| Rfc | 3984 |
|---|
| Title | RTP Payload Format for H.264 Video |
|---|
| Author | S. Wenger, M.M. Hannuksela, T.
Stockhammer, M. Westerlund, D. Singer |
|---|
| Date | February 2005 |
|---|
| Format: | TXT,
HTML |
|---|
| Obsoleted by | RFC6184 |
|---|
| Status: | PROPOSED STANDARD |
|---|
|
Network Working Group S. Wenger
Request for Comments: 3984 M.M. Hannuksela
Category: Standards Track T. Stockhammer
M. Westerlund
D. Singer
February 2005
RTP Payload Format for H.264 Video
Status of This Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2005).
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, as it supports
applications from simple low bit-rate conversational usage, to
Internet video streaming with interleaved transmission, to high bit-
rate video-on-demand.
Table of Contents
1. Introduction.................................................. 3
1.1. The H.264 Codec......................................... 3
1.2. Parameter Set Concept................................... 4
1.3. Network Abstraction Layer Unit Types.................... 5
2. Conventions................................................... 6
3. Scope......................................................... 6
4. Definitions and Abbreviations................................. 6
4.1. Definitions............................................. 6
5. RTP Payload Format............................................ 8
5.1. RTP Header Usage........................................ 8
5.2. Common Structure of the RTP Payload Format.............. 11
5.3. NAL Unit Octet Usage.................................... 12
5.4. Packetization Modes..................................... 14
5.5. Decoding Order Number (DON)............................. 15
5.6. Single NAL Unit Packet.................................. 18
5.7. Aggregation Packets..................................... 18
5.8. Fragmentation Units (FUs)............................... 27
6. Packetization Rules........................................... 31
6.1. Common Packetization Rules.............................. 31
6.2. Single NAL Unit Mode.................................... 32
6.3. Non-Interleaved Mode.................................... 32
6.4. Interleaved Mode........................................ 33
7. De-Packetization Process (Informative)........................ 33
7.1. Single NAL Unit and Non-Interleaved Mode................ 33
7.2. Interleaved Mode........................................ 34
7.3. Additional De-Packetization Guidelines.................. 36
8. Payload Format Parameters..................................... 37
8.1. MIME Registration....................................... 37
8.2. SDP Parameters.......................................... 52
8.3. Examples................................................ 58
8.4. Parameter Set Considerations............................ 60
9. Security Considerations....................................... 62
10. Congestion Control............................................ 63
11. IANA Considerations........................................... 64
12. Informative Appendix: Application Examples.................... 65
12.1. Video Telephony according to ITU-T Recommendation H.241
Annex A................................................. 65
12.2. Video Telephony, No Slice Data Partitioning, No NAL
Unit Aggregation........................................ 65
12.3. Video Telephony, Interleaved Packetization Using NAL
Unit Aggregation........................................ 66
12.4. Video Telephony with Data Partitioning.................. 66
12.5. Video Telephony or Streaming with FUs and Forward
Error Correction........................................ 67
12.6. Low Bit-Rate Streaming.................................. 69
12.7. Robust Packet Scheduling in Video Streaming............. 70
13. Informative Appendix: Rationale for Decoding Order Number..... 71
13.1. Introduction............................................ 71
13.2. Example of Multi-Picture Slice Interleaving............. 71
13.3. Example of Robust Packet Scheduling..................... 73
13.4. Robust Transmission Scheduling of Redundant Coded
Slices.................................................. 77
13.5. Remarks on Other Design Possibilities................... 77
14. Acknowledgements.............................................. 78
15. References.................................................... 78
15.1. Normative References.................................... 78
15.2. Informative References.................................. 79
Authors' Addresses................................................ 81
Full Copyright Statement.......................................... 83
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 [2] (both also known as Advanced
Video Coding, or AVC). Recommendation H.264 was approved by ITU-T on
May 2003, and the approved draft specification is available for
public review [8]. 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 nearly lossless coding. Compared to the current
state of technology, the overall performance of H.264 is such that
bit rate savings of 50% or more 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 [9].
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, and motion compensated
prediction; and a loop filter. It follows the general concept of
most of today's video codecs, a macroblock-based coder that uses
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 information). Macroblocks in
slices are arranged in scan order unless a different macroblock
allocation is specified, by 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 transmission over packet networks or
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
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 is common
in the form of the Temporal Reference in earlier video compression
standards). Also, there are NAL units that affect many pictures and
that are, therefore, inherently timeless. For this reason, the
handling of the RTP timestamp requires some special considerations
for 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 [10] or MPEG-4's Header Extension Code (HEC) [11]
unnecessary. This was achieved by decoupling information 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 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.
To be able to change picture parameters (such as the picture size)
without having 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 be possible
that they are 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 [12], [13],
and [14].
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.
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 BCP 14, RFC 2119 [3].
This specification uses the notion of setting and clearing a bit when
bit fields are handled. 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 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 by 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.
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 to 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
signaling (e.g., to learn about the payload type mappings of
the media streams), and in that it has to be trusted when
working with SRTP. The advantage of using MANEs is that they
allow packets to be dropped according to the needs of the media
coding. For example, if a MANE has 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 them in order to remove the congestion
and/or keep the delay low.
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 one NAL unit is encapsulated 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.
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 to 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, 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 been 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. 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 has 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.
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.
If 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, in which film
content that consists of coded frames is displayed on a 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: Because 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.
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 as follows:
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.
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 violations may be present in the
payload or in the NAL unit type octet. The simplest decoder
reaction 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 from 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 they do less important NAL units. The highest
transport priority is 11, followed by 10, and then by 01; finally,
00 is the lowest.
Informative note: Any non-zero value of NRI is handled
identically in H.264 decoders. Therefore, receivers need not
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.
For NAL units having nal_unit_type equal to 7 or 8 (indicating a
sequence parameter set or a picture parameter set, respectively),
an H.264 encoder SHOULD set the value of NRI to 11 (in binary
format). 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), an H.264 encoder SHOULD set the
value of NRI to 11 (in binary format).
For a mapping of the remaining nal_unit_types to NRI values, the
following example MAY be used and has been shown to be efficient
in a certain environment [13]. 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 bitstream 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.
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,
as 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 [15] (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.
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 examples of 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.
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,
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.
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, incremented
by one.
An example of the 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 will be equal to one, even in error-free
transmission. An increment by one is not required, as 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 bit rate 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. When transmitting the first
intra picture, the originator does not exactly know how many NAL
units will 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 when they are transmitted, and gaps in
values of DON may occur.
5.6. Single NAL Unit Packet
The single NAL unit packet defined here MUST contain 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
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. 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 including DON (STAP-B).
o Multi-time aggregation packet (MTAP): aggregates NAL units with
potentially differing NALU-time. Two different MTAPs are defined,
differing 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 four different
aggregation units and their characteristics.
The structure of the RTP payload format for aggregation packets is
presented in Figure 3.
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 that 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 sections 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 so 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.
5.7.1. Single-Time Aggregation Packet
Single-time aggregation packet (STAP) SHOULD be used whenever NAL
units are aggregated 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. For each successive NAL unit in
appearance order in an STAP-B, the value of DON 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
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 an 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.
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 in which
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
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 the flexibility of the MTAP, but
the overhead is also higher.
The structure of the multi-time aggregation units for MTAP16 and
MTAP24 are presented in Figures 10 and 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. The 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
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 would have 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 wraparound of the timestamp field, thus enabling one to
determine the smallest value if the timestamp wraps. Such an
"earliest" aggregation unit may not be the first one in the order
in which 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.
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 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.
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|MTAP24 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 RTP packet including a multi-time aggregation
packet of type MTAP24 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 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 a 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
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
When set to one, the Start bit indicates the start of a fragmented
NAL unit. 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
When set to one, the End bit 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. 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.
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., the
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 so that if the fragmentation unit payloads of consecutive
FUs are sequentially concatenated, the payload of the fragmented NAL
unit can be 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 nearly lossless environments. These
senders can be characterized in that they packetize NALU fragments
before the NALU is completely generated and, hence, before the
NALU size is 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 weighed against the increased risk of the loss
of the NALU because of the additional packets 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 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.
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 thus sharing the same RTP
timestamp value) MAY be sent in any order permitted by the
applicable profile defined in [1]; however, 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, as 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 using 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 [18], 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, the packetization-mode is not present,
or no other packetization mode is signaled by external 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 [15] (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. Therefore,
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
The receiver includes a receiver buffer to compensate for
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 in which 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 under the assumption that there
is no transmission delay jitter. To make a difference from a
practical receiver buffer that is also used for compensation of
transmission delay jitter, the receiver buffer is here after 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
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, in the deinterleaving buffer as follows.
NAL units of aggregation packets are stored in 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:
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.
The order in which 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
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 have 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
bitstream. 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 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 will 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
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 to
indicate 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
it decodes 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
supported for the signaled profile. The
profile-iop byte indicates whether the codec
has additional limitations whereby only the
common subset of the algorithmic features and
limitations of the profiles signaled with the
profile-iop byte and of 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
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 [7]).
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
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 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 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.
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 is
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
is 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 are 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 is
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
[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 bit rate
than 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 of the
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 of 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
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 is 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,
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 4036458 bits
(1550000 / 384000 * 1000 * 1000).
The value of max-br MUST be greater 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 that 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; therefore, 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
decoded picture (at least partly), and therefore
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 a 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
any 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, but a picture
parameter set NAL unit can contain several
hundreds of bytes.
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). Therefore, 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 an
RTP payload type or the capabilities of a
receiver implementation. Only a single
configuration point can be indicated; thus,
when capabilities to support more than one
packetization-mode are declared, 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 3984, MUST be used. This mode is in use in
standards using ITU-T Recommendation H.241 [15]
(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 3984, MUST be used. When the value of
packetization-mode is equal to 2, the
interleaved mode, as defined in section 6.4 of
RFC 3984, 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.
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 3984. 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 4294967295,
inclusive.
Informative note: sprop-deint-buf-req
indicates the required size of the
deinterleaving buffer only. When network
jitter can occur, an appropriately sized
jitter buffer has to be provisioned for
as well.
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
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 4294967295, inclusive.
Informative note: deint-buf-cap indicates
the maximum possible size of the
deinterleaving buffer of the receiver only.
When network jitter can occur, an
appropriately sized jitter buffer has to
be provisioned for as well.
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
that decoding starts 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 in the
following series:
0 -1 1 0 -1 1 0 -1 1 ...
Thus, in terms of intervals of NAL unit
transmission times, the value of
sprop-init-buf-time in this
example is 1.
The parameter is coded as a non-negative base10
integer representation in clock ticks of a 90-
kHz clock. If the parameter is not present,
then no initial buffering time value is
defined. Otherwise the value of sprop-init-
buf-time MUST be an integer in the range of 0
to 4294967295, inclusive.
In addition to the signaled sprop-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 a 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)
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 3984.
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 a
higher cost than NALUs conforming to the
limitation.
The value of max-rcmd-nalu-size MUST be an
integer in the range of 0 to 4294967295,
inclusive. If this parameter is not specified,
no known limitation to the NALU size exists.
Senders still have 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.
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
[29]. This definition is utilized by other
file formats, such as the 3GPP multimedia file
format (MIME type video/3gpp) [30] or the MP4
file format (MIME type video/mp4).
Security considerations:
See section 9 of RFC 3984.
Public specification:
Please refer to RFC 3984 and its section 15.
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:
stewe@stewe.org
Change controller:
IETF Audio/Video Transport working group
delegated from the IESG.
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.
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 H.264 is offered over RTP using SDP in an Offer/Answer model [7]
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 [7]. 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, has 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.
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 that 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. Rather then being bound to
the payload type, the values may have to be applied to another
payload type when being sent, as they apply for the
configuration.
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 has 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 used when the value for "sprop-deint-buf-req"
is selected 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.
Informative note: care must be taken when parameter sets are
added 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.
Thus, 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 for which "a=sendrecv" or no direction
attribute is used, or in offers and answers for which "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
- 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 for which 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
Furthermore, 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 sender's 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"
parameter. This expresses values a receiver expects to be used
and must be used verbatim on the sender side.
o When a sender's capabilities are declared, 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. Thus, 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 format's 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 their usage, in most cases a
second offer is required from the offerer to provide the stream
properties parameters that the media sender will use. This also
has the effect that the offerer has to be able to receive this
media format configuration, not only to 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 H.264 over RTP is offered with SDP in a declarative style, as in
RTSP [27] or SAP [28], the following considerations are necessary.
o All parameters capable of indicating the properties of both a NAL
unit stream and 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 the 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
values of the parameters provided; otherwise, 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 wherein 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
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 presents the same codec configuration in three
different packetization formats. PT 98 represents single NALU mode,
PT 99 non-interleaved mode; PT 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, whereas 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.
The answerer also accepts the reception of the two configurations
that payload types 99 and 100 represent. It provides the initial
parameter sets for the answerer-to-offerer direction, and for
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 do not
represent 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. Therefore, 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 on how principles A and B must be
implemented within session control protocols. It is independent of
the particular protocol used. Principle C is supported by the RTP
payload format defined in this specification.
The picture and sequence parameter set NALUs SHOULD NOT be
transmitted in the RTP payload unless reliable transport is provided
for RTP, as a loss of a parameter set of either type will likely
prevent 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 and that 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). Therefore, it 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 parameter sets are added or updated, principle C is
vulnerable to transmission errors as described above, and
therefore principle B is RECOMMENDED.
- When parameter sets are added or updated, 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 RECOMMENDED that a sender does not start sending NALUs
requiring the updated parameter sets prior to acknowledgement of
delivery from the signaling protocol.
- When parameter sets are updated, the following synchronization
issue should be taken into account. When overwriting a parameter
set at the receiver, the sender has to ensure that the parameter
set in question is not needed by any NALU present in the network
or receiver buffers. Otherwise, decoding with a wrong parameter
set may occur. To lessen this problem, it is RECOMMENDED either
to 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 new parameter sets are added, previously unused parameter set
identifiers are used. This avoids the problem identified in the
previous paragraph. However, in a multiparty session, unless a
synchronized control protocol is used, there is a risk that
multiple entities try to add different parameter sets for the same
identifier, which has 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, principles 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 have 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 has to be taken into account.
The loss probability should be reduced using the mechanisms discussed
above.
- When parameter sets are initially provided using principle A and
then later added or updated in-band (principle C), there is a risk
associated with updating the parameter sets delivered out-of-band.
If receivers miss some in-band updates (for example, because of a
loss or a late tune-in), those receivers attempt to decode the
bitstream using out-dated parameters. It is RECOMMENDED that
parameter set IDs be 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 in any appropriate RTP profile (for example,
[16]). This implies that confidentiality of the media streams is
achieved by encryption; for example, through the application of SRTP
[26]. Because the data compression used with this payload format is
applied end-to-end, any encryption needs to be performed after
compression.
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 pathological datagrams
into the stream that are complex to decode and that cause the
receiver to be overloaded. H.264 is particularly vulnerable to such
attacks, as it is extremely simple to generate datagrams containing
NAL units that affect the decoding process of many future NAL units.
Therefore, the usage of data origin authentication and data integrity
protection of at least the RTP packet is RECOMMENDED; for example,
with SRTP [26].
Note that the appropriate mechanism to ensure confidentiality and
integrity of RTP packets and their payloads is very dependent on the
application and on the transport and signaling protocols employed.
Thus, although SRTP is given as an example above, other possible
choices exist.
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.
End-to-End security with either authentication, integrity or
confidentiality protection will prevent a MANE from performing
media-aware operations other than discarding complete packets. And
in the case of confidentiality protection it will even be prevented
from performing discarding of packets in a media aware way. To allow
any MANE to perform its operations, it will be required to be a
trusted entity which is included in the security context
establishment.
10. Congestion Control
Congestion control for RTP SHALL be used in accordance with RFC 3550
[4], and with any applicable RTP profile; e.g., RFC 3551 [16]. An
additional requirement if best-effort service is being used is:
users of this payload format MUST monitor packet loss to ensure that
the packet loss rate is within acceptable parameters. Packet loss is
considered acceptable if a TCP flow across the same network path, and
experiencing the same network conditions, would achieve an average
throughput, measured on a reasonable timescale, that is not less than
the RTP flow is achieving. This condition can be satisfied by
implementing congestion control mechanisms to adapt the transmission
rate (or the number of layers subscribed for a layered multicast
session), or by arranging for a receiver to leave the session if the
loss rate is unacceptably high.
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 [22] 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 when non-
downgradable parameters (such as the profile part of the
profile/level ID) are required to be changed does it become necessary
to terminate and re-start the media stream. This may be accomplished
by using a different RTP payload type.
MANEs MAY follow the suggestions outlined in section 7.3 and remove
certain unusable packets from the packet stream when that stream was
damaged due to previous packet losses. This can help reduce the
network load in certain special cases.
11. IANA Consideration
IANA has registered one new MIME type; see section 8.1.
12. Informative Appendix: Application Examples
This payload specification is very flexible in its use, in order to
cover the extremely wide application space anticipated for H.264.
However, this 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 [15]
as a packetization scheme. The packetization mechanism defined in
this Annex is technically identical with a small subset of this
specification.
When a system operates according to H.241 Annex A, parameter set 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, picture parameters
such as picture size or optional modes never change during the
lifetime of a connection. Therefore, 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. As all necessary
parameter set information is established before the RTP session
starts, there is no need for sending any parameter set NAL units.
Slice data partitioning is not used, either. Thus, the RTP packet
stream basically consists of NAL units that carry single coded
slices.
The encoder chooses the size of coded slice NAL units so 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 designs using RFC 2429 packetization [10]. It has been
implemented, and good results were reported [12].
The VCL encoder codes the source picture so 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. 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, as 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 [17]. Coding individual slices according to
this packetization scheme provides 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 into several RTP packets with only one NAL unit, which
are preferred in a wireless network, and vice versa.
12.4. Video Telephony with Data Partitioning
This scheme has been implemented and has been shown to offer good
performance, especially at higher packet loss rates [12].
Data Partitioning is known to be useful only when some form of
unequal error protection is available. Normally, in single-session
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 [18], for example, specifies which media
packets are associated with the FEC packet.
In all cases, the incurred overhead is substantial but is in the same
order of magnitude as the number of bits that have otherwise been
spent for intra information. However, this mechanism does not add
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 has been implemented and has been shown to provide good
performance, especially at higher packet loss rates [19].
The most efficient means to combat packet losses for scenarios where
retransmissions are not applicable is forward error correction (FEC).
Although application layer, end-to-end use of FEC is often less
efficient than an 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 [18] 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 a code is used 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 a packet loss probability
of 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 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 for which each video frame is
coded in a single slice. Applying FEC, one could use a simple
code; e.g., (n=2, k=1). That is, each NAL unit would basically
just be repeated. The disadvantage is obviously the bad code
performance according to d), above, and the low flexibility, as
only (n, k=1) codes can be used.
2) The video encoder produces NAL units for which 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 when more
RTP packets are used per frame. Packets of completely different
length might also be connected, which decreases bit rate
efficiency according to b), above. 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, for which a certain frame
contains k slices of possibly almost equal length. Then, applying
FEC, a better code, e.g., (n=24, k=12), can be used over the
sequence of NAL units for each frame. The delay compared to that
of 2), above, may be reduced, but several disadvantages are
obvious. First, the coding efficiency of the encoded video is
lowered significantly, as slice-structured coding reduces intra-
frame prediction and additional slice overhead is necessary.
Second, 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; therefore, FEC with a reasonable k and
n can be applied, even if the encoder made no effort to produce
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 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
layer redundancy (such as periodic intra frames) is present.
Consequently, the same coded video sequence can be used to achieve
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 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.
If the error probability depends on the length of the transmitted
packet (e.g., in case of mobile transmission [14]), the benefits of
applying FUs with FEC are even more obvious. Basically, the
flexibility of the size of FUs allows appropriate FEC to be applied
for each NAL unit and unequal error protection of NAL units.
When FUs and FEC are used, the incurred overhead is substantial but
is 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 [19],
it was shown that the overall performance of the FEC-based approach
enhanced quality when using the same error rate and same overall bit
rate, including the overhead.
12.6. Low Bit-Rate Streaming
This scheme has been implemented with H.263 and non-standard RTP
packetization and has given good results [20]. 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 for 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 interleaved slices from several
pictures. 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 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
be relatively successful.
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 [21]. 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. Rather, it typically buffers the incoming
data for a few seconds. This buffering helps maintain continuous
playback, as, 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 recovered perfectly. 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, as
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 [22]. 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
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 from left to right is 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,
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 bit
rate control purposes, for example. In some cases, the pre-encoding
buffer may not exist; instead, 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:
... 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, 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, as 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 that 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.
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. As 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 as follows:
... 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
Note 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 using the redundant codec picture feature include
the video redundancy coding [23] and the protection of "key pictures"
in multicast streaming [24].
One property of many error-prone video communications systems is that
transmission errors are often bursty. Therefore, they may affect
more than one consecutive transmission packets in transmission order.
In low bit-rate 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. To make the transmission scheme more tolerant of
bursty transmission errors, it is beneficial to transmit the primary
coded picture and redundant coded picture separated by more than a
single packet. 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).
o Coded slices from multiple coded video sequences cannot be
interleaved, as 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
[25] 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 comprise all NAL units
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, Gary Sullivan, Joerg Ott, and Colin Perkins for
careful review.
15. References
15.1. 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] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[4] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[5] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[6] Josefsson, S., "The Base16, Base32, and Base64 Data Encodings",
RFC 3548, July 2003.
[7] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
15.2. Informative References
[8] "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 http://ftp3.itu.int/av-
arch/jvt-site/2003_03_Pattaya/JVT-G050r1.zip, May 2003.
[9] Luthra, A., Sullivan, G.J., and T. Wiegand (eds.), Special Issue
on H.264/AVC. IEEE Transactions on Circuits and Systems on Video
Technology, July 2003.
[10] Bormann, C., Cline, L., Deisher, G., Gardos, T., Maciocco, C.,
Newell, D., Ott, J., Sullivan, G., Wenger, S., and C. Zhu, "RTP
Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
(H.263+)", RFC 2429, October 1998.
[11] ISO/IEC IS 14496-2.
[12] Wenger, S., "H.26L over IP", IEEE Transaction on Circuits and
Systems for Video technology, Vol. 13, No. 7, July 2003.
[13] Wenger, S., "H.26L over IP: The IP Network Adaptation Layer",
Proceedings Packet Video Workshop 02, April 2002.
[14] Stockhammer, T., Hannuksela, M.M., and S. Wenger, "H.26L/JVT
Coding Network Abstraction Layer and IP-based Transport" in
Proc. ICIP 2002, Rochester, NY, September 2002.
[15] ITU-T Recommendation H.241, "Extended video procedures and
control signals for H.300 series terminals", 2004.
[16] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[17] ITU-T Recommendation H.223, "Multiplexing protocol for low bit
rate multimedia communication", July 2001.
[18] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
Generic Forward Error Correction", RFC 2733, December 1999.
[19] Stockhammer, T., Wiegand, T., Oelbaum, T., 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.
[20] Varsa, V. and M. Karczewicz, "Slice interleaving in compressed
video packetization", Packet Video Workshop 2000.
[21] Kang, S.H. and A. Zakhor, "Packet scheduling algorithm for
wireless video streaming," International Packet Video Workshop
2002.
[22] Hannuksela, M.M., "Enhanced concept of GOP", JVT-B042, available
http://ftp3.itu.int/av-arch/video-site/0201_Gen/JVT-B042.doc,
January 2002.
[23] Wenger, S., "Video Redundancy Coding in H.263+", 1997
International Workshop on Audio-Visual Services over Packet
Networks, September 1997.
[24] Wang, Y.-K., Hannuksela, M.M., and M. Gabbouj, "Error Resilient
Video Coding Using Unequally Protected Key Pictures", in Proc.
International Workshop VLBV03, September 2003.
[25] van der Meer, J., Mackie, D., Swaminathan, V., Singer, D., and
P. Gentric, "RTP Payload Format for Transport of MPEG-4
Elementary Streams", RFC 3640, November 2003.
[26] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
3711, March 2004.
[27] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
[28] Handley, M., Perkins, C., and E. Whelan, "Session Announcement
Protocol", RFC 2974, October 2000.
[29] ISO/IEC 14496-15: "Information technology - Coding of audio-
visual objects - Part 15: Advanced Video Coding (AVC) file
format".
[30] Castagno, R. and D. Singer, "MIME Type Registrations for 3rd
Generation Partnership Project (3GPP) Multimedia files", RFC
3839, July 2004.
Authors' Addresses
Stephan Wenger
TU Berlin / Teles AG
Franklinstr. 28-29
D-10587 Berlin
Germany
Phone: +49-172-300-0813
EMail: stewe@stewe.org
Miska M. Hannuksela
Nokia Corporation
P.O. Box 100
33721 Tampere
Finland
Phone: +358-7180-73151
EMail: miska.hannuksela@nokia.com
Thomas Stockhammer
Nomor Research
D-83346 Bergen
Germany
Phone: +49-8662-419407
EMail: stockhammer@nomor.de
Magnus Westerlund
Multimedia Technologies
Ericsson Research EAB/TVA/A
Ericsson AB
Torshamsgatan 23
SE-164 80 Stockholm
Sweden
Phone: +46-8-7190000
EMail: magnus.westerlund@ericsson.com
David Singer
QuickTime Engineering
Apple
1 Infinite Loop MS 302-3MT
Cupertino
CA 95014
USA
Phone +1 408 974-3162
EMail: singer@apple.com
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