   frames in a packet exceeds the 'stride' (the distance between them).
   This shortens the buffering needed, smooths the data-flow, and gives
   slightly larger packets -- and thus lower overhead -- for the same
   interleave.  For example, here is a continuous interleave also over a
   stride of 3 frames, but with 4 frames per packet, for a run of 20
   frames.  This shows both how the scheme 'starts up' and how it
   finishes.

   Packet   Time-stamp   Frame Numbers       AU-Index, AU-Index-delta 
   0        T[0]                     0       0
   1        T[1]                 1   4       0  2
   2        T[2]             2   5   8       0  2  2
   3        T[3]          3   6   9  12      0  2  2  2
   4        T[7]          7  10  13  16      0  2  2  2
   5        T[11]        11  14  17  20      0 2  2  2
   6        T[15]        15  18              0 2
   7        T[19]        19                  0

   In this case, the receiver has to buffer only 3 frames, not 4.  Say
   we are waiting for packet 4.  We can flush frames 0, 1, 2, 3, 4, 5,
   6;  we are holding therefore 8, 9, 12.   Packet 4 arrives, allowing
   us to emit 7,8,9,10, and we are holding 12,13,16.  Each arriving
   packet contains 4 frames, and allows 4 frames to be flushed.

   In the above example the AU-Index is coded with the value 0, as 
   required for the modes defined in this document. To reconstruct the 
   original order, the RTP time stamp and the AU-Index-delta are used. 
   See also 3.2.3.2.

   If there is loss, again the receiver has to wait to emit the erasure
   frames.  In this case, say packet 3 is lost.  We were holding frames
   4, 5, and 8.  On the arrival of packet 4, (time-stamp of frame 7), we
   now know frame 3 was lost, we can emit frames 4,5, and we know 6 must
   be lost, and emit 7, which is in the packet that arrived.  Then on
   the arrival of packet 5 (time-stamp 11) we can emit 8, indicate loss
   of 9, and emit 10 and 11.  Finally, the arrival of packet 6
   (time-stamp 15) indicates that 12 must be lost;  we have now detected
   all the lost frames.