Network Working Group S. Wenger
Internet Draft M.M. Hannuksela
Document: draft-ietf-avt-rtp-h264-10.txt T. Stockhammer
Expires: January 2005 M. Westerlund
D. Singer
July 2004
RTP payload Format for H.264 Video
Status of this Memo
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Abstract
This memo describes an RTP Payload format for the ITU-T
Recommendation H.264 video codec and the technically identical
ISO/IEC International Standard 14496-10 video codec. The RTP
payload format allows for packetization of one or more Network
Abstraction Layer Units (NALUs), produced by an H.264 video encoder,
in each RTP payload. The payload format has wide applicability
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supporting from simple low-bit rate conversational usage to Internet
video streaming with interleaved transmission, all the way to high
bit-rate video-on-demand applications.
Table of Contents
1. Introduction.......................................................4
1.1. The H.264 codec................................................4
1.2. Parameter Set Concept..........................................5
1.3. Network Abstraction Layer Unit Types...........................6
2. Conventions........................................................7
3. Scope..............................................................7
4. Definitions and Abbreviations......................................7
4.1. Definitions....................................................7
4.2. Abbreviations..................................................9
5. RTP Payload Format.................................................9
5.1. RTP Header Usage...............................................9
5.2. Common structure of the RTP payload format....................12
5.3. NAL Unit Octet Usage..........................................13
5.4. Packetization Modes...........................................15
5.5. Decoding Order Number (DON)...................................16
5.6. Single NAL Unit Packet........................................18
5.7. Aggregation Packets...........................................19
5.8. Fragmentation Units (FUs).....................................27
6. Packetization Rules...............................................30
6.1. Common Packetization Rules....................................31
6.2. Single NAL Unit Mode..........................................31
6.3. Non-Interleaved Mode..........................................32
6.4. Interleaved Mode..............................................32
7. De-Packetization Process (Informative)............................32
7.1. Single NAL Unit and Non-Interleaved Mode......................32
7.2. Interleaved Mode..............................................33
7.3. Additional De-Packetization Guidelines........................35
8. Payload Format Parameters.........................................36
8.1. MIME Registration.............................................36
8.2. SDP Parameters................................................49
8.3. Examples......................................................55
8.4. Parameter Set Considerations..................................57
9. Security Considerations...........................................59
10. Congestion Control...............................................60
11. IANA Consideration...............................................61
12. Informative Appendix: Application Examples.......................61
12.1. Video Telephony according to ITU-T Recommendation H.241 Annex
A..................................................................61
12.2. Video Telephony, No Slice Data Partitioning, No NAL Unit
Aggregation........................................................61
12.3. Video Telephony, Interleaved Packetization Using NAL Unit
Aggregation........................................................62
12.4. Video Telephony, with Data Partitioning......................62
12.5. Video Telephony or Streaming, with FUs and Forward Error
Correction.........................................................63
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12.6. Low-Bit-Rate Streaming.......................................65
12.7. Robust Packet Scheduling in Video Streaming..................66
13. Informative Appendix: Rationale for Decoding Order Number........66
13.1. Introduction.................................................66
13.2. Example of Multi-Picture Slice Interleaving..................67
13.3. Example of Robust Packet Scheduling..........................68
13.4. Robust Transmission Scheduling of Redundant Coded Slices.....72
13.5. Remarks on Other Design Possibilities........................72
14. Acknowledgements.................................................73
15. Full Copyright Statement.........................................73
16. Intellectual Property Notice.....................................73
17. References.......................................................74
17.1. Normative References.........................................74
17.2. Informative References.......................................74
18. RFC Editor Considerations........................................76
Annex A: Changes relative to draft-ietf-avt-rtp-h264-07.txt..........77
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1. Introduction
1.1. The H.264 codec
This memo specifies an RTP payload specification for the video
coding standard known as ITU-T Recommendation H.264 [1] and ISO/IEC
International Standard 14496 Part 10 (both also known as Advanced
Video Coding, AVC) [2]. Recommendation H.264 was approved by ITU-T
on May 2003, and the approved draft specification is available for
public review [9]. In this memo the H.264 acronym is used for the
codec and the standard, but the memo is equally applicable to the
ISO/IEC counterpart of the coding standard.
The H.264 video codec has a very broad application range that covers
all forms of digital compressed video from low bit rate Internet
streaming applications to HDTV broadcast and Digital Cinema
applications with near loss-less coding. The overall performance of
H.264 is as such that bit rate savings of 50% or more, compared to
the current state of technology, are reported. Digital Satellite TV
quality, for example, was reported to be achievable at 1.5 Mbit/s,
compared to the current operation point of MPEG 2 video at around
3.5 Mbit/s [10].
The codec specification [1] itself distinguishes conceptually
between a video coding layer (VCL), and a network abstraction layer
(NAL). The VCL contains the signal processing functionality of the
codec, mechanisms such as transform, quantization, motion
compensated prediction, and a loop filter. It follows the general
concept of most of today's video codecs, a macroblock-based coder
that utilizes inter picture prediction with motion compensation, and
transform coding of the residual signal. The VCL encoder outputs
slices: a bit string that contains the macroblock data of an integer
number of macroblocks, and the information of the slice header
(containing the spatial address of the first macroblock in the
slice, the initial quantization parameter, and similar).
Macroblocks in slices are ordered in scan order unless a different
macroblock allocation is specified, using the so-called Flexible
Macroblock Ordering syntax. In-picture prediction is used only
within a slice. More information is provided in [9].
The Network Abstraction Layer (NAL) encoder encapsulates the slice
output of the VCL encoder into Network Abstraction Layer Units (NAL
units), which are suitable for the transmission over packet networks
or the use in packet oriented multiplex environments. Annex B of
H.264 defines an encapsulation process to transmit such NAL units
over byte-stream oriented networks. In the scope of this memo Annex
B is not relevant.
Internally, the NAL uses NAL units. A NAL unit consists of a one-
byte header and the payload byte string. The header indicates the
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type of the NAL unit, the (potential) presence of bit errors or
syntax violations in the NAL unit payload, and information regarding
the relative importance of the NAL unit for the decoding process.
This RTP payload specification is designed to be unaware of the bit
string in the NAL unit payload.
One of the main properties of H.264 is the complete decoupling of
the transmission time, the decoding time, and the sampling or
presentation time of slices and pictures. The decoding process
specified in H.264 is unaware of time, and the H.264 syntax does not
carry information such as the number of skipped frames (as common in
the form of the Temporal Reference in earlier video compression
standards). Also, there are NAL units that affect many pictures and
are, hence, inherently time-less. For this reason, the handling of
the RTP timestamp requires some special considerations for those NAL
units for which the sampling or presentation time is not defined,
or, at transmission time, unknown.
1.2. Parameter Set Concept
One very fundamental design concept of H.264 is to generate self-
contained packets, to make mechanisms such as the header duplication
of RFC 2429 [12] or MPEG-4's Header Extension Code (HEC) [13]
unnecessary. The way that this was achieved is to decouple
information that is relevant to more than one slice from the media
stream. This higher layer meta information should be sent reliably,
asynchronously and in advance from the RTP packet stream that
contains the slice packets. (Provisions for sending this
information in-band are also available for such applications that do
not have an out-of-band transport channel appropriate for the
purpose.) The combination of the higher-level parameters is called
a parameter set. The H.264 specification includes two types of
parameter sets: sequence parameter set and picture parameter set.
An active sequence parameter set remains unchanged throughout a
coded video sequence, and an active picture parameter set remains
unchanged within a coded picture. The sequence and picture
parameter set structures contain information such as picture size,
optional coding modes employed, and macroblock to slice group map.
In order to be able to change picture parameters (such as the
picture size), without having the need to transmit parameter set
updates synchronously to the slice packet stream, the encoder and
decoder can maintain a list of more than one sequence and picture
parameter set. Each slice header contains a codeword that indicates
the sequence and picture parameter set to be used.
This mechanism allows the decoupling of the transmission of
parameter sets from the packet stream, and the transmission of them
by external means, e.g. as a side effect of the capability exchange,
or through a (reliable or unreliable) control protocol. It may even
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be possible that they get never transmitted but are fixed by an
application design specification.
1.3. Network Abstraction Layer Unit Types
Tutorial information on the NAL design can be found in [14],
[15] and [16].
All NAL units consist of a single NAL unit type octet, which also
co-serves as the payload header of this RTP payload format. The
payload of a NAL unit follows immediately.
The syntax and semantics of the NAL unit type octet are specified in
[1], but the essential properties of the NAL unit type octet are
summarized below. The NAL unit type octet has the following format:
+---------------+
|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+
|F|NRI| Type |
+---------------+
The semantics of the components of the NAL unit type octet, as
specified in the H.264 specification, are described briefly below.
F: 1 bit
forbidden_zero_bit. The H.264 specification declares a value of
1 as a syntax violation.
NRI: 2 bits
nal_ref_idc. A value of 00 indicates that the content of the
NAL unit is not used to reconstruct reference pictures for inter
picture prediction. Such NAL units can be discarded without
risking the integrity of the reference pictures. Values greater
than 00 indicate that the decoding of the NAL unit is required
to maintain the integrity of the reference pictures.
Type: 5 bits
nal_unit_type. This component specifies the NAL unit payload
type as defined in table 7-1 of [1], and later within this memo.
For a reference of all currently defined NAL unit types and
their semantics please refer to section 7.4.1 in [1].
This memo introduces new NAL unit types, which are presented in
section 5.2. The NAL unit types defined in this memo are marked as
unspecified in [1]. Moreover, this specification extends the
semantics of F and NRI as described in section 5.3.
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2. Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [3].
This specification uses the notion of setting and clearing a bit
when handling bit fields. Setting a bit is the same as assigning
that bit the value of 1 (On). Clearing a bit is the same as
assigning that bit the value of 0 (Off).
3. Scope
This payload specification can only be used to carry the "naked"
H.264 NAL unit stream over RTP, and not the bitstream format
discussed in Annex B of H.264. Likely, the first applications of
this specification will be in the conversational multimedia field,
video telephony or video conferencing, but the payload format also
covers other applications such as Internet streaming and TV over IP.
4. Definitions and Abbreviations
4.1. Definitions
This document uses the definitions of [1]. The following terms
defined in [1] are summed up below for convenience:
access unit: A set of NAL units always containing a primary
coded picture. In addition to the primary coded picture, an
access unit may also contain one or more redundant coded
pictures or other NAL units not containing slices or slice data
partitions of a coded picture. The decoding of an access unit
always results in a decoded picture.
coded video sequence: A sequence of access units that consists,
in decoding order, of an instantaneous decoding refresh (IDR)
access unit followed zero or more non-IDR access units including
all subsequent access units up to but not including any
subsequent IDR access unit.
IDR access unit: An access unit in which the primary coded
picture is an IDR picture.
IDR picture: A coded picture containing only slices with I or SI
slice types that causes a "reset" in the decoding process.
After the decoding of an IDR picture all following coded
pictures in decoding order can be decoded without inter
prediction from any picture decoded prior to the IDR picture.
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primary coded picture: The coded representation of a picture to
be used by the decoding process for a bitstream conforming to
H.264. The primary coded picture contains all macroblocks of
the picture.
redundant coded picture: A coded representation of a picture or
a part of a picture. The content of a redundant coded picture
shall not be used by the decoding process for a bitstream
conforming to H.264. The content of a redundant coded picture
may be used by the decoding process for a bitstream that
contains errors or losses.
VCL NAL unit: A collective term used to refer to coded slice and
coded data partition NAL units.
In addition, the following definitions apply:
decoding order number (DON): A field in the payload structure or
a derived variable indicating NAL unit decoding order. Values
of DON are in the range of 0 to 65535, inclusive. After
reaching the maximum value, the value of DON wraps around to 0.
NAL unit decoding order: A NAL unit order that conforms to the
constraints on NAL unit order given in section 7.4.1.2 in [1].
transmission order: The order of packets in ascending RTP
sequence number order (in modulo arithmetic). Within an
aggregation packet, the NAL unit transmission order is the same
as the order of appearance of NAL units in the packet.
Media aware network element (MANE): A network element, such as a
middlebox or (application layer) gateway that is capable of
parsing certain aspects of the RTP payload headers or the RTP
payload, and reacting on the contents.
Informative note: The concept of a MANE goes beyond normal
routers or gateways in that a MANE has to be aware of the
signalling (e.g. to learn about the payload type mappings of
the media streams) and that it has to be trusted when working
with SRTP. The advantage of using MANEs is that they allow to
drop packets according to the needs of the media coding. For
example, if a MANE needs to drop packets due to congestion on
a certain link, it can identify those packets whose dropping
has the smallest negative impact on the user experience, and
remove those in order to remove the congestion and/or keep
the delay low.
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Abbreviations
DON: Decoding Order Number
DONB: Decoding Order Number Base
DOND: Decoding Order Number Difference
FEC: Forward Error Correction
FU: Fragmentation Unit
IDR: Instantaneous Decoding Refresh
IEC: International Electrotechnical Commission
ISO: International Organization for Standardization
ITU-T: International Telecommunication Union, Telecommunication
Standardization Sector
MANE: Media Aware Network Element
MTAP: Multi-Time Aggregation Packet
MTAP16: MTAP with 16-bit timestamp offset
MTAP24: MTAP with 24-bit timestamp offset
NAL: Network Abstraction Layer
NALU: NAL Unit
SEI: Supplemental Enhancement Information
STAP: Single-Time Aggregation Packet
STAP-A: STAP type A
STAP-B: STAP type B
TS: Timestamp
VCL: Video Coding Layer
5. RTP Payload Format
5.1. RTP Header Usage
The format of the RTP header is specified in RFC 3550 [4] and
reprinted in Figure 1 for convenience. This payload format uses the
fields of the header in a manner consistent with that specification.
When encapsulating one NAL unit per RTP packet, the RECOMMENDED RTP
payload format is specified in section 5.6. The RTP payload (and
the settings for some RTP header bits) for aggregation packets and
fragmentation units are specified in sections 5.7 and 5.8,
respectively.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: RTP header according RFC 3550.
The RTP header information to be set according to this RTP payload
format is set as follows:
Marker bit (M): 1 bit
Set for the very last packet of the access unit indicated by the
RTP timestamp, in line with the normal use of the M bit in video
formats and to allow an efficient playout buffer handling. For
aggregation packets (STAP and MTAP) the marker bit in the RTP
header MUST be set to the value that the marker bit of the last
NAL unit of the aggregation packet would have if it were
transported in its own RTP packet. Decoders MAY use this bit as
an early indication of the last packet of an access unit, but
MUST NOT rely on this property.
Informative note: Only one M bit is associated with an
aggregation packet carrying multiple NAL units, and thus if a
gateway has re-packetized an aggregation packet into several
packets, it cannot reliably set the M bit of those packets.
Payload type (PT): 7 bits
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. The assignment of a payload type needs to be performed
either through the profile used or in a dynamic way.
Sequence number (SN): 16 bits
Set and used in accordance with RFC 3550. For the single NALU
and non-interleaved packetization mode, the sequence number is
used to determine decoding order for the NALU.
Timestamp: 32 bits
The RTP timestamp is set to the sampling timestamp of the
content. A 90 kHz clock rate MUST be used.
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If the NAL unit has no timing properties of its own (e.g.
parameter set and SEI NAL units), the RTP timestamp is set to
the RTP timestamp of the primary coded picture of the access
unit in which the NAL unit is included according to section
7.4.1.2 of [1].
The setting of the RTP Timestamp for MTAPs is defined in section
5.7.2.
Receivers SHOULD ignore any picture timing SEI messages included
in access units that have only one display timestamp. Instead,
receivers SHOULD use the RTP timestamp for synchronizing the
display process.
RTP senders SHOULD NOT transmit picture timing SEI messages for
pictures that are not supposed to be displayed as multiple
fields.
In case that one access unit has more than one display timestamp
carried in a picture timing SEI message, then the information in
the SEI message SHOULD be treated as relative to the RTP
timestamp, with the earliest event occurring at the time given
by the RTP timestamp, and subsequent events later, as given by
the difference in SEI message picture timing values. Let tSEI1,
tSEI2, ..., tSEIn be the display timestamps carried in the SEI
message of an access unit, where tSEI1 is the earliest of all
such timestamps. Let tmadjst() be a function that adjusts the
SEI messages time scale to a 90-kHz time scale. Let TS be the
RTP timestamp. Then, the display time for the event associated
with tSEI1 is TS. The display time for the event with tSEIx,
where x is [2..n] is TS + tmadjst (tSEIx - tSEI1).
Informative note: Displaying coded frames as fields is needed
commonly in an operation known as 3:2 pulldown where film
content that consists of coded frames is displayed on an
display using interlaced scanning. The picture timing SEI
message enables carriage of multiple timestamps for the same
coded picture, and therefore the 3:2 pulldown process is
perfectly controlled. The picture timing SEI message
mechanism is necessary, because only one timestamp per coded
frame can be conveyed in the RTP timestamp.
Informative note: Due to the fact that H.264 allows the
decoding order to be different from the display order, values
of RTP timestamps may not be monotonically non-decreasing as
a function of RTP sequence numbers. Furthermore, the value
for interarrival jitter reported in the RTCP reports may not
be a trustworthy indication of the network performance, as
the calculation rules for interarrival jitter (section 6.4.1
of RFC 3550) assume that the RTP timestamp of a packet is
directly proportional to its transmission time.
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5.2. Common structure of the RTP payload format
The payload format defines three different basic payload structures.
A receiver can identify the payload structure by the first byte of
the RTP payload, which co-serves as the RTP payload header and in
some cases as the first byte of the payload. This byte is always
structured as a NAL unit header. The NAL unit type field indicates
which structure is present. The possible structures are:
Single NAL Unit Packet: Contains only a single NAL unit in the
payload. The NAL header type field will be equal to the original
NAL unit type, i.e., in the range of 1 to 23, inclusive. Specified
in section 5.6.
Aggregation packet: Packet type used to aggregate multiple NAL units
into a single RTP payload. This packet exists in four versions, the
Single-Time Aggregation Packet type A (STAP-A), the Single-Time
Aggregation Packet type B (STAP-B), Multi-Time Aggregation Packet
(MTAP) with 16 bit offset (MTAP16), and Multi-Time Aggregation
Packet (MTAP) with 24 bit offset (MTAP24). The NAL unit type
numbers assigned for STAP-A, STAP-B, MTAP16, and MTAP24 are 24, 25,
26, and 27, respectively. Specified in section 5.7.
Fragmentation unit: Used to fragment a single NAL unit over multiple
RTP packets. Exists with two versions, FU-A and FU-B, identified
with the NAL unit type numbers 28 and 29, respectively. Specified
in section 5.8.
Table 1. Summary of NAL unit types and their payload structures.
Type Packet Type name Section
---------------------------------------------------------
0 undefined -
1-23 NAL unit Single NAL unit packet per H.264 5.6
24 STAP-A Single-time aggregation packet 5.7.1
25 STAP-B Single-time aggregation packet 5.7.1
26 MTAP16 Multi-time aggregation packet 5.7.2
27 MTAP24 Multi-time aggregation packet 5.7.2
28 FU-A Fragmentation unit 5.8
29 FU-B Fragmentation unit 5.8
30-31 undefined -
Informative note: This specification does not limit the size of
NAL units encapsulated in single NAL unit packets and
fragmentation units. The maximum size of a NAL unit
encapsulated in any aggregation packet is 65535 bytes.
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5.3. NAL Unit Octet Usage
The structure and semantics of the NAL unit octet were introduced in
section 1.3. For convenience, the format of the NAL unit type octet
is reprinted below:
+---------------+
|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+
|F|NRI| Type |
+---------------+
This section specifies the semantics of F and NRI according to this
specification.
F: 1 bit
forbidden_zero_bit. A value of 0 indicates that the NAL unit
type octet and payload should not contain bit errors or other
syntax violations. A value of 1 indicates that the NAL unit
type octet and payload may contain bit errors or other syntax
violations.
MANEs SHOULD set the F bit to indicate detected bit errors in
the NAL unit. The H.264 specification requires that the F bit
is equal to 0. When the F bit is set, the decoder is advised
that bit errors or any other syntax violation may be present in
the payload or in the NAL unit type octet. The simplest decoder
reaction to respond to a NAL unit in which the F bit is equal to
1 is to discard such a NAL unit and to conceal the lost data in
the discarded NAL unit.
NRI: 2 bits
nal_ref_idc. The semantics of value 00 and a non-zero value
remain unchanged compared to the H.264 specification. In other
words, a value of 00 indicates that the content of the NAL unit
is not used to reconstruct reference pictures for inter picture
prediction. Such NAL units can be discarded without risking the
integrity of the reference pictures. Values greater than 00
indicate that the decoding of the NAL unit is required to
maintain the integrity of the reference pictures.
In addition to the specification above, according to this RTP
payload specification, values of NRI greater than 00 indicate
the relative transport priority, as determined by the encoder.
MANEs can use this information to protect more important NAL
units better than less important NAL units. 11 is the highest
transport priority, followed by 10, then by 01 and, finally, 00
is the lowest.
Informative note: Any non-zero value of NRI is handled
identically in H.264 decoders. Therefore, receivers need not
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manipulate the value of NRI when passing NAL units to the
decoder.
An H.264 encoder MUST set the value of NRI according to the
H.264 specification (subclause 7.4.1), when the value of
nal_unit_type is in the range of 1 to 12, inclusive. In
particular, the H.264 specification requires that the value of
NRI SHALL be equal to 0 for all NAL units having nal_unit_type
equal to 6, 9, 10, 11, or 12.
An H.264 encoder SHOULD set the value of NRI for NAL units
having nal_unit_type equal to 7 or 8 (indicating a sequence
parameter set or a picture parameter set respectively) to 11 (in
binary format). An H.264 encoder SHOULD set the value of NRI for
coded slice NAL units of a primary coded picture having
nal_unit_type equal to 5 (indicating a coded slice belonging to
an IDR picture) to 11 (in binary format).
The following example for a mapping of the remaining
nal_unit_types to NRI values MAY be used and has been shown as
efficient in a certain environment [15]. Other mappings MAY also
be desirable, depending on the application and the H.264/AVC
Annex A profile in use.
Informative Note: Data Partitioning is not available in
certain profiles, e.g. in the Main or Baseline profiles.
Consequently, the nal unit types 2, 3, and 4 can occur only
if the video bit stream conforms to a profile in which data
partitioning is allowed, and not in streams that conform to
the Main or Baseline profiles.
Table 2: Example of NRI values for coded slices and coded slice
data partitions of primary coded reference pictures
NAL Unit Type Content of NAL unit NRI
(binary)
----------------------------------------------------------------
1 non-IDR coded slice 10
2 Coded slice data partition A 10
3 Coded slice data partition B 01
4 Coded slice data partition C 01
Informative note: As mentioned before, the NRI value of non-
reference pictures is 00 as mandated by H.264/AVC.
An H.264 encoder SHOULD set the value of NRI for coded slice and
coded slice data partition NAL units of redundant coded
reference pictures equal to 01 (in binary format).
Definitions of the values for NRI for NAL unit types 24 to 29,
inclusive, are given in sections 5.7 and 5.8 of this memo.
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No recommendation for the value of NRI is given for NAL units
having nal_unit_type in the range of 13 to 23, inclusive,
because these values are reserved for ITU-T and ISO/IEC. No
recommendation for the value of NRI is given for NAL units
having nal_unit_type equal to 0 or in the range of 30 to 31,
inclusive, because the semantics of these values are not
specified in this memo.
5.4. Packetization Modes
This memo specifies three cases of packetization modes:
o Single NAL unit mode
o Non-interleaved mode
o Interleaved mode
The single NAL unit mode is targeted for conversational systems that
comply with ITU-T Recommendation H.241 [17] (see section 12.1). The
non-interleaved mode is targeted for conversational systems that may
not comply with ITU-T Recommendation H.241. In the non-interleaved
mode NAL units are transmitted in NAL unit decoding order. The
interleaved mode is targeted for systems that do not require very
low end-to-end latency. The interleaved mode allows transmission of
NAL units out of NAL unit decoding order.
The packetization mode in use MAY be signaled by the value of the
OPTIONAL packetization-mode MIME parameter or by external means.
The used packetization mode governs which NAL unit types are allowed
in RTP payloads. Table 3 summarizes the allowed NAL unit types for
each packetization mode. Some NAL unit type values (indicated as
undefined in Table 3) are reserved for future extensions. NAL units
of those types SHOULD NOT be sent by a sender, and MUST be ignored
by a receiver. For example, the Types 1-23, with the associated
packet type "NAL unit", are allowed in "Single NAL Unit Mode" and in
"Non-Interleaved Mode", but disallowed in "Interleaved Mode".
Packetization modes are explained in more detail in section 6.
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Table 3. Summary of allowed NAL unit types for each packetization
mode (yes = allowed, no = disallowed, ig = ignore).
Type Packet Single NAL Non-Interleaved Interleaved
Unit Mode Mode Mode
-------------------------------------------------------------
0 undefined ig ig ig
1-23 NAL unit yes yes no
24 STAP-A no yes no
25 STAP-B no no yes
26 MTAP16 no no yes
27 MTAP24 no no yes
28 FU-A no yes yes
29 FU-B no no yes
30-31 undefined ig ig ig
5.5. Decoding Order Number (DON)
In the interleaved packetization mode, the transmission order of NAL
units is allowed to differ from the decoding order of the NAL units.
Decoding order number (DON) is a field in the payload structure or a
derived variable that indicates the NAL unit decoding order.
Rationale and example use cases for transmission out of decoding
order and for the use of DON are given in section 13.
The coupling of transmission and decoding order is controlled by the
OPTIONAL sprop-interleaving-depth MIME parameter as follows. When
the value of the OPTIONAL sprop-interleaving-depth MIME parameter is
equal to 0 (explicitly or per default) or transmission of NAL units
out of their decoding order is disallowed by external means, the
transmission order of NAL units MUST conform to the NAL unit
decoding order. When the value of the OPTIONAL sprop-interleaving-
depth MIME parameter is greater than 0 or transmission of NAL units
out of their decoding order is allowed by external means,
o the order of NAL units in an MTAP16 and an MTAP24 is NOT REQUIRED
to be the NAL unit decoding order, and
o the order of NAL units generated by decapsulating STAP-Bs, MTAPs,
and FUs in two consecutive packets is NOT REQUIRED to be the NAL
unit decoding order.
The RTP payload structures for a single NAL unit packet, an STAP-A,
and an FU-A do not include DON. STAP-B and FU-B structures include
DON, and the structure of MTAPs enables derivation of DON as
specified in section 5.7.2.
Informative note: When an FU-A occurs in interleaved mode, it
always follows an FU-B which sets its DON.
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Informative note: If a transmitter wants to encapsulate a single
NAL unit per packet and transmit packets out of their decoding
order, STAP-B packet type can be used.
In the single NAL unit packetization mode, the transmission order of
NAL units, determined by the RTP sequence number, MUST be the same
as their NAL unit decoding order. In the non-interleaved
packetization mode, the transmission order of NAL units in single
NAL unit packets and STAP-As, and FU-As MUST be the same as their
NAL unit decoding order. The NAL units within an STAP MUST appear
in the NAL unit decoding order. Thus the decoding order is first
provided through the implicit order within a STAP, and second
provided through the RTP sequence number for the order between
STAPs, FUs, and single NAL unit packets.
Signaling of the value of DON for NAL units carried in STAP-B, MTAP,
and a series of fragmentation units starting with an FU-B is
specified in sections 5.7.1, 5.7.2, and 5.8 respectively. The DON
value of the first NAL unit in transmission order MAY be set to any
value. Values of DON are in the range of 0 to 65535, inclusive.
After reaching the maximum value, the value of DON wraps around to
0.
The decoding order of two NAL units contained in any STAP-B, MTAP,
or a series of fragmentation units starting with an FU-B is
determined as follows. Let DON(i) be the decoding order number of
the NAL unit having index i in the transmission order. Function
don_diff(m,n) is specified as follows:
If DON(m) == DON(n), don_diff(m,n) = 0
If (DON(m) < DON(n) and DON(n) - DON(m) < 32768),
don_diff(m,n) = DON(n) - DON(m)
If (DON(m) > DON(n) and DON(m) - DON(n) >= 32768),
don_diff(m,n) = 65536 - DON(m) + DON(n)
If (DON(m) < DON(n) and DON(n) - DON(m) >= 32768),
don_diff(m,n) = - (DON(m) + 65536 - DON(n))
If (DON(m) > DON(n) and DON(m) - DON(n) < 32768),
don_diff(m,n) = - (DON(m) - DON(n))
A positive value of don_diff(m,n) indicates that the NAL unit having
transmission order index n follows, in decoding order, the NAL unit
having transmission order index m. When don_diff(m,n) is equal to
0, then the NAL unit decoding order of the two NAL units can be in
either order. A negative value of don_diff(m,n) indicates that the
NAL unit having transmission order index n precedes, in decoding
order, the NAL unit having transmission order index m.
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Values of DON related fields (DON, DONB, and DOND, see section 5.7)
MUST be such that the decoding order determined by the values of DON
as specified above conforms to the NAL unit decoding order. If the
order of two NAL units in NAL unit decoding order is switched and
the new order does not conform to the NAL unit decoding order, the
NAL units MUST NOT have the same value of DON. If the order of two
consecutive NAL units in the NAL unit stream is switched and the new
order still conforms to the NAL unit decoding order, the NAL units
MAY have the same value of DON. For example, when arbitrary slice
order is allowed by the video coding profile in use, all the coded
slice NAL units of a coded picture are allowed to have the same
value of DON. Consequently, NAL units having the same value of DON
can be decoded in any order, and two NAL units having a different
value of DON should be passed to the decoder in the order specified
above. When two consecutive NAL units in the NAL unit decoding
order have a different value of DON, the value of DON for the second
NAL unit in decoding order SHOULD be the value of DON for the first
NAL unit in decoding order incremented by one.
An example decapsulation process to recover the NAL unit decoding
order is given in section 7.
Informative note: Receivers should not expect that the absolute
difference of values of DON for two consecutive NAL units in the
NAL unit decoding order is equal to one even in case of error-
free transmission. An increment by one is not required, because
at the time of associating values of DON to NAL units, it may
not be known, whether all NAL units are delivered to the
receiver. For example, a gateway may not forward coded slice
NAL units of non-reference pictures or SEI NAL units, when there
is a shortage of bitrate in the network to which the packets are
forwarded. In another example a live broadcast is interrupted
by pre-encoded content such as commercials from time to time.
The first intra picture of a pre-encoded clip is transmitted in
advance to ensure that it is readily available in the receiver.
At the time of transmitting the first intra picture, the
originator does not exactly know how many NAL units are going to
be encoded before the first intra picture of the pre-encoded
clip follows in decoding order. Thus, the values of DON for the
NAL units of the first intra picture of the pre-encoded clip
have to be estimated at the time of transmitting them and gaps
in values of DON may occur.
5.6. Single NAL Unit Packet
The single NAL unit packet defined here MUST contain one and only
one NAL unit of the types defined in [1]. This means that neither
an aggregation packet nor a fragmentation unit can be used within a
single NAL unit packet. A NAL unit stream composed by decapsulating
single NAL unit packets in RTP sequence number order MUST conform to
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the NAL unit decoding order. The structure of the single NAL unit
packet is shown in Figure 2.
Informative note: The first byte of a NAL unit co-serves as the
RTP payload header.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|NRI| type | |
+-+-+-+-+-+-+-+-+ |
| |
| Bytes 2..n of a Single NAL unit |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2. RTP payload format for single NAL unit packet.
5.7. Aggregation Packets
Aggregation packets are the NAL unit aggregation scheme of this
payload specification. The scheme is introduced to reflect the
dramatically different MTU sizes of two key target networks --
wireline IP networks (with an MTU size that is often limited by the
Ethernet MTU size -- roughly 1500 bytes), and IP or non-IP (e.g.
ITU-T H.324/M) based wireless communication systems with preferred
transmission unit sizes of 254 bytes or less. In order to prevent
media transcoding between the two worlds, and to avoid undesirable
packetization overhead, a NAL unit aggregation scheme is introduced.
Two types of aggregation packets are defined by this specification:
o Single-time aggregation packet (STAP) aggregates NAL units with
identical NALU-time. Two types of STAPs are defined, one without
DON (STAP-A) and another one including DON (STAP-B).
o Multi-time aggregation packet (MTAP) aggregates NAL units with
potentially differing NALU-time. Two different MTAPs are defined
that differ in the length of the NAL unit timestamp offset.
The term NALU-time is defined as the value that the RTP timestamp
would have if that NAL unit would be transported in its own RTP
packet.
Each NAL unit to be carried in an aggregation packet is encapsulated
in an aggregation unit. Please see below for the three different
aggregation units and their characteristics.
The structure of the RTP payload format for aggregation packets is
presented in Figure 3.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|NRI| type | |
+-+-+-+-+-+-+-+-+ |
| |
| one or more aggregation units |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3. RTP payload format for aggregation packets.
MTAPs and STAPs share the following packetization rules: The RTP
timestamp MUST be set to the earliest of the NALU times of all the
NAL units to be aggregated. The type field of the NAL unit type
octet MUST be set to the appropriate value as indicated in Table 4.
The F bit MUST be cleared if all F bits of the aggregated NAL units
are zero, otherwise it MUST be set. The value of NRI MUST be the
maximum of all the NAL units carried in the aggregation packet.
Table 4. Type field for STAPs and MTAPs
Type Packet Timestamp offset DON related fields
field length (DON, DONB, DOND)
(in bits) present
--------------------------------------------------------
24 STAP-A 0 no
25 STAP-B 0 yes
26 MTAP16 16 yes
27 MTAP24 24 yes
The marker bit in the RTP header is set to the value the marker bit
of the last NAL unit of the aggregated packet would have if it were
transported in its own RTP packet.
The payload of an aggregation packet consists of one or more
aggregation units. See section 5.7.1 and 5.7.2 for the four
different types of aggregation units. An aggregation packet can
carry as many aggregation units as necessary, however the total
amount of data in an aggregation packet obviously MUST fit into an
IP packet, and the size SHOULD be chosen such that the resulting IP
packet is smaller than the MTU size. An aggregation packet MUST NOT
contain fragmentation units specified in section 5.8. Aggregation
packets MUST NOT be nested, i.e., an aggregation packet MUST NOT
contain another aggregation packet.
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5.7.1. Single-Time Aggregation Packet
Single-time aggregation packet (STAP) SHOULD be used whenever
aggregating NAL units that all share the same NALU-time. The
payload of an STAP-A does not include DON and consists of at least
one single-time aggregation unit as presented in Figure 4. The
payload of an STAP-B consists of a 16-bit unsigned decoding order
number (DON) (in network byte order) followed by at least one
single-time aggregation unit as presented in Figure 5.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: |
+-+-+-+-+-+-+-+-+ |
| |
| single-time aggregation units |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4. Payload format for STAP-A.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: decoding order number (DON) | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
| single-time aggregation units |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5. Payload format for STAP-B.
The DON field specifies the value of DON for the first NAL unit in
an STAP-B in transmission order. The value of DON for each
successive NAL unit in appearance order in an STAP-B is equal to
(the value of DON of the previous NAL unit in the STAP-B + 1) %
65536, in which '%' stands for the modulo operation.
A single-time aggregation unit consists of 16-bit unsigned size
information (in network byte order) that indicates the size of the
following NAL unit in bytes (excluding these two octets, but
including the NAL unit type octet of the NAL unit), followed by the
NAL unit itself including its NAL unit type byte. A single-time
aggregation unit is byte-aligned within the RTP payload but it may
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not be aligned on a 32-bit word boundary. Figure 6 presents the
structure of the single-time aggregation unit.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: NAL unit size | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
| NAL unit |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6. Structure for single-time aggregation unit.
Figure 7 presents an example of an RTP packet that contains an STAP-
A. The STAP contains two single-time aggregation units, labeled as
1 and 2 in the figure.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP Header |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|STAP-A NAL HDR | NALU 1 Size | NALU 1 HDR |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 1 Data |
: |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | NALU 2 Size | NALU 2 HDR |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 2 Data |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7. An example of an RTP packet including a STAP-A and two
single-time aggregation units.
Figure 8 presents an example of an RTP packet that contains an STAP-
B. The STAP contains two single-time aggregation units, labeled as
1 and 2 in the figure.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP Header |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|STAP-B NAL HDR | DON | NALU 1 Size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 1 Size | NALU 1 HDR | NALU 1 Data |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
: |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | NALU 2 Size | NALU 2 HDR |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 2 Data |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8. An example of an RTP packet including an STAP-B and two
single-time aggregation units.
5.7.2. Multi-Time Aggregation Packets (MTAPs)
The NAL unit payload of MTAPs consists of a 16-bit unsigned decoding
order number base (DONB) (in network byte order) and one or more
multi-time aggregation units as presented in Figure 9. DONB MUST
contain the value of DON for the first NAL unit in the NAL unit
decoding order among the NAL units of the MTAP.
Informative note: The first NAL unit in the NAL unit decoding
order is not necessarily the first NAL unit in the order the NAL
units are encapsulated in an MTAP.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: decoding order number base | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
| multi-time aggregation units |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 9. NAL unit payload format for MTAPs.
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Two different multi-time aggregation units are defined in this
specification. Both of them consist of 16 bits unsigned size
information of the following NAL unit (in network byte order), an 8-
bit unsigned decoding order number difference (DOND), and n bits (in
network byte order) of timestamp offset (TS offset) for this NAL
unit, whereby n can be 16 or 24. The choice between the different
MTAP types (MTAP16 and MTAP24) is application dependent -- the
larger the timestamp offset is, the higher is the flexibility of the
MTAP, but the higher is also the overhead.
The structure of the multi-time aggregation units for MTAP16 and
MTAP24 are presented in Figure 10 and Figure 11 respectively. The
starting or ending position of an aggregation unit within a packet
is NOT REQUIRED to be on a 32-bit word boundary. DON of the
following NAL unit is equal to (DONB + DOND) % 65536, in which %
denotes the modulo operation. This memo does not specify how the
NAL units within an MTAP are ordered, but, in most cases, NAL unit
decoding order SHOULD be used.
The timestamp offset field MUST be set to a value equal to the value
of the following formula: If the NALU-time is larger than or equal
to the RTP timestamp of the packet, then the timestamp offset equals
(the NALU-time of the NAL unit - the RTP timestamp of the packet).
If the NALU-time is smaller than the RTP timestamp of the packet,
then the timestamp offset is equal to the NALU-time + (2^32 - the
RTP timestamp of the 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: NAL unit size | DOND | TS offset |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| TS offset | |
+-+-+-+-+-+-+-+-+ NAL unit |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 10. Multi-time aggregation unit for MTAP16
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: NALU unit size | DOND | TS offset |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| TS offset | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| NAL unit |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 11. Multi-time aggregation unit for MTAP24
For the "earliest" multi-time aggregation unit in an MTAP the
timestamp offset MUST be zero. Hence, the RTP timestamp of the MTAP
itself is identical to the earliest NALU-time.
Informative note: The "earliest" multi-time aggregation unit is
the one that has the smallest extended RTP timestamp among all
the aggregation units of an MTAP if the aggregation units were
encapsulated in single NAL unit packets. An extended timestamp
is a timestamp that has more than 32 bits and is capable of
counting the wrap around of the timestamp field, thus enabling
one to actually determine the smallest value if the timestamp
wraps. Such an "earliest" aggregation unit may not be the first
one in the order the aggregation units are encapsulated in an
MTAP. The "earliest" NAL unit need not be the same as the first
NAL unit in the NAL unit decoding order either.
Figure 12 presents an example of an RTP packet that contains a
multi-time aggregation packet of type MTAP16 that contains two
multi-time aggregation units, labeled as 1 and 2 in the figure.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP Header |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|MTAP16 NAL HDR | decoding order number base | NALU 1 Size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 1 Size | NALU 1 DOND | NALU 1 TS offset |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 1 HDR | NALU 1 DATA |
+-+-+-+-+-+-+-+-+ +
: |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | NALU 2 SIZE | NALU 2 DOND |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 2 TS offset | NALU 2 HDR | NALU 2 DATA |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 12. An example of an RTP packet including a multi-time
aggregation packet of type MTAP16 and two multi-time aggregation
units.
Figure 13 presents an example of an RTP packet that contains a
multi-time aggregation packet of type MTAP24 that contains two
multi-time aggregation units, labeled as 1 and 2 in the figure.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP Header |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|MTAP16 NAL HDR | decoding order number base | NALU 1 Size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 1 Size | NALU 1 DOND | NALU 1 TS offs |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|NALU 1 TS offs | NALU 1 HDR | NALU 1 DATA |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
: |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | NALU 2 SIZE | NALU 2 DOND |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 2 TS offset | NALU 2 HDR |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NALU 2 DATA |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 13. An example of an RTP packet including a multi-time
aggregation packet of type MTAP16 and two multi-time aggregation
units.
5.8. Fragmentation Units (FUs)
This payload type allows fragmenting a NAL unit into several RTP
packets. Doing so on the application layer instead of relying on
lower layer fragmentation (e.g. by IP) has the following advantages:
o The payload format is capable of transporting NAL units bigger
than 64 kbytes over an IPv4 network that may be present in pre-
recorded video, particularly in High Definition formats (there is
a limit of the number of slices per picture, which results in a
limit of NAL units per picture, which may result in big NAL units)
o The fragmentation mechanism allows fragmenting a single picture
and applying generic forward error correction as described in
section 12.5.
Fragmentation is defined only for a single NAL unit, and not for any
aggregation packets. A fragment of a NAL unit consists of an
integer number of consecutive octets of that NAL unit. Each octet
of the NAL unit MUST be part of exactly one fragment of that NAL
unit. Fragments of the same NAL unit MUST be sent in consecutive
order with ascending RTP sequence numbers (with no other RTP packets
within the same RTP packet stream being sent between the first and
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last fragment). Similarly, a NAL unit MUST be reassembled in RTP
sequence number order.
When a NAL unit is fragmented and conveyed within fragmentation
units (FUs), it is referred to as fragmented NAL unit. STAPs and
MTAPs MUST NOT be fragmented. FUs MUST NOT be nested, i.e., an FU
MUST NOT contain another FU.
The RTP timestamp of an RTP packet carrying an FU is set to the NALU
time of the fragmented NAL unit.
Figure 14 presents the RTP payload format for FU-As. An FU-A
consists of a fragmentation unit indicator of one octet, a
fragmentation unit header of one octet, and a fragmentation unit
payload.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| FU indicator | FU header | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
| FU payload |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 14. RTP payload format for FU-A.
Figure 15 presents the RTP payload format for FU-Bs. An FU-B
consists of a fragmentation unit indicator of one octet, a
fragmentation unit header of one octet, a decoding order number
(DON) (in network byte order), and a fragmentation unit payload. In
other words, the structure of FU-B is the same as the structure of
FU-A except for the additional DON field.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| FU indicator | FU header | DON |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-|
| |
| FU payload |
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 15. RTP payload format for FU-B.
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NAL unit type FU-B MUST be used in the interleaved packetization
mode for the first fragmentation unit of a fragmented NAL unit. NAL
unit type FU-B MUST NOT be used in any other case. In other words,
in the interleaved packetization mode, each NALU that is fragmented
has an FU-B as the first fragment, followed by one or more FU-A
fragments.
The FU indicator octet has the following format:
+---------------+
|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+
|F|NRI| Type |
+---------------+
Values equal to 28 and 29 in the Type field of the FU indicator
octet identify an FU-A and an FU-B, respectively. The use of the F
bit is described in section 5.3. The value of the NRI field MUST be
set according to the value of the NRI field in the fragmented NAL
unit.
The FU header has the following format:
+---------------+
|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+
|S|E|R| Type |
+---------------+
S: 1 bit
The Start bit, when one, indicates the start of a fragmented NAL
unit. Otherwise, when the following FU payload is not the start
of a fragmented NAL unit payload, the Start bit is set to zero.
E: 1 bit
The End bit, when one, indicates the end of a fragmented NAL
unit, i.e., the last byte of the payload is also the last byte
of the fragmented NAL unit. Otherwise, when the following FU
payload is not the last fragment of a fragmented NAL unit, the
End bit is set to zero.
R: 1 bit
The Reserved bit MUST be equal to 0 and MUST be ignored by the
receiver.
Type: 5 bits
The NAL unit payload type as defined in table 7-1 of [1].
The value of DON in FU-Bs is selected as described in section 5.5.
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Informative note: The DON field in FU-Bs allows gateways to
fragment NAL units to FU-Bs without organizing the incoming NAL
units to the NAL unit decoding order.
A fragmented NAL unit MUST NOT be transmitted in one FU, i.e., Start
bit and End bit MUST NOT both be set to one in the same FU header.
The FU payload consists of fragments of the payload of the
fragmented NAL unit such that if the fragmentation unit payloads of
consecutive FUs are sequentially concatenated, the payload of the
fragmented NAL unit is reconstructed. The NAL unit type octet of
the fragmented NAL unit is not included as such in the fragmentation
unit payload, but rather the information of the NAL unit type octet
of the fragmented NAL unit is conveyed in F and NRI fields of the FU
indicator octet of the fragmentation unit and in the type field of
the FU header. A FU payload MAY have any number of octets and MAY
be empty.
Informative note: Empty FUs are allowed to reduce the latency of
a certain class of senders in near loss-less environments.
Those senders can be characterized in that they packetize NALU
fragments before the NALU is completely generated and hence,
before the NALU size if known. If zero-length NALU fragments
were not allowed, the sender would have to generate at least one
bit of data of the following fragment before the current
fragment could be sent. Due to the characteristics of H.264,
where sometimes several macroblocks occupy zero bits, this is
undesirable and can add delay. However, the (potential) use of
zero-length NALUs should be carefully weighted against the
increase of the risk of the loss of the NALU, because of the
additional packets that are employed for its transmission.
If a fragmentation unit is lost, the receiver SHOULD discard all
following fragmentation units in transmission order corresponding to
the same fragmented NAL unit.
A receiver in an endpoint or in a MANE MAY aggregate the first n-1
fragments of a NAL unit to an (incomplete) NAL unit even if fragment
n of that NAL unit is not received. In this case the
forbidden_zero_bit of the NAL unit MUST be set to one to indicate a
syntax violation.
6. Packetization Rules
The packetization modes are introduced in section 5.2. The
packetization rules that are common to more than one of the
packetization modes are specified in section 6.1. The packetization
rules for the single NAL unit mode, the non-interleaved mode, and
the interleaved mode are specified in sections 6.2, 6.3, and 6.4
respectively.
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6.1. Common Packetization Rules
All senders MUST enforce the following packetization rules
regardless of the packetization mode in use:
o Coded slice NAL units or coded slice data partition NAL units
belonging to the same coded picture (and hence sharing the same
RTP timestamp value) MAY be sent in any order permitted by the
applicable profile defined in [1], although, for delay-critical
systems, they SHOULD be sent in their original coding order to
minimize the delay. Note that the coding order is not necessarily
the scan order, but the order the NAL packets become available to
the RTP stack.
o Parameter sets are handled in accordance with the rules and
recommendations given in section 8.4.
o MANEs MUST NOT duplicate any NAL unit except for sequence or
picture parameter set NAL units, because neither this memo nor the
H.264 specification provides means to identify duplicated NAL
units. Sequence and picture parameter set NAL units MAY be
duplicated to make their correct reception more probable, but any
such duplication MUST NOT affect the contents of any active
sequence or picture parameter set. Duplication SHOULD be
performed on the application layer, and not by duplicating RTP
packets (with identical sequence numbers).
Senders according to the non-interleaved mode and the interleaved
mode MUST enforce the following packetization rule:
o MANEs MAY convert single NAL unit packets into one aggregation
packet, convert an aggregation packet into several single NAL unit
packets, or mix both concepts, in an RTP translator. The RTP
translator SHOULD take into account at least the following
parameters: path MTU size, unequal protection mechanisms (e.g.
through packet-based FEC according to RFC 2733 [21], especially
for sequence and picture parameter set NAL units and coded slice
data partition A NAL units), bearable latency of the system, and
buffering capabilities of the receiver.
Informative note: An RTP translator is required to handle RTCP
as per RFC 3550.
6.2. Single NAL Unit Mode
This mode is in use when the value of the OPTIONAL packetization-
mode MIME parameter is equal to 0 or packetization-mode is not
present or no other packetization mode is signaled by external
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means. All receivers MUST support this mode. It is primarily
intended for low-delay applications that are compatible with systems
using ITU-T Recommendation H.241 [17] (see section 12.1). Only
single NAL unit packets MAY be used in this mode. STAPs, MTAPs, and
FUs MUST NOT be used. The transmission order of single NAL unit
packets MUST comply with the NAL unit decoding order.
6.3. Non-Interleaved Mode
This mode is in use when the value of the OPTIONAL packetization-
mode MIME parameter is equal to 1 or the mode is turned on by
external means. This mode SHOULD be supported. It is primarily
intended for low-delay applications. Only single NAL unit packets,
STAP-As and FU-As MAY be used in this mode. STAP-Bs, MTAPs, and FU-
Bs MUST NOT be used. The transmission order of NAL units MUST
comply with the NAL unit decoding order.
6.4. Interleaved Mode
This mode is in use when the value of the OPTIONAL packetization-
mode MIME parameter is equal to 2 or the mode is turned on by
external means. Some receivers MAY support this mode. STAP-Bs,
MTAPs, FU-As, and FU-Bs MAY be used. STAP-As and single NAL unit
packets MUST NOT be used. The transmission order of packets and NAL
units is constrained as specified in section 5.5.
7. De-Packetization Process (Informative)
The de-packetization process is implementation dependent. Hence,
the following description should be seen as an example of a suitable
implementation. Other schemes may be used as well. Optimizations
relative to the described algorithms are likely possible. Section
7.1 presents the de-packetization process for the single NAL unit
and non-interleaved packetization modes, whereas section 7.2
describes the process for the interleaved mode. Section 7.3
includes additional decapsulation guidelines for intelligent
receivers.
All normal RTP mechanisms related to buffer management apply. In
particular, duplicated or outdated RTP packets (as indicated by the
RTP sequences number and the RTP timestamp) are removed. To
determine the exact time for decoding, factors such as a possible
intentional delay to allow for proper inter-stream synchronization
must be factored in.
7.1. Single NAL Unit and Non-Interleaved Mode
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The receiver includes a receiver buffer to compensate transmission
delay jitter. The receiver stores incoming packets in reception
order into the receiver buffer. Packets are decapsulated in RTP
sequence number order. If a decapsulated packet is a single NAL
unit packet, the NAL unit contained in the packet is passed directly
to the decoder. If a decapsulated packet is an STAP-A, the NAL
units contained in the packet are passed to the decoder in the order
they are encapsulated in the packet. If a decapsulated packet is an
FU-A, all the fragments of the fragmented NAL unit are concatenated
and passed to the decoder.
Informative note: If the decoder supports Arbitrary Slice Order,
coded slices of a picture can be passed to the decoder in any
order regardless of their reception and transmission order.
7.2. Interleaved Mode
The general concept behind these de-packetization rules is to
reorder NAL units from transmission order to the NAL unit decoding
order.
The receiver includes a receiver buffer, which is used to compensate
for transmission delay jitter and to reorder packets from
transmission order to the NAL unit decoding order. In this section,
the receiver operation is described assuming that there is no
transmission delay jitter. To make a difference between a practical
receiver buffer that is also used for compensation of transmission
delay jitter, the receiver buffer is hereinafter called the
deinterleaving buffer in this section. Receivers SHOULD also
prepare for transmission delay jitter, i.e., either reserve separate
buffers for transmission delay jitter buffering and deinterleaving
buffering or use a receiver buffer for both transmission delay
jitter and deinterleaving. Moreover, receivers SHOULD take
transmission delay jitter into account in the buffering operation,
e.g., by additional initial buffering before starting of decoding
and playback.
This section is organized as follows: Subsection 7.2.1 presents how
to calculate the size of the deinterleaving buffer. Subsection
7.2.2 specifies the receiver process how to organize received NAL
units to the NAL unit decoding order.
7.2.1. Size of the Deinterleaving Buffer
When SDP Offer/Answer model or any other capability exchange
procedure is used in session setup, the properties of the received
stream SHOULD be such that the receiver capabilities are not
exceeded. In the SDP Offer/Answer model, the receiver can indicate
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its capabilities to allocate a deinterleaving buffer with the deint-
buf-cap MIME parameter. The sender indicates the requirement for
the deinterleaving buffer size with the sprop-deint-buf-req MIME
parameter. It is therefore RECOMMENDED to set the deinterleaving
buffer size, in terms of number of bytes, equal to or greater than
the value of sprop-deint-buf-req MIME parameter. See section 8.1
for further information on deint-buf-cap and sprop-deint-buf-req
MIME parameters and section 8.2.2 for further information on their
use in SDP Offer/Answer model.
When a declarative session description is used in session setup, the
sprop-deint-buf-req MIME parameter signals the requirement for the
deinterleaving buffer size. It is therefore RECOMMENDED to set the
deinterleaving buffer size, in terms of number of bytes, equal to or
greater than the value of sprop-deint-buf-req MIME parameter.
7.2.2. Deinterleaving Process
There are two buffering states in the receiver: initial buffering
and buffering while playing. Initial buffering occurs when the RTP
session is initialized. After initial buffering, decoding and
playback is started and the buffering-while-playing mode is used.
Regardless of the buffering state the receiver stores incoming NAL
units in reception order into the deinterleaving buffer as follows.
NAL units of aggregation packets are stored into the deinterleaving
buffer individually. The value of DON is calculated and stored for
all NAL units.
The receiver operation is described below with the help of the
following functions and constants:
o Function AbsDON is specified in section 8.1.
o Function don_diff is specified in section 5.5.
o Constant N is the value of the OPTIONAL sprop-interleaving-depth
MIME type parameter (see section 8.1) incremented by 1.
Initial buffering lasts until one of the following conditions is
fulfilled:
o There are N VCL NAL units in the deinterleaving buffer.
o If sprop-max-don-diff is present, don_diff(m,n) is greater than
the value of sprop-max-don-diff, in which n corresponds to the NAL
unit having the greatest value of AbsDON among the received NAL
units and m corresponds to the NAL unit having the smallest value
of AbsDON among the received NAL units.
o Initial buffering has lasted for the duration equal to or greater
than the value of the OPTIONAL sprop-init-buf-time MIME parameter.
The NAL units to be removed from the deinterleaving buffer are
determined as follows:
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o If the deinterleaving buffer contains at least N VCL NAL units,
NAL units are removed from the deinterleaving buffer and passed to
the decoder in the order specified below until the buffer contains
N-1 VCL NAL units.
o If sprop-max-don-diff is present, all NAL units m for which
don_diff(m,n) is greater than sprop-max-don-diff are removed from
the deinterleaving buffer and passed to the decoder in the order
specified below. Herein, n corresponds to the NAL unit having the
greatest value of AbsDON among the received NAL units.
o Variable ts is set to the value of system timer that was
initialized to 0 when the first packet of the NAL unit stream was
received. If the deinterleaving buffer contains a NAL unit whose
reception time tr fulfills the condition that ts - tr > sprop-
init-buf-time, NAL units are passed to the decoder (and removed
from the deinterleaving buffer) in the order specified below until
the deinterleaving buffer contains no NAL unit whose reception
time tr fulfills the specified condition. Note that transmission
delay jitter should be taken into account in the calculations with
timestamps.
The order that NAL units are passed to the decoder is specified as
follows:
o Let PDON be a variable that is initialized to 0 at the beginning
of the an RTP session.
o For each NAL unit associated with a value of DON, a DON distance
is calculated as follows. If the value of DON of the NAL unit is
larger than the value of PDON, the DON distance is equal to DON -
PDON. Otherwise, the DON distance is equal to 65535 - PDON + DON
+ 1.
o NAL units are delivered to the decoder in ascending order of DON
distance. If several NAL units share the same value of DON
distance, they can be passed to the decoder in any order.
o When a desired number of NAL units have been passed to the
decoder, the value of PDON is set to the value of DON for the last
NAL unit passed to the decoder.
7.3. Additional De-Packetization Guidelines
The following additional de-packetization rules may be used to
implement an operational H.264 de-packetizer:
o Intelligent RTP receivers (e.g. in gateways) may identify lost
coded slice data partitions A (DPAs). If a lost DPA is found, a
gateway may decide not to send the corresponding coded slice data
partitions B and C, as their information is meaningless for H.264
decoders. In this way a MANE can reduce network load by
discarding useless packets, without parsing a complex bitstream.
o Intelligent RTP receivers (e.g. in gateways) may identify lost
FUs. If a lost FU is found, a gateway may decide not to send the
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following FUs of the same fragmented NAL unit, as their
information is meaningless for H.264 decoders. In this way a MANE
can reduce network load by discarding useless packets, without
parsing a complex bitstream.
o Intelligent receivers having to discard packets or NALUs should
first discard all packets/NALUs in which the value of the NRI
field of the NAL unit type octet is equal to 0. This will
minimize the impact on user experience and keep the reference
pictures intact. If more packets need to be discarded, then
packets with a numerically lower NRI value should be discarded
before packets with a numerically higher NRI value. However,
discarding any packets with an NRI bigger than 0 very likely leads
to decoder drift and SHOULD be avoided.
8. Payload Format Parameters
This section specifies the parameters that MAY be used to select
optional features of the payload format and certain features of the
bit stream. The parameters are specified here as part of the MIME
subtype registration for the ITU-T H.264 | ISO/IEC 14496-10 codec.
A mapping of the parameters into the Session Description Protocol
(SDP) [5] is also provided for those applications that use SDP.
Equivalent parameters could be defined elsewhere for use with
control protocols that do not use MIME or SDP.
Some parameters provide a receiver with the properties of the stream
that is going to be sent. The name of all these parameters starts
with "sprop" for stream properties. Some of these "sprop"
parameters are limited by other payload or codec configuration
parameters. For example, the sprop-parameter-sets parameter is
constrained by the profile-level-id parameter. The media sender
selects all "sprop" parameters rather than the receiver. This
uncommon characteristic of the "sprop" parameters may not be
compatible with some signaling protocol concepts, in which case the
use of these parameters SHOULD be avoided.
8.1. MIME Registration
The MIME subtype for the ITU-T H.264 | ISO/IEC 14496-10 codec is
allocated from the IETF tree.
The receiver MUST ignore any unspecified parameter.
Media Type name: video
Media subtype name: H264
Required parameters: none
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OPTIONAL parameters:
profile-level-id: A base16 [6] (hexadecimal) representation of
the following three bytes in the sequence
parameter set NAL unit specified in [1]: 1)
profile_idc, 2) a byte herein referred to as
profile-iop, composed of the values of
constraint_set0_flag, constraint_set1_flag,
constraint_set2_flag, and reserved_zero_5bits
in bit-significance order starting from the
most significant bit, and 3) level_idc. Note
that reserved_zero_5bits is required to be
equal to 0 in [1], but other values for it may
be specified in the future by ITU-T or ISO/IEC.
If the profile-level-id parameter is used for
indicating properties of a NAL unit stream, it
indicates the profile and level that a decoder
has to support in order to comply with [1] when
decoding the stream. The profile-iop byte
indicates whether the NAL unit stream also
obeys all constraints of the indicated profiles
as follows. If bit 7 (the most significant
bit), bit 6, or bit 5 of profile-iop is equal
to 1, all constraints of the Baseline profile,
the Main profile, or the Extended profile,
respectively, are obeyed in the NAL unit
stream.
If the profile-level-id parameter is used for
capability exchange or session setup procedure,
it indicates the profile that the codec
supports and the highest level that is
supported for the signaled profile. The
profile-iop byte indicates whether the codec
has such additional limitations that only the
common subset of the algorithmic features and
limitations of the profiles signaled with the
profile-iop byte and the profile indicated by
profile_idc is supported by the codec. For
example, if a codec supports only the common
subset of the coding tools of the Baseline
profile and the Main profile at level 2.1 and
below, the profile-level-id becomes 42E015, in
which 42 stands for the Baseline profile, E0
indicates that only the common subset for all
profiles is supported, and 15 indicates level
2.1.
Informative note: Capability exchange and
session setup procedures should provide
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means to list the capabilities for each
supported codec profile separately. For
example, the one-of-N codec selection
procedure of the SDP offer/answer model can
be used (section 10.2 of [8]).
If no profile-level-id is present, the Baseline
Profile without additional constraints at Level
1 MUST be implied.
max-mbps, max-fs, max-cpb, max-dpb, and max-br:
These parameters MAY be used to signal the
capabilities of a receiver implementation.
These parameters MUST NOT be used for any other
purpose. The profile-level-id parameter MUST
be present in the same receiver capability
description that contains any of these
parameters. The level conveyed in the value of
the profile-level-id parameter MUST be such
that the receiver is fully capable of
supporting. max-mbps, max-fs, max-cpb, max-
dpb, and max-br MAY be used to indicate such
capabilities of the receiver that extend the
required capabilities of the signaled level as
specified below.
When more than one parameter from the set
(max_mbps, max-fs, max-cpb, max_dpb, max-br) is
present, the receiver MUST support all signaled
capabilities simultaneously. For example, if
both max-mbps and max-br are present, the
signaled level with the extension of both the
frame rate and bit rate is supported. That is,
the receiver is able to decode such NAL unit
streams in which the macroblock processing rate
is up to max-mbps (inclusive), the bit rate is
up to max-br (inclusive), the coded picture
buffer size is derived as specified in the
semantics of the max-br parameter below, and
other properties comply with the level
specified in the value of the profile-level-id
parameter.
A receiver MUST NOT signal such values of max-
mbps, max-fs, max-cpb, max-dpb, and max-br that
meet the requirements of a higher level,
referred to as level A herein, compared to the
level specified in the value of the profile-
level-id parameter, if the receiver can support
all the properties of level A.
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Informative note: When the OPTIONAL MIME
type parameters are used to signal the
properties of a NAL unit stream, max-mbps,
max-fs, max-cpb, max-dpb, and max-br are
not present, and the value of profile-
level-id must always be such that the NAL
unit stream complies fully with the
specified profile and level.
max-mbps: The value of max-mbps is an integer indicating
the maximum macroblock processing rate in units
of macroblocks per second. The max-mbps
parameter signals that the receiver is capable
of decoding video at a higher rate than
required by the signaled level conveyed in the
value of the profile-level-id parameter. When
max-mbps is signaled, the receiver MUST be able
to decode NAL unit streams that conform to the
signaled level with the exception that the
MaxMBPS value in Table A-1 of [1] for the
signaled level is replaced with the value of
max-mbps. The value of max-mbps MUST be
greater than or equal to the value of MaxMBPS
for the level given in Table A-1 of [1].
Senders MAY use this knowledge to send pictures
of a given size at a higher picture rate than
indicated in the signaled level.
max-fs: The value of max-fs is an integer indicating
the maximum frame size in units of macroblocks.
The max-fs parameter signals that the receiver
is capable of decoding larger picture sizes
than required by the signaled level conveyed in
the value of the profile-level-id parameter.
When max-fs is signaled, the receiver MUST be
able to decode NAL unit streams that conform to
the signaled level with the exception that the
MaxFS value in Table A-1 of [1] for the
signaled level is replaced with the value of
max-fs. The value of max-fs MUST be greater
than or equal to the value of MaxFS for the
level given in Table A-1 of [1]. Senders MAY
use this knowledge to send larger pictures at a
proportionally lower frame rate than indicated
in the signaled level.
max-cpb The value of max-cpb is an integer indicating
the maximum coded picture buffer size in units
of 1000 bits for the VCL HRD parameters (see
A.3.1 item i of [1]) and in units of 1200 bits
for the NAL HRD parameters (see A.3.1 item j of
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[1]). The max-cpb parameter signals that the
receiver has more memory than the minimum
amount of coded picture buffer memory required
by the signaled level conveyed in the value of
the profile-level-id parameter. When max-cpb
is signaled, the receiver MUST be able to
decode NAL unit streams that conform to the
signaled level with the exception that the
MaxCPB value in Table A-1 of [1] for the
signaled level is replaced with the value of
max-cpb. The value of max-cpb MUST be greater
than or equal to the value of MaxCPB for the
level given in Table A-1 of [1]. Senders MAY
use this knowledge to construct coded video
streams with greater variation of bitrate
compared to which can be achieved with the
MaxCPB value in Table A-1 of [1].
Informative note: The coded picture buffer
is used in the hypothetical reference
decoder (Annex C) of H.264. The use
hypothetical reference decoder is
recommended in H.264 encoders to verify
that the produced bitstream conforms to the
standard and to control the output bitrate.
Thus, the coded picture buffer is
conceptually independent from any other
potential buffers in the receiver,
including de-interleaving and de-jitter
buffers. The coded picture buffer need not
be implemented in decoders as specified in
Annex C of H.264, but rather standard-
compliant decoders can have any buffering
arrangements provided that they can decode
standard-compliant bitstreams. Thus, in
practice, the input buffer for video
decoder can be integrated with de-
interleaving and de-jitter buffers of the
receiver.
max-dpb: The value of max-dpb is an integer indicating
the maximum decoded picture buffer size in
units of 1024 bytes. The max-dpb parameter
signals that the receiver has more memory than
the minimum amount of decoded picture buffer
memory required by the signaled level conveyed
in the value of the profile-level-id parameter.
When max-dpb is signaled, the receiver MUST be
able to decode NAL unit streams that conform to
the signaled level with the exception that the
MaxDPB value in Table A-1 of [1] for the
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signaled level is replaced with the value of
max-dpb. Consequently, a receiver that signals
max-dpb MUST be capable of storing the
following number of decoded frames,
complementary field pairs, and non-paired
fields in its decoded picture buffer:
Min(1024 * max-dpb / ( PicWidthInMbs *
FrameHeightInMbs * 256 * ChromaFormatFactor ),
16)
PicWidthInMbs, FrameHeightInMbs, and
ChromaFormatFactor are defined in [1].
The value of max-dpb MUST be greater than or
equal to the value of MaxDPB for the level
given in Table A-1 of [1]. Senders MAY use
this knowledge to construct coded video streams
with improved compression.
Informative note: This parameter was added
primarily to complement a similar codepoint
in the ITU-T Recommendation H.245, so as to
facilitate signaling gateway designs. The
decoded picture buffer stores reconstructed
samples, and is a property of the video
decoder only. There is no relationship
between the size of the decoded picture
buffer and the buffers used in RTP,
especially de-interleaving and de-jitter
buffers.
max-br: The value of max-br is an integer indicating
the maximum video bit rate in units of 1000
bits per second for the VCL HRD parameters (see
A.3.1 item i of [1]) and in units of 1200 bits
per second for the NAL HRD parameters (see
A.3.1 item j of [1]).
The max-br parameter signals that the video
decoder of the receiver is capable of decoding
video at a higher bit rate than required by the
signaled level conveyed in the value of the
profile-level-id parameter. The value of max-
br MUST be greater than or equal to the value
of MaxBR for the level given in Table A-1 of
[1].
When max-br is signaled, the video codec of the
receiver MUST be able to decode NAL unit
streams that conform to the signaled level,
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conveyed in the profile-level-id parameter,
with the following exceptions in the limits
specified by the level:
o The value of max-br replaces the MaxBR value
of the signaled level (in Table A-1 of [1]).
o When the max-cpb parameter is not present,
the result of the following formula replaces
the value of MaxCPB in Table A-1 of [1]:
(MaxCPB of the signaled level) * max_br /
(MaxBR of the signaled level).
For example, if a receiver signals capability
for Level 1.2 with max-br equal to 1550, this
indicates a maximum video bitrate of 1550
kbits/sec for VCL HRD parameters, a maximum
video bitrate of 1860 kbits/sec for NAL HRD
parameters, and a CPB size of 4,036,458 bits
(1550000 / 384000 * 1000 * 1000).
The value of max-br MUST be grater than or
equal to the value MaxBR for the signaled level
given in Table A-1 of [1].
Senders MAY use this knowledge to send higher
bitrate video as allowed in the level
definition of Annex A of H.264, to achieve
improved video quality.
Informative note: This parameter was added
primarily to complement a similar codepoint
in the ITU-T Recommendation H.245, so as to
facilitate signaling gateway designs. No
assumption can be made from the value of
this parameter that the network is capable
of handling such bit rates at any given
time. In particular, no conclusion can be
drawn that the signaled bit rate is
possible under congestion control
constraints.
redundant-pic-cap: This parameter signals the capabilities of a
receiver implementation. When equal to 0, the
parameter indicates the receiver makes no
attempt to use redundant coded pictures to
correct incorrectly decoded primary coded
pictures. When equal to 0, the receiver is not
capable of using redundant slices, hence a
sender SHOULD avoid sending redundant slices to
save bandwidth. When equal to 1, the receiver
is capable of decoding any such redundant slice
that covers a corrupted area in a primary
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decoded picture (at least partly), and hence a
sender MAY send redundant slices. When the
parameter is not present, then a value of 0
MUST be used for redundant-pic-cap. When
present, the value of redundant-pic-cap MUST be
either 0 or 1.
When the profile-level-id parameter is present
in the same capability signaling as the
redundant-pic-cap parameter and the profile
indicated in profile-level-id is such that it
disallows the use of redundant coded pictures
(e.g., Main Profile), the value of redundant-
pic-cap MUST be equal to 0. When a receiver
indicates redundant-pic-cap equal to 0, the
received stream SHOULD NOT contain redundant
coded pictures.
Informative note: Even if redundant-pic-cap
is equal to 0, the decoder is able to
ignore redundant codec pictures provided
that the decoder supports such profile
(Baseline, Extended) in which redundant
coded pictures are allowed.
Informative note: Even if redundant-pic-cap
is equal to 1, the receiver may also choose
other error concealment strategies to
replace or complement decoding of redundant
slices.
sprop-parameter-sets: This parameter MAY be used to convey
such sequence and picture parameter set NAL
units, herein referred to as the initial
parameter set NAL units, that MUST precede any
other NAL units in decoding order. The
parameter MUST NOT be used to indicate codec
capability in any capability exchange
procedure. The value of the parameter is the
base64 [6] representation of the initial
parameter set NAL units as specified in
sections 7.3.2.1 and 7.3.2.2 of [1]. The
parameter sets are conveyed in decoding order
and no framing of the parameter set NAL units
takes place. A comma is used to separate any
pair of parameter sets in the list. Note that
the number of bytes in a parameter set NAL unit
is typically less than 10 bytes, but a picture
parameter set NAL unit can contain several
hundreds of bytes.
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Informative Note: When several payload
types are offered in the SDP Offer/Answer
model, each with its own sprop-parameter-
sets parameter, then the receiver cannot
assume that those parameter sets do not use
conflicting storage locations (i.e.,
identical values of parameter set
identifiers). Hence, a receiver should
double-buffer all sprop-parameter-sets and
make them available to the decoder instance
that decodes a certain payload type.
parameter-add: This parameter MAY be used to signal whether
the receiver of this parameter is allowed to
add parameter sets in its signaling response
using the sprop-parameter-sets MIME parameter.
The value of this parameter is either 0 or 1.
0 is equal to false, i.e., it is not allowed to
add parameter sets. 1 is equal to true, i.e.
it is allowed to add parameter sets. If the
parameter is not present, its value MUST be 1.
packetization-mode: This parameter signals the properties of a
RTP payload type or the capabilities of a
receiver implementation. Only a single
configuration point can be indicated, thus for
when declaring capabilities to support more
than one packetization-mode, multiple
configuration points (RTP payload types) must
be used.
When the value of packetization-mode is equal
to 0 or packetization-mode is not present, the
single NAL mode as defined in section 6.2 of
RFC XXXX MUST be used. This mode is in use in
standards using ITU-T Recommendation H.241 [17]
(see section 12.1). When the value of
packetization-mode is equal to 1, the non-
interleaved mode as defined in section 6.3 of
RFC XXXX MUST be used. When the value of
packetization-mode is equal to 2, the
interleaved mode as defined in section 6.4 of
RFC XXXX MUST be used. The value of
packetization mode MUST be an integer in the
range of 0 to 2, inclusive.
sprop-interleaving-depth: This parameter MUST NOT be present
when packetization-mode is not present or the
value of packetization-mode is equal to 0 or 1.
This parameter MUST be present when the value
of packetization-mode is equal to 2.
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This parameter signals the properties of a NAL
unit stream. It specifies the maximum number
of VCL NAL units that precede any VCL NAL unit
in the NAL unit stream in transmission order
and follow the VCL NAL unit in decoding order.
Consequently, it is guaranteed that receivers
can reconstruct NAL unit decoding order, when
the buffer size for NAL unit decoding order
recovery is at least the value of sprop-
interleaving-depth + 1 in terms of VCL NAL
units.
The value of sprop-interleaving-depth MUST be
an integer in the range of 0 to 32767,
inclusive.
sprop-deint-buf-req: This parameter MUST NOT be present when
packetization-mode is not present or the value
of packetization-mode is equal to 0 or 1. It
MUST be present when the value of
packetization-mode is equal to 2.
sprop-deint-buf-req signals the required size
of the deinterleaving buffer for the NAL unit
stream. The value of the parameter MUST be
greater than or equal to the maximum buffer
occupancy (in units of bytes) required in such
a deinterleaving buffer that is specified in
section 7.2 of RFC XXXX. It is guaranteed that
receivers can perform the deinterleaving of
interleaved NAL units into NAL unit decoding
order, when the deinterleaving buffer size is
at least the value of sprop-deint-buf-req in
terms of bytes.
The value of sprop-deint-buf-req MUST be an
integer in the range of 0 to 4 294 967 295,
inclusive.
Informative note: deint_buf_req indicates
the required size of the deinterleaving
buffer only. When network jitter can
occur, additionally an appropriately sized
jitter buffer has to be provisioned for.
deint-buf-cap: This parameter signals the capabilities of a
receiver implementation, and indicates the
amount of deinterleaving buffer space in units
of bytes that the receiver has available for
reconstructing the NAL unit decoding order. A
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receiver is able to handle any stream for which
the value of the sprop-deint-buf-req parameter
is smaller than or equal to this parameter.
If the parameter is not present, then a value
of 0 MUST be used for deint-buf-cap. The value
of deint-buf-cap MUST be an integer in the
range of 0 to 4 294 967 295, inclusive.
Informative note: deint_buf_cap indicates
the maximum possible size of the
deinterleaving buffer of the receiver only.
When network jitter can occur, additionally
an appropriately sized jitter buffer has to
be provisioned for.
sprop-init-buf-time: This parameter MAY be used to signal the
properties of a NAL unit stream. The parameter
MUST NOT be present, if the value of
packetization-mode is equal to 0 or 1.
The parameter signals the initial buffering
time that a receiver MUST buffer before
starting decoding to recover the NAL unit
decoding order from the transmission order.
The parameter is the maximum value of
(transmission time of a NAL unit - decoding
time of the NAL unit) assuming reliable and
instantaneous transmission, the same timeline
for transmission and decoding, and starting of
decoding when the first packet arrives.
An example of specifying the value of sprop-
init-buf-time follows: A NAL unit stream is
sent in the following interleaved order, in
which the value corresponds to the decoding
time and the transmission order is from left to
right:
0 2 1 3 5 4 6 8 7 ...
Assuming a steady transmission rate of NAL
units, the transmission times are:
0 1 2 3 4 5 6 7 8 ...
Subtracting the decoding time from the
transmission time column-wise results into the
following series:
0 -1 1 0 -1 1 0 -1 1 ...
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Thus, the value of sprop-init-buf-time in this
example is 1 in terms of intervals of NAL unit
transmission times.
The parameter is coded as a decimal
representation in clock ticks of a 90-kHz
clock. If the parameter is not present, then a
value of 0 MUST be used for sprop-init-buf-
time. The value of sprop-init-buf-time MUST be
an integer in the range of 0 to 4 294 967 295,
inclusive.
In addition to the signaled init_buf_time,
receivers SHOULD take into account the
transmission delay jitter buffering, including
buffering for the delay jitter caused by
mixers, translators, gateways, proxies,
traffic-shapers and other network elements.
sprop-max-don-diff: This parameter MAY be used to signal the
properties of a NAL unit stream. It MUST NOT
be used to signal transmitter or receiver or
codec capabilities. The parameter MUST NOT be
present, if the value of packetization-mode is
equal to 0 or 1. sprop-max-don-diff is an
integer in the range of 0 to 32767, inclusive.
If sprop-max-don-diff is not present, the value
of the parameter is unspecified. sprop-max-
don-diff is calculated as follows:
sprop-max-don-diff = max{AbsDON(i) -
AbsDON(j)},
for any i and any j>i,
where i and j indicate the index of the NAL
unit in the transmission order and AbsDON
denotes such decoding order number of the NAL
unit that does not wrap around to 0 after
65535. In other words, AbsDON is calculated as
follows: Let m and n be consecutive NAL units
in transmission order. For the very first NAL
unit in transmission order (whose index is 0),
AbsDON(0) = DON(0). For other NAL units,
AbsDON is calculated as follows:
If DON(m) == DON(n), AbsDON(n) = AbsDON(m)
If (DON(m) < DON(n) and DON(n) - DON(m) <
32768),
AbsDON(n) = AbsDON(m) + DON(n) - DON(m)
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If (DON(m) > DON(n) and DON(m) - DON(n) >=
32768),
AbsDON(n) = AbsDON(m) + 65536 - DON(m) + DON(n)
If (DON(m) < DON(n) and DON(n) - DON(m) >=
32768),
AbsDON(n) = AbsDON(m) - (DON(m) + 65536 -
DON(n))
If (DON(m) > DON(n) and DON(m) - DON(n) <
32768),
AbsDON(n) = AbsDON(m) - (DON(m) - DON(n))
where DON(i) is the decoding order number of
the NAL unit having index i in the transmission
order. The decoding order number is specified
in section 5.5 of RFC XXXX.
Informative note: Receivers may use sprop-
max-don-diff to trigger which NAL units in
the receiver buffer can be passed to the
decoder.
max-rcmd-nalu-size: This parameter MAY be used to signal the
capabilities of a receiver. The parameter MUST
NOT be used for any other purposes. The value
of the parameter indicates the largest NALU
size in bytes that the receiver can handle
efficiently. The parameter value is a
recommendation, not a strict upper boundary.
The sender MAY create larger NALUs but must be
aware that the handling of these may come at
higher cost than NALUs following the
limitation.
The value of max-rcmd-nalu-size MUST be an
integer in the range of 0 to 4 294 967 295,
inclusive. If this parameter is not specified,
no known limitation to the NALU size exists.
Senders still need to consider the MTU size
available between the sender and the receiver
and SHOULD run MTU discovery for this purpose.
This parameter is motivated by, for example, an
IP to H.223 video telephony gateway, where
NALUs smaller than the H.223 transport data
unit will be more efficient. A gateway may
terminate IP, thus MTU discovery will normally
not work beyond the gateway.
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Informative note: Setting this parameter to
a lower than necessary value may have a
negative impact.
Encoding considerations:
This type is only defined for transfer via RTP
(RFC 3550).
A file format of H.264/AVC video is defined in
[32]. This definition is utilized by other
file formats such as the 3GPP multimedia file
format (MIME type video/3gpp) [33] or the MP4
file format (MIME type video/mp4).
Security considerations:
See section 9 of RFC XXXX.
Public specification:
Please refer to RFC XXXX and its section 17.
Additional information:
None
File extensions: none
Macintosh file type code: none
Object identifier or OID: none
Person & email address to contact for further information:
stewe@stewe.org
Intended usage: COMMON.
Author/Change controller:
stewe@stewe.org
IETF Audio/Video transport working group
8.2. SDP Parameters
8.2.1. Mapping of MIME Parameters to SDP
The MIME media type video/H264 string is mapped to fields in the
Session Description Protocol (SDP) [5] as follows:
o The media name in the "m=" line of SDP MUST be video.
o The encoding name in the "a=rtpmap" line of SDP MUST be H264 (the
MIME subtype).
o The clock rate in the "a=rtpmap" line MUST be 90000.
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o The OPTIONAL parameters "profile-level-id", "max-mbps", "max-fs",
"max-cpb", "max-dpb", "max-br", "redundant-pic-cap", "sprop-
parameter-sets", "parameter-add", "packetization-mode", "sprop-
interleaving-depth", "deint-buf-cap", "sprop-deint-buf-req",
"sprop-init-buf-time", "sprop-max-don-diff", and "max-rcmd-nalu-
size", when present, MUST be included in the "a=fmtp" line of SDP.
These parameters are expressed as a MIME media type string, in the
form of a semicolon separated list of parameter=value pairs.
An example of media representation in SDP is as follows (Baseline
Profile, Level 3.0, some of the constraints of the Main profile may
not be obeyed):
m=video 49170 RTP/AVP 98
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=42A01E; sprop-parameter-
sets=Z0IACpZTBYmI,aMljiA==
8.2.2. Usage with the SDP Offer/Answer Model
When offering H.264 over RTP using SDP in an Offer/Answer model [8]
for negotiation for unicast usage, the following limitations and
rules apply:
o The parameters identifying a media format configuration for H.264
are "profile-level-id", "packetization-mode", and, if required by
"packetization-mode", "sprop-deint-buf-req". These three
parameters MUST be used symmetrically, i.e. the answerer MUST
either maintain all configuration parameters or remove the media
format (payload type) completely, if one or more of the parameter
values are not supported.
Informative note: The requirement for symmetric use applies
only for the above three parameters, and not for the other
stream properties and capability parameters.
To simplify handling and matching of these configurations, the
same RTP payload type number used in the offer SHOULD also be used
in the answer, as specified in [8]. An answer MUST NOT contain a
payload type number used in the offer unless the configuration
("profile-level-id", "packetization-mode", and if present "sprop-
deint-buf-req") is the same as in the offer.
Informative note: An offerer, when receiving the answer, needs
to compare payload types not declared in the offer based on
media type (i.e. video/h264) and the above three parameters
with any payload types it has already declared, in order to
determine whether the configuration in question is new or
equivalent to a configuration already offered.
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o The parameters "sprop-parameter-sets", "sprop-deint-buf-req",
"sprop-interleaving-depth", "sprop-max-don-diff", and "sprop-init-
buf-time" describe the properties of the NAL unit stream that the
offerer or answerer is sending for this media format
configuration. This differs from the normal usage of the
offer/answer parameters: normally such parameters declare the
properties of the stream the offerer or the answerer is able to
receive. When dealing with H.264, the offerer assumes that the
answerer will be able to receive media encoded using the
configuration being offered.
Informative note: The above parameters apply for any stream
sent by the declaring entity with the same configuration, i.e.
they are dependent on their source. As they apply for the
configuration, rather then being bound to the payload type,
the values may need to be applied to another payload type when
sending.
o The capability parameters ("max-mbps", "max-fs", "max-cpb", "max-
dpb", "max-br", ,"redundant-pic-cap", "max-rcmd-nalu-size") MAY be
used to declare further capabilities. Their interpretation
depends on the direction attribute. When the direction attribute
is sendonly, then the parameters describe the limits of the RTP
packets and the NAL unit stream that the sender is capable of
producing. When the direction attribute is sendrecv or recvonly,
then the parameters describe the limitations of what the receiver
accepts.
o As specified above, an offerer needs to include the size of the
deinterleaving buffer in the offer for an interleaved H.264
stream. To enable the offerer and answerer to inform each other
about their capabilities for deinterleaving buffering, both
parties are RECOMMENDED to include "deint-buf-cap". This
information MAY be utilized when selecting the value for "sprop-
deint-buf-req" in a second round of offer and answer. For
interleaved streams, it is also RECOMMENDED to consider offering
multiple payload types with different buffering requirements when
the capabilities of the receiver are unknown.
o The "sprop-parameter-sets" parameter is used as described above.
In addition, an answerer MUST maintain all parameter sets received
in the offer in its answer. Depending on the value of the
"parameter-add" parameter different rules apply: If "parameter-
add" is false (0), the answer MUST NOT add any additional
parameter sets. If "parameter-add" is true (1), the answerer, in
its answer, MAY add additional parameter sets to the "sprop-
parameter-sets" parameter. The answerer MUST also, independent of
the value of "parameter-add", accept to receive a video stream
using the sprop-parameter-sets it declared in the answer.
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Informative note: care must be taken when adding parameter
sets not to cause overwriting of already transmitted parameter
sets by using conflicting parameter set identifiers.
For streams being delivered over multicast, the following rules
apply in addition.
o The stream properties parameters ("sprop-parameter-sets", "sprop-
deint-buf-req", "sprop-interleaving-depth", "sprop-max-don-diff",
and "sprop-init-buf-time") MUST NOT be changed by the answerer.
Hence, a payload type can either be accepted unaltered, or
removed.
o The receiver capability parameters "max-mbps", "max-fs", "max-
cpb", "max-dpb", "max-br", and "max-rcmd-nalu-size" MUST be
supported by the answerer for all streams declared as sendrecv or
recvonly, otherwise one of the following actions MUST be
performed: the media format is removed, or the session rejected.
o The receiver capability parameter redundant-pic-cap SHOULD be
supported by the answerer for all streams declared as sendrecv or
recvonly as follows: The answerer SHOULD NOT include redundant
coded pictures in the transmitted stream, if the offerer indicated
redundant-pic-cap equal to 0. Otherwise (when redundant_pic_cap
is equal to 1), it is beyond the scope of this memo to recommend
how the answerer should use redundant coded pictures.
Below are the complete lists of how the different parameters shall
be interpreted in the different combinations of offer or answer and
direction attribute.
o In offers and answers when "a=sendrecv", or no direction attribute
is used, or in offers and answers where "a=recvonly" is used, the
following interpretation of the parameters MUST be used.
Declaring actual configuration or properties for receiving:
- profile-level-id
- packetization-mode
Declaring actual properties of the stream to be sent (applicable
only when "a=sendrecv" or no direction attribute is used):
- sprop-deint-buf-req
- sprop-interleaving-depth
- sprop-parameter-sets
- sprop-max-don-diff
- sprop-init-buf-time
Declaring receiver implementation capabilities:
- max-mbps
- max-fs
- max-cpb
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- max-dpb
- max-br
- redundant-pic-cap
- deint-buf-cap
- max-rcmd-nalu-size
Declaring how Offer/Answer negotiation shall be performed:
- parameter-add
o In an Offer or Answer where the direction attribute "a=sendonly"
is included for the media stream, the following interpretation of
the parameters MUST be used:
Declaring actual configuration and properties of stream proposed
to be sent:
- profile-level-id
- packetization-mode
- sprop-deint-buf-req
- sprop-max-don-diff
- sprop-init-buf-time
- sprop-parameter-sets
- sprop-interleaving-depth
Declaring the capabilities of the sender when it receives a
stream:
- max-mbps
- max-fs
- max-cpb
- max-dpb
- max-br
- redundant-pic-cap
- deint-buf-cap
- max-rcmd-nalu-size
Declaring how Offer/Answer negotiation shall be performed:
- parameter-add
Further the following considerations are necessary:
o Parameters used for declaring receiver capabilities are in general
downgradable, i.e. they express the upper limit for a sender's
possible behavior. Thus a sender MAY select to set its encoder
using only lower/lesser or equal values of these parameters.
"sprop-parameter-sets" MUST NOT be used in a senders declaration
of its capabilities, as the limits of the values that are carried
inside the parameter sets are implicit with the profile and level
used.
o Parameters declaring a configuration point are not downgradable,
with the exception of the level part of the "profile-level-id"
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parameter. They express values a receiver expects to be used, and
must be used verbatim on the sender side.
o When declaring sender's capabilities, and non-downgradable
parameters are used in this declaration, then these parameters
express a configuration that is acceptable. In order to achieve
high interoperability levels, it is often advisable to offer
multiple alternative configurations, e.g. for the packetization
mode. It is impossible to offer multiple configurations in a
single payload type. Hence, when multiple configuration offers
are made, each offer requires its own RTP payload type associated
with the offer.
o A receiver SHOULD understand all MIME parameters even if it only
supports a subset of the payload formats functionality. This
ensures that a receiver is capable of understanding when an offer
to receive media can be downgraded to what is supported by the
receiver of the offer.
o An answerer MAY extend the offer with additional media format
configurations. However, to enable the usage of these, a second
offer from the offerer is required in most cases to provide the
stream properties parameters that the media sender will use. This
also has the effect that the offerer needs to be able to receive
this media format configuration, not only send it.
o If an offerer wishes to have non-symmetric capabilities between
sending and receiving, the offerer has to offer different RTP
sessions, i.e. different media lines declared as "recvonly" and
"sendonly" respectively. This may have further implications on
the system.
8.2.3. Usage in Declarative Session Descriptions
When offering H.264 over RTP using SDP in a declarative style as
used in RTSP [30] or SAP [31], the following considerations are
necessary.
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o All parameters that are capable of indicating both the
properties of a NAL unit stream and the capabilities of a
receiver are used to indicate the properties of a NAL unit
stream. For example, in this case, the parameter "profile-
level-id" declares the values used by the stream, instead of
capabilities of the sender. This results in that the following
interpretation of the parameters MUST be used:
Declaring actual configuration or properties:
- profile-level-id
- sprop-parameter-sets
- packetization-mode
- sprop-interleaving-depth
- sprop-deint-buf-req
- sprop-max-don-diff
- sprop-init-buf-time
Not usable:
- max-mbps
- max-fs
- max-cpb
- max-dpb
- max-br
- redundant-pic-cap
- max-rcmd-nalu-size
- parameter-add
- deint-buf-cap
o A receiver of the SDP is required to support all parameters and
all values of the parameters provided, or the receiver MUST reject
(RTSP) or not participate in (SAP) the session. It falls on the
creator of the session to use values that are expected to be
supported by the receiving application.
8.3. Examples
A SIP Offer/Answer exchange where both parties are expected to both
send and receive could look like the following. Only the media
codec specific parts of the SDP are shown. Some lines are wrapped
due to text constraints.
Offerer -> Answer SDP message:
m=video 49170 RTP/AVP 100 99 98
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=42A01E; packetization-mode=0;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42A01E; packetization-mode=1;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
a=rtpmap:100 H264/90000
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a=fmtp:100 profile-level-id=42A01E; packetization-mode=2;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==;
sprop-interleaving-depth=45; sprop-deint-buf-req=64000;
sprop-init-buf-time=102478; deint-buf-cap=128000
The above offer offers the same codec configuration in three
different packetization formats. PT 98 represents single NALU mode,
99 non-interleaved mode, and 100 indicates the interleaved mode. In
the interleaved mode case, the interleaving parameters that the
offerer would use if the answer indicates support for PT 100 are
also included. In all three cases the parameter "sprop-parameter-
sets" conveys the initial parameter sets that are required for the
answerer when receiving a stream from the offerer when this
configuration (profile-level-id and packetization mode) is accepted.
Note that the value for "sprop-parameter-sets", although identical
in the example above, could be different for each payload type.
Answerer -> Offerer SDP message:
m=video 49170 RTP/AVP 100 99 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42A01E; packetization-mode=0;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==,As0DEWlsIOp==,
KyzFGleR
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42A01E; packetization-mode=1;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==,As0DEWlsIOp==,
KyzFGleR; max-rcmd-nalu-size=3980
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=42A01E; packetization-mode=2;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==,As0DEWlsIOp==,
KyzFGleR; sprop-interleaving-depth=60;
sprop-deint-buf-req=86000; sprop-init-buf-time=156320;
deint-buf-cap=128000; max-rcmd-nalu-size=3980
As the offer/answer negotiation covers both sending and receiving
streams, an offer indicates the exact parameters for what the
offerer is willing to receive, while the answer indicates the same
for what the answerer accepts to receive. In this case the offerer
declared that it is willing to receive payload type 98. The
answerer accepts this by declaring a equivalent payload type 97,
i.e. it has identical values for the three parameters "profile-
level-id", packetization-mode, and "sprop-deint-buf-req". This has
the following implications for both the offerer and the answerer
concerning the parameters that declare properties. The offerer
initially declared a certain value of the "sprop-parameter-sets" in
the payload definition for PT=98. However, as the answerer accepted
this as PT=97, the values of "sprop-parameter-sets" in PT=98 must
now be used instead when the offerer sends PT=97. Similarly, when
the answerer sends PT=98 to the offerer, it has to use the
properties parameters it declared in PT=97.
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The answerer also accepts the reception of the two configurations
that payload types 99 and 100 represents. It provides the initial
parameter sets for the answerer-to-offerer direction, and buffering
related parameters that it will use to send the payload types. It
also provides the offerer with its memory limit for deinterleaving
operations by providing a "deint-buf-cap" parameter. This is only
useful if the offerer decides on making a second offer, where it can
take the new value into account. The "max-rcmd-nalu-size" indicates
that the answerer can efficiently process NALUs up to the size of
3980 bytes. However, there is no guarantee that the network
supports this size.
Please note that the parameter sets in the above example are not
representing a legal operation point of an H.264 codec -- the base64
strings are only used for illustration.
8.4. Parameter Set Considerations
The H.264 parameter sets are a fundamental part of the video codec
and vital to its operation, see section 1.2. Due to their
characteristics and their importance for the decoding process, lost
or erroneously transmitted parameter sets can hardly be concealed
locally at the receiver. A reference to a corrupt parameter set has
normally fatal results to the decoding process. Corruption could
occur, for example, due to the erroneous transmission or loss of a
parameter set data structure, but also due to the untimely
transmission of a parameter set update. Hence, the following
recommendations are provided as a guideline for the implementer of
the RTP sender.
Parameter set NALUs can be transported using three different
principles:
A. Using a session control protocol (out-of-band) prior to the
actual RTP session.
B. Using a session control protocol (out-of-band) during an ongoing
RTP session.
C. Within the RTP stream in the payload (in-band) during an ongoing
RTP session.
It is necessary to implement principles A and B within a session
control protocol. SIP and SDP can be used as described in the SDP
Offer/Answer model and in the previous sections of this memo. This
section contains guidelines how principles A and B must be
implemented within session control protocols, and is independent of
the particular protocol used. Principle C is supported by the RTP
payload format defined in this specification.
Picture and sequence parameter set NALUs SHOULD NOT be transmitted
in the RTP payload unless reliable transport is provided for RTP, as
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a loss of a parameter set of either type likely prevents decoding of
a considerable portion of the corresponding RTP stream. Thus, the
transmission of parameter sets using a reliable session control
protocol, i.e. usage of principle A or B above, is RECOMMENDED.
In the rest of the section it is assumed that out-of-band signaling
provides reliable transport of parameter set NALUs, while in-band
transport does not. If in-band signaling of parameter sets is used,
the sender SHOULD take the error characteristics into account and
use mechanisms to provide a high probability for delivering the
parameter sets correctly. Mechanisms that increase the probability
for a correct reception include packet repetition, FEC, and
retransmission. The use of an unreliable, out-of-band control
protocol has similar disadvantages as the in-band signaling
(possible loss) and, in addition, may also lead to difficulties in
the synchronization (see below) and is NOT RECOMMENDED.
Parameter sets MAY be added or updated during the lifetime of a
session using principles B and C. It is required that parameter
sets are present at the decoder prior to the NAL units that refer to
them. Updating or adding of parameter sets can result in further
problems, and therefore the following recommendations should be
considered.
- When adding or updating parameter sets, principle C is vulnerable
to transmission errors as described above, and therefore principle
B is RECOMMENDED.
- When adding or updating parameter sets, care SHOULD be taken to
ensure that any parameter set is delivered prior to its usage. It
is common that no synchronization is present between out-of-band
signaling and in-band traffic. If out-of-band signaling is used,
it is RECOMMEDED that a sender does not start sending NALUs
requiring the updated parameter sets prior to acknowledgement of
delivery from the signaling protocol.
- When updating parameter sets, the following synchronization issue
should be taken into account. When overwriting a parameter set at
the receiver, the sender needs ensure that the parameter set in
question is not needed by any NALU present in the network or
receiver buffers. Otherwise decoding using a wrong parameter set
may occur. To lessen this problem, it is RECOMMENDED to either
overwrite only those parameter sets that have not been used for a
sufficiently long time (to ensure that all related NALUs have been
consumed), or to add a new parameter set instead (which may have
negative consequences for the efficiency of the video coding).
- When adding new parameter sets, previously unused parameter set
identifiers are used. This avoids the problem identified in the
previous paragraph. However, in a multiparty session and unless a
synchronized control protocol is used, there is a risk that
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multiple entities try to add different parameter sets for the same
identifier, which needs to be avoided.
- Adding or modifying parameter sets by using both principles B and
C in the same RTP session may lead to inconsistencies of the
parameter sets because of the lack of synchronization between the
control and the RTP channel. Therefore principle B and C MUST NOT
both be used in the same session, unless sufficient
synchronization can be provided.
In some scenarios, e.g. when only the subset of this payload format
specification corresponding to H.241 is used, it is not possible to
employ out-of-band parameter set transmission. In this case,
parameter sets need to be transmitted in-band. Here, the
synchronization with the non-parameter-set-data in the bitstream is
implicit, but the possibility of a loss needs to be taken into
account and the loss probability should be reduced using the
mechanisms discussed above.
- When parameter sets are both provided initially using principle A
and then later added or updated in-band (principle C), then there
is a risk associated with updating the parameter sets delivered
out-of-band. If receivers miss some in-band updates, because of a
loss or a late tune-in, for example, those receivers attempt to
decode the bitstream using out-dated parameters. It is
RECOMMENDED that parameter set IDs are partitioned between the
out-of-band and in-band parameter sets.
To allow for maximum flexibility and best performance from the H.264
coder, it is recommended if possible to allow any sender to add its
own parameter sets to be used in a session. Setting the "parameter-
add" parameter to false should only be done in cases where the
session topology prevents a participant to add its own parameter
sets.
9. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification [4], and any appropriate RTP profile (for example
[18]). This implies that confidentiality of the media streams is
achieved by encryption, for example through the application of SRTP
[29]. Because the data compression used with this payload format is
applied end-to-end, encryption may be performed after compression so
there is no conflict between the two operations.
A potential denial-of-service threat exists for data encodings using
compression techniques that have non-uniform receiver-end
computational load. The attacker can inject such pathological
datagrams into the stream that are complex to decode and cause the
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receiver to be overloaded. H.264 is particularly vulnerable to such
attacks because it is extremely simple to generate datagrams
containing NAL units that affect the decoding process of many future
NAL units. Therefore the usage of authentication of at least the
RTP packet is RECOMMENDED, for example with SRTP [29].
Note that the appropriate mechanism to ensure confidentiality and
integrity of RTP packets and their payloads are very dependent on
the application and the transport and signaling protocols employed.
Hence, although SRTP is given as example above, other possible
choices exist.
As with any IP-based protocol, in some circumstances a receiver may
be overloaded simply by the receipt of too many packets, either
desired or undesired. Network-layer authentication may be used to
discard packets from undesired sources, but the processing cost of
the authentication itself may be too high. In a multicast
environment, pruning of specific sources may be implemented in
future versions of IGMP [19] and in multicast routing protocols to
allow a receiver to select which sources are allowed to reach it.
Decoders MUST exercise caution with respect to the handling of user
data SEI messages, particularly if they contain active elements, and
MUST restrict their domain of applicability to the presentation
containing the stream.
10. Congestion Control
Congestion control for RTP SHALL be used in accordance with RFC 3550
[4], and any applicable RTP profile, e.g. RFC 3551 [18]. This means
that congestion control is required for any transmission over
unmanaged best-effort networks.
The bit rate adaptation necessary for obeying the congestion control
principle is easily achievable when real-time encoding is used.
However, when pre-encoded content is being transmitted, bandwidth
adaptation requires the availability of more than one coded
representation of the same content, at different bit rates, or the
existence of non-reference pictures or sub-sequences [25] in the
bitstream. The switching between the different representations can
normally be performed in the same RTP session, e.g. by employing a
concept known as SI/SP slices of the Extended Profile, or by
switching streams at IDR picture boundaries. Only if non-
downgradable parameters, such as the profile part of the
profile/level ID change, it becomes necessary to terminate and re-
start the media stream, possibly using a different RTP payload type.
MANEs MAY follow the suggestions outlined in section 7.3 and remove
certain not usable packets from the packet stream when that stream
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was damaged due to previous packet losses. This can help reducing
the network load in certain special cases.
11. IANA Consideration
IANA is kindly requested to register one new MIME type, see section
8.1.
12. Informative Appendix: Application Examples
This payload specification is very flexible in its use, to cover the
extremely wide application space that is anticipated for H.264.
However, such a great flexibility also makes it difficult for an
implementer to decide on a reasonable packetization scheme. Some
information on how to apply this specification to real-world
scenarios is likely to appear in the form of academic publications
and a test model software and description in the near future.
However, some preliminary usage scenarios are described here as
well.
12.1. Video Telephony according to ITU-T Recommendation H.241
Annex A
H.323-based video telephony systems that use H.264 as an optional
video compression scheme are required to support H.241 Annex A [17]
as a packetization scheme. The packetization mechanism defined in
this Annex is technically identical with a small subset of this
specification.
When operating according to H.241 Annex A, parameter sets NAL units
are sent in-band. Only Single NAL unit packets are used. Many such
systems are not sending IDR pictures regularly, but only when
required by user interaction or by control protocol means, e.g. when
switching between video channels in a Multipoint Control Unit or for
error recovery requested by feedback.
12.2. Video Telephony, No Slice Data Partitioning, No NAL Unit
Aggregation
The RTP part of this scheme is implemented and tested (though not
the control-protocol part, see below).
In most real-world video telephony applications, the picture
parameters such as picture size or optional modes never change
during the lifetime of a connection. Hence, all necessary parameter
sets (usually only one) are sent as a side effect of the capability
exchange/announcement process e.g. according to the SDP syntax
specified in section 8.2 of this document. Since all necessary
parameter set information is established before the RTP session
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starts, there is no need for sending any parameter set NAL units.
Slice data partitioning is not used either. Hence, the RTP packet
stream consists basically of NAL units that carry single coded
slices.
The encoder chooses the size of coded slice NAL units such that they
offer the best performance. Often, this is done by adapting the
coded slice size to the MTU size of the IP network. For small
picture sizes this may result in a one-picture-per-one-packet
strategy. Intra refresh algorithms clean up the loss of packets and
the resulting drift-related artifacts.
12.3. Video Telephony, Interleaved Packetization Using NAL Unit
Aggregation
This scheme allows better error concealment and is used in H.263
based designed using RFC 2429 packetization [12]. It is also
implemented and good results were reported [14].
The VCL encoder codes the source picture such that all macroblocks
(MBs) of one MB line are assigned to one slice. All slices with
even MB row addresses are combined into one STAP, and all slices
with odd MB row addresses into another STAP. Those STAPs are
transmitted as RTP packets. The establishment of the parameter sets
is performed as discussed above.
Note that the use of STAPs is essential here, because the high
number of individual slices (18 for a CIF picture) would lead to
unacceptably high IP/UDP/RTP header overhead (unless the source
coding tool FMO is used, which is not assumed in this scenario).
Furthermore, some wireless video transmission systems, such as
H.324M and the IP-based video telephony specified in 3GPP, are
likely to use relatively small transport packet size. For example,
a typical MTU size of H.223 AL3 SDU is around 100 bytes [20].
Coding individual slices according to this packetization scheme
provides a further advantage in communication between wired and
wireless networks, as individual slices are likely to be smaller
than the preferred maximum packet size of wireless systems.
Consequently, a gateway can convert the STAPs used in a wired
network to several RTP packets with only one NAL unit that are
preferred in a wireless network and vice versa.
12.4. Video Telephony, with Data Partitioning
This scheme is implemented and was shown to offer good performance
especially at higher packet loss rates [14].
Data Partitioning is known to be useful only when some form of
unequal error protection is available. Normally, in single-session
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RTP environments, even error characteristics are assumed, i.e., the
packet loss probability of all packets of the session is the same
statistically. However, there are means to reduce the packet loss
probability of individual packets in an RTP session. A FEC packet
according to RFC 2733 [21], for example, specifies which media
packets are associated with the FEC packet.
In all cases, the incurred overhead is substantial, but in the same
order of magnitude as the number of bits that have otherwise be
spent for intra information. However, this mechanism is not adding
any delay to the system.
Again, the complete parameter set establishment is performed through
control protocol means.
12.5. Video Telephony or Streaming, with FUs and Forward Error
Correction
This scheme is implemented and was shown to provide good performance
especially at higher packet loss rates [22].
The most efficient means to combat packet-losses for scenarios where
retransmissions are not applicable is forward error correction
(FEC). Although the application layer, end-to-end use of FEC is
often less efficient when compared to a FEC-based protection of
individual links (especially when links of different characteristics
are in the transmission path), application layer, end-to-end FEC is
unavoidable in some scenarios. RFC 2733 [21] provides means to use
generic, application layer, end-to-end FEC in packet-loss
environments. A binary forward error correcting code is generated
by applying the XOR operation to the bits at the same bit position
in different packets. The binary code can be specified by the
parameters (n,k) in which k is the number of information packets
used in the connection and n is the total number of packets
generated for k information packets, i.e., n-k parity packets are
generated for k information packets.
When using a code with parameters (n,k) within the RFC 2733
framework, the following properties are well-known:
a) If applied over one RTP packet, RFC 2733 provides only packet
repetition.
b) RFC 2733 is most bit-rate efficient if XOR-connected packets have
equal length.
c) At the same packet loss probability p and for a fixed k, the
greater the value of n is, the smaller the residual error
probability becomes. For example, for packet loss probability
10%, k=1, and n=2, the residual error probability is about 1%,
whereas for n=3, the residual error probability is about 0.1%.
d) At the same packet loss probability p and for a fixed code rate
k/n, the greater the value of n is, the smaller the residual
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error probability becomes. For example, at a packet loss
probability of p=10%, k=1 and n=2, the residual error rate is
about 1%, whereas for an extended Golay code with k=12 and n=24,
the residual error rate is about 0.01%.
For applying RFC 2733 in combination with H.264 baseline coded video
without using FUs several options might be considered:
1) The video encoder produces NAL units where each video frame is
coded in a single slice. Applying FEC, one could use a simple
code, e.g. (n=2, k=1), i.e., each NAL unit would basically just
be repeated. The disadvantage is obviously the bad code
performance according to (d) and the low flexibility as only (n,
k=1) codes can be used.
2) The video encoder produces NAL units where each video frame is
encoded in one or more consecutive slices. Applying FEC, one
could use a better code, e.g. (n=24, k=12), over a sequence of
NAL units. Depending on the number of RTP packets per frame, a
loss may introduce a significant delay, which is reduced the more
RTP packets per frame are used. Packets of completely different
length might also be connected, which decreases bit-rate
efficiency according to (b). However with some care and for
slices of 1kb or larger, similar length (100-200 bytes
difference) may be produced, which will not lower the bit-
efficiency catastrophically.
3) The video encoder produces NAL units, where a certain frame
contains k slices of possibly almost equal length. Then,
applying FEC, a better code, e.g. (n=24, k=12), over the sequence
of NAL units for each frame can be used. The delay compared to
(2) may be reduced, but several disadvantages are obvious.
Firstly, the coding efficiency of the encoded video is lowered
significantly as slice-structured coding reduces intra-frame
prediction and additional slice overhead is necessary. Secondly,
pre-encoded content or, when operating over a gateway, the video
is usually not appropriately coded with k slices such that FEC
can be applied. Finally, the encoding of video producing k
slices of equal length is not straightforward and might require
more than one encoding pass.
Many of the mentioned disadvantages can be avoided by applying FUs
in combination with FEC. Each NAL unit can be split into any number
of FUs of basically equal length, and therefore FEC with a
reasonable k and n can be applied even if the encoder made no effort
of producing slices of equal length. For example, a coded slice NAL
unit containing an entire frame can be split to k FUs and a parity
check code (n=k+1, k) can be applied. However this has the
disadvantage that unless all created fragments can be recovered the
whole slice will be lost. Thus a larger section is lost, than would
be the case if the frame had been split into several slices.
The presented technique makes it possible to achieve good
transmission error tolerance even if no additional source coding
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layer redundancy, such as periodic intra frames, is present.
Consequently, the same coded video sequence can be used for
achieving the maximum compression efficiency and quality over error-
free transmission and for transmission over error-prone networks.
Furthermore, the technique allows the application of FEC to pre-
encoded sequences without adding delay. In addition, in this case
pre-encoded sequences that are not encoded for error-prone networks
can still be transmitted almost reliably without adding extensive
delays. In addition, FUs of equal length result in a bit-rate
efficient use of RFC 2733.
In case that the error probability depends on the length of the
transmitted packet, e.g. in case of mobile transmission [16], the
benefits of applying FUs with FEC are even more obvious. Basically,
the flexibility of the size of FUs allows applying appropriate FEC
for each NAL unit and even unequal error protection of NAL units.
The incurred overhead when using FUs and FEC is substantial, but in
the same order of magnitude as the number of bits that have to be
spent for intra coded macroblocks if no FEC is applied. In [22] it
was shown that the overall performance at the same error rate and
the same overall bit-rate including the overhead, the FEC-based
approach can enhance the quality.
12.6. Low-Bit-Rate Streaming
This scheme has been implemented with H.263 and non-standard RTP
packetization and gave good results [23]. There is no technical
reason why similarly good results could not be achievable with
H.264.
In today's Internet streaming, some of the offered bit-rates are
relatively low in order to allow terminals with dial-up modems to
access the content. In wired IP networks, relatively large packets,
say 500 - 1500 bytes, are preferred to smaller and more frequently
occurring packets in order to reduce network congestion. Moreover,
use of large packets decreases the amount of RTP/UDP/IP header
overhead. For low-bit-rate video, the use of large packets means
that sometimes up to few pictures should be encapsulated in one
packet.
However, loss of a packet including many coded pictures would have
drastic consequences in visual quality, as there is practically no
other way to conceal a loss of an entire picture than to repeat the
previous one. One way to construct relatively large packets and
maintain possibilities for successful loss concealment is to
construct MTAPs that contain slices from several pictures in an
interleaved manner. An MTAP should not contain spatially adjacent
slices from the same picture or spatially overlapping slices from
any picture. If a packet is lost, it is likely that a lost slice is
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surrounded by spatially adjacent slices of the same picture and
spatially corresponding slices of the temporally previous and
succeeding pictures. Consequently, concealment of the lost slice is
likely to succeed relatively well.
12.7. Robust Packet Scheduling in Video Streaming
Robust packet scheduling has been implemented with MPEG-4 Part 2 and
simulated in a wireless streaming environment [24]. There is no
technical reason why similar or better results could not be
achievable with H.264.
Streaming clients typically have a receiver buffer that is capable
of storing a relatively large amount of data. Initially, when a
streaming session is established, a client does not start playing
the stream back immediately, but rather it typically buffers the
incoming data for a few seconds. This buffering helps to maintain
continuous playback, because, in case of occasional increased
transmission delays or network throughput drops, the client can
decode and play buffered data. Otherwise, without initial
buffering, the client has to freeze the display, stop decoding, and
wait for incoming data. The buffering is also necessary for either
automatic or selective retransmission in any protocol level. If any
part of a picture is lost, a retransmission mechanism may be used to
resend the lost data. If the retransmitted data is received before
its scheduled decoding or playback time, the loss is perfectly
recovered. Coded pictures can be ranked according to their
importance in the subjective quality of the decoded sequence. For
example, non-reference pictures, such as conventional B pictures,
are subjectively least important, because their absence does not
affect decoding of any other pictures. In addition to non-reference
pictures, the ITU-T H.264 | ISO/IEC 14496-10 standard includes a
temporal scalability method called sub-sequences [25]. Subjective
ranking can also be made on coded slice data partition or slice
group basis. Coded slices and coded slice data partitions that are
subjectively the most important can be sent earlier than their
decoding order indicates, whereas coded slices and coded slice data
partitions that are subjectively the least important can be sent
later than their natural coding order indicates. Consequently, any
retransmitted parts of the most important slices and coded slice
data partitions are more likely to be received before their
scheduled decoding or playback time compared to the least important
slices and slice data partitions.
13. Informative Appendix: Rationale for Decoding Order Number
13.1. Introduction
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The Decoding Order Number (DON) concept was introduced mainly to
enable efficient multi-picture slice interleaving (see section 12.6)
and robust packet scheduling (see section 12.7). In both of these
applications NAL units are transmitted out of decoding order. DON
indicates the decoding order of NAL units and should be used in the
receiver to recover the decoding order. Example use cases for
efficient multi-picture slice interleaving and for robust packet
scheduling are given in sections 13.2 and 13.3 respectively.
Section 13.4 describes the benefits of the DON concept in error
resiliency achieved by redundant coded pictures. Section 13.5
summarizes considered alternatives to DON and justifies why DON was
chosen to this RTP payload specification.
13.2. Example of Multi-Picture Slice Interleaving
An example of multi-picture slice interleaving follows. A subset of
a coded video sequence is depicted below in output order. R denotes
a reference picture, N denotes a non-reference picture, and the
number indicates a relative output time.
... R1 N2 R3 N4 R5 ...
The decoding order of these pictures is from left to right as
follows:
... R1 R3 N2 R5 N4 ...
The NAL units of pictures R1, R3, N2, R5, and N4 are marked with a
DON equal to 1, 2, 3, 4, and 5, respectively.
Each reference picture consists of three slice groups that are
scattered as follows (a number denotes the slice group number for
each macroblock in a QCIF frame):
0 1 2 0 1 2 0 1 2 0 1
2 0 1 2 0 1 2 0 1 2 0
1 2 0 1 2 0 1 2 0 1 2
0 1 2 0 1 2 0 1 2 0 1
2 0 1 2 0 1 2 0 1 2 0
1 2 0 1 2 0 1 2 0 1 2
0 1 2 0 1 2 0 1 2 0 1
2 0 1 2 0 1 2 0 1 2 0
1 2 0 1 2 0 1 2 0 1 2
For the sake of simplicity, we assume that all the macroblocks of a
slice group are included in one slice. Three MTAPs are constructed
from three consecutive reference pictures so that each MTAP contains
three aggregation units, each of which contains all the macroblocks
from one slice group. The first MTAP contains slice group 0 of
picture R1, slice group 1 of picture R3, and slice group 2 of
picture R5. The second MTAP contains slice group 1 of picture R1,
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slice group 2 of picture R3, and slice group 0 of picture R5. The
third MTAP contains slice group 2 of picture R1, slice group 0 of
picture R3, and slice group 1 of picture R5. Each non-reference
picture is encapsulated into an STAP-B.
Consequently, the transmission order of NAL units is the following:
R1, slice group 0, DON 1, carried in MTAP, RTP SN: N
R3, slice group 1, DON 2, carried in MTAP, RTP SN: N
R5, slice group 2, DON 4, carried in MTAP, RTP SN: N
R1, slice group 1, DON 1, carried in MTAP, RTP SN: N+1
R3, slice group 2, DON 2, carried in MTAP, RTP SN: N+1
R5, slice group 0, DON 4, carried in MTAP, RTP SN: N+1
R1, slice group 2, DON 1, carried in MTAP, RTP SN: N+2
R3, slice group 1, DON 2, carried in MTAP, RTP SN: N+2
R5, slice group 0, DON 4, carried in MTAP, RTP SN: N+2
N2, DON 3, carried in STAP-B, RTP SN: N+3
N4, DON 5, carried in STAP-B, RTP SN: N+4
The receiver is able to organize the NAL units back in decoding
order based on the value of DON associated with each NAL unit.
If one of the MTAPs is lost, the spatially adjacent and temporally
co-located macroblocks are received and can be used to conceal the
loss efficiently. If one of the STAPs is lost, the effect of the
loss does not propagate temporally.
13.3. Example of Robust Packet Scheduling
An example of robust packet scheduling follows. The communication
system used in the example consists of the following components in
the order that the video is processed from source to sink:
o camera and capturing
o pre-encoding buffer
o encoder
o encoded picture buffer
o transmitter
o transmission channel
o receiver
o receiver buffer
o decoder
o decoded picture buffer
o display
The video communication system used in the example operates as
follows. Note that processing of the video stream happens gradually
and at the same time in all components of the system. The source
video sequence is shot and captured to a pre-encoding buffer. The
pre-encoding buffer can be used to order pictures from sampling
order to encoding order or to analyze multiple uncompressed frames
for bitrate rate control purposes, for example. In some cases the
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pre-encoding buffer may not exist, but rather the sampled pictures
are encoded right away. The encoder encodes pictures from the pre-
encoding buffer and stores the output, i.e., coded pictures, to the
encoded picture buffer. The transmitter encapsulates the coded
pictures from the encoded picture buffer to transmission packets and
sends them to a receiver through a transmission channel. The
receiver stores the received packets to the receiver buffer. The
receiver buffering process typically includes buffering for
transmission delay jitter. The receiver buffer can also be used to
recover correct decoding order of coded data. The decoder reads
coded data from the receiver buffer and produces decoded pictures as
output into the decoded picture buffer. The decoded picture buffer
is used to recover the output (or display) order of pictures.
Finally, pictures are displayed.
In the following example figures, I denotes an IDR picture, R
denotes a reference picture, N denotes a non-reference picture, and
the number after I, R, or N indicates the sampling time relative to
the previous IDR picture in decoding order. Values below the
sequence of pictures indicate scaled system clock timestamps. The
system clock is initialized arbitrarily in this example, and time
runs from left to right. Each I, R, and N picture is mapped into
the same timeline compared to the previous processing step, if any,
assuming that encoding, transmission, and decoding take no time.
Thus, events happening at the same time are located in the same
column throughout all example figures.
A subset of a sequence of coded pictures is depicted below in
sampling order.
... N58 N59 I00 N01 N02 R03 N04 N05 R06 ... N58 N59 I00 N01 ...
... --|---|---|---|---|---|---|---|---|- ... -|---|---|---|- ...
... 58 59 60 61 62 63 64 65 66 ... 128 129 130 131 ...
Figure 16. Sequence of pictures in sampling order
The sampled pictures are buffered in the pre-encoding buffer to
arrange them in encoding order. In this example, we assume that the
non-reference pictures are predicted from both the previous and the
next reference picture in output order except for the non-reference
pictures immediately preceding an IDR picture, which are predicted
only from the previous reference picture in output order. Thus, the
pre-encoding buffer has to contain at least two pictures and the
buffering causes a delay of two picture intervals. The output of
the pre-encoding buffering process and the encoding (and decoding)
order of the pictures are as follows:
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... N58 N59 I00 R03 N01 N02 R06 N04 N05 ...
... -|---|---|---|---|---|---|---|---|- ...
... 60 61 62 63 64 65 66 67 68 ...
Figure 17. Re-ordered pictures in the pre-encoding buffer
The encoder or the transmitter can set the value of DON for each
picture to a value of DON for the previous picture in decoding order
plus one.
For the sake of simplicity, let us assume that:
o the frame rate of the sequence is constant,
o each picture consists of only one slice,
o each slice is encapsulated in a single NAL unit packet,
o there is no transmission delay, and
o pictures are transmitted at constant intervals (that is equal to 1
/ frame rate).
When pictures are transmitted in decoding order, they are received
as follows:
... N58 N59 I00 R03 N01 N02 R06 N04 N05 ...
... -|---|---|---|---|---|---|---|---|- ...
... 60 61 62 63 64 65 66 67 68 ...
Figure 18. Received pictures in decoding order
The OPTIONAL sprop-interleaving-depth MIME type parameter is set to
0, because the transmission (or reception) order is identical to the
decoding order.
The decoder has to buffer for one picture interval initially in its
decoded picture buffer to organize pictures from decoding order to
output order as depicted below:
... N58 N59 I00 N01 N02 R03 N04 N05 R06 ...
... -|---|---|---|---|---|---|---|---|- ...
... 61 62 63 64 65 66 67 68 69 ...
Figure 19. Output order
The amount of required initial buffering in the decoded picture
buffer can be signaled in the buffering period SEI message or with
the num_reorder_frames syntax element of H.264 video usability
information. num_reorder_frames indicates the maximum number of
frames, complementary field pairs, or non-paired fields that precede
any frame, complementary field pair, or non-paired field in the
sequence in decoding order and follow it in output order. For the
sake of simplicity, we assume that num_reorder_frames is used to
indicate the initial buffer in the decoded picture buffer. In this
example, num_reorder_frames is equal to 1.
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It can be observed that if the IDR picture I00 is lost during
transmission and a retransmission request is issued when the value
of the system clock is 62, there is one picture interval of time
(until the system clock reaches timestamp 63) to receive the
retransmitted IDR picture I00.
Let us then assume that IDR pictures are transmitted two frame
intervals earlier than their decoding position, i.e., the pictures
are transmitted as follows:
... I00 N58 N59 R03 N01 N02 R06 N04 N05 ...
... --|---|---|---|---|---|---|---|---|- ...
... 62 63 64 65 66 67 68 69 70 ...
Figure 20. Interleaving: early IDR pictures in sending order
The OPTIONAL sprop-interleaving-depth MIME type parameter is set
equal to 1 according to its definition. (The value of sprop-
interleaving-depth in this example can be derived as follows:
Picture I00 is the only picture preceding picture N58 or N59 in
transmission order and following it in decoding order. Except for
pictures I00, N58, and N59, the transmission order is the same as
the decoding order of pictures. Since a coded picture is
encapsulated into exactly one NAL unit, the value of sprop-
interleaving-depth is equal to the maximum number of pictures
preceding any picture in transmission order and following the
picture in decoding order.)
The receiver buffering process contains two pictures at a time
according to the value of the sprop-interleaving-depth parameter and
orders pictures from the reception order to the correct decoding
order based on the value of DON associated with each picture. The
output of the receiver buffering process is the following:
... N58 N59 I00 R03 N01 N02 R06 N04 N05 ...
... -|---|---|---|---|---|---|---|---|- ...
... 63 64 65 66 67 68 69 70 71 ...
Figure 21. Interleaving: Receiver Buffer
Again, an initial buffering delay of one picture interval is needed
to organize pictures from decoding order to output order as depicted
below:
... N58 N59 I00 N01 N02 R03 N04 N05 ...
... -|---|---|---|---|---|---|---|- ...
... 64 65 66 67 68 69 70 71 ...
Figure 22. Interleaving: Receiver buffer after reordering
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It can be observed that the maximum delay that IDR pictures can
undergo during transmission, including possible application,
transport, or link layer retransmission, is equal to three picture
intervals. Thus, the loss resiliency of IDR pictures is improved in
systems supporting retransmission compared to the case in which
pictures were transmitted in their decoding order.
13.4. Robust Transmission Scheduling of Redundant Coded Slices
A redundant coded picture is a coded representation of a picture or
a part of a picture that is not used in the decoding process if the
corresponding primary coded picture is correctly decoded. There
should be no noticeable difference between any area of the decoded
primary picture and a corresponding area that would result from
application of the H.264 decoding process for any redundant picture
in the same access unit. A redundant coded slice is a coded slice
that is a part of a redundant coded picture.
Redundant coded pictures can be used to provide unequal error
protection in error-prone video transmission. If a primary coded
representation of a picture is decoded incorrectly, a corresponding
redundant coded picture can be decoded. Examples of applications
and coding techniques utilizing the redundant codec picture feature
include the video redundancy coding [26] and protection of "key
pictures" in multicast streaming [27].
One property of many error-prone video communications systems is
that transmission errors are often bursty and therefore they may
affect more than one consecutive transmission packets in
transmission order. In low bitrate video communication it is
relatively common that an entire coded picture can be encapsulated
into one transmission packet. Consequently, a primary coded picture
and the corresponding redundant coded pictures may be transmitted in
consecutive packets in transmission order. In order to make the
transmission scheme more tolerant of bursty transmission errors, it
is beneficial to transmit a primary coded picture further apart from
the corresponding redundant coded pictures. The DON concept enables
this.
13.5. Remarks on Other Design Possibilities
The slice header syntax structure of the H.264 coding standard
contains the frame_num syntax element that can indicate the decoding
order of coded frames. However, the usage of the frame_num syntax
element is not feasible or desirable to recover the decoding order
due to the following reasons:
o The receiver is required to parse at least one slice header per
coded picture (before passing the coded data to the decoder).
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o Coded slices from multiple coded video sequences cannot be
interleaved, because the frame number syntax element is reset to 0
in each IDR picture.
o The coded fields of a complementary field pair share the same
value of the frame_num syntax element. Thus, the decoding order
of the coded fields of a complementary field pair cannot be
recovered based on the frame_num syntax element or any other
syntax element of the H.264 coding syntax.
The RTP payload format for transport of MPEG-4 elementary streams
[28] enables interleaving of access units and transmission of
multiple access units in the same RTP packet. An access unit is
specified in the H.264 coding standard to consist of all NAL units
that are associated with a primary coded picture according to
subclause 7.4.1.2 of [1]. Consequently, slices of different
pictures cannot be interleaved and the multi-picture slice
interleaving technique (see section 12.6) for improved error
resilience cannot be used.
14. Acknowledgements
The authors thank Roni Even, Dave Lindbergh, Philippe Gentric,
Gonzalo Camarillo, Joerg Ott, and Colin Perkins for careful review.
15. Full Copyright Statement
Copyright (C) The Internet Society (2004). This document is subject
to the rights, licenses and restrictions contained in BCP 78, and
except as set forth therein, the authors retain all their rights.
This document and the information contained herein are provided on
an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE
INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
16. Intellectual Property Notice
The IETF takes no position regarding the validity or scope of any
Intellectual Property Rights or other rights that might be claimed
to pertain to the implementation or use of the technology described
in this document or the extent to which any license under such
rights might or might not be available; nor does it represent that
it has made any independent effort to identify any such rights.
Information on the procedures with respect to rights in RFC
documents can be found in BCP 78 and BCP 79.
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Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
attempt made to obtain a general license or permission for the use
of such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository
at http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at ietf-
ipr@ietf.org.
17. References
17.1. Normative References
[1] ITU-T Recommendation H.264, "Advanced video coding for generic
audiovisual services", May 2003.
[2] ISO/IEC International Standard 14496-10:2003.
[3] S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[4] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD
64, RFC 3550, July 2003.
[5] M. Handley and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[6] S. Josefsson, "The Base16, Base32, and Base64 Data Encodings",
RFC 3548, July 2003.
[7] ITU-T Recommendation T.35, "Procedure for the allocation of
ITU-T defined codes for non-standard facilities", February
2000.
[8] J. Rosenberg, and H. Schulzrinne, "An Offer/Answer Model with
the Session Description Protocol (SDP)", RFC 3264, June 2002.
17.2. Informative References
[9] "Draft ITU-T Recommendation and Final Draft International
Standard of Joint Video Specification (ITU-T Rec. H.264 |
ISO/IEC 14496-10 AVC)", available from ftp://ftp.imtc-
files.org/jvt-experts/2003_03_Pattaya/JVT-G050r1.zip, May
2003.
[10] A. Luthra, G.J. Sullivan, and T. Wiegand (eds.), Special Issue
on H.264/AVC. IEEE Transactions on Circuits and Systems on
Video Technology, July 2003.
[11] P. Borgwardt, "Handling Interlaced Video in H.26L", VCEG-
N57r2, available from http://ftp3.itu.int/av-arch/video-
site/0109_San/VCEG-N57r2.doc, September 2001.
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[12] C. Borman et. Al., "RTP Payload Format for the 1998 Version of
ITU-T Rec. H.263 Video (H.263+)", RFC 2429, October 1998.
[13] ISO/IEC IS 14496-2.
[14] S. Wenger, "H.26L over IP", IEEE Transaction on Circuits and
Systems for Video technology, July 2003.
[15] S. Wenger, "H.26L over IP: The IP Network Adaptation Layer",
Proceedings Packet Video Workshop 02, April 2002
[16] T. Stockhammer, M.M. Hannuksela, and S. Wenger, "H.26L/JVT
Coding Network Abstraction Layer and IP-based Transport" in
Proc. ICIP 2002, Rochester, NY, September 2002.
[17] ITU-T Recommendation H.241, "Extended video procedures and
control signals for H.300 series terminals", 2004.
[18] H. Schulzrinne and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July
2003.
[19] B. Cain, S. Deering, I. Kouvelas, B. Fenner, and A.
Thyagarajan, "Internet Group Management Protocol, Version 3",
RFC 3376, October 2002.
[20] ITU-T Recommendation H.223, "Multiplexing protocol for low bit
rate multimedia communication", July 2001.
[21] J. Rosenberg, H. Schulzrinne, "An RTP Payload Format for
Generic Forward Error Correction", RFC 2733, December 1999.
[22] T. Stockhammer, T. Wiegand, T. Oelbaum, and F. Obermeier,
"Video Coding and Transport Layer Techniques for H.264/AVC-
Based Transmission over Packet-Lossy Networks", IEEE
International Conference on Image Processing (ICIP 2003),
Barcelona, Spain, September 2003.
[23] V. Varsa, M. Karczewicz, "Slice interleaving in compressed
video packetization", Packet Video Workshop 2000.
[24] S.H. Kang and A. Zakhor, "Packet scheduling algorithm for
wireless video streaming," International Packet Video Workshop
2002, available http://www.pv2002.org.
[25] M.M. Hannuksela, "Enhanced concept of GOP", JVT-B042,
available http://ftp3.itu.int/av-arch/video-site/0201_Gen/JVT-
B042.doc , January 2002.
[26] S. Wenger, "Video Redundancy Coding in H.263+", 1997
International Workshop on Audio-Visual Services over Packet
Networks, September 1997.
[27] Y.-K. Wang, M.M. Hannuksela, and M. Gabbouj, "Error Resilient
Video Coding Using Unequally Protected Key Pictures", in Proc.
International Workshop VLBV03, September 2003.
[28] J. van der Meer, D. Mackie, V. Swaminathan, D. Singer, and P.
Gentric, "RTP Payload Format for Transport of MPEG-4
Elementary Streams", RFC 3640, November 2003.
[29] Baugher, McGrew, Carrara, Naslund, and Norrman, "The Secure
Real-time Transport Protocol," RFC 3711, Internet Engineering
Task Force, March 2004.
[30] H. Schulzrinne, A. Rao, R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, Internet Engineering Task Force,
April 1998.
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[31] M. Handley, C. Perkins, E. Whelan, "Session Announcement
Protocol", RFC 2974, Internet Engineering Task Force, June
2001.
[32] ISO/IEC 14496-15: "Information technology - Coding of audio-
visual objects - Part 15: Advanced Video Coding (AVC) file
format".
[33] D. Singer, and R. Castagno, "MIME Type Registrations for 3GPP
Multimedia files", Internet Draft,
draft-singer-avt-3gpp-mime-01, Sep 2003.
Author's Addresses
Stephan Wenger Phone: +49-172-300-0813
TU Berlin / Teles AG Email: stewe@stewe.org
Franklinstr. 28-29
D-10587 Berlin
Germany
Miska M. Hannuksela Phone: +358-7180-73151
Nokia Corporation Email: miska.hannuksela@nokia.com
P.O. Box 100
33721 Tampere
Finland
Thomas Stockhammer Phone: +49-89-28923474
Institute for Communications Eng. Email: stockhammer@ei.tum.de
Munich University of Technology
D-80290 Munich
Germany
Magnus Westerlund Phone: +46-8-4048287
Multimedia Technologies Email:
Ericsson Research EAB/TVA/A magnus.westerlund@ericsson.com
Ericsson AB
Torshamsgatan 23
SE-164 80 Stockholm
Sweden
David Singer Phone +1 408 974-3162
QuickTime Engineering Email: singer@apple.com
Apple
1 Infinite Loop MS 302-3MT
Cupertino
CA 95014
USA
18. RFC Editor Considerations
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The RFC editor is requested to remove this section and Annex A
before publications as a RFC. The RFC editor is also requested to
replace all occurrences of XXXX with the RFC number this document
receive.
If available at the time of publication please do update reference
33 with the assigned RFC number.
Annex A: Changes relative to draft-ietf-avt-rtp-h264-08.txt
[This section will be removed in a future version of this draft.]
This memo contains the following technical changes relative to the
previous I-D:
o Editorial fixes as requested by the I-D review
o Fixed table and figure numbering
o Clarified the term network element and introduced MANE
abbreviation
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