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  standard. MPEG defines tools to compress content such as audio-visual information into elementary streams. In  a generic RTP payload format is defined for transport of any non-multiplexed MPEG-4 elementary stream. To achieve the generic MPEG-4 functionality,  addresses detailed issues related to the MPEG-4 SL layer. However, many initial applications will not use the SL Layer. To facilitate usage of  by such applications, this document describes how to use  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 firstname.lastname@example.org 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 . 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 , a general payload format is defined for transport of a single MPEG-4 elementary stream over RTP. The RTP payload format specified in  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  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  is highly desirable for all MPEG-4 applications, to understand  may be difficult without knowledge of the MPEG-4 SL layer. Therefore in this document the use of  is described without requiring knowledge of the SL layer to understand its functionality. Sophisticated features on interleaving of fragmented Access Units are defined in . Because initial applications do not require these complicated features, these features are not supported in this document. Hence, only a functional subset of  is supported. In , 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  without requiring knowledge of the SL layer; - defines a functional but true subset of ; - defines modes how to use this specification for transport of MPEG-4 CELP and AAC streams. The use of  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 . 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  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  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 . 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  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 . 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 , 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" . 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 . Receivers that comply with  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  but not to this document. Such receivers may also not be capable of exploiting some of features of the SL layer supported in , 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  (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 . CC and CSRC fields are used as described in RFC 1889 . RTCP SHOULD be used as defined in RFC 1889 . 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 . For mapping between section names in  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. 184.108.40.206 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 . 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. 220.127.116.11 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. 18.104.22.168 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 = 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 22.214.171.124. 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. 126.96.36.199.1 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 , and hence no further MIME type registration is required. In  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 . 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  ISO/IEC International Standard 14496 (MPEG-4); "Information technology - Coding of audio-visual objects", January 2000  Schulzrinne, Casner, Frederick, Jacobson RTP: A Transport Protocol for Real Time Applications RFC 1889, Internet Engineering Task Force, January 1996.  S. Bradner, Key words for use in RFCs to Indicate Requirement Levels, RFC 2119, March 1997.  D. Hoffman, G. Fernando, V. Goyal, M. Civanlar, RTP payload format for MPEG1/MPEG2 Video, RFC 2250, January 1998.  Y. Kikuchi, T. Nomura, S. Fukunaga, Y. Matsui, H. Kimata, RTP payload format for MPEG-4 Audio/Visual streams, RFC 3016.  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.  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.  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@example.com David Mackie Cisco Systems Inc. 170 West Tasman Dr. San Jose, CA 95034 Email: firstname.lastname@example.org Viswanathan Swaminathan Sun Microsystems Inc. 901 San Antonio Road, M/S UMPK15-214 Palo Alto, CA 94303 Email: email@example.com David Singer Apple Computer, Inc. One Infinite Loop, MS:302-3MT Cupertino CA 95014 Email: firstname.lastname@example.org 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, 3, 6 0, 2, 2 1 T 1, 4, 7 0, 2, 2 2 T 2, 5, 8 0, 2, 2 3 T 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 188.8.131.52. 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 1 T 1 4 0 2 2 T 2 5 8 0 2 2 3 T 3 6 9 12 0 2 2 2 4 T 7 10 13 16 0 2 2 2 5 T 11 14 17 20 0 2 2 2 6 T 15 18 0 2 7 T 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 184.108.40.206. 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.