One document matched: draft-schmidt-avt-rfc3016bis-02.xml
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<rfc category="std"
docName="draft-schmidt-avt-rfc3016bis-02.txt"
ipr="trust200902"
obsoletes="3016"
submissionType="IETF"
updates=""
xml:lang="en">
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<!-- ***** FRONT MATTER ***** -->
<front>
<!-- The abbreviated title is used in the page header - it is only necessary if the
full title is longer than 39 characters -->
<title abbrev="RTP Payload Format for MPEG-4 Streams">RTP Payload Format
for MPEG-4 Audio/Visual Streams</title>
<!-- add 'role="editor"' below for the editors if appropriate -->
<!-- Another author who claims to be an editor -->
<author fullname="Malte Schmidt"
initials="M.S."
surname="Schmidt">
<organization>Dolby Laboratories</organization>
<address>
<postal>
<street>Deutschherrnstr. 15-19</street>
<!-- Reorder these if your country does things differently -->
<city>90537 Nuernberg</city>
<region></region>
<country>DE</country>
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<phone>+49 911 928 91 42</phone>
<email>malte.schmidt@dolby.com</email>
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</address>
</author>
<author fullname="Frans de Bont"
initials="F.d.B."
surname="de Bont">
<organization>Philips Electronics</organization>
<address>
<postal>
<street>High Tech Campus 5</street>
<!-- Reorder these if your country does things differently -->
<city>5656 AE Eindhoven</city>
<region></region>
<country>NL</country>
</postal>
<phone>+31 40 2740234</phone>
<email>frans.de.bont@philips.com</email>
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</address>
</author>
<author fullname="Stefan Doehla"
initials="S.D."
surname="Doehla">
<organization>Fraunhofer IIS</organization>
<address>
<postal>
<street>Am Wolfmantel 33</street>
<!-- Reorder these if your country does things differently -->
<city>91058 Erlangen</city>
<region></region>
<country>DE</country>
</postal>
<phone>+49 9131 776 6042</phone>
<email>stefan.doehla@iis.fraunhofer.de</email>
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</address>
</author>
<author fullname="Jaehwan Kim"
initials="Jaehwan"
surname="Kim">
<organization>Vidiator (Korea) Inc.</organization>
<address>
<postal>
<street>7th Fl. AnnJay BLDG 718-2 YeokSam-Dong, KangNam-Gu</street>
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<city>135-920, Seoul</city>
<region></region>
<country>Korea</country>
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<phone>+82 70 7012 2540</phone>
<email>jaehwan@vidiator.com, kjh1905m@naver.com</email>
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<author fullname="Yoshihiro Kikuchi"
initials="Y."
surname="Kikuchi">
<organization>Toshiba corporation</organization>
<address>
<postal>
<street>1, Komukai Toshiba-cho, Saiwai-ku</street>
<city>Kawasaki, 212-8582</city>
<region></region>
<country>JP</country>
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<email>yoshihiro.kikuchi@toshiba.co.jp</email>
</address>
</author>
<author fullname="Yoshinori Matsui"
initials="Y."
surname="Matsui">
<organization>Matsushita Electric Industrial Co., LTD.</organization>
<address>
<postal>
<street>1006, Kadoma, Kadoma-shi, </street>
<city>Osaka</city>
<region></region>
<country>JP</country>
</postal>
<email>matsui@drl.mei.co.jp</email>
</address>
</author>
<author fullname="Toshiyuki Nomura"
initials="T."
surname="Nomura">
<organization>NEC Corporation</organization>
<address>
<postal>
<street>4-1-1, Miyazaki, Miyamae-ku</street>
<city>Kawasaki</city>
<region></region>
<country>JP</country>
</postal>
<email>t-nomura@ccm.cl.nec.co.jp</email>
</address>
</author>
<author fullname="Shigeru Fukunaga"
initials="S."
surname="Fukunaga">
<organization>Oki Electric Industry Co., Ltd.</organization>
<address>
<postal>
<street>1-2-27 Shiromi, Chuo-ku</street>
<city>Osaka 540-6025</city>
<region></region>
<country>JP</country>
</postal>
<email>fukunaga444@oki.co.jp</email>
</address>
</author>
<author fullname="Hideaki Kimata"
initials="H."
surname="Kimata">
<organization>Nippon Telegraph and Telephone Corporation</organization>
<address>
<postal>
<street>1-1, Hikari-no-oka, Yokosuka-shi</street>
<city>Kanagawa</city>
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<email>kimata@nttvdt.hil.ntt.co.jp</email>
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<date year="2009" />
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<area>Real-time Applications and Infrastructure</area>
<workgroup>AVT</workgroup>
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If this element is not present, the default is "Network Working Group",
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<keyword>RFC3016, RTP, MPEG-4, Audio, Visual, Video, AAC, HE AAC, HE AAC v2, MPEG
Surround</keyword>
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<abstract>
<t>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. For the purpose of directly mapping MPEG-4
Audio/Visual bitstreams onto RTP packets, it 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 Session Description Protocol (SDP).</t>
<t>Comments are solicited and should be addressed to the working group's
mailing list at avt@ietf.org and/or the author(s).</t>
</abstract>
</front>
<!-- ***************************************************************** -->
<middle>
<section title="Introduction" toc="default">
<t>The RTP payload formats described in this document specify how MPEG-4
Audio <xref target="14496-3" /> and MPEG-4 Visual streams <xref
target="14496-2" /> <xref target="14496-2/Amd.1" /> are to be fragmented
and mapped directly onto RTP packets.</t>
<t>These RTP payload formats enable transport of MPEG-4 Audio/Visual
streams without using the synchronization and stream management
functionality of MPEG-4 Systems <xref target="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 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. The streams are directly mapped onto RTP
packets without using the MPEG-4 Systems Sync Layer. Other examples
are SIP and 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 an 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.</t>
<t>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 resilience tools provided inside the
MPEG-4 Video stream.</t>
<section title="MPEG-4 Visual RTP payload format" toc="default">
<t>MPEG-4 Visual is a visual coding standard with many new features:
high coding efficiency; high error resiliency; multiple, arbitrary
shape object-based coding; etc. <xref target="14496-2" />. It covers a
wide range of bitrate from scores of Kbps to several Mbps. It also
covers a wide variety of networks, ranging from those guaranteed to be
almost error-free to mobile networks with high error rates.</t>
<t>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 resilience
functionalities of MPEG-4 Visual.</t>
<t>The fragmentation rule recommends not to map more than one VOP in
an RTP packet 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 3.2 and marker bit and
timestamp in section 3.1.)</t>
<t>While the additional media specific RTP header defined for such
video coding tools as H.261 or MPEG-1/2 is effective in helping to
recover picture headers corrupted by packet losses, MPEG-4 Visual has
already error resilience 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/TS, etc.). Therefore, no extra RTP
header fields are defined in this MPEG-4 Visual RTP payload
format.</t>
</section>
<section title="MPEG-4 Audio RTP payload format" toc="default">
<t>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.</t>
<t>While LATM has several multiplexing features as follows;
<list hangIndent="0" style="symbols">
<t>Carrying configuration information with audio data,</t>
<t>Concatenation of multiple audio frames in one audio stream,</t>
<t>Multiplexing multiple objects (programs),</t>
<t>Multiplexing scalable layers,</t>
</list>
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.</t>
<t>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
<xref target="I-D.ietf-mmusic-decoding-dependency" /> SHOULD be used
(see section 4.2).</t>
<t>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 (<xref target="RFC5109"> RFC 5109
</xref>) and Redundant Audio Data (<xref target="RFC2198"> RFC 2198
</xref>), MAY be applied to improve error resiliency.</t>
</section>
<section title="Differences to RFC 3016" toc="default">
<t>The RTP payload format for MPEG-4 Audio as specified in RFC 3016 is used by the
<xref target="3GPP" >3GPP PSS service</xref>. However,
there are some misalignments between RFC 3016 and the 3GPP PSS
specification that are addressed by this update:
<list hangIndent="0" style="symbols">
<t>The audio payload format (LATM) referenced in RFC 3016 is binary
incompatible to the format used in 3GPP.</t>
<t>The audio signaling format (StreamMuxConfig) referenced in RFC
3016 is binary incompatible to the format used in 3GPP.</t>
<t>The audio parameter "SBR-enabled" is not defined within RFC 3016
but used by 3GPP</t>
<t>The rate parameter specification is ambiguous in the presence of
SBR (Spectral Band Replication)</t>
<t>The number of audio channel parameter specification is ambiguous
in the presence of PS (Parametric Stereo) </t>
</list>
Furthermore some comments have been addressed and signaling
support for MPEG surround <xref target="23003-1" /> was added. It should
be noted that the audio payload format described here has some known
limitations. For new system designs <xref target="RFC3640"> RFC 3640
</xref> is recommended.</t>
</section>
</section>
<section title="Definitions and Abbreviations" toc="default">
<t>This memo makes use of terms, specified in <xref target="14496-2" />, <xref
target="14496-3" />, and <xref target="23003-1" />. In addition,
the following terms are used in this document and have specific
meaning within the context of this document.</t>
<t>Core codec sampling rate:
<list hangIndent="0" style="empty">
<t>Audio codec sampling rate. When SBR (Spectral Band Replication)
is used, typically the double value of this will be regarded as the definitive
sampling rate (i.e., the decoder's output sampling rate) </t>
<t>Note: The exception is downsampled SBR mode in which the
SBR sampling rate equals the core codec sampling rate.</t>
</list>
</t>
<t>Core codec channel configuration:
<list hangIndent="0" style="empty">
<t>Audio codec channel configuration. When PS (Parametric Stereo)
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.</t>
</list>
</t>
<t>SBR sampling rate:
<list hangIndent="0" style="empty">
<t>When SBR is used, typically the sampling rate is the double value
of the core codec sampling rate, with the exception of downsampled
SBR mode, where the SBR sampling rate and core codec sampling rate
are identical.</t>
</list>
</t>
<t>Abbreviations:
<list hangIndent="0" style="empty">
<t>AAC: Advanced Audio Coding</t>
<t>ASC: AudioSpecificConfig</t>
<t>HE AAC: High Efficiency AAC</t>
<t>LATM: Low-overhead MPEG-4 Audio Transport Multiplex</t>
<t>PS: Parametric Stereo</t>
<t>SBR: Spectral Band Replication</t>
<t>VOP: Video Object Plane</t>
</list>
</t>
<t>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 <xref
target="RFC2119" />.</t>
</section>
<section title="RTP Packetization of MPEG-4 Visual bitstream"
toc="default">
<t>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 6.2.1 "Start codes" of ISO/IEC
14496-2 <xref target="14496-2" /> <xref target="14496-2/Cor.1" /> <xref
target="14496-2/Amd.1" />) The configuration information MAY
additionally be specified by some out-of-band means. If needed for an
H.323 terminal, H.245 codepoint "decoderConfigurationInformation" MUST
be used for this purpose. 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 5.1 and
5.2).</t>
<t>When the short video header mode is used, the RTP payload format for
H.263 SHOULD be used (the format defined in <xref target="RFC4629">
RFC 4629 </xref> is RECOMMENDED, but the <xref target="RFC4628">
RFC 4628 </xref> format MAY be used for compatibility with
older implementations).</t>
<t>
<figure>
<artwork>
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
</artwork>
</figure>
</t>
<section title="Use of RTP header fields for MPEG-4 Visual"
toc="default">
<t>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, etc).</t>
<t>Extension (X) bit: Defined by the RTP profile used.</t>
<t>Sequence Number: Incremented by one for each RTP data packet sent,
starting, for security reasons, with a random initial value.</t>
<t>Marker (M) bit: The marker bit is set to one 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 one.</t>
<t>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.
<list hangIndent="1" style="symbols">
<t>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 are derived from the timestamp fields in the VOP header
(modulo_time_base and vop_time_increment).</t>
<t>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.</t>
<t>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.</t>
</list></t>
<t>The resolution of the timestamp is set to its default value of
90kHz, unless specified by an out-of-band means (e.g., SDP parameter
or Media Type parameter as defined in section 5).</t>
<t>Other header fields are used as described in RFC 3550 <xref
target="RFC3550" />.</t>
</section>
<section title="Fragmentation of MPEG-4 Visual bitstream" toc="default">
<t>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. The Combined Configuration/Elementary
streams mode is used. The following rules apply for the
fragmentation.</t>
<t>In the following, header means one of the following: <list
hangIndent="1" style="symbols">
<t>Configuration information (Visual Object Sequence Header,
Visual Object Header and Video Object Layer Header)</t>
<t>visual_object_sequence_end_code</t>
<t>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())</t>
<t>The video packet header (video_packet_header() excluding
next_resync_marker())</t>
<t>The header of gob_layer()</t>
<t>See 6.2.1 "Start codes" of ISO/IEC 14496-2 <xref
target="14496-2" /> <xref target="14496-2/Cor.1" /> <xref
target="14496-2/Amd.1" /> for the definition of the configuration
information and the entry point functions.</t>
</list></t>
<t>(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.</t>
<t>(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.</t>
<t>(3) A header SHALL NOT be split into a plurality of RTP
packets.</t>
<t>(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.</t>
<t>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 one and the beginning of RTP payload corresponds
to a start code. (See timestamp and marker bit in section 3.1.)</t>
<t>(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. Note: Rule (5) does not apply when 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. In this case, a VOP MAY be split at arbitrary
byte-positions.</t>
<t>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().</t>
</section>
<section title="Examples of packetized MPEG-4 Visual bitstream"
toc="default">
<t>Figure 2 shows examples of RTP packets generated based on the
criteria described in 3.2</t>
<t>(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
3.2 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.</t>
<t>(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 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 immediately
following GOV and/or (a part of) VOP can be packetized into a single
RTP packet as in this example.</t>
<t>(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, (a part of) the following VOP can
be placed in the same RTP packet as shown in (c).</t>
<t>(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.</t>
<t>(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 bit-rate 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
considering the packet-loss rate and the bit-rate of the underlying
network.</t>
<t>(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. On the other hand, it is not recommended to use it in
error-prone environment since it provides only poor packet loss
resiliency.</t>
<t>Figure 3 shows examples of RTP packets prohibited by the criteria
of 3.2.</t>
<t>Fragmentation of a header into multiple RTP packets, as in (a),
will not only increase the overhead of RTP/IP headers but also
decrease the error resiliency. Therefore, it is prohibited by the
criterion (3).</t>
<t>When concatenating more than one video packets into an RTP packet,
VOP header or video_packet_header() shall not be placed in the middle
of the RTP payload. The packetization as in (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).</t>
<t>
<figure>
<artwork>
+------+------+------+------+
(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
</artwork>
</figure>
</t>
<t>
<figure>
<artwork>
+------+-------------+ +------+------------+------------+
(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
bitstream
</artwork>
</figure>
</t>
</section>
</section>
<section anchor="RTP Packetization of MPEG-4 Audio bitstreams"
title="RTP Packetization of MPEG-4 Audio bitstream">
<t>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) <xref target="14496-3" /> streams, and
the LATM-based streams are then mapped onto RTP packets as described in the
sections below.</t>
<section anchor="RTP Packet Format" title="RTP Packet Format"
toc="default">
<t>LATM-based streams consist of a sequence of audioMuxElements that
include one or more PayloadMux elements which 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.</t>
<figure>
<artwork>
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
</artwork>
</figure>
<t>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 5.3). The following restrictions apply: <list
hangIndent="1" style="symbols">
<t>In the out-of-band configuration case the number of PayloadMux
elements contained in each audioMuxElement can only be set once.
If values greater than one PayloadMux Element are used, special
care is required to ensure that the last RTP packet remains
decodable.</t>
<t>In the in-band configuration case the audio frames are in general
not byte aligned. Hinting RTP payload from MP4 file format <xref
target="14496-12" /> <xref target="14496-14" /> is therefore not
possible.</t>
</list></t>
<t>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-band mode), the StreamMuxConfig
element is required to be transmitted by an out-of-band means. In case
of SDP, Media Type parameter "config" is utilized (see section 5.3).</t>
</section>
<section title="Use of RTP Header Fields for MPEG-4 Audio" toc="default">
<t>Payload Type (PT): The assignment of an RTP payload type for this
new packet format is outside the scope of this document, and will not
be specified here. It is expected that the RTP profile for a
particular class of applications will assign a payload type for this
encoding, or if that is not done then a payload type in the dynamic
range shall be chosen by means of an out-of-band signaling protocol
(e.g., H.245, SIP, etc). In the dynamic assignment of RTP payload
types for scalable streams, a different value SHOULD be assigned to
each layer. The dependency relationships between the enhance layer
and the base layer SHOULD be signaled as specified in
<xref target="I-D.ietf-mmusic-decoding-dependency" />. An example
of the use of such signaling for scalable audio streams
can be found in <xref target="I-D.ietf-avt-rtp-mps" />.</t>
<t>Marker (M) bit: The marker bit indicates audioMuxElement
boundaries. It is set to one to indicate that the RTP packet contains
a complete audioMuxElement or the last fragment of an
audioMuxElement.</t>
<t>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.</t>
<t>Unless specified by an out-of-band means, the resolution of the
timestamp is set to its default value of 90 kHz.</t>
<t>Sequence Number: Incremented by one for each RTP packet sent,
starting, for security reasons, with a random value.</t>
<t>Other header fields are used as described in RFC 3550 <xref
target="RFC3550" />.</t>
</section>
<section title="Fragmentation of MPEG-4 Audio bitstream" toc="default">
<t>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
MAY be fragmented and spread across multiple packets.</t>
</section>
</section>
<section title="Media Type registration for MPEG-4 Audio/Visual streams"
toc="default">
<t>The following sections describe the Media Type registrations for
MPEG-4 Audio/Visual streams. Media Type registration and SDP usage for
the MPEG-4 Visual stream are described in Sections 5.1 and 5.2,
respectively, while Media Type registration and SDP usage for MPEG-4
Audio stream are described in Sections 5.3 and 5.4, respectively.</t>
<section title="Media Type registration for MPEG-4 Visual" toc="default">
<t>Media type name: video</t>
<t>Media subtype name: MP4V-ES</t>
<t>Required parameters: none</t>
<t>Optional parameters: <list hangIndent="1" style="empty">
<t>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
(90kHz) is used.</t>
<t>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 ISO/IEC 14496-2 <xref target="14496-2" />
<xref target="14496-2/Amd.1" />. This parameter MAY be used in the
capability exchange or session setup procedure to indicate 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.</t>
<t>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
ISO/IEC14496-2 <xref target="14496-2" /> <xref
target="14496-2/Amd.1" /> <xref target="14496-2/Cor.1" />. The
configuration information is mapped onto the octet string in an
MSB-first basis. 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 exist, which may vary in the
repeated configuration information inside an MPEG-4 Visual stream
(See 6.2.1 Start codes of ISO/IEC14496-2).</t>
<t>Example usages for these parameters are: <list hangIndent="1"
style="symbols">
<t>MPEG-4 Visual Simple Profile/Level 1: Content-type:
video/mp4v-es; profile-level-id=1</t>
<t>MPEG-4 Visual Core Profile/Level 2: Content-type:
video/mp4v-es; profile-level-id=34</t>
<t>MPEG-4 Visual Advanced Real Time Simple Profile/Level 1:
Content-type: video/mp4v-es; profile-level-id=145</t>
</list></t>
</list></t>
<t>Published specification:</t>
<t>
<list hangIndent="1" style="empty">
<t>The specifications for MPEG-4 Visual streams are presented in
ISO/IEC 14469-2 <xref target="14496-2" /> <xref
target="14496-2/Amd.1" /> <xref target="14496-2/Cor.1" />. The RTP
payload format is described in RFC 3016.</t>
</list>
</t>
<t>Encoding considerations:</t>
<t>
<list hangIndent="1" style="empty">
<t>Video bitstreams MUST be generated according to MPEG-4 Visual
specifications (ISO/IEC 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 RFC 3016.</t>
</list>
</t>
<t>Security considerations:</t>
<t>
<list hangIndent="1" style="empty">
<t>See section 7 of RFC 3016.</t>
</list>
</t>
<t>Interoperability considerations:</t>
<t>
<list hangIndent="1" style="empty">
<t>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:
<list hangIndent="0" style="symbols">
<t>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</t>
<t>checking whether MPEG-4 devices comply with the standard
('conformance testing').</t>
</list></t>
</list>
</t>
<t>
<list hangIndent="1" style="empty">
<t>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 in the capability exchange/announcement procedure to
set this parameter mutually to the same value.</t>
</list>
</t>
<t>Applications which use this Media Type:</t>
<t>
<list hangIndent="1" style="empty">
<t>Audio and visual streaming and conferencing tools</t>
</list>
</t>
<t>Additional information: none</t>
<t>Person and email address to contact for further information: <list
hangIndent="1" style="empty">
<t>See Authors' Address section at the end of this document.</t>
</list></t>
<t>Intended usage: COMMON</t>
<t>Author/Change controller: <list hangIndent="1" style="empty">
<t>See Authors' Address section at the end of this document.</t>
</list></t>
</section>
<section title="SDP usage of MPEG-4 Visual" toc="default">
<t>The Media Type video/MP4V-ES string is mapped to fields in the
Session Description Protocol (SDP) <xref target="RFC4566"> </xref>, as follows:
<list hangIndent="1" style="symbols">
<t>The Media Type (video) goes in SDP "m=" as the media name.</t>
<t>The Media subtype (MP4V-ES) goes in SDP "a=rtpmap" as the
encoding name.</t>
<t>The optional parameter "rate" goes in "a=rtpmap" as the clock
rate.</t>
<t>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 as a semicolon separated
list of parameter=value pairs.</t>
</list></t>
<t>The following are some examples of media representation in SDP:</t>
<t>
<figure>
<artwork>
Simple Profile/Level 1, rate=90000(90kHz), "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=000001B001000001B50900000100000001
20008440FA282C2090A21F
Core Profile/Level 2, rate=90000(90kHz), "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(90kHz),
"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
</artwork>
</figure>
</t>
</section>
<section title="Media Type registration of MPEG-4 Audio" toc="default">
<t>Media type name: audio</t>
<t>Media subtype name: MP4A-LATM</t>
<t>Required parameters: <list hangIndent="1" style="empty">
<t>rate: the rate parameter indicates the RTP time stamp clock
rate. The default value is 90000. Other rates MAY be specified
only if they are set to the same value as the audio sampling rate
(number of samples per second).</t>
<t>In the presence of SBR, the sampling rates for the core
en-/decoder and the SBR tool are different in most cases. This
parameter shall therefore not be considered as the definitive
sampling rate. If this parameter is used, the server must
following the rules below:
<list hangIndent="1" style="symbols">
<t>When the presence of SBR is not explicitly signaled by the
optional SDP parameters such as object parameter,
profile-level-id or config string, this parameter shall be set
to the core codec sampling rate.</t>
<t>When the presence of SBR is explicitly signaled by the
optional SDP parameters such as object parameter,
profile-level-id or config string this parameter shall be set
to the SBR sampling rate.</t>
</list></t>
<t>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
itself is not the criteria to determine whether HE AAC / HE AAC v2
signaling is explicit or not. </t>
</list></t>
<t>Optional parameters: <list hangIndent="1" style="empty">
<t>profile-level-id: a decimal representation of MPEG-4 Audio
Profile Level indication value defined in <xref
target="14496-3">ISO/IEC 14496-3</xref>. 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.</t>
<t>
<figure>
<artwork>
Followings are some examples of this 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
</artwork>
</figure>
</t>
<t>MPS-profile-level-id: a decimal representation of the MPEG
Surround Profile Level indication as defined in <xref
target="14496-3">ISO/IEC 14496-3</xref>. This parameter indicates
the MPEG Surround profile and level that the decoder must be
capable in order to decode the stream.</t>
<t>object: a decimal representation of the MPEG-4 Audio Object
Type value defined in <xref target="14496-3">ISO/IEC
14496-3</xref>. This parameter specifies the tool to be used by
the coder. It CAN be used to limit the capability within the
specified "profile-level-id".</t>
<t>bitrate: the data rate for the audio bit stream.</t>
<t>cpresent: a boolean parameter indicates whether audio payload
configuration data has been multiplexed into an RTP payload (see
section 4.1). A 0 indicates the configuration data has not been
multiplexed into an RTP payload, a 1 indicates that it has. The
default if the parameter is omitted is 1.</t>
<t>config: a hexadecimal representation of an octet string that
expresses the audio payload configuration data "StreamMuxConfig",
as defined in <xref target="14496-3">ISO/IEC 14496-3</xref>.
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.</t>
<t>MPS-asc: a hexadecimal representation of an octet string that
expresses audio payload configuration data "AudioSpecificConfig",
as defined in <xref target="14496-3">ISO/IEC 14496-3</xref>. If
this parameter is not present the relevant signaling is performed
by other means (e.g. in-band or contained in the config
string).</t>
<t>The same mapping rules as for the config parameter apply.</t>
<t>ptime: RECOMMENDED duration of each packet in milliseconds.</t>
<t>SBR-enabled: a boolean parameter which 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 can not be detected from an out-of-band decoder
configuration (e.g. contained in the config string).</t>
<t>If this parameter is set to 0, a decoder SHALL expect that SBR
is not used. If this parameter is set to 1, a decoder SHOULD
upsample the audio data with the SBR tool, regardless whether SBR
data is present in the stream or not.</t>
<t>If the presence of SBR can not 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 may be disabled or run in downsampled
mode.</t>
<t>The timestamp resolution at RTP layer is determined by the
rate parameter.</t>
</list></t>
<t>Published specification: <list hangIndent="1" style="empty">
<t>Payload format specifications are described in this document.
Encoding specifications are provided in <xref
target="14496-3">ISO/IEC 14496-3</xref>.</t>
</list></t>
<t>Encoding considerations: <list hangIndent="1" style="empty">
<t>This type is only defined for transfer via RTP.</t>
</list></t>
<t>Security considerations: <list hangIndent="1" style="empty">
<t>See Section 7 of RFC 3016.</t>
</list></t>
<t>Interoperability considerations: <list hangIndent="1" style="empty">
<t>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 5.1).</t>
<t>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 capability exchange.</t>
</list></t>
<t>Applications which use this media type: <list hangIndent="1"
style="empty">
<t>Audio and video streaming and conferencing tools.</t>
</list></t>
<t>Additional information: none</t>
<t>Personal and email address to contact for further information:
<list hangIndent="1" style="empty">
<t>See Authors' Address section at the end of this document.</t>
</list></t>
<t>Intended usage: COMMON</t>
<t>Author/Change controller: <list hangIndent="1" style="empty">
<t>See Authors' Address section at the end of this document.</t>
</list></t>
</section>
<section title="SDP usage of MPEG-4 Audio" toc="default">
<t>The Media Type audio/MP4A-LATM string is mapped to fields in
the Session Description Protocol (SDP) <xref target="RFC4566"> </xref>, as follows:
<list hangIndent="1" style="symbols">
<t>The Media Type (audio) goes in SDP "m=" as the media name.</t>
<t>The Media subtype (MP4A-LATM) goes in SDP "a=rtpmap" as the
encoding name.</t>
<t>The required parameter "rate" goes in "a=rtpmap" as the clock
rate.</t>
<t>The optional parameter "ptime" goes in SDP "a=ptime"
attribute.</t>
<t>The optional parameters "profile-level-id" and
"MPS-profile-level-id" goes in the "a=fmtp" line to indicate the
coder capability. The "object" parameter goes in the "a=fmtp"
attribute. The payload-format-specific parameters "bitrate",
"cpresent", "config", "MPS-asc" and "SBR-enabled" go in the
"a=fmtp" line. These parameters are expressed as a
string, in the form of as a semicolon separated list of
parameter=value pairs.</t>
</list></t>
<t>The following sections contain some examples of the media
representation in SDP.</t>
<t>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.</t>
<section title="Example: In-band configuration" toc="default">
<t>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).
<figure>
<artwork>
m=audio 49230 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/90000
a=fmtp:96 object=2; cpresent=1
</artwork>
</figure>
</t>
<t>The "clock rate" is set to 90kHz. This is the default value and
the real audio sampling rate is known when the audio configuration
data is received.</t>
</section>
<section title="Example: 6kb/s CELP" toc="default">
<t>6 kb/s CELP bitstreams (with an audio sampling rate of 8 kHz)
<figure>
<artwork>
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
</artwork>
</figure>
</t>
<t>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.</t>
</section>
<section title="Example: 64 kb/s AAC LC stereo" toc="default">
<t>64 kb/s AAC LC stereo bitstream (with an audio sampling rate
of 24 kHz)
<figure>
<artwork>
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
</artwork>
</figure>
</t>
<t>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.</t>
<t>In this example, the presence of SBR can not be determined
by the SDP parameter set. The clock rate represents the core
codec sampling rate. An SBR enabled decoder SHOULD use the SBR tool to
upsample the audio data if complexity and resulting output sampling rate permits.</t>
</section>
<section title="Example: Use of the SBR-enabled parameter" toc="default">
<t>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.</t>
<t>Example with "SBR-enabled=0", definitive and core codec sampling rate 24kHz:
<figure>
<artwork>
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
</artwork>
</figure>
</t>
<t>Example with "SBR-enabled=1", core codec sampling rate 24kHz, definitive and SBR sampling rate 48kHz:
<figure>
<artwork>
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
</artwork>
</figure>
</t>
<t> In this example, the clock rate is still 24000 and
this information should be 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 ourput data with the SBR sampling rate.</t>
</section>
<section title="Example: Hierarchical Signaling of SBR" toc="default">
<t>When the presence of SBR is explicitly signaled by the SDP
parameters object, profile-level-id or the config string as in the
example below, the StreamMuxConfig contains both the core codec
sampling rate and the SBR sampling rate.
<figure>
<artwork>
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
</artwork>
</figure>
</t>
<t>This config string uses the explicit signaling mode 2.A
(hierarchical signaling; See <xref target="14496-3">ISO/IEC 14496-3
</xref>). 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).</t>
</section>
<section title="Example: HE AAC v2 Signaling" toc="default">
<t>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.</t>
<t>Example with "SBR-enabled=1" and 1 channel signaled in the a=rtpmap line and within the config parameter.
Core codec sampling rate is 24kHz, definitive and SBR sampling rate is 48kHz.
Core codec channel configuration is mono, PS channel configuration is stereo.
<figure>
<artwork>
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
</artwork>
</figure>
</t>
</section>
<section title="Example: Hierarchical Signaling of PS" toc="default">
<t>Example: 48khz stereo audio input:
<figure>
<artwork>
m=audio 49230 RTP/AVP 110
a=rtpmap:110 MP4A-LATM/48000/2
a=fmtp:110 profile-level-id=48; cpresent=0; config=4001d613101fe0
</artwork>
</figure>
</t>
<t>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 the coded data.</t>
</section>
<section title="Example: MPEG Surround" toc="default">
<t>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 (24kHz);
extensionSamplingFrequencyIndex=3 (48kHz); 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 (48kHz);
channelConfiguration=6 (5.1 channels); sacPayloadEmbedding=1;
SpatialSpecificConfig=(48 kHz; 32 slots; 525 tree; ResCoding=1;
ResBands=[7,7,7,7]).</t>
<t>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.
<figure>
<artwork>
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=9FF8005192B11880FF2DDE3699F2408C00536C02313CF3CE0FF0
</artwork>
</figure>
</t>
</section>
<section title="Example: MPEG Surround with extended SDP parameters"
toc="default">
<t>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 (48kHz); channelConfiguration=6 (5.1
channels); sacPayloadEmbedding=1; SpatialSpecificConfig=(48 kHz; 32
slots; 525 tree; ResCoding=1; ResBands=[0,13,13,13])].
<figure>
<artwork>
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;
</artwork>
</figure>
</t>
</section>
</section>
</section>
<!-- This PI places the pagebreak correctly (before the section title) in the text output. -->
<?rfc needLines="8" ?>
<section title="IANA Considerations" toc="default">
<t>This memo defines additional optional format parameters to the Media
Type "audio" and its subtype "MP4A-LATM", as defined in
<xref target="RFC3016"> RFC 3016 </xref>.
The Media Type parameters are defined in sections 5.1 and 5.3. </t>
<section anchor="Media Type Registration"
title="Media Type Registration">
<t>This memo defines the following additional optional parameters
which SHOULD be used if SBR or MPEG Surround data is present inside
the payload of an AAC elementary stream. <list hangIndent="2"
style="empty">
<t>MPS-profile-level-id: a decimal representation of the MPEG
Surround Profile Level indication as defined in <xref
target="14496-3">ISO/IEC 14496-3</xref>. This parameter indicates
the MPEG Surround profile and level that the decoder must be
capable in order to decode the stream.</t>
<t>MPS-asc: a hexadecimal representation of an octet string that
expresses audio payload configuration data "AudioSpecificConfig",
as defined in <xref target="14496-3">ISO/IEC 14496-3</xref>. If
this parameter is not present the relevant signaling is performed
by other means (e.g. in-band or contained in the config
string).</t>
<t>SBR-enabled: a boolean parameter which 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 can not be detected from an out-of-band decoder
configuration (e.g. contained in the config string).</t>
</list></t>
</section>
<section title="Usage of SDP" toc="default">
<t>It is assumed that the Media Type parameters are conveyed via an
SDP message as specified in <xref target="RFC3016"> RFC 3016 </xref>,
sections 5.2 and 5.4.</t>
</section>
</section>
<!-- Possibly a 'Contributors' section ... -->
<section anchor="Security" title="Security Considerations" toc="default">
<t>RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification <xref target="RFC3550" />. 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.</t>
<t>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.</t>
<t>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 covert channel may be used to transmit
unwanted data in an otherwise valid stream and it is hence recommended to use
<xref target="RFC3711">SRTP</xref> for stream encryption, authentication,
and integrity check.</t>
</section>
</middle>
<!-- ***************************************************************** -->
<!-- *****BACK MATTER ***** -->
<back>
<!-- References split into informative and normative -->
<!-- There are 2 ways to insert reference entries from the citation libraries:
1. define an ENTITY at the top, and use "ampersand character"RFC2629; here (as shown)
2. simply use a PI "less than character"?rfc include="reference.RFC.2119.xml"?> here
(for I-Ds: include="reference.I-D.narten-iana-considerations-rfc2434bis.xml")
Both are cited textually in the same manner: by using xref elements.
If you use the PI option, xml2rfc will, by default, try to find included files in the same
directory as the including file. You can also define the XML_LIBRARY environment variable
with a value containing a set of directories to search. These can be either in the local
filing system or remote ones accessed by http (http://domain/dir/... ).-->
<references title="Normative References">
&RFC2119;
&RFC3016;
&RFC3550;
&RFC4566;
&RFC4629;
&ID_DDP;
<reference anchor="14496-2">
<front>
<title>ISO/IEC International Standard 14496-2 - Coding of
audio-visual objects, Part 2: Visual</title>
<author initials="" surname="MPEG">
<organization />
</author>
<date year="1999" />
</front>
</reference>
<reference anchor="14496-2/Amd.1">
<front>
<title>ISO/IEC International Standard 14496-2 - Coding of
audio-visual objects, Part 2: Visual, Amendment 1: Visual
extensions</title>
<author initials="" surname="MPEG">
<organization />
</author>
<date year="2000" />
</front>
</reference>
<reference anchor="14496-3">
<front>
<title>ISO/IEC International Standard 14496-3 - Coding of
audio-visual objects, Part 3 Audio</title>
<author initials="" surname="MPEG">
<organization />
</author>
<date year="2005" />
</front>
</reference>
<reference anchor="14496-2/Cor.1">
<front>
<title>ISO/IEC International Standard 14496-2 - Coding of
audio-visual objects, Part 2: Visual, Technical corrigendum
1</title>
<author initials="" surname="MPEG">
<organization />
</author>
<date year="2000" />
</front>
</reference>
<reference anchor="23003-1">
<front>
<title>ISO/IEC International Standard 23003-1 - MPEG Surround (MPEG
D)</title>
<author initials="" surname="MPEG">
<organization />
</author>
<date year="2007" />
</front>
</reference>
</references>
<references title="Informative References">
<reference anchor="14496-1">
<front>
<title>ISO/IEC International Standard 14496-1 - Coding of
audio-visual objects, Part 1 Systems</title>
<author initials="" surname="MPEG">
<organization />
</author>
<date year="2004" />
</front>
</reference>
<reference anchor="14496-12">
<front>
<title>ISO/IEC International Standard 14496-12 - Coding of
audio-visual objects, Part 12 ISO base media file format</title>
<author initials="" surname="MPEG">
<organization />
</author>
</front>
</reference>
<reference anchor="14496-14">
<front>
<title>ISO/IEC International Standard 14496-14 - Coding of
audio-visual objects, Part 12 MP4 file format</title>
<author initials="" surname="MPEG">
<organization />
</author>
</front>
</reference>
&RFC2198;
&RFC3640;
&RFC3711;
&RFC4628;
&RFC5109;
&ID_MPS;
<reference anchor="3GPP">
<front>
<title>3rd Generation Partnership Project;
Technical Specification Group Services and System Aspects;
Transparent end-to-end Packet-switched
Streaming Service (PSS);
Protocols and codecs
(Release 8)</title>
<author initials="" surname="3GPP">
<organization />
</author>
<date year="2008" month="September"/>
</front>
<seriesInfo name="3GPP TS" value="24.234 V8.0.0" />
</reference>
</references>
</back>
</rfc>
| PAFTECH AB 2003-2026 | 2026-04-24 05:41:47 |