| Rfc | 6416 |
|---|
| Title | RTP Payload Format for MPEG-4 Audio/Visual Streams |
|---|
| Author | M. Schmidt, F.
de Bont, S. Doehla, J. Kim |
|---|
| Date | October 2011 |
|---|
| Format: | TXT, HTML |
|---|
| Obsoletes | RFC3016 |
|---|
| Status: | PROPOSED STANDARD |
|---|
|
Internet Engineering Task Force (IETF) M. Schmidt
Request for Comments: 6416 Dolby Laboratories
Obsoletes: 3016 F. de Bont
Category: Standards Track Philips Electronics
ISSN: 2070-1721 S. Doehla
Fraunhofer IIS
J. Kim
LG Electronics Inc.
October 2011
RTP Payload Format for MPEG-4 Audio/Visual Streams
Abstract
This document describes Real-time Transport Protocol (RTP) payload
formats for carrying each of MPEG-4 Audio and MPEG-4 Visual
bitstreams without using MPEG-4 Systems. This document obsoletes RFC
3016. It contains a summary of changes from RFC 3016 and discusses
backward compatibility to RFC 3016. It is a necessary revision of
RFC 3016 in order to correct misalignments with the 3GPP Packet-
switched Streaming Service (PSS) specification regarding the RTP
payload format for MPEG-4 Audio.
For the purpose of directly mapping MPEG-4 Audio/Visual bitstreams
onto RTP packets, this document provides specifications for the use
of RTP header fields and also specifies fragmentation rules. It also
provides specifications for Media Type registration and the use of
the Session Description Protocol (SDP). The audio payload format
described in this document has some limitations related to the
signaling of audio codec parameters for the required multiplexing
format. Therefore, new system designs should utilize RFC 3640, which
does not have these restrictions. Nevertheless, this revision of RFC
3016 is provided to update and complete the specification and to
enable interoperable implementations.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc6416.
Copyright Notice
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. MPEG-4 Visual RTP Payload Format . . . . . . . . . . . . . 4
1.2. MPEG-4 Audio RTP Payload Format . . . . . . . . . . . . . 5
1.3. Interoperability with RFC 3016 . . . . . . . . . . . . . . 6
1.4. Relation with RFC 3640 . . . . . . . . . . . . . . . . . . 6
2. Definitions and Abbreviations . . . . . . . . . . . . . . . . 6
3. Clarifications on Specifying Codec Configurations for
MPEG-4 Audio . . . . . . . . . . . . . . . . . . . . . . . . . 7
4. LATM Restrictions for RTP Packetization of MPEG-4 Audio
Bitstreams . . . . . . . . . . . . . . . . . . . . . . . . . . 7
5. RTP Packetization of MPEG-4 Visual Bitstreams . . . . . . . . 8
5.1. Use of RTP Header Fields for MPEG-4 Visual . . . . . . . . 9
5.2. Fragmentation of MPEG-4 Visual Bitstream . . . . . . . . . 10
5.3. Examples of Packetized MPEG-4 Visual Bitstream . . . . . . 11
6. RTP Packetization of MPEG-4 Audio Bitstreams . . . . . . . . . 15
6.1. RTP Packet Format . . . . . . . . . . . . . . . . . . . . 15
6.2. Use of RTP Header Fields for MPEG-4 Audio . . . . . . . . 16
6.3. Fragmentation of MPEG-4 Audio Bitstream . . . . . . . . . 17
7. Media Type Registration for MPEG-4 Audio/Visual Streams . . . 17
7.1. Media Type Registration for MPEG-4 Visual . . . . . . . . 17
7.2. Mapping to SDP for MPEG-4 Visual . . . . . . . . . . . . . 20
7.2.1. Declarative SDP Usage for MPEG-4 Visual . . . . . . . 20
7.3. Media Type Registration for MPEG-4 Audio . . . . . . . . . 21
7.4. Mapping to SDP for MPEG-4 Audio . . . . . . . . . . . . . 24
7.4.1. Declarative SDP Usage for MPEG-4 Audio . . . . . . . . 25
7.4.1.1. Example: In-Band Configuration . . . . . . . . . . 25
7.4.1.2. Example: 6 kbit/s CELP . . . . . . . . . . . . . . 25
7.4.1.3. Example: 64 kbit/s AAC LC Stereo . . . . . . . . . 26
7.4.1.4. Example: Use of the "SBR-enabled" Parameter . . . 26
7.4.1.5. Example: Hierarchical Signaling of SBR . . . . . . 27
7.4.1.6. Example: HE AAC v2 Signaling . . . . . . . . . . . 27
7.4.1.7. Example: Hierarchical Signaling of PS . . . . . . 28
7.4.1.8. Example: MPEG Surround . . . . . . . . . . . . . . 28
7.4.1.9. Example: MPEG Surround with Extended SDP
Parameters . . . . . . . . . . . . . . . . . . . . 28
7.4.1.10. Example: MPEG Surround with Single-Layer
Configuration . . . . . . . . . . . . . . . . . . 29
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 29
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 30
10. Security Considerations . . . . . . . . . . . . . . . . . . . 30
11. Differences to RFC 3016 . . . . . . . . . . . . . . . . . . . 31
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32
12.1. Normative References . . . . . . . . . . . . . . . . . . . 32
12.2. Informative References . . . . . . . . . . . . . . . . . . 33
1. Introduction
The RTP payload formats described in this document specify how MPEG-4
Audio [14496-3] and MPEG-4 Visual streams [14496-2] are to be
fragmented and mapped directly onto RTP packets.
These RTP payload formats enable transport of MPEG-4 Audio/Visual
streams without using the synchronization and stream management
functionality of MPEG-4 Systems [14496-1]. Such RTP payload formats
will be used in systems that have intrinsic stream management
functionality and thus require no such functionality from MPEG-4
Systems. H.323 [H323] terminals are an example of such systems,
where MPEG-4 Audio/Visual streams are not managed by MPEG-4 Systems
Object Descriptors but by H.245 [H245]. The streams are directly
mapped onto RTP packets without using the MPEG-4 Systems Sync Layer.
Other examples are the Session Initiation Protocol (SIP) [RFC3261]
and Real Time Streaming Protocol (RTSP) where media type and SDP are
used. Media type and SDP usages of the RTP payload formats described
in this document are defined to directly specify the attribute of
Audio/Visual streams (e.g., media type, packetization format, and
codec configuration) without using MPEG-4 Systems. The obvious
benefit is that these MPEG-4 Audio/Visual RTP payload formats can be
handled in a unified way together with those formats defined for non-
MPEG-4 codecs. The disadvantage is that interoperability with
environments using MPEG-4 Systems may be difficult; hence, other
payload formats may be better suited to those applications.
The semantics of RTP headers in such cases need to be clearly
defined, including the association with MPEG-4 Audio/Visual data
elements. In addition, it is beneficial to define the fragmentation
rules of RTP packets for MPEG-4 Video streams so as to enhance error
resiliency by utilizing the error resiliency tools provided inside
the MPEG-4 Video stream.
1.1. MPEG-4 Visual RTP Payload Format
MPEG-4 Visual is a visual coding standard with many features,
including: high coding efficiency; high error resiliency; and
multiple, arbitrary shape object-based coding [14496-2]. It covers a
wide range of bitrates from scores of kbit/s to several Mbit/s. It
also covers a wide variety of networks, ranging from those guaranteed
to be almost error-free to mobile networks with high error rates.
With respect to the fragmentation rules for an MPEG-4 Visual
bitstream defined in this document, since MPEG-4 Visual is used for a
wide variety of networks, it is desirable not to apply too much
restriction on fragmentation, and a fragmentation rule such as "a
single video packet shall always be mapped on a single RTP packet"
may be inappropriate. On the other hand, careless, media-unaware
fragmentation may cause degradation in error resiliency and bandwidth
efficiency. The fragmentation rules described in this document are
flexible but manage to define the minimum rules for preventing
meaningless fragmentation while utilizing the error resiliency
functionalities of MPEG-4 Visual.
The fragmentation rule "Different Video Object Planes (VOPs) SHOULD
be fragmented into different RTP packets" is made so that the RTP
timestamp uniquely indicates the VOP time framing. On the other
hand, MPEG-4 video may generate VOPs of very small size, in cases
with an empty VOP (vop_coded=0) containing only VOP header or an
arbitrary shaped VOP with a small number of coding blocks. To reduce
the overhead for such cases, the fragmentation rule permits
concatenating multiple VOPs in an RTP packet. (See fragmentation
rule (4) in Section 5.2 and the descriptions of marker bit and
timestamp in Section 5.1.)
While the additional media-specific RTP header defined for such video
coding tools as H.261 [H261] or MPEG-1/2 is effective in helping to
recover picture headers corrupted by packet losses, MPEG-4 Visual
already has error resiliency functionalities for recovering corrupt
headers, and these can be used on RTP/IP networks as well as on other
networks (H.223/mobile, MPEG-2 Transport Stream, etc.). Therefore,
no extra RTP header fields are defined in this MPEG-4 Visual RTP
payload format.
1.2. MPEG-4 Audio RTP Payload Format
MPEG-4 Audio is an audio standard that integrates many different
types of audio coding tools. Low-overhead MPEG-4 Audio Transport
Multiplex (LATM) manages the sequences of audio data with relatively
small overhead. In audio-only applications, then, it is desirable
for LATM-based MPEG-4 Audio bitstreams to be directly mapped onto RTP
packets without using MPEG-4 Systems.
For MPEG-4 Audio coding tools, as is true for other audio coders, if
the payload is a single audio frame, packet loss will not impair the
decodability of adjacent packets. Therefore, the additional media-
specific header for recovering errors will not be required for MPEG-4
Audio. Existing RTP protection mechanisms, such as Generic Forward
Error Correction [RFC5109] and Redundant Audio Data [RFC2198], MAY be
applied to improve error resiliency.
1.3. Interoperability with RFC 3016
This specification is not backwards compatible with [RFC3016], as a
binary incompatible LATM version is mandated. Existing
implementations of RFC 3016 that use a recent LATM version may
already comply to this specification and must be considered as not
compliant with RFC 3016. The 3GPP PSS service [3GPP] is such an
example, as a more recent LATM version is mandated in the 3GPP PSS
specification. Existing implementations that use the LATM version as
specified in RFC 3016 MUST be updated to comply with this
specification.
1.4. Relation with RFC 3640
In this document a payload format for the transport of MPEG-4
Elementary Streams is specified. For MPEG-4 Audio streams "out-of-
band" signaling is defined such that a receiver is not obliged to
decode the payload data to determine the audio codec and its
configuration. The signaling capabilities specified in this document
are less explicit than those defined in [RFC3640]. But, the use of
the MPEG-4 LATM in various transmission standards justifies its right
to exist; see also Section 1.2.
2. Definitions and Abbreviations
This document makes use of terms, specified in [14496-2], [14496-3],
and [23003-1]. In addition, the following terms are used in this
document and have specific meaning within the context of this
document.
Abbreviations:
AAC: Advanced Audio Coding
ASC: AudioSpecificConfig
HE AAC: High Efficiency AAC
LATM: Low-overhead MPEG-4 Audio Transport Multiplex
PS: Parametric Stereo
SBR: Spectral Band Replication
VOP: Video Object Plane
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 [RFC2119].
3. Clarifications on Specifying Codec Configurations for MPEG-4 Audio
For MPEG-4 Audio [14496-3] streams, the decoder output configuration
can differ from the core codec configuration depending of use of the
SBR and PS tools.
The core codec sampling rate is the default audio codec sampling
rate. When SBR is used, typically the double value of the core codec
sampling rate will be regarded as the definitive sampling rate (i.e.,
the decoder's output sampling rate)
Note: The exception is down-sampled SBR mode, in which case the SBR
sampling rate and core codec sampling rate are identical.
The core codec channel configuration is the default audio codec
channel configuration. When PS is used, the core codec channel
configuration indicates one channel (i.e., mono) whereas the
definitive channel configuration is two channels (i.e. stereo). When
MPEG Surround is used, the definitive channel configuration depends
on the output of the MPEG Surround decoder.
4. LATM Restrictions for RTP Packetization of MPEG-4 Audio Bitstreams
LATM has several multiplexing features as follows:
o carrying configuration information with audio data,
o concatenating multiple audio frames in one audio stream,
o multiplexing multiple objects (programs), and
o multiplexing scalable layers,
However, in RTP transmission, there is no need for the last two
features. Therefore, these two features MUST NOT be used in
applications based on RTP packetization specified by this document.
Since LATM has been developed for only natural audio coding tools,
i.e., not for synthesis tools, it seems difficult to transmit
Structured Audio (SA) data and Text-to-Speech Interface (TTSI) data
by LATM. Therefore, SA data and TTSI data MUST NOT be transported by
the RTP packetization in this document.
For transmission of scalable streams, audio data of each layer SHOULD
be packetized onto different RTP streams allowing for the different
layers to be treated differently at the IP level, for example, via
some means of differentiated service. On the other hand, all
configuration data of the scalable streams are contained in one LATM
configuration data "StreamMuxConfig", and every scalable layer shares
the StreamMuxConfig. The mapping between each layer and its
configuration data is achieved by LATM header information attached to
the audio data. In order to indicate the dependency information of
the scalable streams, the signaling mechanism as specified in
[RFC5583] SHOULD be used (see Section 6.2).
5. RTP Packetization of MPEG-4 Visual Bitstreams
This section specifies RTP packetization rules for MPEG-4 Visual
content. An MPEG-4 Visual bitstream is mapped directly onto RTP
packets without the addition of extra header fields or any removal of
Visual syntax elements. The Combined Configuration/Elementary stream
mode MUST be used so that configuration information will be carried
to the same RTP port as the elementary stream. (See Subclause 6.2.1,
"Start codes", of [14496-2].) The configuration information MAY
additionally be specified by some out-of-band means. If needed by
systems using media type parameters and SDP parameters, e.g., SIP and
RTSP, the optional parameter "config" MUST be used to specify the
configuration information (see Sections 7.1 and 7.2).
When the short video header mode is used, the RTP payload format for
H.263 SHOULD be used. (The format defined in [RFC4629] is
RECOMMENDED, but the [RFC4628] format MAY be used for compatibility
with older implementations.)
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number | RTP
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp | Header
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| .... |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| | RTP
| MPEG-4 Visual stream (byte aligned) | Pay-
| | load
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: An RTP Packet for MPEG-4 Visual Stream
5.1. Use of RTP Header Fields for MPEG-4 Visual
Payload Type (PT): The assignment of an RTP payload type for this
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 by means of an out-of-band signaling protocol (e.g.,
H.245, SIP).
Extension (X) bit: Defined by the RTP profile used.
Sequence Number: Incremented by 1 for each RTP data packet sent,
starting, for security reasons, with a random initial value.
Marker (M) bit: The marker bit is set to 1 to indicate the last RTP
packet (or only RTP packet) of a VOP. When multiple VOPs are carried
in the same RTP packet, the marker bit is set to 1.
Timestamp: The timestamp indicates the sampling instance of the VOP
contained in the RTP packet. A constant offset, which is random, is
added for security reasons.
o When multiple VOPs are carried in the same RTP packet, the
timestamp indicates the earliest of the VOP times within the VOPs
carried in the RTP packet. Timestamp information of the rest of
the VOPs is derived from the timestamp fields in the VOP header
(modulo_time_base and vop_time_increment).
o If the RTP packet contains only configuration information and/or
Group_of_VideoObjectPlane() fields, the timestamp of the next VOP
in the coding order is used.
o If the RTP packet contains only visual_object_sequence_end_code
information, the timestamp of the immediately preceding VOP in the
coding order is used.
The resolution of the timestamp is set to its default value of 90
kHz, unless specified by out-of-band means (e.g., SDP parameter or
media type parameter as defined in Section 7).
Other header fields are used as described in [RFC3550].
5.2. Fragmentation of MPEG-4 Visual Bitstream
A fragmented MPEG-4 Visual bitstream is mapped directly onto the RTP
payload without any addition of extra header fields or any removal of
Visual syntax elements.
In the following, header means one of the following:
o Configuration information (Visual Object Sequence Header, Visual
Object Header, and Video Object Layer Header)
o visual_object_sequence_end_code
o The header of the entry point function for an elementary stream
(Group_of_VideoObjectPlane() or the header of VideoObjectPlane(),
video_plane_with_short_header(), MeshObject(), or FaceObject())
o The video packet header (video_packet_header() excluding
next_resync_marker())
o The header of gob_layer()
o See Subclause 6.2.1 ("Start codes") of [14496-2] for the
definition of the configuration information and the entry point
functions.
The Combined Configuration/Elementary streams mode is used. The
following rules apply for the fragmentation.
(1) Configuration information and Group_of_VideoObjectPlane() fields
SHALL be placed at the beginning of the RTP payload (just after
the RTP header) or just after the header of the syntactically
upper-layer function.
(2) If one or more headers exist in the RTP payload, the RTP payload
SHALL begin with the header of the syntactically highest
function. Note: The visual_object_sequence_end_code is regarded
as the lowest function.
(3) A header SHALL NOT be split into a plurality of RTP packets.
(4) Different VOPs SHOULD be fragmented into different RTP packets
so that one RTP packet consists of the data bytes associated
with a unique VOP time instance (that is indicated in the
timestamp field in the RTP packet header), with the exception
that multiple consecutive VOPs MAY be carried within one RTP
packet in the decoding order if the size of the VOPs is small.
Note: When multiple VOPs are carried in one RTP payload, the
timestamp of the VOPs after the first one may be calculated by
the decoder. This operation is necessary only for RTP packets
in which the marker bit equals to 1 and the beginning of the RTP
payload corresponds to a start code. (See the descriptions of
timestamp and marker bit in Section 5.1.)
(5) It is RECOMMENDED that a single video packet is sent as a single
RTP packet. The size of a video packet SHOULD be adjusted in
such a way that the resulting RTP packet is not larger than the
Path MTU. If the video packet is disabled by the coder
configuration (by setting resync_marker_disable in the VOL
header to 1), or in coding tools where the video packet is not
supported, a VOP MAY be split at arbitrary byte positions.
The video packet starts with the VOP header or the video packet
header, followed by motion_shape_texture(), and ends with
next_resync_marker() or next_start_code().
5.3. Examples of Packetized MPEG-4 Visual Bitstream
Figure 2 shows examples of RTP packets generated based on the
criteria described in Section 5.2
(a) is an example of the first RTP packet or the random access point
of an MPEG-4 Visual bitstream containing the configuration
information. According to criterion (1), the Visual Object Sequence
Header (VS header) is placed at the beginning of the RTP payload,
preceding the Visual Object Header and the Video Object Layer Header
(VO header, VOL header). Since the fragmentation rule defined in
Section 5.2 guarantees that the configuration information, starting
with visual_object_sequence_start_code, is always placed at the
beginning of the RTP payload, RTP receivers can detect the random
access point by checking if the first 32-bit field of the RTP payload
is visual_object_sequence_start_code.
(b) is another example of the RTP packet containing the configuration
information. It differs from example (a) in that the RTP packet also
contains a VOP header and a video packet in the VOP following the
configuration information. Since the length of the configuration
information is relatively short (typically scores of bytes) and an
RTP packet containing only the configuration information may thus
increase the overhead, the configuration information and the
subsequent VOP can be packetized into a single RTP packet.
(c) is an example of an RTP packet that contains
Group_of_VideoObjectPlane (GOV). Following criterion (1), the GOV is
placed at the beginning of the RTP payload. It would be a waste of
RTP/IP header overhead to generate an RTP packet containing only a
GOV whose length is 7 bytes. Therefore, the following VOP (or a part
of it) can be placed in the same RTP packet as shown in (c).
(d) is an example of the case where one video packet is packetized
into one RTP packet. When the packet-loss rate of the underlying
network is high, this kind of packetization is recommended. Even
when the RTP packet containing the VOP header is discarded by a
packet loss, the other RTP packets can be decoded by using the HEC
(Header Extension Code) information in the video packet header. No
extra RTP header field is necessary.
(e) is an example of the case where more than one video packet is
packetized into one RTP packet. This kind of packetization is
effective to save the overhead of RTP/IP headers when the bitrate of
the underlying network is low. However, it will decrease the packet-
loss resiliency because multiple video packets are discarded by a
single RTP packet loss. The optimal number of video packets in an
RTP packet and the length of the RTP packet can be determined by
considering the packet-loss rate and the bitrate of the underlying
network.
(f) is an example of the case when the video packet is disabled by
setting resync_marker_disable in the VOL header to 1. In this case,
a VOP may be split into a plurality of RTP packets at arbitrary byte
positions. For example, it is possible to split a VOP into fixed-
length packets. This kind of coder configuration and RTP packet
fragmentation may be used when the underlying network is guaranteed
to be error-free.
Figure 3 shows examples of RTP packets prohibited by the criteria of
Section 5.2.
Fragmentation of a header into multiple RTP packets, as in Figure
3(a), will not only increase the overhead of RTP/IP headers but also
decrease the error resiliency. Therefore, it is prohibited by
criterion (3).
When concatenating more than one video packet into an RTP packet, the
VOP header or video_packet_header() is not allowed to be placed in
the middle of the RTP payload. The packetization as in Figure 2(b)
is not allowed by criterion (2) due to the aspect of the error
resiliency. Comparing this example with Figure 2(d), although two
video packets are mapped onto two RTP packets in both cases, the
packet-loss resiliency is not identical. Namely, if the second RTP
packet is lost, both video packets 1 and 2 are lost in the case of
Figure 3(b), whereas only video packet 2 is lost in the case of
Figure 2(d).
+------+------+------+------+
(a) | RTP | VS | VO | VOL |
|header|header|header|header|
+------+------+------+------+
+------+------+------+------+------+------------+
(b) | RTP | VS | VO | VOL | VOP |Video Packet|
|header|header|header|header|header| |
+------+------+------+------+------+------------+
+------+-----+------------------+
(c) | RTP | GOV |Video Object Plane|
|header| | |
+------+-----+------------------+
+------+------+------------+ +------+------+------------+
(d) | RTP | VOP |Video Packet| | RTP | VP |Video Packet|
|header|header| (1) | |header|header| (2) |
+------+------+------------+ +------+------+------------+
+------+------+------------+------+------------+------+------------+
(e) | RTP | VP |Video Packet| VP |Video Packet| VP |Video Packet|
|header|header| (1) |header| (2) |header| (3) |
+------+------+------------+------+------------+------+------------+
+------+------+------------+ +------+------------+
(f) | RTP | VOP |VOP fragment| | RTP |VOP fragment|
|header|header| (1) | |header| (2) | . . .
+------+------+------------+ +------+------------+
Figure 2: Examples of RTP Packetized MPEG-4 Visual Bitstream
+------+-------------+ +------+------------+------------+
(a) | RTP |First half of| | RTP |Last half of|Video Packet|
|header| VP header | |header| VP header | |
+------+-------------+ +------+------------+------------+
+------+------+----------+ +------+---------+------+------------+
(b) | RTP | VOP |First half| | RTP |Last half| VP |Video Packet|
|header|header| of VP(1) | |header| of VP(1)|header| (2) |
+------+------+----------+ +------+---------+------+------------+
Figure 3: Examples of Prohibited RTP Packetization for MPEG-4 Visual
6. RTP Packetization of MPEG-4 Audio Bitstreams
This section specifies RTP packetization rules for MPEG-4 Audio
bitstreams. MPEG-4 Audio streams MUST be formatted LATM (Low-
overhead MPEG-4 Audio Transport Multiplex) [14496-3] streams, and the
LATM-based streams are then mapped onto RTP packets as described in
the sections below.
6.1. RTP Packet Format
LATM-based streams consist of a sequence of audioMuxElements that
include one or more PayloadMux elements that carry the audio frames.
A complete audioMuxElement or a part of one SHALL be mapped directly
onto an RTP payload without any removal of audioMuxElement syntax
elements (see Figure 4). The first byte of each audioMuxElement
SHALL be located at the first payload location in an RTP packet.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number |RTP
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |Header
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| .... |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| |RTP
: audioMuxElement (byte aligned) :Payload
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4 - An RTP packet for MPEG-4 Audio
In order to decode the audioMuxElement, the following
muxConfigPresent information is required to be indicated by out-of-
band means. When SDP is utilized for this indication, the media type
parameter "cpresent" corresponds to the muxConfigPresent information
(see Section 7.3). The following restrictions apply:
o In the out-of-band configuration case, the number of PayloadMux
elements contained in each audioMuxElement can only be set once.
If more than one PayloadMux element is contained in each
audioMuxElement, special care is required to ensure that the last
RTP packet remains decodable.
o To construct the audioMuxElement in the in-band configuration
case, non-octet-aligned configuration data is inserted immediately
before the one or more PayloadMux elements. Since the generation
of RTP payloads with non-octet-aligned data is not possible with
RTP hint tracks, as defined by the MP4 file format [14496-12]
[14496-14], this document does not support RTP hint tracks for the
in-band configuration case.
muxConfigPresent: If this value is set to 1 (in-band mode), the
audioMuxElement SHALL include an indication bit "useSameStreamMux"
and MAY include the configuration information for audio compression
"StreamMuxConfig". The useSameStreamMux bit indicates whether the
StreamMuxConfig element in the previous frame is applied in the
current frame. If the useSameStreamMux bit indicates to use the
StreamMuxConfig from the previous frame, but if the previous frame
has been lost, the current frame may not be decodable. Therefore, in
case of in-band mode, the StreamMuxConfig element SHOULD be
transmitted repeatedly depending on the network condition. On the
other hand, if muxConfigPresent is set to 0 (out-of-band mode), the
StreamMuxConfig element is required to be transmitted by an out-of-
band means. In case of SDP, the media type parameter "config" is
utilized (see Section 7.3).
6.2. Use of RTP Header Fields for MPEG-4 Audio
Payload Type (PT): The assignment of an RTP payload type for this
packet format is outside the scope of this document and will only be
restricted 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 by means of an out-of-band signaling protocol
(e.g., H.245, SIP). In the dynamic assignment of RTP payload types
for scalable streams, the server SHALL assign a different value to
each layer. The dependency relationships between the enhanced layer
and the base layer MUST be signaled as specified in [RFC5583]. An
example of the use of such signaling for scalable audio streams can
be found in [RFC5691].
Marker (M) bit: The marker bit indicates audioMuxElement boundaries.
It is set to 1 to indicate that the RTP packet contains a complete
audioMuxElement or the last fragment of an audioMuxElement.
Timestamp: The timestamp indicates the sampling instance of the first
audio frame contained in the RTP packet. Timestamps are RECOMMENDED
to start at a random value for security reasons.
Unless specified by an out-of-band means, the resolution of the
timestamp is set to its default value of 90 kHz.
Sequence Number: Incremented by 1 for each RTP packet sent, starting,
for security reasons, with a random value.
Other header fields are used as described in [RFC3550].
6.3. Fragmentation of MPEG-4 Audio Bitstream
It is RECOMMENDED to put one audioMuxElement in each RTP packet. If
the size of an audioMuxElement can be kept small enough that the size
of the RTP packet containing it does not exceed the size of the Path
MTU, this will be no problem. If it cannot, the audioMuxElement
SHALL be fragmented and spread across multiple packets.
7. Media Type Registration for MPEG-4 Audio/Visual Streams
The following sections describe the media type registrations for
MPEG-4 Audio/Visual streams, which are registered in accordance with
[RFC4855] and use the template of [RFC4288]. Media type registration
and SDP usage for the MPEG-4 Visual stream are described in Sections
7.1 and 7.2, respectively, while media type registration and SDP
usage for MPEG-4 Audio stream are described in Sections 7.3 and 7.4,
respectively.
7.1. Media Type Registration for MPEG-4 Visual
The receiver MUST ignore any unspecified parameter in order to ensure
that additional parameters can be added in any future revision of
this specification.
Type name: video
Subtype name: MP4V-ES
Required parameters: none
Optional parameters:
"rate": This parameter is used only for RTP transport. It
indicates the resolution of the timestamp field in the RTP header.
If this parameter is not specified, its default value of 90000 (90
kHz) is used.
"profile-level-id": A decimal representation of MPEG-4 Visual
Profile and Level indication value (profile_and_level_indication)
defined in Table G-1 of [14496-2]. This parameter MAY be used in
the capability exchange or session setup procedure to indicate the
MPEG-4 Visual Profile and Level combination of which the MPEG-4
Visual codec is capable. If this parameter is not specified by
the procedure, its default value of 1 (Simple Profile/Level 1) is
used.
"config": This parameter SHALL be used to indicate the
configuration of the corresponding MPEG-4 Visual bitstream. It
SHALL NOT be used to indicate the codec capability in the
capability exchange procedure. It is a hexadecimal representation
of an octet string that expresses the MPEG-4 Visual configuration
information, as defined in Subclause 6.2.1 ("Start codes") of
[14496-2]. The configuration information is mapped onto the octet
string most significant bit (MSB) first. The first bit of the
configuration information SHALL be located at the MSB of the first
octet. 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 they
exist), which may vary in the repeated configuration information
inside an MPEG-4 Visual stream. (See Subclause 6.2.1, "Start
codes", of [14496-2].)
Published specification:
The specifications for MPEG-4 Visual streams are presented in
[14496-2]. The RTP payload format is described in [RFC6416].
Encoding considerations:
Video bitstreams MUST be generated according to MPEG-4 Visual
specifications [14496-2]. A video bitstream is binary data and
MUST be encoded for non-binary transport (for email, the Base64
encoding is sufficient). This type is also defined for transfer
via RTP. The RTP packets MUST be packetized according to the
MPEG-4 Visual RTP payload format defined in [RFC6416].
Security considerations:
See Section 10 of [RFC6416].
Interoperability considerations:
MPEG-4 Visual 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 Visual 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').
The visual stream SHALL be compliant with the MPEG-4 Visual
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" or by arranging a
capability exchange/announcement procedure for this parameter.
Applications that use this media type:
Audio and visual streaming and conferencing tools
Additional information: none
Person and email address to contact for further information:
See Authors' Addresses section at the end of [RFC6416].
Intended usage: COMMON
Author:
See Authors' Addresses section at the end of [RFC6416].
Change controller:
IETF Audio/Video Transport Payloads working group delegated from
the IESG.
7.2. Mapping to SDP for MPEG-4 Visual
The media type video/MP4V-ES string is mapped to fields in SDP
[RFC4566], as follows:
o The media type (video) goes in SDP "m=" as the media name.
o The Media subtype (MP4V-ES) goes in SDP "a=rtpmap" as the encoding
name.
o The optional parameter "rate" goes in "a=rtpmap" as the "clock
rate".
o The optional parameter "profile-level-id" and "config" go in the
"a=fmtp" line to indicate the coder capability and configuration,
respectively. These parameters are expressed as a string, in the
form of a semicolon-separated list of parameter=value pairs.
Example usages for the "profile-level-id" parameter are:
1 : MPEG-4 Visual Simple Profile/Level 1
34 : MPEG-4 Visual Core Profile/Level 2
145: MPEG-4 Visual Advanced Real Time Simple Profile/Level 1
7.2.1. Declarative SDP Usage for MPEG-4 Visual
The following are some examples of media representations in SDP:
Simple Profile/Level 1, rate=90000(90 kHz), "profile-level-id" and
"config" are present in "a=fmtp" line:
m=video 49170/2 RTP/AVP 98
a=rtpmap:98 MP4V-ES/90000
a=fmtp:98 profile-level-id=1;config=000001B001000001B50900000100000
00120008440FA282C2090A21F
Core Profile/Level 2, rate=90000(90 kHz), "profile-level-id" is
present in "a=fmtp" line:
m=video 49170/2 RTP/AVP 98
a=rtpmap:98 MP4V-ES/90000
a=fmtp:98 profile-level-id=34
Advance Real Time Simple Profile/Level 1, rate=90000(90 kHz),
"profile-level-id" is present in "a=fmtp" line:
m=video 49170/2 RTP/AVP 98
a=rtpmap:98 MP4V-ES/90000
a=fmtp:98 profile-level-id=145
7.3. Media Type Registration for MPEG-4 Audio
The receiver MUST ignore any unspecified parameter, to ensure that
additional parameters can be added in any future revision of this
specification.
Type name: audio
Subtype name: MP4A-LATM
Required parameters:
"rate": the "rate" parameter indicates the RTP timestamp "clock
rate". The default value is 90000. Other rates MAY be indicated
only if they are set to the same value as the audio sampling rate
(number of samples per second).
In the presence of SBR, the sampling rates for the core encoder/
decoder and the SBR tool are different in most cases. Therefore,
this parameter SHALL NOT be considered as the definitive sampling
rate. If this parameter is used, the server must follow the rules
below:
* When the presence of SBR is not explicitly signaled by the
optional SDP parameters such as "object", "profile-level-id",
or "config", this parameter SHALL be set to the core codec
sampling rate.
* When the presence of SBR is explicitly signaled by the optional
SDP parameters such as "object", "profile-level-id", or
"config", this parameter SHALL be set to the SBR sampling rate.
NOTE: The optional parameter "SBR-enabled" in SDP "a=fmtp" is
useful for implicit HE AAC / HE AAC v2 signaling. But the
"SBR-enabled" parameter can also be used in the case of explicit
HE AAC / HE AAC v2 signaling. Therefore, its existence (in
itself) is not the criteria to determine whether or HE AAC / HE
AAC v2 signaling is explicit.
Optional parameters:
"profile-level-id": a decimal representation of MPEG-4 Audio
Profile Level indication value defined in [14496-3]. This
parameter indicates which MPEG-4 Audio tool subsets the decoder is
capable of using. If this parameter is not specified in the
capability exchange or session setup procedure, its default value
of 30 (Natural Audio Profile/Level 1) is used.
"MPS-profile-level-id": a decimal representation of the MPEG
Surround Profile Level indication as defined in [14496-3]. This
parameter indicates the support of the MPEG Surround profile and
level by the decoder to be capable to decode the stream.
"object": a decimal representation of the MPEG-4 Audio Object Type
value defined in [14496-3]. This parameter specifies the tool to
be used by the decoder. It CAN be used to limit the capability
within the specified "profile-level-id".
"bitrate": the data rate for the audio bitstream.
"cpresent": a boolean parameter that indicates whether audio
payload configuration data has been multiplexed into an RTP
payload (see Section 6.1). A 0 indicates the configuration data
has not been multiplexed into an RTP payload, and in that case,
the "config" parameter MUST be present; a 1 indicates that it has
been multiplexed. The default if the parameter is omitted is 1.
If this parameter is set to 1 and the "config" parameter is
present, the multiplexed configuration data and the value of the
"config" parameter SHALL be consistent.
"config": a hexadecimal representation of an octet string that
expresses the audio payload configuration data "StreamMuxConfig",
as defined in [14496-3]. Configuration data is mapped onto the
octet string in an MSB-first basis. The first bit of the
configuration data SHALL be located at the MSB of the first octet.
In the last octet, zero-padding bits, if necessary, SHALL follow
the configuration data. Senders MUST set the StreamMuxConfig
elements taraBufferFullness and latmBufferFullness to their
largest respective value, indicating that buffer fullness measures
are not used in SDP. Receivers MUST ignore the value of these two
elements contained in the "config" parameter.
"MPS-asc": a hexadecimal representation of an octet string that
expresses audio payload configuration data "AudioSpecificConfig",
as defined in [14496-3]. If this parameter is not present, the
relevant signaling is performed by other means (e.g., in-band or
contained in the "config" string).
The same mapping rules as for the "config" parameter apply.
"ptime": duration of each packet in milliseconds.
"SBR-enabled": a boolean parameter that indicates whether SBR-data
can be expected in the RTP-payload of a stream. This parameter is
relevant for an SBR-capable decoder if the presence of SBR cannot
be detected from an out-of-band decoder configuration (e.g.,
contained in the "config" string).
If this parameter is set to 0, a decoder MAY expect that SBR is
not used. If this parameter is set to 1, a decoder CAN up-sample
the audio data with the SBR tool, regardless of whether or not SBR
data is present in the stream.
If the presence of SBR cannot be detected from out-of-band
configuration and the "SBR-enabled" parameter is not present, the
parameter defaults to 1 for an SBR-capable decoder. If the
resulting output sampling rate or the computational complexity is
not supported, the SBR tool can be disabled or run in down-sampled
mode.
The timestamp resolution at the RTP layer is determined by the
"rate" parameter.
Published specification:
Encoding specifications are provided in [14496-3]. The RTP
payload format specification is described in [RFC6416].
Encoding considerations:
This type is only defined for transfer via RTP.
Security considerations:
See Section 10 of [RFC6416].
Interoperability considerations:
MPEG-4 Audio provides a large and rich set of tools for the coding
of audio objects. For effective implementation of the standard,
subsets of the MPEG-4 Audio tool sets similar to those used in
MPEG-4 Visual have been provided (see Section 7.1).
The audio stream SHALL be compliant with the MPEG-4 Audio Profile@
Level specified by the parameters "profile-level-id" and
"MPS-profile-level-id". Interoperability between a sender and a
receiver may be achieved by specifying the parameters
"profile-level-id" and "MPS-profile-level-id" or by arranging in
the capability exchange procedure to set this parameter mutually
to the same value. Furthermore, the "object" parameter can be
used to limit the capability within the specified Profile@Level in
the capability exchange.
Applications that use this media type:
Audio and video streaming and conferencing tools.
Additional information: none
Personal and email address to contact for further information:
See Authors' Addresses section at the end of [RFC6416].
Intended usage: COMMON
Author:
See Authors' Addresses section at the end of [RFC6416].
Change controller:
IETF Audio/Video Transport Payloads working group delegated from
the IESG.
7.4. Mapping to SDP for MPEG-4 Audio
The media type audio/MP4A-LATM string is mapped to fields in SDP
[RFC4566], as follows:
o The media type (audio) goes in SDP "m=" as the media name.
o The Media subtype (MP4A-LATM) goes in SDP "a=rtpmap" as the
encoding name.
o The required parameter "rate" goes in "a=rtpmap" as the "clock
rate".
o The optional parameter "ptime" goes in SDP "a=ptime" attribute.
o The optional parameters "profile-level-id",
"MPS-profile-level-id", and "object" go in the "a=fmtp" line to
indicate the coder capability.
The following are some examples of the "profile-level-id" value:
1 : Main Audio Profile Level 1
9 : Speech Audio Profile Level 1
15: High Quality Audio Profile Level 2
30: Natural Audio Profile Level 1
44: High Efficiency AAC Profile Level 2
48: High Efficiency AAC v2 Profile Level 2
55: Baseline MPEG Surround Profile (see ISO/IEC 23003-1) Level 3
The optional payload-format-specific parameters "bitrate",
"cpresent", "config", "MPS-asc", and "SBR-enabled" also go in the
"a=fmtp" line. These parameters are expressed as a string, in the
form of a semicolon-separated list of parameter=value pairs.
7.4.1. Declarative SDP Usage for MPEG-4 Audio
The following sections contain some examples of the media
representation in SDP.
Note that the "a=fmtp" line in some of the examples has been wrapped
to fit the page; they would comprise a single line in the SDP file.
7.4.1.1. Example: In-Band Configuration
In this example, the audio configuration data appears in the RTP
payload exclusively (i.e., the MPEG-4 audio configuration is known
when a StreamMuxConfig element appears within the RTP payload).
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/90000
a=fmtp:96 object=2; cpresent=1
The "clock rate" is set to 90 kHz. This is the default value, and
the real audio sampling rate is known when the audio configuration
data is received.
7.4.1.2. Example: 6 kbit/s CELP
This example shows a 6 kbit/s CELP (Code-Excited Linear Prediction)
bitstream (with an audio sampling rate of 8 kHz).
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/8000
a=fmtp:96 profile-level-id=9; object=8; cpresent=0;
config=40008B18388380
a=ptime:20
In this example, audio configuration data is not multiplexed into the
RTP payload and is described only in SDP. Furthermore, the "clock
rate" is set to the audio sampling rate.
7.4.1.3. Example: 64 kbit/s AAC LC Stereo
This example shows a 64 kbit/s AAC LC stereo bitstream (with an audio
sampling rate of 24 kHz).
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/24000/2
a=fmtp:96 profile-level-id=1; bitrate=64000; cpresent=0;
object=2; config=400026203fc0
In this example, audio configuration data is not multiplexed into the
RTP payload and is described only in SDP. Furthermore, the "clock
rate" is set to the audio sampling rate.
In this example, the presence of SBR cannot be determined by the SDP
parameter set. The "clock rate" represents the core codec sampling
rate. An SBR-enabled decoder can use the SBR tool to up-sample the
audio data if the complexity and resulting output sampling rate
permit.
7.4.1.4. Example: Use of the "SBR-enabled" Parameter
These two examples are identical to the example above with the
exception of the "SBR-enabled" parameter. The presence of SBR is not
signaled by the SDP parameters "object", "profile-level-id", and
"config", but instead the "SBR-enabled" parameter is present. The
"rate" parameter and the StreamMuxConfig contain the core codec
sampling rate.
This example shows "SBR-enabled=0", with definitive and core codec
sampling rates of 24 kHz.
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/24000/2
a=fmtp:96 profile-level-id=1; bitrate=64000; cpresent=0;
SBR-enabled=0; config=400026203fc0
This example shows "SBR-enabled=1", with core codec sampling rate of
24 kHz, and definitive and SBR sampling rates of 48 kHz:
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/24000/2
a=fmtp:96 profile-level-id=1; bitrate=64000; cpresent=0;
SBR-enabled=1; config=400026203fc0
In this example, the "clock rate" is still 24000, and this
information is used for RTP timestamp calculation. The value of
24000 is used to support old AAC decoders. This makes the decoder
supporting only AAC understand the HE AAC coded data, although only
plain AAC is supported. A HE AAC decoder is able to generate output
data with the SBR sampling rate.
7.4.1.5. Example: Hierarchical Signaling of SBR
When the presence of SBR is explicitly signaled by the SDP parameters
"object", "profile-level-id", or "config", as in the example below,
the StreamMuxConfig contains both the core codec sampling rate and
the SBR sampling rate.
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/48000/2
a=fmtp:96 profile-level-id=44; bitrate=64000; cpresent=0;
config=40005623101fe0; SBR-enabled=1
This "config" string uses the explicit signaling mode 2.A
(hierarchical signaling; see [14496-3]. This means that the AOT
(Audio Object Type) is SBR (5) and SFI (Sampling Frequency Index) is
6 (24000 Hz), which refers to the underlying core codec sampling
frequency. CC (Channel Configuration) is stereo (2), and the ESFI
(Extension Sampling Frequency Index)=3 (48000) is referring to the
sampling frequency of the extension tool (SBR).
7.4.1.6. Example: HE AAC v2 Signaling
HE AAC v2 decoders are required to always produce a stereo signal
from a mono signal. Hence, there is no parameter necessary to signal
the presence of PS.
This example shows "SBR-enabled=1" with 1 channel signaled in the
"a=rtpmap" line and within the "config" parameter. The core codec
sampling rate is 24 kHz; the definitive and SBR sampling rates are 48
kHz. The core codec channel configuration is mono; the PS channel
configuration is stereo.
m=audio 49230 RTP/AVP 110
a=rtpmap:110 MP4A-LATM/24000/1
a=fmtp:110 profile-level-id=15; object=2; cpresent=0;
config=400026103fc0; SBR-enabled=1
7.4.1.7. Example: Hierarchical Signaling of PS
This example shows 48 kHz stereo audio input.
m=audio 49230 RTP/AVP 110
a=rtpmap:110 MP4A-LATM/48000/2
a=fmtp:110 profile-level-id=48; cpresent=0; config=4001d613101fe0
The "config" parameter indicates explicit hierarchical signaling of
PS and SBR. This configuration method is not supported by legacy AAC
an HE AAC decoders, and these are therefore unable to decode the
coded data.
7.4.1.8. Example: MPEG Surround
The following examples show how MPEG Surround configuration data can
be signaled using SDP. The configuration is carried within the
"config" string in the first example by using two different layers.
The general parameters in this example are: AudioMuxVersion=1;
allStreamsSameTimeFraming=1; numSubFrames=0; numProgram=0;
numLayer=1. The first layer describes the HE AAC payload and signals
the following parameters: ascLen=25; audioObjectType=2 (AAC LC);
extensionAudioObjectType=5 (SBR); samplingFrequencyIndex=6 (24 kHz);
extensionSamplingFrequencyIndex=3 (48 kHz); channelConfiguration=2
(2.0 channels). The second layer describes the MPEG Surround payload
and specifies the following parameters: ascLen=110;
AudioObjectType=30 (MPEG Surround); samplingFrequencyIndex=3 (48
kHz); channelConfiguration=6 (5.1 channels); sacPayloadEmbedding=1;
SpatialSpecificConfig=(48 kHz; 32 slots; 525 tree; ResCoding=1;
ResBands=[7,7,7,7]).
In this example, the signaling is carried by using two different LATM
layers. The MPEG Surround payload is carried together with the AAC
payload in a single layer as indicated by the sacPayloadEmbedding
Flag.
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/48000
a=fmtp:96 profile-level-id=1; bitrate=64000; cpresent=0;
SBR-enabled=1;
config=8FF8004192B11880FF0DDE3699F2408C00536C02313CF3CE0FF0
7.4.1.9. Example: MPEG Surround with Extended SDP Parameters
The following example is an extension of the configuration given
above by the MPEG-Surround-specific parameters. The "MPS-asc"
parameter specifies the MPEG Surround Baseline Profile at Level 3
(PLI55), and the "MPS-asc" string contains the hexadecimal
representation of the MPEG Surround ASC [audioObjectType=30 (MPEG
Surround); samplingFrequencyIndex=0x3 (48 kHz);
channelConfiguration=6 (5.1 channels); sacPayloadEmbedding=1;
SpatialSpecificConfig=(48 kHz; 32 slots; 525 tree; ResCoding=1;
ResBands=[0,13,13,13])].
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/48000
a=fmtp:96 profile-level-id=44; bitrate=64000; cpresent=0;
config=40005623101fe0; MPS-profile-level-id=55;
MPS-asc=F1B4CF920442029B501185B6DA00;
7.4.1.10. Example: MPEG Surround with Single-Layer Configuration
The following example shows how MPEG Surround configuration data can
be signaled using the SDP "config" parameter. The configuration is
carried within the "config" string using a single layer. The general
parameters in this example are: AudioMuxVersion=1;
allStreamsSameTimeFraming=1; numSubFrames=0; numProgram=0;
numLayer=0. The single layer describes the combination of HE AAC and
MPEG Surround payload and signals the following parameters:
ascLen=101; audioObjectType=2 (AAC LC); extensionAudioObjectType=5
(SBR); samplingFrequencyIndex=7 (22.05 kHz);
extensionSamplingFrequencyIndex=7 (44.1 kHz); channelConfiguration=2
(2.0 channels). A backward-compatible extension according to
[14496-3/Amd.1] signals the presence of MPEG Surround payload data
and specifies the following parameters: SpatialSpecificConfig=(44.1
kHz; 32 slots; 525 tree; ResCoding=0).
In this example, the signaling is carried by using a single LATM
layer. The MPEG Surround payload is carried together with the HE AAC
payload in a single layer.
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/44100
a=fmtp:96 profile-level-id=44; bitrate=64000; cpresent=0;
SBR-enabled=1; config=8FF8000652B920876A83A1F440884053620FF0;
MPS-profile-level-id=55
8. IANA Considerations
This document updates the media subtypes "MP4A-LATM" and "MP4V-ES"
from RFC 3016. The new registrations are in Sections 7.1 and 7.3 of
this document.
9. Acknowledgements
The authors would like to thank Yoshihiro Kikuchi, Yoshinori Matsui,
Toshiyuki Nomura, Shigeru Fukunaga, and Hideaki Kimata for their work
on RFC 3016, and Ali Begen, Keith Drage, Roni Even, and Qin Wu for
their valuable input and comments on this document.
10. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification [RFC3550] and in any applicable RTP profile. The main
security considerations for the RTP packet carrying the RTP payload
format defined within this document are confidentiality, integrity,
and source authenticity. Confidentiality is achieved by encryption
of the RTP payload, and integrity of the RTP packets is achieved
through a suitable cryptographic integrity protection mechanism. A
cryptographic system may also allow the authentication of the source
of the payload. A suitable security mechanism for this RTP payload
format should provide confidentiality, integrity protection, and (at
least) source authentication capable of determining whether or not an
RTP packet is from a member of the RTP session.
Note that most MPEG-4 codecs define an extension mechanism to
transmit extra data within a stream that is gracefully skipped by
decoders that do not support this extra data. This may be used to
transmit unwanted data in an otherwise valid stream.
The appropriate mechanism to provide security to RTP and payloads
following this may vary. It is dependent on the application, the
transport, and the signaling protocol employed. Therefore, a single
mechanism is not sufficient, although, if suitable, the usage of the
Secure Real-time Transport Protocol (SRTP) [RFC3711] is recommended.
Other mechanisms that may be used are IPsec [RFC4301] and Transport
Layer Security (TLS) [RFC5246] (e.g., for RTP over TCP), but other
alternatives may also exist.
This RTP payload format and its media decoder do not exhibit any
significant non-uniformity in the receiver-side computational
complexity for packet processing, and thus are unlikely to pose a
denial-of-service threat due to the receipt of pathological data.
The complete MPEG-4 System allows for transport of a wide range of
content, including Java applets (MPEG-J) and scripts. Since this
payload format is restricted to audio and video streams, it is not
possible to transport such active content in this format.
11. Differences to RFC 3016
The RTP payload format for MPEG-4 Audio as specified in RFC 3016 is
used by the 3GPP PSS service [3GPP]. However, there are some
misalignments between RFC 3016 and the 3GPP PSS specification that
are addressed by this update:
o The audio payload format (LATM) referenced in this document is the
newer format specified in [14496-3], which is binary compatible to
the format used in [3GPP]. This newer format is not binary
compatible with the LATM referenced in RFC 3016, which is
specified in [14496-3:1999/Amd.1:2000].
o The audio signaling format (StreamMuxConfig) referenced in this
document is binary compatible to the format used in [3GPP]. The
StreamMuxConfig element has also been revised by MPEG since RFC
3016.
o The use of an audio parameter "SBR-enabled" is now defined in this
document, which is used by 3GPP implementations [3GPP]. RFC 3016
does not define this parameter.
o The "rate" parameter is defined unambiguously in this document for
the case of presence of SBR (Spectral Band Replication). In RFC
3016, the definition of the "rate" parameter is ambiguous.
o The number of audio channels parameter is defined unambiguously in
this document for the case of presence of PS (Parametric Stereo).
At the time RFC 3016 was written, PS was not yet defined.
Furthermore, some comments have been addressed and signaling support
for MPEG Surround [23003-1] was added.
Below is a summary of the changes in requirements by this update:
o In the dynamic assignment of RTP payload types for scalable MPEG-4
Audio streams, the server SHALL assign a different value to each
layer.
o The dependency relationships between the enhanced layer and the
base layer for scalable MPEG-4 Audio streams MUST be signaled as
specified in [RFC5583].
o If the size of an audioMuxElement is so large that the size of the
RTP packet containing it does exceed the size of the Path MTU, the
audioMuxElement SHALL be fragmented and spread across multiple
packets.
o The receiver MUST ignore any unspecified parameter in order to
ensure that additional parameters can be added in any future
revision of this specification.
12. References
12.1. Normative References
[14496-2] MPEG, "ISO/IEC International Standard 14496-2 - Coding of
audio-visual objects, Part 2: Visual", 2003.
[14496-3] MPEG, "ISO/IEC International Standard 14496-3 - Coding of
audio-visual objects, Part 3 Audio", 2009.
[14496-3/Amd.1]
MPEG, "ISO/IEC International Standard 14496-3 - Coding of
audio-visual objects, Part 3: Audio, Amendment 1: HD-AAC
profile and MPEG Surround signaling", 2009.
[23003-1] MPEG, "ISO/IEC International Standard 23003-1 - MPEG
Surround (MPEG D)", 2007.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4288] Freed, N. and J. Klensin, "Media Type Specifications and
Registration Procedures", BCP 13, RFC 4288, December 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4629] Ott, H., Bormann, C., Sullivan, G., Wenger, S., and R.
Even, "RTP Payload Format for ITU-T Rec", RFC 4629,
January 2007.
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding
Dependency in the Session Description Protocol (SDP)",
RFC 5583, July 2009.
12.2. Informative References
[14496-1] MPEG, "ISO/IEC International Standard 14496-1 - Coding of
audio-visual objects, Part 1 Systems", 2004.
[14496-12] MPEG, "ISO/IEC International Standard 14496-12 - Coding of
audio-visual objects, Part 12 ISO base media file format".
[14496-14] MPEG, "ISO/IEC International Standard 14496-14 - Coding of
audio-visual objects, Part 12 MP4 file format".
[14496-3:1999/Amd.1:2000]
MPEG, "ISO/IEC International Standard 14496-3 - Coding of
audio-visual objects, Part 3 Audio, Amendment 1: Audio
extensions", 2000.
[3GPP] 3GPP, "3rd Generation Partnership Project; Technical
Specification Group Services and System Aspects;
Transparent end-to-end Packet-switched Streaming Service
(PSS); Protocols and codecs (Release 9)", 3GPP TS 26.234
V9.5.0, December 2010.
[H245] International Telecommunication Union, "Control protocol
for multimedia communication", ITU Recommendation H.245,
December 2009.
[H261] International Telecommunication Union, "Video codec for
audiovisual services at p x 64 kbit/s", ITU
Recommendation H.261, March 1993.
[H323] International Telecommunication Union, "Packet-based
multimedia communications systems", ITU
Recommendation H.323, December 2009.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[RFC3016] Kikuchi, Y., Nomura, T., Fukunaga, S., Matsui, Y., and H.
Kimata, "RTP Payload Format for MPEG-4 Audio/Visual
Streams", RFC 3016, November 2000.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3640] van der Meer, J., Mackie, D., Swaminathan, V., Singer, D.,
and P. Gentric, "RTP Payload Format for Transport of
MPEG-4 Elementary Streams", RFC 3640, November 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4301] Kent, S. and K. Seo, "Security Architecture for the
Internet Protocol", RFC 4301, December 2005.
[RFC4628] Even, R., "RTP Payload Format for H.263 Moving RFC 2190 to
Historic Status", RFC 4628, January 2007.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC5691] de Bont, F., Doehla, S., Schmidt, M., and R.
Sperschneider, "RTP Payload Format for Elementary Streams
with MPEG Surround Multi-Channel Audio", RFC 5691,
October 2009.
Authors' Addresses
Malte Schmidt
Dolby Laboratories
Deutschherrnstr. 15-19
90537 Nuernberg
DE
Phone: +49 911 928 91 42
EMail: malte.schmidt@dolby.com
Frans de Bont
Philips Electronics
High Tech Campus 36
5656 AE Eindhoven
NL
Phone: +31 40 2740234
EMail: frans.de.bont@philips.com
Stefan Doehla
Fraunhofer IIS
Am Wolfmantel 33
91058 Erlangen
DE
Phone: +49 9131 776 6042
EMail: stefan.doehla@iis.fraunhofer.de
Jaehwan Kim
LG Electronics Inc.
VCS/HE, 16Fl. LG Twin Towers
Yoido-Dong, YoungDungPo-Gu,
Seoul 150-721
Korea
Phone: +82 10 6225 0619
EMail: kjh1905m@naver.com