Network Working Group S. Wenger
Internet Draft Y.-K. Wang
Document: draft-wenger-avt-rtp-svc-03.txt T. Schierl
Expires: April 2007
October 2006
RTP Payload Format for SVC Video
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Copyright Notice
Copyright (C) The Internet Society (2006).
INTERNET-DRAFT RTP Payload Format for SVC Video October 2006
Abstract
This memo describes an RTP Payload format for the scalable extension
of the ITU-T Recommendation H.264 video codec which is the
technically identical to 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 the
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.
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Table of Content
RTP Payload Format for SVC Video...............................1
1. Introduction..............................................5
1.1. SVC -- the scalable extensions of H.264/AVC................5
2. Conventions...............................................5
3. The SVC Codec.............................................6
3.1. Overview................................................6
3.2. Parameter Set Concept....................................7
3.3. Network Abstraction Layer Unit Header......................7
4. Scope...................................................11
5. Definitions and Abbreviations .............................11
5.1. Definitions............................................11
5.2. Abbreviations..........................................14
6. RTP Payload Format.......................................14
6.1. Design Principles.......................................14
6.2. RTP Header Usage........................................15
6.3. Common Structure of the RTP Payload Format................16
6.4. NAL Unit Header Usage...................................17
6.5. Packetization Modes.....................................18
6.6. Decoding Order Number (DON)..............................18
6.7. Single NAL Unit Packet..................................19
6.8. Aggregation Packets.....................................19
6.9. Fragmentation Units (FUs)................................19
6.10. Payload Content Scalability Information (PACSI) NAL Unit..19
7. Packetization Rules ......................................22
8. De-Packetization Process (Informative).....................22
9. Payload Format Parameters.................................22
9.1. MIME Registration.......................................23
9.2. SDP Parameters .........................................25
9.2.1. Mapping of MIME Parameters to SDP.......................25
9.2.2. Usage with the SDP Offer/Answer Model...................25
9.2.3. Usage with Session and SSRC multiplexing.................26
9.2.4. Usage in Declarative Session Descriptions................26
9.3. Examples...............................................26
9.4. Parameter Set Considerations.............................26
10. Security Considerations.................................26
11. Congestion Control......................................26
12. IANA Consideration......................................27
13. Informative Appendix: Application Examples................27
13.1. Introduction..........................................28
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13.2. Layered Multicast.....................................28
13.3. Streaming of an SVC scalable stream.....................29
13.4. Multicast to MANE, SVC scalable stream to endpoint........30
13.5. SSRC Multiplexing in case of using SRTP .................32
13.6. Scenarios currently not considered for complexity reasons.34
13.7. Scenarios currently not considered for being unaligned with
IP philosophy...............................................34
14. Acknowledgements........................................36
15. References.............................................36
15.1. Normative References...................................36
15.2. Informative References.................................37
16. Author's Addresses......................................37
17. Intellectual Property Statement..........................38
18. Disclaimer of Validity..................................38
19. Copyright Statement.....................................38
20. RFC Editor Considerations................................39
21. Open Issues............................................39
22. Changes Log............................................39
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1. Introduction
1.1. SVC -- the scalable extensions of H.264/AVC
This memo specifies an RTP [RFC3550] payload format for a
forthcoming new mode of the H.264/AVC video codec, known as Scalable
Video Coding (SVC). Formally, SVC will take the form of an Amendment
to ISO/IEC 14496 Part 10 [MPEG4-10], and likely as one or more new
Annexes of ITU-T Rec. H.264 [H.264]. It is planned to keep the
technical alignment between the two mentioned specifications, as
well as backward compatibility with previous versions of H.264/AVC.
The current working draft of SVC is available for public review
[SVC]. In this memo, SVC is used as an acronym for the mentioned
scalable extensions of H.264/AVC.
SVC covers all of H.264/AVC's applications, ranging from all forms
of digital compressed video from, low bit-rate Internet streaming
applications to HDTV broadcast and Digital Cinema applications with
nearly lossless coding.
This memo tries to follow a backward compatible enhancement
philosophy similar to what the video coding standardization
committees implement, by keeping as close an alignment to the
H.264/AVC payload RFC [RFC3984] as possible. It basically documents
the enhancements relevant from an RTP transport viewpoint, defines
signaling support for SVC, and deprecates the single NAL unit
packetization mode of RFC 3984.
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
[RFC2119].
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).
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3. The SVC Codec
3.1. Overview
SVC provides scalable video bitstreams. In SVC, a scalable video
bitstream contains a base layer conforming to the existing profiles
of H.264 as defined in [H.264] and one or more enhancement layers.
An enhancement layer may enhance the temporal resolution (i.e. the
frame rate), the spatial resolution, or the quality of the video
content represented by the lower layer or part thereof. The
scalable layers can be aggregated to a single RTP packet stream, or
transported independently.
The concept of video coding layer (VCL) and network abstraction
layer (NAL) is inherited from H.264. The VCL contains the signal
processing functionality of the codec; mechanisms such as transform,
quantization, motion-compensated prediction, loop filtering and
inter-layer prediction. A coded picture of a base or enhancement
layer consists of one or more slices. The Network Abstraction Layer
(NAL) encapsulates each slice generated by the VCL into one or more
Network Abstraction Layer Units (NAL units). Please consult RFC 3984
for a more in-depth discussion of the NAL unit concept. SVC
specifies the decoding order of these NAL units.
[Edt. Note: The definition of a ''coded picture'' is currently
under discussion in JVT. For now, we apply the same
definition as in the AVC specification within a give scalable
layer. That is, a ''coded picture'' consists of all the coded
slices having identical values of dependency_id,
quality_level and redundant_pic_cnt, respectively, in one
access unit.]
The term ''Layer'' in Video Coding Layer and Network Abstraction
Layer refers to a conceptual distinction, and is closely related to
syntax layers (block, macroblock, slice, ... layers). ''Layer'' here
describes a syntax level of the bitstream in contrast to the meaning
of layer as a nested part of the bitstream which may be discarded.
It should not be confused with base and enhancement layers.
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The concept of temporal scalability is not newly introduced by SVC,
as H.264 already supports it. In [H.264], sub-sequences have been
introduced in order to allow optional use of temporal layers. [SVC]
extends this approach by advertising the temporal layer information
within the NAL unit header, or suffix NAL units, as discussed in
section 3.3 and [SVC]. By our definition, the base layer may be
scalable in the temporal dimension (only).
The concept of scaling the visual content quality in the granularity
of complete enhancement layers, i.e. through omitting the transport
and decoding of entire enhancement layers, is denoted as coarse-
grained scalability (CGS). This is what is commonly understood as
scalability in the IETF community. According to SVC, a CGS layer
may be a spatial or quality (SNR) enhancement layer.
In some cases, the bit rate of a given enhancement layer may be
reduced by truncating bits from individual NAL units. Truncation
leads to a graceful degradation of the video quality of the
reproduced enhancement layer. This concept is known as Fine
Granularity Scalability (FGS). In SVC, FGS is provided by a concept
known as progressive refinement slices.
3.2. Parameter Set Concept
The parameter set concept is inherited from [H.264]. Please see
section 1.2 of RFC 3984 for more details.
In SVC, pictures from different layers may use the same sequence or
picture parameter set, but may also use different sequence or
picture parameter sets. If different sequence or picture parameter
sets are used, then, at any time instant during the decoding
process, there may be more than one active sequence or picture
parameter set. Any specific active sequence parameter set remains
unchanged throughout a coded video sequence in the layer in which
the active sequence parameter set is referred to. The active
picture parameter set remains unchanged within a coded picture.
3.3. Network Abstraction Layer Unit Header
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An SVC NAL unit consists of a header of four bytes and the payload
byte string. SVC extends by that the NAL unit header defined in
[H.264] by three additional bytes. The header indicates the type of
the NAL unit, the (potential) presence of bit errors or syntax
violations in the NAL unit payload, information regarding the
relative importance of the NAL unit for the decoding process, the
layer decoding dependency information, and FGS fragmentation
information. This RTP payload specification is designed to be
unaware of the bit string in the NAL unit payload.
The NAL unit header 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 header are formally
specified in [SVC], but the essential properties of the NAL unit
header are summarized below.
The first byte of the NAL unit header has the following format (the
bit fields are the same as in [H.264] and [RFC3984], while the
semantics have changed slightly, in a backward compatible way):
+---------------+
|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+
|F|NRI| Type |
+---------------+
F: 1 bit
forbidden_zero_bit. H.264 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 in the same layer. 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
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nal_unit_type. This component specifies the NAL unit payload type
as defined in table 7-1 of [SVC], 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 [SVC].
Previously, NAL unit types 20 and 21 (among others) have been
reserved for future extensions. SVC is using these two NAL unit
types. They indicate the presence of three more bytes as shown
below.
+---------------+---------------+---------------+
|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|RR | PRID | TL | DID | QL|R|B|U|D|G|L| O |
+---------------+---------------+---------------+
RR: 2 bits
reserved_zero_two_bits. Reserved bits for future extension. RR
MUST be zero.
PRID: 6 bits
simple_priority_id. This component specifies a priority identifier
for the NAL unit. A lower value of PRID indicates a higher
priority.
TL: 3 bits
temporal_level indicates the temporal layer (or frame rate)
hierarchy. Informally put, a layer consisted of pictures of a
smaller temporal_level value has a smaller frame rate. A given
temporal layer typically depends on the lower temporal layers (i.e.
the temporal layers with smaller temporal_level values) but never
depends on any higher temporal layer.
DID: 3 bits
dependency_id denotes the inter-layer coding dependency hierarchy.
At any temporal location, a picture of a smaller dependency_id value
may be used for inter-layer prediction for coding of a picture of a
larger dependency_id value, while a picture of a larger
dependency_id value is disallowed to be used for inter-layer
prediction for coding of a picture of a smaller dependency_id value.
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QL: 2 bits
quality_level designates the quality level hierarchy of a
progressive refinement (PR) or quality (SNR) enhancement layer
slice. At any temporal location and with identical dependency_id
value, a picture with quality_level equal to ql uses a picture with
quality_level equal to ql-1 for inter-layer prediction.
R: 1 bit
reserved_zero_bit. Reserved bit for future extension. R MUST be
zero.
B: 1 bit
layer_base_flag indicates that no inter-layer prediction (of coding
mode, motion, sample value, and/or residual prediction) is used for
the current slice otherwise inter-layer prediction may be used.
U: 1 bit
use_base_prediction_flag indicates that the base representation of
the reference pictures (i.e. only NAL units of the reference
pictures with QL equal to zero are used for inter prediction) is
used during the inter prediction process.
D: 1 bit
discardable_flag. A value of 1 indicates that the content of the
NAL unit with dependency_id equal to currDependencyId is not used in
the decoding process of NAL units with dependency_id larger than
currDependencyId. Such NAL units can be discarded without risking
the integrity of higher scalable layers with larger values of
dependency_id. discardable_flag equal to 0 indicates that the
decoding of the NAL unit is required to maintain the integrity of
higher scalable layers with larger values of dependency_id.
G: 1 bit
fragmented_flag indicates that the current NAL unit is fragmented,
which may be the case for partitions of an FGS (progressive
refinement) slice.
L: 1 bit
last_fragemented_flag indicates, that the NAL unit is the last
fragment of a fragmented NAL unit.
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O: 2 bits
fragemnet_order indicates the order in which the NAL units with
fragmented_flag equal to 1 shall be ordered before the parsing
process is started, starting from lower values.
This memo introduces the same additional NAL unit types as RFC 3984,
which are presented in section 6.3. The NAL unit types defined in
this memo are marked as unspecified in [SVC]. Moreover, this
specification extends the semantics of F, NRI, PRID, D, TL, DID and
QL as described in section 6.4.
4. Scope
This payload specification can only be used to carry the "naked" SVC
NAL unit stream over RTP, and not the byte stream format according
to Annex B of [SVC]. Likely, the applications of this specification
will be in the IP based multimedia communications fields including
conversational multimedia, video telephony or video conferencing,
Internet streaming and TV over IP.
This specification allows, in a given RTP session, to encapsulate
NAL units belonging to
o the base layer only, detailed specification in [RFC3984], or
o one or more enhancement layers, or
o the base layer and one or more enhancement layers
5. Definitions and Abbreviations
5.1. Definitions
This document uses the definitions of [SVC] and [H.264]. The
following terms, defined in [SVC], are summed up for convenience:
scalable bitstream: An SVC compliant bit stream containing a base
layer and at least one enhancement layer.
suffix NAL unit: A NAL unit that immediately follows another NAL
unit in decoding order and contains descriptive information of the
preceding NAL unit, which is referred to as the associated NAL unit.
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A suffix NAL unit shall have nal_ref_idc equal to 20 or 21, shall
have dependency_id and quality_level both equal to 0, and shall not
contain a coded slice. A suffix NAL unit belongs to the same coded
picture as the associated NAL unit. A suffix NAL unit may be used
for indicating temporal levels within the base layer.
base layer: The base layer is typically representing the minimal
spatial resolution and, or minimal quality of an SVC bitstream. The
base layer must be fully complying with [H.264]. The base layer is
independently decodable without the requirement of using any other
layer of the SVC bitstream. In SVC context each slice NAL unit in
the base layer is associated with a suffix NAL unit, which has a
four-byte NAL unit header containing all the syntax elements
described in section 3.3.
[Edt. Note: The definition of ''base layer'' is not deadly
clear, mainly because of temporal scalability. One definition
is to call all the coded pictures in the lowest inter-layer
coding hierarchy (i.e. having both dependency_id and
quality_level equal to 0) as the base layer. This concept
works perfectly if there is no temporal scalability. Another
definition is to call all the coded pictures having
temporal_level, dependency_id and quality_level all equal to
0 as the base layer. Yet another definition is to define the
layer for which the bitstream of the scalable layer
representation is non-scalable as the base layer. However,
the absolutely non-scalable stream is the bitstream
consisting of only one IDR picture having both dependency_id
and quality_level equal to 0.]
operation point: An operation point of a SVC bitstream represents a
certain level of temporal, spatial and quality scalability. An
operation point contains all NAL units required for restoring a
valid bitstream (conforming to [SVC]) up to a certain SVC layer.
The operation point is further described by simple_priority_id,
temporal_level, dependency_id, and quality_level values of that
layer.
scalable enhancement layer: An SVC enhancement layer is identified
by simple_priority_id, temporal_level, dependency_id, and
quality_level as defined in [SVC] and summarized in section 3.3.
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access unit: A set of NAL units pertaining to a certain temporal
location. An access unit includes the slice data of the pictures of
all scalable layers at that temporal location and possibly other
associated data, e.g. SEI messages and parameter sets.
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 all the primary coded
pictures are IDR pictures. Such an access unit allows for random
access to any layer combination.
IDR picture: A coded picture with the property that the decoding of
this coded picture and all the following coded pictures in decoding
order, with the same value of dependency_id, can be performed
without inter prediction from any picture prior to the coded picture
in decoding order with the same value of dependency_id. Thus an IDR
picture allows for random access to the scalable layer, which it
belongs to. An IDR picture causes a "reset" in the decoding process
of the scalable layer containing the IDR picture.
progressive refinement (PR) slice: A progressive refinement slice
is contained in an SVC NAL unit that may be truncated since the end
of the slice header for bit-rate and quality reduction. PR slices
provide Fine Granularity Scalability (FGS).
The following terms are itemized for clarification on RTP
multiplexing strategies. For further information and discussion on
RTP multiplexing, we refer to section 5.2 of [RFC3550]:
RTP packet stream: A sequence of RTP packets with increasing
sequence numbers, identical PT and SSRC, carried in one RTP session,
and utilized to transport an integer number of SVC layers (which may
be FGS scalable).
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Single-Sender RTP Session: an (perhaps multicasted) RTP session in
which all RTP packet streams in the session stem from entities that
are in close cooperation, and can coordinate SSRC values. By
definition, in Single-Sender RTP Sessions, SSRC collisions on the
forward media path cannot occur. Note that, in practice, the
''entities in close cooperation'' likely run on the same machine and
communicate through non-protocol means, or they communicate by
protocols outside the RTP/SIP/SDP environment.
Session multiplexing: The scalable SVC bitstream is distributed
onto different RTP sessions, whereby each RTP session carries one
RTP packet stream. Each RTP session requires a separate signaling
and has a separate Timestamp, Sequence Number, and SSRC space.
Dependency between sessions MUST be signaled according to
[SDPsiglay].
SSRC multiplexing: The scalable SVC bitstream is distributed in a
single RTP session, but that session comprises more than one RTP
packet stream, identified by its SSRC.
The use of SSRC multiplexing MUST be signaled according to
[SDPsiglay].
5.2. Abbreviations
In addition to the abbreviations defined in [RFC3984], the following
ones are defined.
CGS: Coarse Granularity Scalability
FGS: Fine Granularity Scalability
6. RTP Payload Format
6.1. Design Principles
The authors observed the following design principles:
o Backward compatibility with RFC 3984 wherever possible.
o As the SVC base layer is H.264/AVC compatible, we assume the base
layer (when transmitted in its own session) to be
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encapsulated using RFC 3984. Requiring this has the desirable
side effect that it can be used by RFC 3984 legacy devices.
o MANEs are signaling aware and rely on signaling information.
MANEs have state.
o MANEs can terminate RTP sessions, and create different RTP
sessions
with perhaps modified content. This form of a MANE acts as an RTP
mixer. Mixer-MANEs necessarily need to be in the SRTP security
context.
o MANEs can also perform very limited functionality, namely
aggregate
multiple RTP packet streams into a single RTP stream within the
same session, by utilizing SSRC multiplexing. In this case, a
MANE
acts as a translator, and does not necessarily need to be in the
security context.
o Packet integrity needs to be preserved end-to-end (whereby
end-to-end can mean endpoint to endpoint but also endpoint to
MANE, if (and only if) the MANE acts as a Mixer).
o In case of layered multicast transmission as motivated in section
13.2, SVC layers are transported in different RTP sessions
(Session multiplexing). If the application should require a
layered transmission on session level, the SVC layers are
transported in different RTP packet streams within a single RTP
session, each stream identified by a unique SSRC (SSRC
multiplexing). SSRC multiplexing may further allow for adaptation
of an RTP session in the security context, further discussion can
be found in section 13.5.
6.2. RTP Header Usage
Please see section 5.1 of RFC 3984 [RFC3984]. The following applies
in addition.
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When different layers of a SVC bitstream are transported over more
than one RTP session, e.g. in layered multicast, for which the use
case is given in 13.2, SSRC multiplexing, as described below, MAY be
applied.
When SSRC multiplexing is in use the same IP address and port number
are shared between all RTP streams and all layers, while the
relative importance for the decoding process of each RTP stream
and/or layer is differentiated by the SSRC values. The SSRC value
space is evenly allocated to a number of sub value spaces, with the
number of sub value spaces being equal to the number of RTP packet
streams forming the RTP session for which SSRC multiplexing is used.
The first RTP packet stream conveying the lowest layers is mapped to
the first sub SSRC value space with the lowest SSRC values, the
second RTP packet stream conveying the second lowest layers is
mapped to the second sub SSRC value space with the second lowest
SSRC values, and so on. For the RTP packets of a certain RTP packet
stream, the SSRC value is randomly selected from the corresponding
sub SSRC value space. This way, a packet with a higher SSRC value
contains data belonging to higher layers or layers of lower
transport priority.
SSRC multiplexing as discussed above, in conjunction with multicast
from multiple senders requires that a) all streams SSRC multiplexed
in the same session carry data of the same layered bitstream, and b)
that the different senders are aware (by unspecified means of
signaling) of the relative importance of the RTP packet streams they
emit. Otherwise, it would be impossible to enforce the allocation
of SSRC numbering spaces according to the importance for the
decoding process. In other words, SSRC multiplexing as discussed
above works only for Single-Sender RTP sessions.
Note: in practice, it appears that SSRC multiplexing, due to the
above limitation, results in requiring a single entity to send all
RTP packet streams. No signaling means are currently available that
would allow different senders to coordinate the SSRC value spaces to
use.
6.3. Common Structure of the RTP Payload Format
Please see section 5.2 of RFC 3984 [RFC3984].
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6.4. NAL Unit Header Usage
The structure and semantics of the NAL unit header were introduced
in section 3.3. This section specifies the semantics of F, NRI,
PRID, D, TL, DID, QL, B, U, G, L, and O according to this
specification.
The semantics of F specified in section 5.3 of [RFC3984] also
applies herein.
For NRI, for the bitstream that is compliant with [H.264], the
semantics specified in section 5.3 of [RFC3984] are applicable,
otherwise only the semantics specified in SVC [SVC] is applicable.
For PRID, the semantics specified in [SVC] applies. MANEs
implementing unequal error protection may use this information to
protect NAL units with smaller PRID values better than those with
larger PRID values, for example by including only the more important
NAL units in a FEC protection mechanism. The desirable transport
priority increases as the PRID value increases.
For D, MANEs may use this information to protect NAL units with D
equal to 0 better than NAL units with D equal to 1. Furthermore a
MANE or a receiver may determine whether a given NAL unit is
required for successfully decoding a certain operation point of the
SVC bitstream.
For TL, DID and QL, in addition to the semantics specified in [SVC],
according to this memo, values of TL, DID or QL indicate the
relative priority in their respective dimension. A higher value of
TL, DID or QL indicates a higher priority if the other two
components are identical correspondingly. MANEs may use this
information to protect more important NAL units better than less
important NAL units.
Informative note: PRID, D, TL, DID, and QL, in combination,
provide complete information of the relative priority of a NAL
unit compared to any other NAL unit. [Edt. note: examples may be
provided in Informative Appendix 13 in future versions.]
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For B, in addition to the semantics specified in [SVC], according to
this memo, a MANE or receiver may use this information in order to
identify the [H.264] conforming base layer NAL units (if marked by a
suffix NAL unit) and may determine the temporal layer (by the TL
value of the suffix NAL unit) of it. Thus it allows for generating
an outgoing RTP stream, with a certain temporal scalability layer
that conforms to [RFC3984] and [H.264].
For U, the semantics specified in [SVC] apply.
For G, L and O, in addition to the semantics specified in [SVC],
according to this memo, a MANE or receiver may detect a fragmented
PR slice by G, L and O. Using this knowledge may let the MANE do
FGS adaptation on the PR slice, by forwarding not all of the
fragments in fragement_order (O).
6.5. Packetization Modes
Please see section 5.4 of RFC 3984 [RFC3984]. The single NAL unit
packetization mode SHALL NOT be used.
Informative note: The non-interleaved mode allows an application
to encapsulate a single NAL unit in a single RTP packet.
Historically, the single NAL unit mode has been included into
[RFC3984] only for compatibility with ITU-T Rec. H.241 Annex A.
There is no point in carrying this historic ballast towards a new
application space such as the one provided with SVC. More
technically speaking, the implementation complexity increase for
providing the additional mechanisms of the non-interleaved mode
(namely STAPs) is so minor, and the benefits are so great, that we
require STAP implementation.
6.6. Decoding Order Number (DON)
Please see section 5.5 of RFC 3984 [RFC3984]. The following applies
in addition.
When different layers of a SVC bitstream are transported in more
than one RTP packet stream (regardless of the use of session or SSRC
multiplexing, or a combination thereof), the interleaved
packetization mode MUST be used, and the DON values of all the NAL
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units MUST indicate the correct NAL unit decoding order over all the
RTP packet streams. If Session multiplexing is used, each session
MUST signal the same value for the (marked as optional, but for this
use case mandatory) MIME parameters sprop-interleaving-depth, sprop-
max-don-diff, sprop-deint-buf-req, and sprop-init-buf-time. Further
these values must be valid for the reception capabilities over all
sessions. A receiver MUST signal the same (marked as optional, but
for this use case mandatory) MIME parameter deint-buf-cap for all
sessions used for Session multiplexing.
6.7. Single NAL Unit Packet
Please see section 5.6 of RFC 3984 [RFC3984].
6.8. Aggregation Packets
Please see section 5.7 of RFC 3984 [RFC3984].
6.9. Fragmentation Units (FUs)
Please see section 5.8 of RFC 3984 [RFC3984].
6.10. Payload Content Scalability Information (PACSI) NAL Unit
A new NAL unit type is specified, and referred to as payload content
scalability information (PACSI) NAL unit. The PACSI NAL unit, if
present, MUST be the first NAL unit in an aggregation packet, and it
MUST NOT be present in other types of packets. The PACSI NAL unit
indicates scalability characteristics that are common for all the
remaining NAL units in the payload, thus making it easier for MANEs
to decide whether to forward or discard the packet. Senders MAY
create PACSI NAL units and receivers can ignore them.
Informative note: The NAL unit type for the PACSI NAL unit is
selected among those values that are unspecified in the H.264/AVC
specification and in RFC 3984 -- and therefore are ignored by
receiver. Hence an SVC stream, even when including PACSI NAL
units, can be processed with RFC 3984 receivers and H.264/AVC
decoders.
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When the first aggregation unit of an aggregation packet contains a
PACSI NAL unit, there MUST be at least one additional aggregation
unit present in the same packet. The RTP header fields are set
according to the remaining NAL units in the aggregation packet.
When a PACSI NAL unit is included in a multi-time aggregation
packet, the decoding order number for the PACSI NAL unit MUST be set
to indicate that the PACSI NAL unit is the first NAL unit in
decoding order among the NAL units in the aggregation packet or the
PACSI NAL unit has an identical decoding order number to the first
NAL unit in decoding order among the remaining NAL units in the
aggregation packet.
The structure of PACSI NAL unit is exactly the same as the four-byte
SVC NAL unit header specified in 3.3, and reproduced here once more
for convenience:.
+---------------+---------------+---------------+---------------+
|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|NRI| Type |RR | PRID | TL | DID | QL|R|B|U|D|G|L| O |
+---------------+---------------+---------------+---------------+
The values of the fields in PACSI NAL unit MUST be set as follows.
o The F bit MUST be set to 1 if the F bit in at least one remaining
NAL unit in the payload is equal to 1. Otherwise, the F bit MUST
be set to 0.
o The NRI field MUST be set to the highest value of NRI field among
all the remaining NAL units in the payload.
o The Type field MUST be set to 30.
o The RR field or reserved_zero_two_bits field (2 bits) MUST be set
to 0.
o The PRID field MUST be set to the lowest value of the PRID values
associated with all the remaining NAL units in the payload.
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o The TL field MUST be set to the lowest value of the TL values
associated with all the remaining NAL units in the payload.
o The DID field MUST be set to the lowest value of the DID values
associated with all the remaining NAL units in the payload.
o The QL field MUST be set to the lowest value of the QL values
associated with all the remaining NAL units in the payload.
o The R field or reserved_zero_bit field (1 bit) MUST be set to 0.
o The B field or layer_base_flag field (1 bit) MUST be set to 1 if
the layer_base_flag associated with all the remaining NAL units in
the payload is equal to 1. Otherwise, layer_base_flag MUST be set
to 0.
o The U field or use_base_prediction_flag field (1 bit)MUST be set
to 1 if the use_base_prediction_flag associated with all the
remaining NAL units in the payload is equal to 1. Otherwise,
use_base_prediction_flag MUST be set to 0.
o The D bit MUST be set to 0 if the D value associated with at least
one remaining NAL unit in the payload is equal to 0. Otherwise,
the D bit MUST be set to 1.
o The G field or fragmented_flag field (1 bit) MUST be set to 1 if
the fragmented_flag associated with all the remaining NAL units in
the payload is equal to 1. Otherwise, fragmented_flag MUST be set
to 0.
o The L field or last_fragment_flag field (1 bit) MUST be set to 1
if
the last_fragment_flag associated with all the remaining NAL units
in the payload is equal to 1. Otherwise, last_fragment_flag MUST
be set to 0.
o The O field or fragment_order field (2 bits) MUST be set to the
lowest value of frame_order associated with all the remaining NAL
units in the payload.
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7. Packetization Rules
Please see section 6 of RFC 3984 [RFC3984]. The following rules
apply in addition.
The single NAL unit mode SHALL NOT be used. (See also section 6.5
for the motivation).
When a suffix NAL unit is encapsulated for transmission, it SHOULD
be aggregated to the same transmission packet as the NAL unit
preceding the suffix NAL unit in decoding order.
When different layers of a SVC bitstream are transported in more
than one RTP packet stream, the interleaved packetization mode MUST
be used.
8. De-Packetization Process (Informative)
Please see section 7 of RFC 3984 [RFC3984]. The following rules
apply in addition.
[Edt. Do we need here more information about cross layer DON? Maybe
in the next version.]
9. Payload Format Parameters
[Edt. note: this section 9 and its subsections will be updated
according to the changes listed below, a little later in the
process. For now, we just list the adjustments necessary, so not to
bury any new information in the RFC 3984 text.]
Section 8 of [RFC3984] applies with the following modification.
The sentence
''The parameters are specified here as part of the MIME subtype
registration for the ITU-T H.264 | ISO/IEC 14496-10 codec.''
is replaced with
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''The parameters are specified here as part of the MIME subtype
registration for the SVC codec.''
9.1. MIME Registration
Editor's note: this needs to be updated by copy-pasting the
RFC 3984 MIME registration into this document, so to make it
self-contained. Will be done later in the process.
The MIME subtype for the SVC codec is allocated from the IETF tree.
The receiver MUST ignore any unspecified parameter.
Media Type name: video
Media subtype name: H.264-SVC
Required parameters: none
OPTIONAL parameters:
The optional MIME parameters specified in [RFC3984] apply, with the
following constraints (to be edited in at the appropriate time):
sprop-interleaving-depth:
In case of using Session multiplexing, the same sprop-interleaving-
depth value MUST be signaled for all sessions and MUST be valid over
all sessions of the multiplex.
sprop-max-don-diff:
In case of using Session multiplexing, the same sprop-max-don-diff
value MUST be signaled for all sessions and MUST be valid over all
sessions of the multiplex.
sprop-deint-buf-req:
In case of using Session multiplexing, the same sprop-deint-buf-req
value MUST be signaled for all sessions and MUST be valid over all
sessions of the multiplex.
sprop-init-buf-time:
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In case of using Session multiplexing, the same sprop-init-buf-time
value MUST be signaled for all sessions and MUST be valid over all
sessions of the multiplex.
deint-buf-cap:
In case of using Session multiplexing, the same deint-buf-cap value
MUST be signaled by the receiver for all sessions and MUST be valid
over all sessions of the multiplex.
In addition the following optional MIME parameters apply:
sprop-scalability-info:
This parameter MAY be used to convey the NAL unit containing the
scalability information SEI message 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 representation of the NAL unit
containing the scalability information SEI message as specified in
[SVC].
sprop-transport-priority:
This parameter MAY be used to signal the transport priority
indicator value(s) in terms of second and third bytes of the SVC NAL
unit header for one or more SVC layer(s) conveyed in one RTP
session. A transport priority indicator is base64 coded. If more
than one layer is transmitted within one RTP session, the transport
priority indicator value of each layer MUST be itemized with
decreasing importance for decoding and MUST be comma-separated.
Encoding considerations:
This type is only defined for transfer
via RTP (RFC 3550).
Security considerations:
See section 9 of this specification.
Public specification:
Please refer to section 15 of this
specification.
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Additional information:
None
File extensions: none
Macintosh file type code: none
Object identifier or OID: none
Person & email address to contact for further information:
Intended usage: COMMON
Author:
Change controller:
IETF Audio/Video Transport working group
delegated from the IESG.
9.2. SDP Parameters
9.2.1. Mapping of MIME Parameters to SDP
The MIME media type video/SVC string is mapped to fields in the
Session Description Protocol (SDP) as follows:
* The media name in the "m=" line of SDP MUST be video.
* The encoding name in the "a=rtpmap" line of SDP MUST be SVC (the
MIME subtype).
* The clock rate in the "a=rtpmap" line MUST be 90000.
* 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", "max-rcmd-nalu-
size'', ''sprop-transport-priority'', and ''sprop-scalability-
info'', 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.
9.2.2. Usage with the SDP Offer/Answer Model
TBD.
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9.2.3. Usage with Session and SSRC multiplexing
If Session or SSRC multiplexing is used, the rules on signaling
media decoding dependency in SDP as defined in [SDPsiglay] apply.
Further the use of SSRC multiplexing must be signaled according to
[SDPsiglay].
9.2.4. Usage in Declarative Session Descriptions
TBD.
9.3. Examples
TBD.
9.4. Parameter Set Considerations
Please see section 10 of RFC 3984 [RFC3984].
10. Security Considerations
Please see section 11 of RFC 3984 [RFC3984].
11. Congestion Control
Within any given RTP session carrying payload according to this
specification, the provisions of section 12 of RFC 3984 [RFC3984]
apply.
One key motivation for the recent attention to scalable codecs has
been the increasing awareness of media codec designers to network
congestion. While CGS scalability cannot reduce congestion for the
transport path of a given RTP session, MANEs and layered multicast
technologies can be used to alleviate congestion on a larger scale.
FGS scalability can be helpful to reduce session bandwidth both end-
to-end (with pre-coded content) and in network segments, again
assuming the use of MANEs.
MANEs MAY alleviate congestion on their outgoing network path by
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a) removing the NAL units belonging to hierarchically ''highest''
enhancement layer (or set of enhancement layers) from an RTP
stream carrying base and enhancement layers.
b) removing some or all bits of a given FGS NAL unit as long as the
remaining bits still form a conforming SVC NAL unit.
[Edt. Note: In the following paragraph, ''translator'' and ''mixer''
are not used consistently with RFC 3550. What we think we would
need is a ''mixer'' that mixes only a single input in a single output
(as a mixer terminates sessions). A ''Translator'' (that does not
terminate the RTP session) carries certain unnecessary baggage which
appears to make it undesirable for MANEs. The following paragraph
can either be fixed into RFC 3550 style and logic (thereby removing
an operation point we consider desirable), or we would need to
explain in detail what we want to do (not really congestion control
related and long). Perhaps we refer to the detailed discussions in
the CCM draft... Added to open issues.
In both cases, the incoming RTP session is terminated in the MANE,
and a second RTP session originates at the MANE. The MANE acts as
an RTP translator. The concept of scalability keeps the
implementation and computational effort within the MANE low, and
avoids expensive and delay-intensive full transcoding (in the sense
of reconstruction and re-encoding).]
When scalable layers are transported in their own RTP sessions, an
RTP receiver SHOULD unsubscribe to one or more enhancement layers
when it senses congestion, similar to what has been described in
[McCanne/Vetterli]. This behavior could perhaps be sufficient to
ease the network load to an acceptable level of congestion.
Nevertheless, it MUST follow the mechanisms described in section 12
of [RFC3984].
12. IANA Consideration
[Edt. Note: A new MIME type should be registered from IANA.]
13. Informative Appendix: Application Examples
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13.1. Introduction
Scalable video coding is a concept that has been around at least
since MPEG-2 [MPEG2], which goes back as early as 1993.
Nevertheless, it has never gained wide acceptance; perhaps partly
because applications didn't materialize in the form envisioned
during standardization.
MPEG and JVT, respectively, performed a requirement analysis before
the SVC project was launched. Dozens of scenarios have been
studied. While some of the scenarios appear not to follow the most
basic design principles of the Internet -- and are therefore not
appropriate for IETF standardization -- others are clearly in the
scope of IETF work. Of these, this draft chooses the following
subset for immediate consideration. Note that we do not reference
the MPEG and JVT documents directly; partly, because at least the
MPEG documents have a limited lifespan and are not publicly
available, and partly because the language used in these documents
is inappropriately video centric and imprecise, when it comes to
protocol matters.
With these remarks, we now introduce three main application
scenarios that we consider as relevant, and that are implementable
with this specification.
13.2. Layered Multicast
This well-understood form of the use of layered coding
[McCanne/Vetterli] implies that all layers are individually conveyed
in their own RTP packet streams, each carried in its own RTP session
using the IP (multicast) address and port number as the single
demultiplexing point. Receivers ''tune'' into the layers by
subscribing to the IP multicast, normally by using IGMP [IGMP].
Layered Multicast has the great advantage of simplicity and easy
implementation. However, it has also the great disadvantage of
utilizing many different transport addresses. While we consider
this not to be a major problem for a professionally maintained
content server, receiving client endpoints need to open many ports
to IP multicast addresses in their firewalls. This is a practical
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problem from a firewall/NAT viewpoint. Furthermore, even today IP
multicast is not as widely deployed as many wish.
We consider layered multicast an important application scenario for
three reasons. First, it is well understood and the implementation
constraints are well known. There may well by large scale IP
networks outside the immediate Internet context that may wish to
employ layered multicast in the future. One possible example could
be a combination of content creation and core-network distribution
for the various mobile TV services, e.g. those being developed by
3GPP (MBMS) [MBMS] and DVB (DVB-H) [DVB-H]. Finally, when one base
and one enhancement layer is in use and are being conveyed
separately, that represents one operation point of layered
multicast.
13.3. Streaming of an SVC scalable stream
In this scenario, a streaming server has a repository of stored SVC
coded layers for a given content. At the time of streaming, and
according to the capabilities and connectivity of the client(s), the
streaming server generates a scalable stream. This scalable stream
is served to the client(s). Both unicast and multicast serving is
possible. At the same time, the streaming server may use the same
repository of stored layers to compose different streams (with a
different set of layers) intended for different audiences.
As every endpoint receives only a single SVC RTP session, the number
of firewall pinholes can be optimized. In fact, only a single
firewall pinhole is required.
The main difference between this scenario and straightforward
simulcasting lies in the architecture and the requirements of the
streaming server, and is therefore out of the scope of IETF
standardization. However, compelling arguments can be made why such
a streaming server design makes sense. One possible argument is
related to storage space and channel bandwidth. Another is
bandwidth adaptivity without transcoding -- a considerable advantage
in a congestion controlled network. When the streaming server
learns about congestion, it can reduce sending bitrate by choosing
fewer layers when composing the layered stream. SVC is designed to
gracefully support both bandwidth rampdown and bandwidth rampup with
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a considerable dynamic range. This payload format is designed to
allow for bandwidth flexibility in the mentioned sense, both for CGS
and FGS layers. While, in theory, a transcoding step could achieve
a similar dynamic range, the computational demands are impractically
high and video quality is typically lowered -- therefore, few (if
any) streaming servers implement full transcoding.
13.4. Multicast to MANE, SVC scalable stream to endpoint
This final scenario is a bit more complex, and designed to optimize
the network traffic in a core network, while still requiring only a
single pinhole in the endpoint's firewall. One of its key
applications is the mobile TV market.
Consider a large IP network, e.g. the core network of 3GPP.
Streaming servers within this core network can be assumed to be
professionally maintained. We assume that these servers can have
many ports open to the network and that layered multicast is a real
option. Therefore, we assume that the streaming server multicasts
SVC scalable layers, instead of simulcasting different
representations of the same content at different bit rates.
Also consider many endpoints of different classes. Some of these
endpoints may not have the processing power or the display size to
meaningfully decode all layers; other may have these capabilities.
Users of some endpoints may not wish to pay for high quality and are
happy with a base service, which may be cheaper or even free. Other
users are willing to pay for high quality. Finally, some connected
users may have a bandwidth problem in that they can't receive the
bandwidth they would want to receive -- be it through congestion,
connectivity, change of service quality, or for whatever other
reasons. However, all these users have in common that they don't
want to be exposed too much, and therefore the number of firewall
pinholes need to be small.
This situation can be handled best by introducing middleboxes close
to the edge of the core network, which receive the layered multicast
streams and compose the single SVC scalable bit stream according to
the needs of the endpoint connected. These middleboxes are called
MANEs throughout this specification. In practice, we envision the
MANE to be part of (or at least physically and topologically close
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to) the base station of a mobile network, where all the signaling
and media traffic necessarily are multiplexed on the same physical
link. This is why we do not worry too much about decomposition
aspects of the MANE as such.
MANEs necessarily need to be fairly complex devices. They certainly
need to understand the signaling, so, for example, to associate the
PT octet in the RTP header with the SVC payload type.
A MANE may terminate the multicasted layered RTP sessions incoming
from the core network side, and create new RTP sessions (perhaps
even multicast sessions) to the endpoints connected to them. In RTP
terminology, these types of MANEs are RTP mixers. This implies, per
RFC 3550, a very loose relationship between the incoming and
outgoing RTP sessions. In particular, there is no direct
relationship between the incoming and outgoing RTP sequence numbers,
RTP timestamps, payload types used, etc.
Mixer-based MANEs are conceptually easy to implement and can offer
powerful features, primarily because they necessarily can ''see'' the
payload (including the RTP payload headers), utilize the wealth of
layering information available therein, and manipulate it.
While a mixer-based MANE operation in its most trivial form
(combining multiple RTP packet streams into a single one) can be
implemented comparatively simply -- reordering the incoming packets
according to the DON and sending them in the appropriate order --
more complex forms can also be envisioned. For example, a mixer-
type MANE can be optimizing the outgoing RTP stream to the MTU size
of the outgoing path by utilizing the aggregation and fragmentation
mechanisms of this memo.
A MANE can also act as a translator. In this case, we envision its
functionality to be limited to the manipulation of the transport
addresses, so to enable SSRC multiplexing. The most compelling use
case appears to be to forward multiple incoming RTP packets streams
(conveyed to their own transport addresses) to a single firewall
pinhole. The translator variant of the MANE does not terminate RTP
sessions, but rather ''translate'' them in a very simple way -- by
changing the transport address -- so to SSRC-multiplex multiple
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sessions onto a single transport address. What sounds trivial at
the first glance is in reality a highly complex process primarily
due to the need of appropriate RTCP processing. This is
particularly true when individual packets are intentionally being
pruned or removed from the incoming session, which may be necessary
to support FGS.
Translator-based MANEs appear to be able to offer a limited amount
of functionality without being in the security context, which opens
up additional application range. Whether this form of a Translator
based MANE is actually feasible, and whether it offers sufficient
benefits to warrant the additional specification burden is open for
discussion, and input is solicited.
While the implementation complexity of either case of a MANE, as
discussed above, is fairly high, the computational demands are
comparatively low. In particular, SVC and/or this specification
contain means to easily generate the correct inter-layer decoding
order of NAL units. It is also simple to identify the fine
granularity scalable bits in a given NAL unit. No serious bit-
oriented processing is required and no significant state information
(beyond that of the signaling and perhaps the SVC sequence parameter
sets) need to be kept.
13.5. SSRC Multiplexing in case of using SRTP
When SRTP is in use, it is not possible to take advantage of the in-
band information (SEI messages, NAL unit headers, PACSI NAL units)
when processing layered streams. Therefore, a MANE outside the
security context cannot make informed decisions when aggregating
information. Some relevant information must be available in the RTP
header to make meaningful decisions.
The first, and most obvious, choice is to map SSRC values directly
to certain layers by the means of signaling. As MANEs need to be in
the signaling context, this appears to be sensible. However, it
requires a per-SSRC signaling mechanism -- a demultiplexing point
that is currently not envisioned in SDP.
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A second design choice is to somehow make available the information
about the properties of a specific layer -- to the extent a MANE can
make a meaningful decision -- in the SSRC value. In other words,
SSRC is no more fully randomly chosen, but selected based on context.
This is possible only when limiting the scope to a single sender to
a multicast group, because the various senders have no means to
coordinate their choice of SSRC values. In practice, that's not a
major limitation.
Any form of such a selection of SSRC values has two major drawbacks:
First, without a sufficiently large random component the probability
for SSRC collisions increases to a point that becomes unacceptable.
We address this point by discouraging the use of multi-sender
multicast. When only a single sender emits packets in a given RTP
session, it can be expected that this sender is able to avoid SSRC
collisions. In addition, we require a sufficiently large random
component in the SSRC generation, which is constant for each layer
stemming from the same sender. While the probability for SSRC
collisions is still lowered, the random component can be kept as
large as 26 bits assumes that the SVC bitstream in question contains
64 layers.
Second, and more critical, a straightforward copy of values known to
be present at fixed locations in the RTP payload would make it easy
for codebreakers to attack an SRTP encrypted stream, because an
unencrypted representation of a encrypted known value would both be
present in the same packet. This is outright unacceptable from a
security viewpoint.
Therefore, we do not allow to simply copy information from the
bitstream into the SSRC field. Instead, we rely on a non-reversible
function, that also necessarily contains the aforementioned random
component, that, when executed, indicates the relative priority
difference between two layers (signaled by two SSRC values).
The SSRC value space is evenly allocated to a number of sub value
spaces, with the number of sub value spaces being equal to the
number of RTP sessions for which SSRC multiplexing is used. Then
the first RTP session conveying the lowest layers is mapped to the
first sub SSRC value space with the lowest SSRC values, and the
second RTP session conveying the second lowest layers is mapped to
the second sub SSRC value space with the second lowest SSRC values,
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and so on. For the RTP packets of a certain RTP session, the SSRC
value is randomly selected from the corresponding sub SSRC value
space. This way, a packet with a higher SSRC value contains data
belonging to higher layers or layers of lower transport priority.
A translator-based MANE can make use of the aforementioned SSRC
values as follows. Suppose that the MANE has identified, through
sensed congestion or other unspecified means, that it needs to
discard packets belonging to higher layers, say K of the N buffered
packets, to maintain a packet sending rate, it identifies the K
packets with the highest SSRC values, and discards them.
13.6. Scenarios currently not considered for complexity reasons
-- vacat --
13.7. Scenarios currently not considered for being unaligned with
IP philosophy
Remarks have been made that the current draft does not take into
consideration at least one application scenario which some JVT folks
consider important. In particular, their idea is to make the RTP
payload format (or the media stream itself) self-contained enough
that a stateless, non signaling aware device can ''thin'' an RTP
session to meet the bandwidth demands of the endpoint. They call
this device a ''Router'' or ''Gateway'', and sometimes a MANE.
Obviously, it's not a Router or Gateway in the IETF sense. To
distinguish it from a MANE as defined in RFC 3984 and in this
specification, let's call it a MDfH (Magic Device from Heaven).
To simplify discussions, let's assume point-to-point traffic only.
The endpoint has a signaling relationship with the streaming server,
but it is known that the MDfH is somewhere in the media path (e.g.
because the physical network topology ensures this). It has been
requested, at least implicitly through MPEG's and JVT's requirements
document, that the MDfH should be capable to intercept the SVC
scalable bit stream, modify it by dropping packets or parts thereof,
and forwarding the resulting packet stream to the receiving
endpoint. It has been requested that this payload specification
contains protocol elements facilitating such an operation, and the
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INTERNET-DRAFT RTP Payload Format for SVC Video October 2006
argument has been made that the NRI field of RFC 3984 serves exactly
the same purpose.
The authors of this I-D do not consider the scenario above to be
aligned with the most basic design philosophies the IETF follows,
and therefore have not addressed the comments made (except through
this section). In particular, we see the following problems with
the MDfH approach):
- As the very minimum, the MDfH would need to know which RTP streams
are carrying SVC. We don't see how this could be accomplished but
by using a static payload type. None of the IETF defined RTP
profiles envision static payload types for SVC, and even the de-
facto profiles developed by some application standard
organizations (3GPP for example) do not use this outdated concept.
Therefore, the MDfH necessarily needs to be at least ''listening''
to the signaling.
- If the RTP packet payload were encrypted, it would be impossible
to interpret the payload header and/or the first bytes of the
media stream. We understand that there are crypto schemes under
discussion that encrypt only the last n bytes of an RTP payload,
but we are more than unsure that this is fully in line with the
IETF's security vision.
Even if the above two problems would have been overcome through
standardization outside of the IETF, we still foresee serious design
flaws:
- An MDfH can't simply dump RTP packets it doesn't want to forward.
It either needs to act as a full RTP Translator (implying that it
patches RTCP RRs and such), or it needs to patch the RTP sequence
numbers to fulfill the RTP specification. Not doing either would,
for the receiver, look like the gaps in the sequence numbers
occurred due to unintentional erasures, which has interesting
effects on congestion control (if implemented), will break pretty
much every meta-payload ever developed, and so on. (Many more
points could be made here).
- An MDfH also can't ''prune'' FGS packets. Again, doing so would
not be compatible with meta payloads, and would mess up RTCP RRs
and congestion control (if the congestion control is based on
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INTERNET-DRAFT RTP Payload Format for SVC Video October 2006
octet count and not on packet count; there are discussions related
to the former at least in the context of TFRC).
In summary, based on our current knowledge we are not willing to
specify protocol mechanisms that support an operation point that has
so little in common with classic RTP use.
14. Acknowledgements
Funding for the RFC Editor function is currently provided by the
Internet Society. Further, the author Thomas Schierl of Fraunhofer
HHI is sponsored by the European Commission under the contract
number FP6-IST-0028097, project ASTRALS.
15. References
15.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[MPEG4-10] ISO/IEC International Standard 14496-10:2003.
[H.264] ITU-T Recommendation H.264, "Advanced video coding for
generic audiovisual services", May 2003.
[SDPsiglay] Schierl, T., ''Signaling media decoding dependency in
Session
Description Protocol (SDP)'', IETF internet draft
draft-schierl-mmusic-layered-codec-01, October 2006.
[SVC] Joint Video Team, ''Annex G of Joint Draft 7 of SVC
Amendment
(with proposed changes)'', available from
http://ftp3.itu.ch
/av-arch/jvt-site/2006_07_Klagenfurt/JVT-T202.zip ,
July 2006
[RFC3984] Wenger, S., Hannuksela, M, Stockhammer, T, Westerlund, M,
Singer, D, ''RTP Payload Format for H.264 Video'', RFC 3984,
February 2005
Wenger, Wang, Schierl Standards Track [page 36]
INTERNET-DRAFT RTP Payload Format for SVC Video October 2006
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
15.2. Informative References
[DVB-H] DVB - Digital Video Broadcasting (DVB); DVB-H
Implementation Guidelines, ETSI TR 102 377, 2005
[IGMP] Cain, B., Deering S., Kovenlas, I., Fenner, B. and
Thyagarajan, A., ''Internet Group Management Protocol,
Version 3'', RFC 3376, October 2002.
[McCanne/Vetterli]
V. Jacobson, S. McCanne and M. Vetterli. Receiver-
driven layered multicast. In Proc. of ACM SIGCOMM'96, pages
117--130, Stanford, CA, August 1996.
[MBMS] 3GPP - Technical Specification Group Services and System
Aspects; Multimedia Broadcast/Multicast Service (MBMS);
Protocols and codecs (Release 6), December 2005.
[MPEG2] ISO/IEC International Standard 13818-2:1993.
[SRTP] Baugher, M., McGrew, D, Naslund, M, Carrara, E,
Norrman, K, ''The secure real-time transport protocol
(SRTP)'', RFC 3711, March 2004.
16. Author's Addresses
Stephan Wenger Phone: +358-50-486-0637
Nokia Research Center Email: stewe@stewe.org
P.O. Box 100
FIN-33721 Tampere
Finland
Ye-Kui Wang Phone: +358-50-486-7004
Nokia Research Center Email: ye-kui.wang@nokia.com
P.O. Box 100
FIN-33721 Tampere
Finland
Thomas Schierl Phone: +49-30-31002-227
Fraunhofer HHI Email: schierl@hhi.fhg.de
Einsteinufer 37
Wenger, Wang, Schierl Standards Track [page 37]
INTERNET-DRAFT RTP Payload Format for SVC Video October 2006
D-10587 Berlin
Germany
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19. Copyright Statement
Wenger, Wang, Schierl Standards Track [page 38]
INTERNET-DRAFT RTP Payload Format for SVC Video October 2006
Copyright (C) The Internet Society (2006). This document is subject
to the rights, licenses and restrictions contained in BCP 78, and
except as set forth therein, the authors retain all their rights.
20. RFC Editor Considerations
none
21. Open Issues
1. Need to double check MANE, Mixers, and Translators throughout the
document (consistently with RFC 3550).
2. Packetization rules need work.
3. Alignment with the SVC specification (ongoing)
4. In context of SSRC multiplexing: make consistent higher/lower
layers vs. RTP packet streams of higher/lower importance.
22. Changes Log
From -00 to -01
- 04.02.2006, StW: Added details to scope
- 04.02.2006, StW: Added short subsection 6.1 ''Design Principles''
- 04.02.2006, StW: Added section 15, ''Application Examples''
- 06.02 - 03.03.2006, YkW: Various modifications throughout the
document
- 13.02.2006 - 03.03.2006 , ThS: Added definitions and additional
information to section 3.3, 5.1, 7 and 8, parameters in section 9.1 and
added section 14 for NAL unit re-ordering for layered multicast.
Further modifications throughout the document
From -01 to -02
- 06.03.2006, StW: Editorial improvements
- 26.05.2006, YkW: Updated NAL unit header syntax and semantics
according to the latest draft SVC spec
- 20.06.2006, Miska/YkW: Added section 6.10 ''Payload Content
Scalability Information (PACSI) NAL Unit''
- 20.06.2006, YkW: Updated the NAL unit reordering process for layered
multicast (removed the old section 14 ''Informative Appendix: NAL Unit
Wenger, Wang, Schierl Standards Track [page 39]
INTERNET-DRAFT RTP Payload Format for SVC Video October 2006
Re-ordering for Layered Multicast'' and added the new section 13 ''NAL
Unit Reordering for Layered Multicast'')
From -02 to -03
- 05.09.2006, YkW: Updated the NAL unit header syntax, definitions,
etc., according to the foreseen July JVT output. Updated possible MANE
adaptation operations according to SPID, TL, DID and QL. Clarified the
removal of single NAL unit packetiztaion mode. Added the support of
SSRC multiplexing in layered multicast.
- 08.09.2006, StW: Editorial changes throughout the document
- 08.09.2006, YkW: Added the packetization rule for suffix NAL unit.
- 19.09.2006, YkW: Moved/updated SSRC multiplexing support to section
6.2 ''RTP header usage''. Moved/updated the cross layer DON constraint
to Section 6.6 ''Decoding order number''. Moved/updated the
packetization rule when a SVC bistream is transported over more than
one RTP session to Section 7 ''Packetization rules''. Removed Section 13
''Support of layered multicast''.
- 16.10, TS: Added detailed four-byte NAL unit header description.
Change ''AVC'' to ''H.264'' conforming to 3984. Modifications throughout
the document. Extended description of 3rd byte of PACSI NAL unit.
Corrected terms RTP session and RTP packet stream in case of SSRC
multiplexing. Added terms in definition section on RTP multiplexing.
Constraints on optional MIME parameters of 3984 for cross-layer DON
(DON section and MIME parameters). Copied parts of SI paper regarding
mixer, translator and SSRC mux with SRTP to section application
examples. Added section on SDP usage with Session and SSRC
multiplexing. Added points in Design principles on translator/mixer and
RTP multiplexing. Added additional founding information in Ack-
section. Corrected reference for SVC and added reference for generic
signaling.
17.10, StW: Fixed many editorials, clarified MANE, mixer, translator
and RTP packet stream throughout doc (hopefully consistently)
18.10., removed comments, clarified B-Bit, changed definition of base-
layer (do not need to be of the lowest temporal resolution),
Wenger, Wang, Schierl Standards Track [page 40]