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From rem-conf-request@es.net Mon Oct 2 02:22:07 2000
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From: "Toshiyuki Nomura"
To: "John Lazzaro"
Cc: ,
References: <200009251800.LAA05183@snap.CS.Berkeley.EDU>
Subject: Re: LATM specified in a public MPEG document?
Date: Mon, 2 Oct 2000 15:14:20 +0900
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Dear John and members,
As you pointed out, it seems difficult to transmit SA data
by LATM. I have forwarded this issue to MPEG Audio Experts
and have not been received any objections.
Therefore, I'll modify the I/D text so that SA data
MUST NOT be handled by the LATM-based RTP specification,
following the suggestion below:
John Lazzaro wrote:
> it sounds like the earlier suggestions I
> made on the draft-ietf-avt-rtp-mpeg4-es-04.txt document about
> Structured Audio issues, which made it into the current draft,
> are inappropriate, since they assume the header described in
> "4.1 RTP Packet Format" is meant for all audio, not only
> "natural" audio. Perhaps the I-D should simply state that
> Structured Audio isn't currently transportable over any of
> the packet formats described in the document.
The modified text will be published as
draft-ietf-avt-rtp-mpeg4-es-05.txt.
Best Regards,
Toshiyuki Nomura, NEC
----------------------
From rem-conf-request@es.net Mon Oct 2 06:35:20 2000
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Date: Mon, 02 Oct 2000 03:21:53 -0700
To: rem-conf@es.net
From: Ross Finlayson
Subject: Announcing: An open-source library for RTP/RTCP streaming
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[This is something that I've been promising people for a long time. I'm
pleased to report that I finally have it in a releasable state. (Not
perfect, but releasable..)]
The C++ source code that forms the basis of my "liveCaster" (&
"playRTPMPEG") media streaming tools is now available as "open source"
(released under the LGPL). This code forms a set of libraries that can be
used within RTP/RTCP streaming applications, and/or can be extended to
support additional media types and codecs (in addition to MP3 audio). For
a description of these libraries, and instructions for how to install and
build them, see .
You can use this code right now to build real applications - in fact, the
distribution includes code for building two demo applications: (i) a server
that reads a MP3 file and streams it (using RTP/RTCP), and (ii) a client
that reads from the same RTP/RTCP stream, and outputs the resulting MP3
stream to stdout.
These libraries can be built and run on Unix (various flavors) and
Windows. (However, the demo applications mentioned above don't yet compile
on Windows.)
These libraries are similar to existing toolkits such as "Open Mash",
although - for now, at least - not nearly as rich in functionality. Also,
they do not require the use of a specific scripting language - or any
scripting language at all. (However, the libraries can, and have, been
used with scripting languages - e.g., "liveCaster" and "playRTPMPEG" were
built using these libraries in combination with Tcl.)
Once again, the URL for this code is .
Ross.
From rem-conf-request@es.net Mon Oct 2 09:24:31 2000
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From: "Nikos Laoutaris"
To:
Subject: video frame arrival traces
Date: Mon, 2 Oct 2000 16:06:40 +0300
Organization: University of Athens, Networks Operation Center
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Dear list members
I am searching for traces of interarrival distributions or (better) =
one-way delay distributions of streaming video, at the frame level. That =
is, i would like to have a real measurement of the distortion of =
interstream synchronization of a series of equally spaced frames (at the =
source) as they travel over a (multi hop) network path. This information =
is to be used for the evaluation of an adaptive playout controller for =
packet video receivers.
Thanking you in advance
Nikos Laoutaris
Communication Networks Laboratory
National and Kapodestrian University of Athens, Greece
e-mail : laoutaris@noc.uoa.gr
tel. : +301 7275603
fax : +301 7275601
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Dear list members
I am searching for traces of =
interarrival distributions or (better) one-way delay distributions of =
streaming=20
video, at the frame level. That is, i would like to have a real =
measurement of=20
the distortion of interstream synchronization of a series of =
equally=20
spaced frames (at the source) as they travel over a =
(multi hop)=20
network path. This information is to be used for the evaluation of an =
adaptive=20
playout controller for packet video receivers.
Thanking you in advance
Nikos Laoutaris
Communication Networks=20
Laboratory
National and Kapodestrian University of Athens, =
Greece
e-mail :=20
laoutaris@noc.uoa.gr
tel. =
: +301=20
7275603
fax : +301 7275601
------=_NextPart_000_0132_01C02C8A.BFC8F0E0--
From rem-conf-request@es.net Mon Oct 2 09:39:45 2000
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Message-ID: <39D88E7D.DFCD1BC7@era.ericsson.se>
Date: Mon, 02 Oct 2000 15:32:45 +0200
From: Magnus Westerlund
Organization: Ericsson Research
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To: Qiaobing Xie
CC: rem-conf@es.net
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
References: <39D203BA.561299A1@era.ericsson.se> <200009271749.MAA02664@agevole.cig.mot.com>
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Hello,
The payload format is designed with IP networks in mind. The problem we see is
that IP is transported over a wide range of different links with different
behavior, e.g. different radio and wireline links. Radio and wire do not have
the same error characteristics. AMR will be widly used over wireless links and
on these links bandwidth efficiency is extremely important. This lead to the
bitwise sorting. Octet sorting needs octet alignement for each block of the
payload, i.e. padding at the end of each block. Which we considered to be to
expensive. The optional sensitivity sorting facilitates the use of different
error robustness schemes. The need for different robustness enhancing schemes
may depend on the application and the lower transport layers. The application
can be for example conversational speech or streaming. Lower layers are here
layer 1 and 2.
Regards
Magnus W.
Qiaobing Xie wrote:
> Magnus,
>
> Correct me if I am wrong, but I find hard to understand your reasoning.
>
> What you are saying is basically that, in order to make the current
> payload format design general enough and flexible enough to be used "for
> a wide range of networks and applications" and with "any mechanism that
> exploit" unequal bit sensitivity, you've chosen to compromise the
> implementation efficiency and performance (and even some functionality)
> for AMR over RTP/whatever-partial-checksum-UDP/IP.
>
> I think it should be the other way around - the AMR payload design
> should primarily focus on finding the optimal solution for
> AMR/RTP/UDP*/IP. It doesn't sound right to me for IETF AVT WG to design
> something that is meant for a wide range of networks but is
> suboptimal/compromised (even functionally handicapped) for IP networks.
>
> Regards,
> -Qiaobing Xie
>
> Magnus Westerlund wrote:
> >
> > Dear Qiaobing, everyone,
> >
> > The unequal error protection/correction scheme is not designed just for
> > UDP-Lite. Any mechanism that can exploit the fact that the data is
> > sorted in sensitivity order is possible to use, e.g. a partial FEC
> > scheme based on the same techniques as RFC2733.
> >
> > We have made some design choices that makes the payload format flexible
> > and bit efficient, i.e. usefull for a wide range of networks and
> > applications.
> >
> > Best regards
> >
> > Magnus Westerlund
> >
> > Qiaobing Xie wrote:
> >
> > > Hello, everyone:
> > >
> > > I have some serious concerns on the design of the AMR payload format as
> > > defined in . In brief, the design seems
> > > very limiting in terms of taking advantage of the error tolerant
> > > features built into the AMR codec algorithm. Also, it seems that not
> > > enough consideration has been given to implemenation efficiency and
> > > performance.
> > >
> > > Since my concerns are rather fundamental to the design, the email is a
> > > little long, please bear with me..
> > >
> > > 1) Basic idea
> > >
> > > The basic idea of the draft is to transport AMR frames over IP in an RTP
> > > payload, i.e., AMR-frame/RTP/UDP/IP. I have no problem with that.
> > >
> > > Therefore, the packet *passed to the UDP layer* will contain three types
> > > of bits:
> > >
> > > +--------------+
> > > | header bits |
> > > +--------------+
> > > | Class A bits |
> > > | and/or |
> > > | other bits |
> > > +--------------+
> > >
> > > Here, the "header bits" will include all the AMR speech frame control
> > > bits (ie., frame header), the RTP payload header bits, and the RTP
> > > header bits.
> > >
> > > "Class A bits" is the most sensitive coded speech data bits, as defined
> > > by the AMR codec. The "other bits" (also called Class B and C bits in
> > > AMR terminology) are less sensitive coded speech data bits.
> > >
> > > 2) Problems with the current AMR payload format design
> > >
> > > The design in the current draft apparently was guided under the
> > > misconception that the "Class A bits" of an AMR speech frame need to be
> > > delivered error-free, and if that part of the frame is corrupted the
> > > whole frame should be tossed. This is not correct.
> > >
> > > What the AMR codec really wants from the transport is: "if Class A bits
> > > is found corrupted in a frame, raise a Bad Frame flag but still pass the
> > > whole frame to the decoder since the decoder still may want to use Class
> > > B and C bits to help error concealment." The purpose of the flag is to
> > > tell the decoder that Class A bits are bad and do not use them.
> > >
> > > However, the draft seems to suggest on the use of a partial checksum
> > > (such as the one provided by the proposed UDP-lite) to protect the
> > > "header bits" AND the "Class A bits" at the transport leverl. This means
> > > if Class A bits have any error, the whole packet will be tossed by the
> > > transport. Even worse, if the packet is a compound payload, you will
> > > have all the AMR frames tossed by the transport if a bit error occurs in
> > > the Class A bits of any one of the enclosed frames.
> > >
> > > It becomes more problematic when, in order to support multiple AMR
> > > frames in a compound RTP payload and to use
> > > the partial checksum, a *very ugly* bit-wise shuffling scheme is
> > > proposed to copy (bit by bit) all the "Class A bits" of each enclosed
> > > AMR speech frame to the front of the packet!! The hugely computational
> > > intensive operation is required no each RTP packet on both ends of the
> > > RTP session.
> > >
> > > 3) A better way to do it:
> > >
> > > To get what AMR codec wants, we should only checksum the "header bits",
> > > and carry an 8-bit CRC in the AMR frame header to cover the "Class A
> > > bits" of the frame. This CRC is for the RTP receiver (that is above the
> > > transport) to verify the integrity of the Class A bits and to raise the
> > > Bad Frame flag before passing the AMR frame to the decoder.
> > >
> > > In other words,
> > >
> > > 1. Partial checksum at the transport level to cover all the "header
> > > bits" -- if checksum fails, toss the whole packet.
> > > 2. CRC in AMR frame header to cover the "Class A bits" -- if CRC fails,
> > > raise the bad frame flag.
> > > 3. No error checking on the "other bits".
> > >
> > > When bundling multiple AMR frames in one RTP packet (the compound
> > > payload), we should use a table of contents (TOC) structure to avoid
> > > bit-wise copying (AMR frames are bit-stuffed to byte boundary). and then
> > > use the transport level checksum to cover the header bits and TOC bits
> > > only (the Class A CRC of each AMR frame is in the TOC).
> > >
> > > +------------------+
> > > | RTP header bits | -- RTP and payload headers
> > > +------------------+
> > > | Table of Contents| -- list of frame headers of the
> > > +==================+ \ included AMR frames
> > > | Fmr1 Cls A bits | \
> > > +------------------+ |
> > > | Fmr1 Cls B bits | |
> > > +------------------+ |
> > > | Fmr1 Cls C bits | |
> > > +==================+ |
> > > | Fmr2 Cls A bits | > not covered by transport level chksum
> > > +------------------+ |
> > > | Fmr2 Cls B bits | |
> > > +------------------+ |
> > > | Fmr2 Cls C bits | |
> > > +==================+ |
> > > | Fmr3 Cls A bits | /
> > > +------------------+ /
> > > .....
> > > .....
> > >
> > > This way, you get what the AMR codec asks for and yet avoid bit-wise
> > > copying.
> > >
> > > ===============================================================
> > > Qiaobing Xie, Ph.D
> > > Principal Staff Engineer
> > > Network Architecture & Technology
> > > Motorola, Inc.
> > > TEL: (847) 632-3028
> > > ===============================================================
> >
> > --
> >
> > Magnus Westerlund
> >
> > Audio Technology, Ericsson Research
> > ----------------------------------------------------------------------
> > Ericsson Radio Systems AB | Phone +46 8 4048287
> > Torshamsgatan 23 | Fax +46 8 7575550
> > S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
--
Magnus Westerlund
Audio Technology, Ericsson Research
----------------------------------------------------------------------
Ericsson Radio Systems AB | Phone +46 8 4048287
Torshamsgatan 23 | Fax +46 8 7575550
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From rem-conf-request@es.net Mon Oct 2 13:17:56 2000
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Date: Mon, 02 Oct 2000 18:56:42 +0200
To: ddma@enst-bretagne.fr
Subject: DDMA workshop CFP
Mime-Version: 1.0
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--=====================_970498602==_
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Dear Colleague,
You are invited to submit a paper to the International "Distributed Dynamic
Multiservice Architectures" (DDMA) Workshop. This workshop is in conjunction
with the 21st International Conference on Distributed Computing Systems
(ICDCS2001), one of the major conferences in the Distributed Computing
Systems Area.
DDMA workshop covers aspects concerning adaptive architectures and service
composition and management.
The submission deadline is November 1st. Further details are at the web
site http://ddma.enst-bretagne.fr/. A text version is also in attachment.
The DDMA workshop will be held in Phoenix, Arizona; April 16-19, 2001.
[We apologize if you receive multiple copies of this announcement]
--=====================_970498602==_
Content-Type: text/plain; charset="iso-8859-1"
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Call for Papers
International Workshop on
Distributed Dynamic Multiservice Architectures
http://ddma.enst-bretagne.fr/
in conjunction with ICDCS2001
sponsored by IEEE Computer Society
April 16 =96 19, 2001
Phoenix, Arizona, USA
The growing need to integrate new services while complying with the=
requirements of end=20
users puts strong emphasis on the management and combination of future=
services and the QoS=20
they should be able to provide and guarantee. Dynamic Multiservice=
Architectures constitute a=20
challenging paradigm for distribution theory and practice due to their=
unique requirements in terms=20
of adaptability, behavioural integration, separation of concerns, service=
composition and service=20
management. In particular, the deployment of multimedia telecommunication=
services is a highly=20
challenging area where those paradigms should apply because advanced=
services require that the=20
user's profiles, preferences and context be taken into account. "Distributed=
Dynamic=20
Multiservice Architectures" hence add a new dimension to distributed=
computing to help the=20
composition and deployment of new services guaranteeing final user's=
requirements.
The goals of this workshop are to bring together users and researchers to=
present their=20
recent work related to diverse aspects of dynamic multiservice=
architectures, their fundamental=20
issues, their paradigms and some appropriate approaches and experiments. The=
workshop will=20
thus present opportunities for discussing further evolutions and their=
expected benefits.
Scope of Workshop
-----------------
The scope of this workshop encompasses but is not limited to :
Evolution of distributed multiservice architectures
Separation of aspects and composition of aspects.
Methodological approaches for aspect oriented programming.
Components as cornerstones for dynamic composition.
Composition requirements for reflection.
Auto-adaptive ad hoc architectures and protocols.
Impact on the management and control
Management of adaptive architectures
Management and monitoring of composed services.
Platforms, tools, languages, ...=20
Languages, tools and platforms for auto-adaptive systems.
Environments for specification and design of dynamic architectures.
Experience with design, implementation and use of auto-adaptive systems.
Application to telecommunication services
Open critical issues in terms of composition of services.
Panorama of QoS integration approaches for multimedia telecom services.
Paradigms and requirements for telecom services architectures.
Architectures for 3rd generation multimedia services.
Distribution protocols for telecom (WAP, CORBA, CAMEL, mobile agents).
Submission Instructions and Workshop Format
-------------------------------------------
Interested participants should submit a 4-8 pages position paper that=
focuses on one of the=20
relevant topics or related issues. Papers will be reviewed by the program=
committee. Electronic=20
submissions should be sent to zied.choukair@enst-bretagne.fr. Mail body=
should include name,=20
address, affiliation and a brief biography along with paper's title and=
abstract.
The workshop will follow the following format: The morning and part of the=
afternoon will=20
be dedicated to presentations and discussions of accepted proposals,=
including demonstrations of=20
existing software. The afternoon session will end with an identification of=
issues, followed by=20
discussion. The organisers will then elaborate a summary of the workshop=
based upon a summary=20
of the papers and the major outlined points of the discussion.
Program Committee Members :
Mehmet Aksit, Twente University, NL
Zi=E8d Choukair, ENST Bretagne, FR
Jaime Delgado, Universitat Pompeu Fabra, ES
Byung-Guk Kim, University of Massachusetts Lowell, USA
Marwan Krunz, University of Arizona, USA
Abdul H. Sadka, University of Surrey, GB
Antonio Rito Silva, University of Lisbon, PT
Timothy K. Shih, Tamkang University, TW
Katsuya Tanaka, Tokyo Denki University, JP
Important Dates :
Paper submission due : Nov 1, 2000
Notification : Dec 30, 2000
Camera ready : Jan 20, 2001
Conference : April 16-19, 2001
Web Pages :
Conference URL : http://cactus.eas.asu.edu/ICDCS2001/
--=====================_970498602==_--
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To: rem-conf@es.net
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From: "Doug Solomon"
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Call for Papers
2nd IP Telephony Workshop
April 2-3, 2001 - Columbia University, New York City
http://www.fokus.gmd.de/events/iptel2001/
Objectives
----------
Internet telephony is rapidly evolving from research to design
and deployment. The objectives of the IP Telephony Workshop are
to bring together researchers, developers, vendors and service
providers active in this area and stimulate discussion on
innovation, research, implementation, deployment experiences and
future directions.
Scope & Topics
--------------
Original technical articles related to IP telephony are solicited.
Only papers with significant technical content, not "white papers"
or tutorials, will be considered for publication:
- Research papers (unique ideas, novel algorithms,
architectures, measurements, theoretical and/or analytical
contributions)
- Surveys, state-of-the-art studies, technology comparisons
- Implementation and deployment reports
- Standardization reports
Particular areas of interest include, but are not limited to, the
following:
- Integration with Internet services (e.g., web, instant
messaging, games)
- Added-value services (e.g., call centers, conferencing)
- Mobility and 3rd generation wireless
- Authentication, authorization, accounting, charging,
settlement
- QoS support
- Security (e.g., privacy, authentication, certification
authorities, firewall traversal)
- Call signaling & processing
- Feature creation
- Supporting services (e.g., call routing, lookup services)
- Audio & video encoding and transmission
- Management and provisioning
- Interworking with the PSTN
- Design and deployment considerations (e.g., performance,
scalability, reliability)
Important Dates
---------------
Full paper due November 27th, 2000
Notification of acceptance January 12th, 2001
Final version due January 31st, 2001
Program published and February 5th, 2001
registration opens
Workshop April 2nd-3rd, 2001
iptel2001 Organizing Committee
------------------------------
Program Chair
H. Schulzrinne Columbia University
Program Committee
M. Arango Sun Microsystems
F. Baker Cisco
W. Bauerfeld T-Nova
G. Bond AT&T Research
S. Bradner Harvard University
G. Carle GMD FOKUS
J. Crowcroft UCL
C. Huitema Microsoft
G. S. Kuo National Central University, Taiwan
J. Kuthan GMD Fokus
T. Magedanz IKV++ GmbH
W. Marshall AT&T Research
K. Mehdi University of Kansas
D. Oran Cisco
J. Ott University of Bremen
T. La Porta Bell Labs
B. Rosen Marconi
J. Rosenberg dynamicsoft
H. Sinnreich MCI WorldCom
R. Steinmetz Technical University of Darmstadt
H. Stüttgen NEC CCRLE
W. Wimmreuter Siemens
L. Wolf University of Karlsruhe
A. Wolisz Technical University of Berlin
M. Zitterbart Technical University of Braunschweig
Submission Instructions
-----------------------
Authors are invited to submit full papers written in English
before November 27th, 2000. The submissions will be reviewed, and
accepted papers will be included in the program. Notifications of
acceptance will be sent out on January 12th, 2000. Deadline for
submission of camera-ready copies is January 31st, 2001. Authors
of accepted papers will need to sign a Copyright Transfer Form and
submit a Netbib entry.
Papers must be submitted electronically using the Web site at
http://www.cs.columbia.edu/iptel
Submissions must be in PDF or Postscript; any other documents
cannot be accepted. Postscript papers must use only standard
PostScript fonts: Times Roman, Courier, Symbol, and Helvetica.
Papers must be formatted according to the IEEE Transactions format
except for the font size, which MUST be 11pt. Templates are
available at the Web site
http://www.fokus.gmd.de/events/iptel2001/cfp/
Because of the size limitation on the final manuscript, and to
ensure that the reviewed paper and the final version have a
similar size, papers with more than 11 pages cannot be reviewed.
Submissions must include: title, authors, affiliation, abstract,
list of keywords, and contact information. One of the authors of
each accepted paper must present the paper at iptel'2001.
Contact Address
---------------
Please, send all your inquiries regarding iptel2001 to
iptel2001@egroups.com.
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Date: Mon, 02 Oct 00 18:20:06 EST
To: jhzk9@2s31hg.com
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1-800-827-5722 for free help and answers to questions. Also, the
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Practices, Federal Trade Commission, Bureau of Consumer Protection in
Washington, DC.
/////////////////////////////////////////////////////////////////
Remove at derr9@goplay.com
/////////////////////////////////////////////////////////////////
From rem-conf-request@es.net Mon Oct 2 19:54:12 2000
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Date: Mon, 02 Oct 2000 18:49:07 -0500
From: Qiaobing Xie
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Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
References: <39D203BA.561299A1@era.ericsson.se> <200009271749.MAA02664@agevole.cig.mot.com> <39D88E7D.DFCD1BC7@era.ericsson.se>
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Hi, Magnus,
See my comments below....
Magnus Westerlund wrote:
>
> Hello,
>
> The payload format is designed with IP networks in mind. The problem we see is
> that IP is transported over a wide range of different links with different
> behavior, e.g. different radio and wireline links. Radio and wire do not have
> the same error characteristics. AMR will be widly used over wireless links and
> on these links bandwidth efficiency is extremely important. This lead to the
> bitwise sorting. Octet sorting needs octet alignement for each block of the
> payload, i.e. padding at the end of each block. Which we considered to be to
> expensive.
This further convinces me that the current design presented in the AMR
draft is a radio-link-centric design, which is wrong in my opinion,
because:
1) bandwidth efficiency over wireless links is more of a concern for
ROHC working group. Architecturally, it should be delt with below IP by
header compression/stripping, as being worked upon in rohc. It doesn't
belong to RTP payload layer (unless it is something facilitating the
underlying header compression/stripping).
2) AMR itself is not specfic to radio links, it is a general purpose
codec. It is in the best interest of the AVT working group to define a
*general* RTP payload format for carrying AMR speech over any IP
connection, not something optimized for radio links only. This is why I
am against the bitwise sorting - it may be inevitable for a radio link
(even for radio links, it is expensive and often done at a lower layer
by DSP or with hardware assistance), but it is a magnitude more
expensive than byte-oriented handling to most applications running on a
general purpose machine.
> The optional sensitivity sorting facilitates the use of different
> error robustness schemes. The need for different robustness enhancing schemes
> may depend on the application and the lower transport layers. The application
> can be for example conversational speech or streaming. Lower layers are here
> layer 1 and 2.
Since the bitwise sorting is so expensive, I think it is more than
necessary that you justify it by showing exactly what "robustness
echancing schemes" you have in mind and exactly how they would work
together.
Here you also brought out another concern of mine: the current design
put no consideration on frame quality classification support (you would
need CRC and bad frame indicator to do so, as I pointed out in my
original email). Is this because you assume those functions are provided
by "the lower transport layers", such as by the Iu-UP protocols defined
in 3GPP? Unfortunately, in a general IP network, there is no such
support from the transport layer. In order to make the AMR over IP work
effectively over any IP connections (not only for a 3GPP Iu-UP
interface), you will need to support those functions in your RTP
payload format.
regards,
-Qiaobing
>
> Regards
>
> Magnus W.
>
> Qiaobing Xie wrote:
>
> > Magnus,
> >
> > Correct me if I am wrong, but I find hard to understand your reasoning.
> >
> > What you are saying is basically that, in order to make the current
> > payload format design general enough and flexible enough to be used "for
> > a wide range of networks and applications" and with "any mechanism that
> > exploit" unequal bit sensitivity, you've chosen to compromise the
> > implementation efficiency and performance (and even some functionality)
> > for AMR over RTP/whatever-partial-checksum-UDP/IP.
> >
> > I think it should be the other way around - the AMR payload design
> > should primarily focus on finding the optimal solution for
> > AMR/RTP/UDP*/IP. It doesn't sound right to me for IETF AVT WG to design
> > something that is meant for a wide range of networks but is
> > suboptimal/compromised (even functionally handicapped) for IP networks.
> >
> > Regards,
> > -Qiaobing Xie
> >
> > Magnus Westerlund wrote:
> > >
> > > Dear Qiaobing, everyone,
> > >
> > > The unequal error protection/correction scheme is not designed just for
> > > UDP-Lite. Any mechanism that can exploit the fact that the data is
> > > sorted in sensitivity order is possible to use, e.g. a partial FEC
> > > scheme based on the same techniques as RFC2733.
> > >
> > > We have made some design choices that makes the payload format flexible
> > > and bit efficient, i.e. usefull for a wide range of networks and
> > > applications.
> > >
> > > Best regards
> > >
> > > Magnus Westerlund
> > >
> > > Qiaobing Xie wrote:
> > >
> > > > Hello, everyone:
> > > >
> > > > I have some serious concerns on the design of the AMR payload format as
> > > > defined in . In brief, the design seems
> > > > very limiting in terms of taking advantage of the error tolerant
> > > > features built into the AMR codec algorithm. Also, it seems that not
> > > > enough consideration has been given to implemenation efficiency and
> > > > performance.
> > > >
> > > > Since my concerns are rather fundamental to the design, the email is a
> > > > little long, please bear with me..
> > > >
> > > > 1) Basic idea
> > > >
> > > > The basic idea of the draft is to transport AMR frames over IP in an RTP
> > > > payload, i.e., AMR-frame/RTP/UDP/IP. I have no problem with that.
> > > >
> > > > Therefore, the packet *passed to the UDP layer* will contain three types
> > > > of bits:
> > > >
> > > > +--------------+
> > > > | header bits |
> > > > +--------------+
> > > > | Class A bits |
> > > > | and/or |
> > > > | other bits |
> > > > +--------------+
> > > >
> > > > Here, the "header bits" will include all the AMR speech frame control
> > > > bits (ie., frame header), the RTP payload header bits, and the RTP
> > > > header bits.
> > > >
> > > > "Class A bits" is the most sensitive coded speech data bits, as defined
> > > > by the AMR codec. The "other bits" (also called Class B and C bits in
> > > > AMR terminology) are less sensitive coded speech data bits.
> > > >
> > > > 2) Problems with the current AMR payload format design
> > > >
> > > > The design in the current draft apparently was guided under the
> > > > misconception that the "Class A bits" of an AMR speech frame need to be
> > > > delivered error-free, and if that part of the frame is corrupted the
> > > > whole frame should be tossed. This is not correct.
> > > >
> > > > What the AMR codec really wants from the transport is: "if Class A bits
> > > > is found corrupted in a frame, raise a Bad Frame flag but still pass the
> > > > whole frame to the decoder since the decoder still may want to use Class
> > > > B and C bits to help error concealment." The purpose of the flag is to
> > > > tell the decoder that Class A bits are bad and do not use them.
> > > >
> > > > However, the draft seems to suggest on the use of a partial checksum
> > > > (such as the one provided by the proposed UDP-lite) to protect the
> > > > "header bits" AND the "Class A bits" at the transport leverl. This means
> > > > if Class A bits have any error, the whole packet will be tossed by the
> > > > transport. Even worse, if the packet is a compound payload, you will
> > > > have all the AMR frames tossed by the transport if a bit error occurs in
> > > > the Class A bits of any one of the enclosed frames.
> > > >
> > > > It becomes more problematic when, in order to support multiple AMR
> > > > frames in a compound RTP payload and to use
> > > > the partial checksum, a *very ugly* bit-wise shuffling scheme is
> > > > proposed to copy (bit by bit) all the "Class A bits" of each enclosed
> > > > AMR speech frame to the front of the packet!! The hugely computational
> > > > intensive operation is required no each RTP packet on both ends of the
> > > > RTP session.
> > > >
> > > > 3) A better way to do it:
> > > >
> > > > To get what AMR codec wants, we should only checksum the "header bits",
> > > > and carry an 8-bit CRC in the AMR frame header to cover the "Class A
> > > > bits" of the frame. This CRC is for the RTP receiver (that is above the
> > > > transport) to verify the integrity of the Class A bits and to raise the
> > > > Bad Frame flag before passing the AMR frame to the decoder.
> > > >
> > > > In other words,
> > > >
> > > > 1. Partial checksum at the transport level to cover all the "header
> > > > bits" -- if checksum fails, toss the whole packet.
> > > > 2. CRC in AMR frame header to cover the "Class A bits" -- if CRC fails,
> > > > raise the bad frame flag.
> > > > 3. No error checking on the "other bits".
> > > >
> > > > When bundling multiple AMR frames in one RTP packet (the compound
> > > > payload), we should use a table of contents (TOC) structure to avoid
> > > > bit-wise copying (AMR frames are bit-stuffed to byte boundary). and then
> > > > use the transport level checksum to cover the header bits and TOC bits
> > > > only (the Class A CRC of each AMR frame is in the TOC).
> > > >
> > > > +------------------+
> > > > | RTP header bits | -- RTP and payload headers
> > > > +------------------+
> > > > | Table of Contents| -- list of frame headers of the
> > > > +==================+ \ included AMR frames
> > > > | Fmr1 Cls A bits | \
> > > > +------------------+ |
> > > > | Fmr1 Cls B bits | |
> > > > +------------------+ |
> > > > | Fmr1 Cls C bits | |
> > > > +==================+ |
> > > > | Fmr2 Cls A bits | > not covered by transport level chksum
> > > > +------------------+ |
> > > > | Fmr2 Cls B bits | |
> > > > +------------------+ |
> > > > | Fmr2 Cls C bits | |
> > > > +==================+ |
> > > > | Fmr3 Cls A bits | /
> > > > +------------------+ /
> > > > .....
> > > > .....
> > > >
> > > > This way, you get what the AMR codec asks for and yet avoid bit-wise
> > > > copying.
> > > >
> > > > ===============================================================
> > > > Qiaobing Xie, Ph.D
> > > > Principal Staff Engineer
> > > > Network Architecture & Technology
> > > > Motorola, Inc.
> > > > TEL: (847) 632-3028
> > > > ===============================================================
> > >
> > > --
> > >
> > > Magnus Westerlund
> > >
> > > Audio Technology, Ericsson Research
> > > ----------------------------------------------------------------------
> > > Ericsson Radio Systems AB | Phone +46 8 4048287
> > > Torshamsgatan 23 | Fax +46 8 7575550
> > > S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
> --
>
> Magnus Westerlund
>
> Audio Technology, Ericsson Research
> ----------------------------------------------------------------------
> Ericsson Radio Systems AB | Phone +46 8 4048287
> Torshamsgatan 23 | Fax +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
From rem-conf-request@es.net Mon Oct 2 21:48:01 2000
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From: "Yoshihiro Kikuchi"
To: "IETF AVT" ,
"4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "Yoshihiro Kikuchi"
Subject: draft text for MPEG-4 Audio/Visual RTP I-D revision
Date: Tue, 3 Oct 2000 10:36:25 +0900
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Dear experts,
Attached find please a draft text for the revision of MPEG-4 Audio/Visual
RTP Internet-Draft.
The revisions are mostly to reflect suggestions by the AVT chair. Most of
them are editorial and there are some minor revisions and clarifications.
The following is a summary of the changes:
(1) MIME subtype names were changed to "video/MP4V-ES" for video and
"audio/MP4A-LATM" for audio.
(2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
video header mode was changed to "MAY" to "SHOULD".
(3) Clarification of timestamp definition
There was a suggestion not to describe the detailed definition of the
video timestamp in the RTP spec. and change it as a reference to MPEG-4
visual document, so as to keep consistency with the MPEG-4 visual. A
description about the RTP timestamp in the RTCP SR packet was added.
(4) SA/TTSI over LATM
A description was added to restrict the LATM usage so as not to transmit
SA data by LATM.
(5) Other editorial and wording changes.
Please let us know if there is comment on the attached draft. The authors
need to submit it soon as an updated I-D and hopefully final before RFC.
Best regards,
Yoshihiro Kikuchi
and authors of the Internet Draft
----
Yoshihiro Kikuchi
E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
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Internet Engineering Task Force Yoshihiro Kikuchi - =
Toshiba
Internet Draft Toshiyuki Nomura - =
NEC
Document: draft-ietf-avt-rtp-mpeg4-es-05.txt Shigeru Fukunaga - =
Oki
Yoshinori Matsui - =
Matsushita
Hideaki Kimata - =
NTT
September 29, =
2000
RTP payload format for MPEG-4 Audio/Visual streams
Status of this Memo
This document is an Internet-Draft and is in full conformance with =
all
provisions of Section 10 of RFC2026 [1].
Internet-Drafts are working documents of the Internet Engineering =
Task
Force (IETF), its areas, and its working groups. Note that other =
groups
may also distribute working documents as Internet-Drafts. =
Internet-Drafts
are draft documents valid for a maximum of six months and may be =
updated,
replaced, or obsoleted by other documents at any time. It is
inappropriate to use Internet- Drafts as reference material or to =
cite
them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
Abstract
This document describes RTP payload formats for carrying each of =
MPEG-4
Audio and MPEG-4 Visual bitstreams without using MPEG-4 Systems. For =
the
purpose of directly mapping MPEG-4 Audio/Visual bitstreams onto RTP
packets, it provides specifications for the use of RTP header fields =
and
also specifies fragmentation rules. It also provides specifications =
for
MIME type registrations and the use of SDP.
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 1]
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1. Introduction
The RTP payload formats described in this document specify how MPEG-4
Audio [3][5] and MPEG-4 Visual streams [2][4] are to be fragmented =
and
mapped directly onto RTP packets.
These RTP payload formats enable transport of MPEG-4 Audio/Visual =
streams
without using the synchronization and stream management functionality =
of
MPEG-4 Systems [6]. Such RTP payload formats will be used in systems =
that
have intrinsic stream management functionality and thus require no =
such
functionality from MPEG-4 Systems. H.323 terminals are an example of =
such
a systems, where MPEG-4 Audio/Visual streams are not managed by =
MPEG-4
Systems Object Descriptors but by H.245. The streams are directly =
mapped
onto RTP packets without using MPEG-4 Systems Sync Layer. Other =
examples
are SIP and RTSP where MIME and SDP are used. MIME types and SDP =
usages
of the RTP payload formats described in this document are defined to
directly specify the attribute of Audio/Visual streams (e.g. media =
type,
packetization format and codec configuration) without using MPEG-4
Systems. The obvious benefit is that these MPEG-4 Audio/Visual RTP
payload formats can be handled in an unified way together with those
formats defined for non-MPEG-4 codecs. The disadvantage is that
interoperability with environments using MPEG-4 Systems may be =
difficult,
other payload formats may be better suited to those applications.
The semantics of RTP headers in such cases need to be clearly =
defined,
including the association with MPEG-4 Audio/Visual data elements. In
addition, it is beneficial to define the fragmentation rules of RTP
packets for MPEG-4 Video streams so as to enhance error resiliency by
utilizing the error resilience tools provided inside the MPEG-4 Video
stream.
1.1 MPEG-4 Visual RTP payload format
MPEG-4 Visual is a visual coding standard with many new features: =
high
coding efficiency; high error resiliency; multiple, arbitrary shape
object-based coding; etc. [2]. It covers a wide range of bitrates =
from
scores of Kbps to several Mbps. It also covers a wide variety of
networks, ranging from those guaranteed to be almost error-free to =
mobile
networks with high error rates.
With respect to the fragmentation rules for an MPEG-4 visual =
bitstream
defined in this document, since MPEG-4 Visual is used for a wide =
variety
of networks, it is desirable not to apply too much restriction on
fragmentation, and a fragmentation rule such as "a single video =
packet
shall always be mapped on a single RTP packet" may be inappropriate. =
On
the other hand, careless, media unaware fragmentation may cause
degradation in error resiliency and bandwidth efficiency. The
fragmentation rules described in this document are flexible but =
manage to
define the minimum rules for preventing meaningless fragmentation =
while
utilizing the error resilience functionalities of MPEG-4 Visual.
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 2]
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The fragmentation rule recommends not to map more than one VOP in an =
RTP
packet so that the RTP timestamp uniquely indicates the VOP time =
framing.
On the other hand, MPEG-4 video may generate VOPs of very small size, =
in
cases with an empty VOP (vop_coded=3D0) containing only VOP header or =
an
arbitrary shaped VOP with a small number of coding blocks. To reduce =
the
overhead for such cases, the fragmentation rule permits concatenating
multiple VOPs in an RTP packet. (See fragmentation rule (4) in =
section
3.2 and marker bit and timestamp in section 3.1.)
While the additional media specific RTP header defined for such video
coding tools as H.261 or MPEG-1/2 is effective in helping to recover
picture headers corrupted by packet losses, MPEG-4 Visual has already
error resilience functionalities for recovering corrupt headers, and
these can be used on RTP/IP networks as well as on other networks
(H.223/mobile, MPEG-2/TS, etc.). Therefore, no extra RTP header =
fields
are defined in this MPEG-4 Visual RTP payload format.
1.2 MPEG-4 Audio RTP payload format
MPEG-4 Audio is a new kind of audio standard that integrates many
different types of audio coding tools. Low-overhead MPEG-4 Audio
Transport Multiplex (LATM) manages the sequences of audio data with
relatively small overhead. In audio-only applications, then, it is
desirable for LATM-based MPEG-4 Audio bitstreams to be directly =
mapped
onto the RTP packets without using MPEG-4 Systems.
While LATM has several multiplexing features as follows;
- Carrying configuration information with audio data,
- Concatenation of multiple audio frames in one audio stream,
- Multiplexing multiple objects (programs),
- Multiplexing scalable layers,
in RTP transmission there is no need for the last two features.
Therefore, these two features MUST NOT be used in applications based =
on
RTP packetization specified by this document. Since LATM has been
developed for only natural audio coding tools, i.e. not for synthesis
tools, it seems difficult to transmit Structured Audio (SA) data and =
Text
to Speech Interface (TTSI) data by LATM. Therefore, SA data and TTSI =
data
MUST NOT be transported by the RTP packetization in this document
For transmission of scalable streams, audio data of each layer should =
be
packetized onto different RTP packets allowing for the different =
layers
to be treated differently at the IP level, for example via some means =
of
differentiated service. On the other hand, all configuration data of =
the
scalable streams are contained in one LATM configuration data
"StreamMuxConfig" and every scalable layer shares the =
StreamMuxConfig.
The mapping between each layer and its configuration data is achieved =
by
LATM header information attached to the audio data. In order to =
indicate
the dependency information of the scalable streams, a restriction is
applied to the dynamic assignment rule of payload type (PT) values =
(see
section 4.2).
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 3]
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For MPEG-4 Audio coding tools, as is true for other audio coders, if =
the
payload is a single audio frame, packet loss will not impair the
decodability of adjacent packets. Therefore, the additional media
specific header for recovering errors will not be required for MPEG-4
Audio. Existing RTP protection mechanisms, such as Generic Forward =
Error
Correction (RFC 2733) and Redundant Audio Data (RFC 2198), MAY be =
applied
to improve error resiliency.
2. Conventions used in this document
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC-2119 [7].
3. RTP Packetization of MPEG-4 Visual bitstream
This section specifies RTP packetization rules for MPEG-4 Visual =
content.
An MPEG-4 Visual bitstream is mapped directly onto RTP packets =
without
the addition of extra header fields or any removal of Visual syntax
elements. The Combined Configuration/Elementary stream mode MUST be =
used
so that configuration information will be carried to the same RTP =
port as
the elementary stream. (see 6.2.1 "Start codes" of ISO/IEC 14496-2
[2][9][4]) The configuration information MAY additionally be =
specified by
some out-of-band means. If needed for an H.323 terminal, H.245 =
codepoint
"decoderConfigurationInformation" MUST be used for this purpose. If
needed by systems using MIME content type and SDP parameters, e.g. =
SIP
and RTSP, the optional parameter "config" MUST be used to specify the
configuration information. (see 5.1 and 5.2)
When the short video header mode is used, RTP payload format for =
H.263
(e.g., RFC 2190 or RFC 2429) SHOULD be used.
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 4]
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=3D2|P|X| CC |M| PT | sequence number | =
RTP
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp | =
Header
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
=
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
| contributing source (CSRC) identifiers |
| .... |
=
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
| | RTP
| MPEG-4 Visual stream (byte aligned) | =
Payload
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1 - An RTP packet for MPEG-4 Visual stream
3.1 Use of RTP header fields for MPEG-4 Visual
Payload Type (PT): The assignment of an RTP payload type for this new
packet format is outside the scope of this document, and will not be
specified here. It is expected that the RTP profile for a particular
class of applications will assign a payload type for this encoding, =
or if
that is not done then a payload type in the dynamic range shall be =
chosen
by means of an out of band signaling protocol (e.g. H.245, SIP, etc).
Extension (X) bit: Defined by the RTP profile used.
Sequence Number: Incremented by one for each RTP data packet sent,
starting, for security reasons, with a random initial value.
Marker (M) bit: The marker bit is set to one to indicate the last RTP
packet (or only RTP packet) of a VOP. When multiple VOPs are carried =
in
the same RTP packet, the marker bit is set to 1.
Timestamp: The timestamp indicates the composition time, or the
presentation time in a no-compositor decoder. It is the same time
instance as that conveyed in the timestamp field (modulo_time_base =
and
vop_time_increment) of the video stream. (See ISO/IEC 14496-2[2], =
section
6 for the detailed definition.) A constant offset, which is random, =
is
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 5]
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2000=0D
added for security reasons. The detailed definition of the timestamp =
is
as follows:
- When multiple VOPs are carried in the same RTP packet, the =
timestamp
indicates the earliest of the presentation times within the VOPs
carried in the RTP packet. Timestamp information of the rest of the
VOPs are derived from the timestamp fields in the VOP header
(modulo_time_base and vop_time_increment).
- If the RTP packet contains only configuration information and/or
Group_of_VideoObjectPlane() fields, the presentation time of the =
next
VOP in the coding order is used.
- If the RTP packet contains only visual_object_sequence_end_code
information, the presentation time of the immediately preceding VOP =
in
the coding order is used.
The resolution of the timestamp is set to its default value of 90KHz,
unless specified by an out-of-band means (e.g. SDP parameter or MIME
parameter as defined in section 5).
Other header fields are used as described in RFC 1889 [8].
RTP timestamps in RTCP SR packets: According to the RTP timing model, =
the
RTP timestamp that is carried into an RTCP SR packet is the same as =
the
presentation time that would be applied to an RTP packet for data =
that
was sampled at the instant the SR packet is being generated and sent. =
The
RTP timestamp value is calculated from the NTP timestamp for the =
current
time which also goes in the RTCP SR packet. To perform that =
calculation,
an implementation needs to periodically establish a correspondence
between the presentation time of a data packet and the NTP time at =
which
that data was sampled.
3.2 Fragmentation of MPEG-4 Visual bitstream
A fragmented MPEG-4 Visual bitstream is mapped directly onto the RTP
payload without any addition of extra header fields or any removal of
Visual syntax elements. The Combined Configuration/Elementary streams
mode is used. The following rules apply for the fragmentation.
In the following header means one of the following:
- Configuration information (Visual Object Sequence Header, Visual =
Object
Header and Video Object Layer Header)
- visual_object_sequence_end_code
- The header of the entry point function for an elementary stream
(Group_of_VideoObjectPlane() or the header of VideoObjectPlane(),
video_plane_with_short_header(), MeshObject() or FaceObject())
- The video packet header (video_packet_header() excluding
next_resync_marker())
- The header of gob_layer()
See 6.2.1 "Start codes" of ISO/IEC 14496-2[2][9][4] for the =
definition of
the configuration information and the entry point functions.
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 6]
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(1) Configuration information and Group_of_VideoObjectPlane() fields
SHALL be placed at the beginning of the RTP payload (just after the =
RTP
header) or just after the header of the syntactically upper layer
function.
(2) If one or more headers exist in the RTP payload, the RTP payload
SHALL begin with the header of the syntactically highest function.
Note: The visual_object_sequence_end_code is regarded as the lowest
function.
(3) A header SHALL NOT be split into a plurality of RTP packets.
(4) Different VOPs SHOULD be fragmented into different RTP packets so
that one RTP packet consists of the data bytes associated with a =
unique
presentation time (that is indicated in the timestamp field in the =
RTP
packet header), with the exception that multiple consecutive VOPs MAY =
be
carried within one RTP packet in the decoding order if the size of =
the
VOPs is small.
Note: When multiple VOPs are carried in one RTP payload, the =
presentation
time of the VOPs after the first one may be calculated by the =
decoder.
This operation is necessary only for RTP packets in which the marker =
bit
equals to one and the beginning of RTP payload corresponds to a start
code. (See timestamp and marker bit in section 3.1)
(5) It is RECOMMENDED that a single video packet is sent as a single =
RTP
packet. The size of a video packet SHOULD be adjusted in such a way =
that
the resulting RTP packet is not larger than the path-MTU.
Note: Rule (5) does not apply when the video packet is disabled by =
the
coder configuration (by setting resync_marker_disable in the VOL =
header
to 1), or in coding tools where the video packet is not supported. In
this case, a VOP MAY be split at arbitrary byte-positions.
The video packet starts with the VOP header or the video packet =
header,
followed by motion_shape_texture(), and ends with =
next_resync_marker() or
next_start_code().
3.3 Examples of packetized MPEG-4 Visual bitstream
Figure 2 shows examples of RTP packets generated based on the =
criteria
described in 3.2
(a) is an example of the first RTP packet or the random access point =
of
an MPEG-4 visual bitstream containing the configuration information.
According to criterion (1), the Visual Object Sequence Header(VS =
header)
is placed at the beginning of the RTP payload, preceding the Visual
Object Header and the Video Object Layer Header(VO header, VOL =
header).
Since the fragmentation rule defined in 3.2 guarantees that the
configuration information, starting with
visual_object_sequence_start_code, is always placed at the beginning =
of
the RTP payload, RTP receivers can detect the random access point by
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 7]
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checking if the first 32-bit field of the RTP payload is
visual_object_sequence_start_code.
(b) is another example of the RTP packet containing the configuration
information. It differs from example (a) in that the RTP packet also
contains a video packet in the VOP following the configuration
information. Since the length of the configuration information is
relatively short (typically scores of bytes) and an RTP packet =
containing
only the configuration information may thus increase the overhead, =
the
configuration information and the immediately following GOV and/or (a
part of) VOP can be packetized into a single RTP packet as in this
example.
(c) is an example of an RTP packet that contains
Group_of_VideoObjectPlane(GOV). Following criterion (1), the GOV is
placed at the beginning of the RTP payload. It would be a waste of =
RTP/IP
header overhead to generate an RTP packet containing only a GOV whose
length is 7 bytes. Therefore, (a part of) the following VOP can be =
placed
in the same RTP packet as shown in (c).
(d) is an example of the case where one video packet is packetized =
into
one RTP packet. When the packet-loss rate of the underlying network =
is
high, this kind of packetization is recommended. Even when the RTP =
packet
containing the VOP header is discarded by a packet loss, the other =
RTP
packets can be decoded by using the HEC(Header Extension Code)
information in the video packet header. No extra RTP header field is
necessary.
(e) is an example of the case where more than one video packet is
packetized into one RTP packet. This kind of packetization is =
effective
to save the overhead of RTP/IP headers when the bit-rate of the
underlying network is low. However, it will decrease the packet-loss
resiliency because multiple video packets are discarded by a single =
RTP
packet loss. The optimal number of video packets in an RTP packet and =
the
length of the RTP packet can be determined considering the =
packet-loss
rate and the bit-rate of the underlying network.
(f) is an example of the case when the video packet is disabled by
setting resync_marker_disable in the VOL header to 1. In this case, a =
VOP
may be split into a plurality of RTP packets at arbitrary =
byte-positions.
For example, it is possible to split a VOP into fixed-length packets.
This kind of coder configuration and RTP packet fragmentation may be =
used
when the underlying network is guaranteed to be error-free. On the =
other
hand, it is not recommended to use it in error-prone environment =
since it
provides only poor packet loss resiliency.
Figure 3 shows examples of RTP packets prohibited by the criteria of =
3.2.
Fragmentation of a header into multiple RTP packets, as in (a), will =
not
only increase the overhead of RTP/IP headers but also decrease the =
error
resiliency. Therefore, it is prohibited by the criterion (3).
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 8]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
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When concatenating more than one video packets into an RTP packet, =
VOP
header or video_packet_header() shall not be placed in the middle of =
the
RTP payload. The packetization as in (b) is not allowed by criterion =
(2)
due to the aspect of the error resiliency. Comparing this example =
with
Figure 2(d), although two video packets are mapped onto two RTP =
packets
in both cases, the packet-loss resiliency is not identical. Namely, =
if
the second RTP packet is lost, both video packets 1 and 2 are lost in =
the
case of Figure 3(b) whereas only video packet 2 is lost in the case =
of
Figure 2(d).
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 9]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
+------+------+------+------+
(a) | RTP | VS | VO | VOL |
|header|header|header|header|
+------+------+------+------+
+------+------+------+------+------------+
(b) | RTP | VS | VO | VOL |Video Packet|
|header|header|header|header| |
+------+------+------+------+------------+
+------+-----+------------------+
(c) | RTP | GOV |Video Object Plane|
|header| | |
+------+-----+------------------+
+------+------+------------+ +------+------+------------+
(d) | RTP | VOP |Video Packet| | RTP | VP |Video Packet|
|header|header| (1) | |header|header| (2) |
+------+------+------------+ +------+------+------------+
=
+------+------+------------+------+------------+------+------------+
(e) | RTP | VP |Video Packet| VP |Video Packet| VP |Video =
Packet|
|header|header| (1) |header| (2) |header| (3) =
|
=
+------+------+------------+------+------------+------+------------+
+------+------+------------+ +------+------------+
(f) | RTP | VOP |VOP fragment| | RTP |VOP fragment|
|header|header| (1) | |header| (2) | ___
+------+------+------------+ +------+------------+
Figure 2 - Examples of RTP packetized MPEG-4 Visual bitstream
+------+-------------+ +------+------------+------------+
(a) | RTP |First half of| | RTP |Last half of|Video Packet|
|header| VP header | |header| VP header | |
+------+-------------+ +------+------------+------------+
+------+------+----------+ =
+------+---------+------+------------+
(b) | RTP | VOP |First half| | RTP |Last half| VP |Video =
Packet|
|header|header| of VP(1) | |header| of VP(1)|header| (2) =
|
+------+------+----------+ =
+------+---------+------+------------+
Figure 3 - Examples of prohibited RTP packetization for MPEG-4 Visual
bitstream
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4. RTP Packetization of MPEG-4 Audio bitstream
This section specifies RTP packetization rules for MPEG-4 Audio
bitstreams. MPEG-4 Audio streams MUST be formatted by LATM =
(Low-overhead
MPEG-4 Audio Transport Multiplex) tool[5], and the LATM-based streams =
are
then mapped onto RTP packets as described the three sections below.
4.1 RTP Packet Format
LATM-based streams consist of a sequence of audioMuxElements that =
include
one or more audio frames. A complete audioMuxElement or a part of one
SHALL be mapped directly onto an RTP payload without any removal of
audioMuxElement syntax elements (see Figure 4). The first byte of =
each
audioMuxElement SHALL be located at the first payload location in an =
RTP
packet.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=3D2|P|X| CC |M| PT | sequence number =
|RTP
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp =
|Header
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
=
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
| contributing source (CSRC) identifiers |
| .... |
=
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
| |RTP
: audioMuxElement (byte aligned) =
:Payload
| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| :...OPTIONAL RTP padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4 - An RTP packet for MPEG-4 Audio
In order to decode the audioMuxElement, the following =
muxConfigPresent
information is required to be indicated by an out-of-band means. When =
SDP
is utilized for this indication, MIME parameter "cpresent" =
corresponds to
the muxConfigPresent information (see section 5.3).
muxConfigPresent: If this value is set to 1 (in-band mode), the
audioMuxElement SHALL include an indication bit "useSameStreamMux" =
and
MAY include the configuration information for audio compression
"StreamMuxConfig". The useSameStreamMux bit indicates whether the
StreamMuxConfig element in the previous frame is applied in the =
current
frame. If the useSameStreamMux bit indicates to use the =
StreamMuxConfig
from the previous frame, but if the previous frame has been lost, the
current frame may not be decodable. Therefore, in case of in-band =
mode,
the StreamMuxConfig element SHOULD be transmitted repeatedly =
depending on
the network condition. On the other hand, if muxConfigPresent is set =
to 0
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 11]
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(out-band mode), the StreamMuxConfig element is required to be
transmitted by an out-of-band means. In case of SDP, MIME parameter
"config" is utilized (see section 5.3).
4.2 Use of RTP Header Fields for MPEG-4 Audio
Payload Type (PT): The assignment of an RTP payload type for this new
packet format is outside the scope of this document, and will not be
specified here. It is expected that the RTP profile for a particular
class of applications will assign a payload type for this encoding, =
or if
that is not done then a payload type in the dynamic range shall be =
chosen
by means of an out of band signaling protocol (e.g. H.245, SIP, etc). =
In
the dynamic assignment of RTP payload types for scalable streams, a
different value should be assigned to each layer. The assigned values
should be in order of enhance layer dependency, where the base layer =
has
the smallest value.
Marker (M) bit: The marker bit indicates audioMuxElement boundaries. =
It
is set to one to indicate that the RTP packet contains a complete
audioMuxElement or the last fragment of an audioMuxElement.
Timestamp: The timestamp indicates the sampling instance of the first
audio frame contained in the RTP packet. Timestamps are recommended =
to
start at a random value for security reasons.
Unless specified by an out-of-band means, the resolution of the =
timestamp
is set to its default value of 90 kHz.
Sequence Number: Incremented by one for each RTP packet sent, =
starting,
for security reasons, with a random value.
Other header fields are used as described in RFC 1889 [8].
4.3 Fragmentation of MPEG-4 Audio bitstream
It is RECOMMENDED to put one audioMuxElement in each RTP packet. If =
the
size of an audioMuxElement can be kept small enough that the size of =
the
RTP packet containing it does not exceed the size of the path-MTU, =
this
will be no problem. If it cannot, the audioMuxElement MAY be =
fragmented
and spread across multiple packets.
5. MIME type registration for MPEG-4 Audio/Visual streams
The following sections describe the MIME type registrations for =
MPEG-4
Audio/Visual streams. MIME type registration and SDP usage for the =
MPEG-4
Visual stream are described in Sections 5.1 and 5.2, respectively, =
while
MIME type registration and SDP usage for MPEG-4 Audio stream are
described in Sections 5.3 and 5.4, respectively.
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 12]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
(In the following sections, the RFC number "XXXX" represents the RFC
number, which should be assigned for this document.)
5.1 MIME type registration for MPEG-4 Visual
MIME media type name: video
MIME subtype name: MP4V-ES
Required parameters: none
Optional parameters:
rate: This parameter is used only for RTP transport. It indicates =
the
resolution of the timestamp field in the RTP header. If this =
parameter
is not specified, its default value of 90000 (90KHz) is used.
profile-level-id: A decimal representation of MPEG-4 Visual Profile
Level indication value (profile_and_level_indication) defined in =
Table
G-1 of ISO/IEC 14496-2 [2][4]. This parameter MAY be used in the
capability exchange or session setup procedure to indicate MPEG-4
Visual Profile and Level combination of which the MPEG-4 Visual =
codec
is capable. If this parameter is not specified by the procedure, =
its
default value of 1 (Simple Profile/Level 1) is used.
config: This parameter indicates the configuration of the
corresponding MPEG-4 visual bitstream. It SHALL NOT be used to
indicate the codec capability in the capability exchange procedure. =
It
is a hexadecimal representation of an octet string that expresses =
the
MPEG-4 Visual configuration information, as defined in subclause =
6.2.1
Start codes of ISO/IEC14496-2[2][4][9]. The configuration =
information
is mapped onto the octet string in an MSB-first basis. The first =
bit
of the configuration information SHALL be located at the MSB of the
first octet. The configuration information indicated by this =
parameter
SHALL be the same as the configuration information in the
corresponding MPEG-4 Visual stream, except for
first_half_vbv_occupancy and latter_half_vbv_occupancy, if exist,
which may vary in the repeated configuration information inside an
MPEG-4 Visual stream (See 6.2.1 Start codes of ISO/IEC14496-2).
Example usages for these parameters are:
- MPEG-4 Visual Simple Profile/Level 1:
Content-type: video/mp4v-es; profile-level-id=3D1
- MPEG-4 Visual Core Profile/Level 2:
Content-type: video/mp4v-es; profile-level-id=3D34
- MPEG-4 Visual Advanced Real Time Simple Profile/Level 1:
Content-type: video/mp4v-es; profile-level-id=3D145
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 13]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
Published specification:
The specifications for MPEG-4 Visual streams are presented in =
ISO/IEC
14469-2[2][4][9]. The RTP payload format is described in RFCXXXX.
Encoding considerations:
Video bitstreams must be generated according to MPEG-4 Visual
specifications (ISO/IEC 14496-2). A video bitstream is binary data =
and
must be encoded for non-binary transport (for Email, the Base64
encoding is sufficient). This type is also defined for transfer =
via
RTP. The RTP packets MUST be packetized according to the MPEG-4 =
Visual
RTP payload format defined in RFCXXXX.
Security considerations:
See section 6 of RFCXXXX.
Interoperability considerations:
MPEG-4 Visual provides a large and rich set of tools for the coding =
of
visual objects. For effective implementation of the standard, =
subsets
of the MPEG-4 Visual tool sets have been provided for use in =
specific
applications. These subsets, called 'Profiles', limit the size of =
the
tool set a decoder is required to implement. In order to restrict
computational complexity, one or more Levels are set for each =
Profile.
A Profile@Level combination allows:
o a codec builder to implement only the subset of the standard he
needs, while maintaining interworking with other MPEG-4 devices
included in the same combination, and
o checking whether MPEG-4 devices comply with the standard
('conformance testing').
The visual stream SHALL be compliant with the MPEG-4 Visual
Profile@Level specified by the parameter "profile-level-id".
Interoperability between a sender and a receiver may be achieved by
specifying the parameter "profile-level-id" in MIME content, or by
arranging in the capability exchange/announcement procedure to set =
this
parameter mutually to the same value.
Applications which use this media type:
Audio and visual streaming and conferencing tools, Internet =
messaging
and Email applications.
Additional information: none
Person & email address to contact for further information:
The authors of RFCXXXX. (See section 8)
Intended usage: COMMON
Author/Change controller:
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 14]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
The authors of RFCXXXX. (See section 8)
5.2 SDP usage of MPEG-4 Visual
The MIME media type video/MP4V-ES string is mapped to fields in the
Session Description Protocol (SDP), RFC 2327, as follows:
o The MIME type (video) goes in SDP "m=3D" as the media name.
o The MIME subtype (MP4V-ES) goes in SDP "a=3Drtpmap" as the encoding =
name.
o The optional parameter "rate" goes in "a=3Drtpmap" as the clock =
rate.
o The optional parameter "profile-level-id" and "config" go in the
"a=3Dfmtp" line to indicate the coder capability and configuration,
respectively. These parameters are expressed as a MIME media type =
string,
in the form of as a semicolon separated list of parameter=3Dvalue =
pairs.
The following are some examples of media representation in SDP:
Simple Profile/Level 1, rate=3D90000(90KHz), "profile-level-id" and
"config" are present in "a=3Dfmtp" line:
m=3Dvideo 49170/2 RTP/AVP 98
a=3Drtpmap:98 MP4V-ES/90000
a=3Dfmtp:98 profile-level-
=
id=3D1;config=3D000001B001000001B5090000010000000120008440FA282C2090A21F
Core Profile/Level 2, rate=3D90000(90KHz), "profile-level-id" is =
present in
"a=3Dfmtp" line:
m=3Dvideo 49170/2 RTP/AVP 98
a=3Drtpmap:98 MP4V-ES/90000
a=3Dfmtp:98 profile-level-id=3D34
Advance Real Time Simple Profile/Level 1, rate=3D25(25Hz), =
"profile-level-
id" is present in "a=3Dfmtp" line:
m=3Dvideo 49170/2 RTP/AVP 98
a=3Drtpmap:98 MP4V-ES/25
a=3Dfmtp:98 profile-level-id=3D145
5.3 MIME type registration of MPEG-4 Audio
MIME media type name: audio
MIME subtype name: MP4A-LATM
Required parameters:
rate: the rate parameter indicates the RTP time stamp clock rate. =
The
default value is 90000. Other rates MAY be specified only if they =
are
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 15]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
set to the same value as the audio sampling rate (number of samples
per second).
Optional parameters:
profile-level-id: a decimal representation of MPEG-4 Audio Profile
Level indication value defined in ISO/IEC 14496-1 [10]. This =
parameter
indicates which MPEG-4 Audio tool subsets the decoder is capable of
using. If this parameter is not specified in the capability =
exchange
or session setup procedure, its default value of 30 (Natural Audio
Profile/Level 1) is used.
object: a decimal representation of the MPEG-4 Audio Object Type =
value
defined in ISO/IEC 14496-3 [5]. This parameter specifies the tool =
to
be used by the coder. It CAN be used to limit the capability within
the specified "profile-level-id".
bitrate: the data rate for the audio bit stream.
cpresent: this parameter indicates whether audio payload =
configuration
data has been multiplexed into an RTP payload (see section 4.1). If
not specified, the default value is 1.
config: a hexadecimal representation of an octet string that =
expresses
the audio payload configuration data "StreamMuxConfig", as defined =
in
ISO/IEC 14496-3 [5] (see section 4.1). Configuration data is mapped
onto the octet string in an MSB-first basis. The first bit of the
configuration data SHALL be located at the MSB of the first octet. =
In
the last octet, zero-padding bits, if necessary, shall follow the
configuration data.
ptime: RECOMMENDED duration of each packet in milliseconds.
Published specification:
Payload format specifications are described in this document. =
Encoding
specifications are provided in ISO/IEC 14496-3 [3][5].
Encoding considerations:
This type is only defined for transfer via RTP.
Security considerations:
See Section 6 of RFCXXXX.
Interoperability considerations:
MPEG-4 Audio provides a large and rich set of tools for the coding =
of
audio objects. For effective implementation of the standard, =
subsets of
the MPEG-4 Audio tool sets similar to those used in MPEG-4 Visual =
have
been provided (see section 5.1).
The audio stream SHALL be compliant with the MPEG-4 Audio
Profile@Level specified by the parameter "profile-level-id".
Interoperability between a sender and a receiver may be achieved by
specifying the parameter "profile-level-id" in MIME content, or by
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 16]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
arranging in the capability exchange procedure to set this =
parameter
mutually to the same value. Furthermore, the "object" parameter can =
be
used to limit the capability within the specified Profile@Level in
capability exchange.
Applications which use this media type:
Audio and video streaming and conferencing tools.
Additional information: none
Personal & email address to contact for further information:
See Section 8 of RFCXXXX.
Intended usage: COMMON
Author/Change controller:
See Section 8 of RFCXXXX.
5.4 SDP usage of MPEG-4 Audio
The MIME media type audio/MP4A-LATM string is mapped to fields in the
Session Description Protocol (SDP), RFC 2327, as follows:
o The MIME type (audio) goes in SDP "m=3D" as the media name.
o The MIME subtype (MP4A-LATM) goes in SDP "a=3Drtpmap" as the =
encoding
name.
o The required parameter "rate" goes in "a=3Drtpmap" as the clock =
rate.
o The optional parameter "ptime" goes in SDP "a=3Dptime" attribute.
o The optional parameter "profile-level-id" goes in the "a=3Dfmtp" =
line to
indicate the coder capability. The "object" parameter goes in the
"a=3Dfmtp" attribute. The payload-format-specific parameters =
"bitrate",
"cpresent" and "config" go in the "a=3Dfmtp" line. These parameters =
are
expressed as a MIME media type string, in the form of as a semicolon
separated list of parameter=3Dvalue pairs.
The following are some examples of the media representation in SDP:
For 6 kb/s CELP bitstreams (with an audio sampling rate of 8 kHz),
m=3Daudio 49230 RTP/AVP 96
a=3Drtpmap:96 MP4A-LATM/8000
a=3Dfmtp:96 =
profile-level-id=3D9;object=3D8;cpresent=3D0;config=3D9128B1071070
a=3Dptime:20
For 64 kb/s AAC LC stereo bitstreams (with an audio sampling rate of =
24
kHz),
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 17]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
m=3Daudio 49230 RTP/AVP 96
a=3Drtpmap:96 MP4A-LATM/24000
a=3Dfmtp:96 profile-level-id=3D1; bitrate=3D64000; cpresent=3D0;
config=3D9122620000
In the above two examples, audio configuration data is not =
multiplexed
into the RTP payload and is described only in SDP. Furthermore, the
"clock rate" is set to the audio sampling rate.
If the clock rate has been set to its default value and it is =
necessary
to obtain the audio sampling rate, this can be done by parsing the
"config" parameter (see the following example).
m=3Daudio 49230 RTP/AVP 96
a=3Drtpmap:96 MP4A-LATM/90000
a=3Dfmtp:96 object=3D8; cpresent=3D0; config=3D9128B1071070
The following example shows that the audio configuration data appears =
in
the RTP payload.
m=3Daudio 49230 RTP/AVP 96
a=3Drtpmap:96 MP4A-LATM/90000
a=3Dfmtp:96 object=3D2; cpresent=3D1
6. Security Considerations
RTP packets using the payload format defined in this specification =
are
subject to the security considerations discussed in the RTP =
specification
[8]. This implies that confidentiality of the media streams is =
achieved
by encryption. Because the data compression used with this payload =
format
is applied end-to-end, encryption may be performed on the compressed =
data
so there is no conflict between the two operations.
The complete MPEG-4 system allows for transport of a wide range of
content, including Java applets (MPEG-J) and scripts. Since this =
payload
format is restricted to audio and video streams, it is not possible =
to
transport such active content in this format.
7. References
1 Bradner, S., "The Internet Standards Process -- Revision 3", BCP =
9,
RFC 2026, October 1996.
2 ISO/IEC 14496-2:1999, "Information technology - Coding of =
audio-visual
objects - Part2: Visual", December 1999.
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 18]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
3 ISO/IEC 14496-3:1999, "Information technology - Coding of =
audio-visual
objects - Part3: Audio", December 1999.
4 ISO/IEC 14496-2:1999/FDAM1:2000, December 1999.
5 ISO/IEC 14496-3:1999/FDAM1:2000, December 1999.
6 ISO/IEC 14496-1:1999, "Information technology - Coding of =
audio-visual
objects - Part1: Systems", December 1999.
7 Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997
8 H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson "RTP: A =
Transport
Protocol for Real Time Applications", RFC 1889, Internet =
Engineering
Task Force, January 1996.
9 ISO/IEC 14496-2:1999/COR1:2000, "Information technology - Coding =
of
audio-visual objects - Part2: Visual, Technical corrigendum 1", =
August
2000.
10 ISO/IEC 14496-1:1999/FDAM1:2000, December 1999.
8. Author's Addresses
Yoshihiro Kikuchi
Toshiba corporation
1, Komukai Toshiba-cho, Saiwai-ku, Kawasaki, 212-8582, Japan
Email: yoshihiro.kikuchi@toshiba.co.jp
Yoshinori Matsui
Matsushita Electric Industrial Co., LTD.
1006, Kadoma, Kadoma-shi, Osaka, Japan
Email: matsui@drl.mei.co.jp
Toshiyuki Nomura
NEC Corporation
4-1-1,Miyazaki,Miyamae-ku,Kawasaki,JAPAN
Email: t-nomura@ccm.cl.nec.co.jp
Shigeru Fukunaga
Oki Electric Industry Co., Ltd.
1-2-27 Shiromi, Chuo-ku, Osaka 540-6025 Japan.
Email: fukunaga444@oki.co.jp
Hideaki Kimata
Nippon Telegraph and Telephone Corporation
1-1, Hikari-no-oka, Yokosuka-shi, Kanagawa, Japan
Email: kimata@nttvdt.hil.ntt.co.jp
Kikuchi/Nomura/Fukunaga/Kimata/Matsui [Page 19]
=0CRTP payload format for MPEG-4 Audio/Visual streams September =
2000=0D
Full Copyright Statement
"Copyright (C) The Internet Society (date). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph
are included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
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copyrights defined in the Internet Standards process must be
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English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
=0C
------=_NextPart_000_0243_01C02D25.C7F165E0--
From rem-conf-request@es.net Tue Oct 3 02:32:04 2000
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Date: Tue, 03 Oct 2000 08:20:06 +0200
From: Franceschini Guido
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
To: IETF AVT ,
4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
"'Yoshihiro Kikuchi'"
Message-id: <85077463E51D6A498C453073E167794039BD69@exc2k02.cselt.it>
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Change (2) means that significantly different packets are generated in case
of Short Video Headers (H263) between the original proposal and this new
version. Am I right, or is this just an editorial matter ?
Best regards
Guido
> ----------
> From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: Tuesday, October 03, 2000 3:36 AM
> To: IETF AVT; 4on2andIP-sys
> Cc: Yoshihiro Kikuchi
> Subject: draft text for MPEG-4 Audio/Visual RTP I-D revision
>
> <>
> Dear experts,
>
> Attached find please a draft text for the revision of MPEG-4 Audio/Visual
> RTP Internet-Draft.
>
> The revisions are mostly to reflect suggestions by the AVT chair. Most of
> them are editorial and there are some minor revisions and clarifications.
> The following is a summary of the changes:
>
> (1) MIME subtype names were changed to "video/MP4V-ES" for video and
> "audio/MP4A-LATM" for audio.
> (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
> video header mode was changed to "MAY" to "SHOULD".
> (3) Clarification of timestamp definition
> There was a suggestion not to describe the detailed definition of the
> video timestamp in the RTP spec. and change it as a reference to MPEG-4
> visual document, so as to keep consistency with the MPEG-4 visual. A
> description about the RTP timestamp in the RTCP SR packet was added.
> (4) SA/TTSI over LATM
> A description was added to restrict the LATM usage so as not to
> transmit
> SA data by LATM.
> (5) Other editorial and wording changes.
>
> Please let us know if there is comment on the attached draft. The authors
> need to submit it soon as an updated I-D and hopefully final before RFC.
>
> Best regards,
> Yoshihiro Kikuchi
> and authors of the Internet Draft
>
> ----
> Yoshihiro Kikuchi
>
> E-mail: kiku@eel.rdc.toshiba.co.jp
> Corporate Research and Development Center
> TOSHIBA Corporation
> TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
>
>
>
From rem-conf-request@es.net Tue Oct 3 05:20:29 2000
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Message-ID: <038001c02d18$6335a920$16c77885@ave>
From: "Yoshihiro Kikuchi"
To: "Franceschini Guido" ,
"IETF AVT" ,
"4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "Yoshihiro Kikuchi"
References: <85077463E51D6A498C453073E167794039BD69@exc2k02.cselt.it>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
Date: Tue, 3 Oct 2000 18:00:30 +0900
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Dear Guido,
Excerpt from the new text:
> When the short video header mode is used, RTP payload format for H.263
< (e.g., RFC 2190 or RFC 2429) SHOULD be used.
This means that it is RECOMMENDED to use the H.263 RTP payload format for
the short video header which is compatible with H.263 baseline. Regardless
of MAY or SHOULD, there is a possibility that the H.263 RTP payload format
may be used, although the possibility may be higher for SHOULD than MAY. If
you really don't want to use the H.263 RTP payload format in some
circumstances, you don't have to use it. The payload format is signaled
through the MIME type.
(IETF experts, please correct me if my understanding on the definition of
SHOULD is wrong.)
Best regards,
Yoshihiro
----- Original Message -----
From: Franceschini Guido
To: IETF AVT ; 4on2andIP-sys
<4on2andIP-sys@advent.ee.columbia.edu>; 'Yoshihiro Kikuchi'
Sent: Tuesday, October 03, 2000 3:20 PM
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
> Change (2) means that significantly different packets are generated in
case
> of Short Video Headers (H263) between the original proposal and this new
> version. Am I right, or is this just an editorial matter ?
>
> Best regards
> Guido
>
> > ----------
> > From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> > Sent: Tuesday, October 03, 2000 3:36 AM
> > To: IETF AVT; 4on2andIP-sys
> > Cc: Yoshihiro Kikuchi
> > Subject: draft text for MPEG-4 Audio/Visual RTP I-D revision
> >
> > <>
> > Dear experts,
> >
> > Attached find please a draft text for the revision of MPEG-4
Audio/Visual
> > RTP Internet-Draft.
> >
> > The revisions are mostly to reflect suggestions by the AVT chair. Most
of
> > them are editorial and there are some minor revisions and
clarifications.
> > The following is a summary of the changes:
> >
> > (1) MIME subtype names were changed to "video/MP4V-ES" for video and
> > "audio/MP4A-LATM" for audio.
> > (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
> > video header mode was changed to "MAY" to "SHOULD".
> > (3) Clarification of timestamp definition
> > There was a suggestion not to describe the detailed definition of the
> > video timestamp in the RTP spec. and change it as a reference to MPEG-4
> > visual document, so as to keep consistency with the MPEG-4 visual. A
> > description about the RTP timestamp in the RTCP SR packet was added.
> > (4) SA/TTSI over LATM
> > A description was added to restrict the LATM usage so as not to
> > transmit
> > SA data by LATM.
> > (5) Other editorial and wording changes.
> >
> > Please let us know if there is comment on the attached draft. The
authors
> > need to submit it soon as an updated I-D and hopefully final before RFC.
> >
> > Best regards,
> > Yoshihiro Kikuchi
> > and authors of the Internet Draft
> >
> > ----
> > Yoshihiro Kikuchi
> >
> > E-mail: kiku@eel.rdc.toshiba.co.jp
> > Corporate Research and Development Center
> > TOSHIBA Corporation
> > TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
> >
> >
> >
>
From rem-conf-request@es.net Tue Oct 3 12:36:17 2000
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Date: Tue, 03 Oct 2000 18:21:15 +0200
From: Franceschini Guido
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
To: Franceschini Guido ,
IETF AVT ,
4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
"'Yoshihiro Kikuchi'"
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Does it mean that a stream with short video header (H263) SHOULD use the
traditional Payload Format for H263, but could as well (not recommended) use
the new Payload Format for MPEG-4 Video, and that the choosen PF is notified
as usual (SDP etc) ?
If so, OK
Guido
> ----------
> From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: Tuesday, October 03, 2000 11:00 AM
> To: Franceschini Guido; IETF AVT; 4on2andIP-sys
> Cc: Yoshihiro Kikuchi
> Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
>
> Dear Guido,
>
> Excerpt from the new text:
>
> > When the short video header mode is used, RTP payload format for H.263
> < (e.g., RFC 2190 or RFC 2429) SHOULD be used.
>
> This means that it is RECOMMENDED to use the H.263 RTP payload format for
> the short video header which is compatible with H.263 baseline. Regardless
> of MAY or SHOULD, there is a possibility that the H.263 RTP payload format
> may be used, although the possibility may be higher for SHOULD than MAY.
> If
> you really don't want to use the H.263 RTP payload format in some
> circumstances, you don't have to use it. The payload format is signaled
> through the MIME type.
>
> (IETF experts, please correct me if my understanding on the definition of
> SHOULD is wrong.)
>
> Best regards,
> Yoshihiro
>
> ----- Original Message -----
> From: Franceschini Guido
> To: IETF AVT ; 4on2andIP-sys
> <4on2andIP-sys@advent.ee.columbia.edu>; 'Yoshihiro Kikuchi'
>
> Sent: Tuesday, October 03, 2000 3:20 PM
> Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
>
>
> > Change (2) means that significantly different packets are generated in
> case
> > of Short Video Headers (H263) between the original proposal and this new
> > version. Am I right, or is this just an editorial matter ?
> >
> > Best regards
> > Guido
> >
> > > ----------
> > > From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> > > Sent: Tuesday, October 03, 2000 3:36 AM
> > > To: IETF AVT; 4on2andIP-sys
> > > Cc: Yoshihiro Kikuchi
> > > Subject: draft text for MPEG-4 Audio/Visual RTP I-D revision
> > >
> > > <>
> > > Dear experts,
> > >
> > > Attached find please a draft text for the revision of MPEG-4
> Audio/Visual
> > > RTP Internet-Draft.
> > >
> > > The revisions are mostly to reflect suggestions by the AVT chair. Most
> of
> > > them are editorial and there are some minor revisions and
> clarifications.
> > > The following is a summary of the changes:
> > >
> > > (1) MIME subtype names were changed to "video/MP4V-ES" for video and
> > > "audio/MP4A-LATM" for audio.
> > > (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
> > > video header mode was changed to "MAY" to "SHOULD".
> > > (3) Clarification of timestamp definition
> > > There was a suggestion not to describe the detailed definition of
> the
> > > video timestamp in the RTP spec. and change it as a reference to
> MPEG-4
> > > visual document, so as to keep consistency with the MPEG-4 visual. A
> > > description about the RTP timestamp in the RTCP SR packet was added.
> > > (4) SA/TTSI over LATM
> > > A description was added to restrict the LATM usage so as not to
> > > transmit
> > > SA data by LATM.
> > > (5) Other editorial and wording changes.
> > >
> > > Please let us know if there is comment on the attached draft. The
> authors
> > > need to submit it soon as an updated I-D and hopefully final before
> RFC.
> > >
> > > Best regards,
> > > Yoshihiro Kikuchi
> > > and authors of the Internet Draft
> > >
> > > ----
> > > Yoshihiro Kikuchi
> > >
> > > E-mail: kiku@eel.rdc.toshiba.co.jp
> > > Corporate Research and Development Center
> > > TOSHIBA Corporation
> > > TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
> > >
> > >
> > >
> >
>
>
From rem-conf-request@es.net Tue Oct 3 12:45:21 2000
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From: Florissa Colina
Subject: 10/4 Studying and Modeling Real Audio Traffic -- John
Heidemann, USC Information Sciences Institute
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Berkeley Multimedia, Graphics and Interfaces Seminar
Studying and Modeling Real Audio Traffic
Wednesday October 4, 2000,
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall)
John Heidemann
USC Information Sciences Institute
The delivery of multimedia content on the Internet is rapidly growing in
importance. While there have been extensive studies on the growth and
effects of Hyper Text Transfer Protocol (HTTP) traffic used on the Web,
little work has been performed in analyzing streaming multimedia traffic.
Based on traces from broadcast.com, we will present a study of Real Audio
traffic. We will describe the characteristics of typical real audio flows
and compare them to web traffic. In addition, we observed consistencies in
audio traffic packet sizes and data rate patterns that may be useful as a
tool for identifying audio data flows. Finally, we will present recent
analysis of real audio traffic and show how it has guided our approaches to
model this traffic in simulation. This is joint work with Art Mena and
Kun-chan Lan at USC.
---------------
The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.
You can connect to the live RealNetworks broadcast at:
http://bmrc.berkeley.edu/bibs/cs298-5
You can also directly put in this url into the Real Player:
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm
To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html
To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr'). If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:
medium bit rate
video: 233.0.25.129/22334
audio: 233.0.25.2/22446
You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:
http://media2.bmrc.berkeley.edu/bibs/archive.cfm
Sponsored by the Berkeley Multimedia Research Center
From rem-conf-request@es.net Tue Oct 3 14:49:41 2000
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Reply-To: "Olivier Avaro"
From: "Olivier Avaro"
To: , "rem-conf" ,
"4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "CURET Dominique FTRD/DMI"
References: <0056890014249640000002L902*@MHS>
Subject: Re: Types of MPEG-4 streams over IP
Date: Tue, 3 Oct 2000 12:14:13 +0200
Organization: France Telecom R&D
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Dear Jan,
> I would like to verify my understanding on the types of MPEG-4 streams
that
> need tools for transport over IP. I summarized my understanding by the
> following list.
Here is the list you can find in V2 :
0x00 Forbidden
0x01 ObjectDescriptorStream (see 8.5)
0x02 ClockReferenceStream (see 10.2.519)
0x03 SceneDescriptionStream (see 9.2.1)
0x04 VisualStream
0x05 AudioStream
0x06 MPEG7Stream
0x07 IPMPStream (see 8.3.2)
0x08 ObjectContentInfoStream (see 8.4.2)
0x09 MPEGJStream
I don't have anything for Flexmux.
cu,
O.
From rem-conf-request@es.net Tue Oct 3 14:59:40 2000
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From: Kris Huber
To: Franceschini Guido
Cc: IETF AVT ,
4on2andIP-sys
<4on2andIP-sys@advent.ee.columbia.edu>,
"'Yoshihiro Kikuchi'"
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
Date: Tue, 3 Oct 2000 12:50:10 -0600
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Hello Guido,
I think you are right. It appears that the other alternative (change MAY to
SHALL or SHOULD) was taken rather than insertion of the MPEG-4 visual
semantics regarding picture rate in short header mode (which lacks
modulo_time_base, vop_time_increment, and vop_time_increment_resolution).
These were spelled out for short header in
draft-ietf-avt-rtp-mpeg4-es-04.txt based on a suggestion I made (e-mail of
9/8/00 "RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt -
timestamp of short header VOP").
Since then, I have learned that a more typical way (I think) that
hardware-assisted compression systems are used with PAL devices is with the
picture clock frequency driving the temporal_reference/TR field set to 25 Hz
rather than giving rounded values of a 30 Hz picture clock. However, my
suggestion does match baseline H.263 and MPEG-4 short-header mode, although
I think H.263 implementers have been a little loose in their interpretation,
perhaps with good reason.
Another opinion I respect is that a better partitioning of functions would
have placed all timing information in the systems part, as (I think) was
done for the audio part. But on the other hand you can't properly decode
video without some timing information (sequence number at least) due to
temporal prediction and B-VOPs.
I heard no discussion of the text I proposed for clarification of the
timestamp relationship to temporal_reference in short header mode after Gary
Sullivan's clarifications and suggestion that MAY be changed to SHALL. I'm
not on the IETF AVT mailing list or at other meetings where the discussion
probably occurred.
Regards,
Kris
-----Original Message-----
From: Franceschini Guido [mailto:Guido.Franceschini@cselt.it]
Sent: Tuesday, October 03, 2000 12:20 AM
To: IETF AVT; 4on2andIP-sys; 'Yoshihiro Kikuchi'
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
Change (2) means that significantly different packets are generated in case
of Short Video Headers (H263) between the original proposal and this new
version. Am I right, or is this just an editorial matter ?
Best regards
Guido
> ----------
> From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: Tuesday, October 03, 2000 3:36 AM
> To: IETF AVT; 4on2andIP-sys
> Cc: Yoshihiro Kikuchi
> Subject: draft text for MPEG-4 Audio/Visual RTP I-D revision
>
> <>
> Dear experts,
>
> Attached find please a draft text for the revision of MPEG-4 Audio/Visual
> RTP Internet-Draft.
>
> The revisions are mostly to reflect suggestions by the AVT chair. Most of
> them are editorial and there are some minor revisions and clarifications.
> The following is a summary of the changes:
>
> (1) MIME subtype names were changed to "video/MP4V-ES" for video and
> "audio/MP4A-LATM" for audio.
> (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
> video header mode was changed to "MAY" to "SHOULD".
> (3) Clarification of timestamp definition
> There was a suggestion not to describe the detailed definition of the
> video timestamp in the RTP spec. and change it as a reference to MPEG-4
> visual document, so as to keep consistency with the MPEG-4 visual. A
> description about the RTP timestamp in the RTCP SR packet was added.
> (4) SA/TTSI over LATM
> A description was added to restrict the LATM usage so as not to
> transmit
> SA data by LATM.
> (5) Other editorial and wording changes.
>
> Please let us know if there is comment on the attached draft. The authors
> need to submit it soon as an updated I-D and hopefully final before RFC.
>
> Best regards,
> Yoshihiro Kikuchi
> and authors of the Internet Draft
>
> ----
> Yoshihiro Kikuchi
>
> E-mail: kiku@eel.rdc.toshiba.co.jp
> Corporate Research and Development Center
> TOSHIBA Corporation
> TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
>
>
>
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------=_NextPart_000_003D_01C02DDA.EC7B5500--
From rem-conf-request@es.net Wed Oct 4 01:24:20 2000
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To: Yoshihiro Kikuchi
cc: Franceschini Guido ,
IETF AVT ,
4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
In-Reply-To: Message from Yoshihiro Kikuchi
of "Tue, 03 Oct 2000 18:00:30 +0900." <038001c02d18$6335a920$16c77885@ave>
Date: Wed, 04 Oct 2000 00:44:46 -0400
From: Colin Perkins
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--> Yoshihiro Kikuchi writes:
>> When the short video header mode is used, RTP payload format for H.263
>< (e.g., RFC 2190 or RFC 2429) SHOULD be used.
You should specify which of RFC 2190 or RFC 2429 is intended.
>This means that it is RECOMMENDED to use the H.263 RTP payload format for
>the short video header which is compatible with H.263 baseline. Regardless
>of MAY or SHOULD, there is a possibility that the H.263 RTP payload format
>may be used, although the possibility may be higher for SHOULD than MAY. If
>you really don't want to use the H.263 RTP payload format in some
>circumstances, you don't have to use it. The payload format is signaled
>through the MIME type.
>
>(IETF experts, please correct me if my understanding on the definition of
>SHOULD is wrong.)
SHOULD = "there may exist valid reasons in particular circumstances to
ignore a particular item, but the full implications must be
understood and carefully weighed before choosing a different
course."
MAY = "an item is truly optional"
[see RFC 2119 for the full wording]
Colin
From rem-conf-request@es.net Wed Oct 4 01:24:41 2000
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To: Kris Huber
cc: Franceschini Guido ,
IETF AVT ,
4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
"'Yoshihiro Kikuchi'"
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
In-Reply-To: Message from Kris Huber
of "Tue, 03 Oct 2000 12:50:10 MDT." <6E031E06378BD311AEF20090273CE1BA81E6EF@el-postino.s-vision.com>
Date: Wed, 04 Oct 2000 00:50:31 -0400
From: Colin Perkins
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Precedence: list
A brief note - the issue of the timestamp is one area where I had some
concerns with this draft, but unfortunately I haven't had enough spare
cycles this past week to follow up on them. This is likely just my mis-
understanding, but input from those who know both MPEG4 and RTP, to ensure
that this is both accurate and consistent with the two timing models, would
be greatly appreciated.
Colin
--> Kris Huber writes:
>Hello Guido,
>
>I think you are right. It appears that the other alternative (change MAY to
>SHALL or SHOULD) was taken rather than insertion of the MPEG-4 visual
>semantics regarding picture rate in short header mode (which lacks
>modulo_time_base, vop_time_increment, and vop_time_increment_resolution).
>These were spelled out for short header in
>draft-ietf-avt-rtp-mpeg4-es-04.txt based on a suggestion I made (e-mail of
>9/8/00 "RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt -
>timestamp of short header VOP").
>
>Since then, I have learned that a more typical way (I think) that
>hardware-assisted compression systems are used with PAL devices is with the
>picture clock frequency driving the temporal_reference/TR field set to 25 Hz
>rather than giving rounded values of a 30 Hz picture clock. However, my
>suggestion does match baseline H.263 and MPEG-4 short-header mode, although
>I think H.263 implementers have been a little loose in their interpretation,
>perhaps with good reason.
>
>Another opinion I respect is that a better partitioning of functions would
>have placed all timing information in the systems part, as (I think) was
>done for the audio part. But on the other hand you can't properly decode
>video without some timing information (sequence number at least) due to
>temporal prediction and B-VOPs.
>
>I heard no discussion of the text I proposed for clarification of the
>timestamp relationship to temporal_reference in short header mode after Gary
>Sullivan's clarifications and suggestion that MAY be changed to SHALL. I'm
>not on the IETF AVT mailing list or at other meetings where the discussion
>probably occurred.
>
>Regards,
>Kris
>
>-----Original Message-----
>From: Franceschini Guido [mailto:Guido.Franceschini@cselt.it]
>Sent: Tuesday, October 03, 2000 12:20 AM
>To: IETF AVT; 4on2andIP-sys; 'Yoshihiro Kikuchi'
>Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
>
>
>Change (2) means that significantly different packets are generated in case
>of Short Video Headers (H263) between the original proposal and this new
>version. Am I right, or is this just an editorial matter ?
>
>Best regards
>Guido
>
>> ----------
>> From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
>> Sent: Tuesday, October 03, 2000 3:36 AM
>> To: IETF AVT; 4on2andIP-sys
>> Cc: Yoshihiro Kikuchi
>> Subject: draft text for MPEG-4 Audio/Visual RTP I-D revision
>>
>> <>
>> Dear experts,
>>
>> Attached find please a draft text for the revision of MPEG-4 Audio/Visual
>> RTP Internet-Draft.
>>
>> The revisions are mostly to reflect suggestions by the AVT chair. Most of
>> them are editorial and there are some minor revisions and clarifications.
>> The following is a summary of the changes:
>>
>> (1) MIME subtype names were changed to "video/MP4V-ES" for video and
>> "audio/MP4A-LATM" for audio.
>> (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
>> video header mode was changed to "MAY" to "SHOULD".
>> (3) Clarification of timestamp definition
>> There was a suggestion not to describe the detailed definition of the
>> video timestamp in the RTP spec. and change it as a reference to MPEG-4
>> visual document, so as to keep consistency with the MPEG-4 visual. A
>> description about the RTP timestamp in the RTCP SR packet was added.
>> (4) SA/TTSI over LATM
>> A description was added to restrict the LATM usage so as not to
>> transmit
>> SA data by LATM.
>> (5) Other editorial and wording changes.
>>
>> Please let us know if there is comment on the attached draft. The authors
>> need to submit it soon as an updated I-D and hopefully final before RFC.
>>
>> Best regards,
>> Yoshihiro Kikuchi
>> and authors of the Internet Draft
>>
>> ----
>> Yoshihiro Kikuchi
>>
>> E-mail: kiku@eel.rdc.toshiba.co.jp
>> Corporate Research and Development Center
>> TOSHIBA Corporation
>> TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
>>
>>
>>
From rem-conf-request@es.net Wed Oct 4 04:41:38 2000
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From: CURET Dominique FTRD/DMI/REN
To: rem-conf ,
4on2andIP-sys
<4on2andIP-sys@advent.ee.columbia.edu>
Cc: CURET Dominique FTRD/DMI ,
jan.vandermeer@philips.com,
"'Olivier Avaro'"
Subject: RE: Types of MPEG-4 streams over IP
Date: Wed, 4 Oct 2000 10:35:35 +0200
MIME-Version: 1.0
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dear all,
*********Jan proposed:
- ObjectDescriptorStream
- ClockReferenceStream (is this needed over IP ?)
- SceneDescriptionStream
- VisualStream (with and without use of MPEG-4 Systems)
- AudioStream (with and without use of MPEG-4 Systems)
- MPEG-7Stream
- IPMPStream
- ObjectContentInfoStream
- FlexMux stream
- MP4 file
*****Olivier found in V2 :
0x00 Forbidden
0x01 ObjectDescriptorStream (see 8.5)
0x02 ClockReferenceStream (see 10.2.519)
0x03 SceneDescriptionStream (see 9.2.1)
0x04 VisualStream
0x05 AudioStream
0x06 MPEG7Stream
0x07 IPMPStream (see 8.3.2)
0x08 ObjectContentInfoStream (see 8.4.2)
0x09 MPEGJStream (or MPEG-J Streams as Viswanathan said)
*******so, to my understanding, at least:
3 streams are missing: Flexmux steeams, SL steams & MP4 file streams
******about the muxcode table in band/out of band:
Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
**it certainly needs to be delivered.
Pierre showed how it is possible to deliver it, accurately in time, by
out of band means
**But is 'out of band' compatible with a multicast or broadcast scenario.
**So, I think agree with Jan that an inband means is necessary.
regards
Dominique
_________________________________________
> Dominique Curet
> France Télécom R&D - DMI/PIA
> % + 33 (0)2 99 12 40 05 Fax : + 33 (0)2 99 12 40 98
> ) CCETT B.P. 59 4, rue du Clos Courtel
> 35512 Cesson-Sevigne Cedex 09 FRANCE
> mailto:dominique.curet@rd.francetelecom.fr
_________________________________________
From rem-conf-request@es.net Wed Oct 4 05:44:22 2000
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To:
Cc: , <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Types of MPEG-4 streams over IP
Message-ID: <0056890014577469000002L992*@MHS>
Date: Wed, 4 Oct 2000 11:38:00 +0200
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Dominique, all,
Dominique wrote:
*******so, to my understanding, at least:
3 streams are missing: Flexmux steeams, SL steams & MP4 file streams
A few comments :
1) I don't think that SL stream is an additional stream type; in my view SL
streams are included in the stream types that use MPEG-4 systems.
2) The wording MP4 file stream is confusing; I don't think there is a need
to "stream" an MP4 file; instead only a mime type is needed for file
transfer.
A question : for which streams we need to specify use of RTP and SDP ? In my
understanding for :
- VisualStream (with and without use of MPEG-4 Systems)
- AudioStream (with and without use of MPEG-4 Systems)
- FlexMux stream
Can we assume that the following streams are probably most efficiently
contained in a FlexMux stream, and consequently that is no need to specify
use of RTP and SDP for those streams:
- ObjectDescriptorStream
- ClockReferenceStream
- SceneDescriptionStream
- MPEG-7Stream
- IPMPStream
- ObjectContentInfoStream
- MPEGJStream
Regards,
Jan
-----Original Message-----
From: dominique.curet@rd.francetelecom.fr
[mailto:dominique.curet@rd.francetelecom.fr]
Sent: woensdag, 04 oktober, 2000 10:39
To: 4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Cc: olivier.avaro@francetelecom.fr; Jan vanderMeer;
dominique.curet@francetelecom.fr
Subject: RE: Types of MPEG-4 streams over IP
dear all,
*********Jan proposed:
- ObjectDescriptorStream
- ClockReferenceStream (is this needed over IP ?)
- SceneDescriptionStream
- VisualStream (with and without use of MPEG-4 Systems)
- AudioStream (with and without use of MPEG-4 Systems)
- MPEG-7Stream
- IPMPStream
- ObjectContentInfoStream
- FlexMux stream
- MP4 file
*****Olivier found in V2 :
0x00 Forbidden
0x01 ObjectDescriptorStream (see 8.5)
0x02 ClockReferenceStream (see 10.2.519)
0x03 SceneDescriptionStream (see 9.2.1)
0x04 VisualStream
0x05 AudioStream
0x06 MPEG7Stream
0x07 IPMPStream (see 8.3.2)
0x08 ObjectContentInfoStream (see 8.4.2)
0x09 MPEGJStream (or MPEG-J Streams as Viswanathan said)
*******so, to my understanding, at least:
3 streams are missing: Flexmux steeams, SL steams & MP4 file streams
******about the muxcode table in band/out of band:
Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
**it certainly needs to be delivered.
Pierre showed how it is possible to deliver it, accurately in time, by
out of band means
**But is 'out of band' compatible with a multicast or broadcast scenario.
**So, I think agree with Jan that an inband means is necessary.
regards
Dominique
_________________________________________
> Dominique Curet
> France Télécom R&D - DMI/PIA
> % + 33 (0)2 99 12 40 05 Fax : + 33 (0)2 99 12 40 98
> ) CCETT B.P. 59 4, rue du Clos Courtel
> 35512 Cesson-Sevigne Cedex 09 FRANCE
> mailto:dominique.curet@rd.francetelecom.fr
_________________________________________
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Message-ID: <067f01c02df4$61442560$cc51c485@eel.rdc.toshiba.co.jp>
From: "Yoshihiro Kikuchi"
To: "Colin Perkins"
Cc: "IETF AVT" ,
"Yoshihiro Kikuchi"
References: <200010040444.AAA09294@purple.east.isi.edu>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
Date: Wed, 4 Oct 2000 20:15:19 +0900
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Dear Colin, all,
> >> When the short video header mode is used, RTP payload format for
H.263
> >< (e.g., RFC 2190 or RFC 2429) SHOULD be used.
>
> You should specify which of RFC 2190 or RFC 2429 is intended.
>
Can I ask suggestions from ITU-T H.263 experts on this selection of RFCs?
Or, if ITU-T does not have any preference, I'm not sure if it is really
necessary to change the text to decide this in "MPEG-4" RTP payload spec.
Best regards,
Yoshihiro
----
Yoshihiro Kikuchi
E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
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From: "Young-Kwon LIM"
To: ,
Cc: , <4on2andIP-sys@advent.ee.columbia.edu>
References: <0056890014577469000002L992*@MHS>
Subject: Re: Types of MPEG-4 streams over IP
Date: Wed, 4 Oct 2000 22:33:20 +0900
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Dear Jan, Dominique and all,
>Dominique wrote:
>*******so, to my understanding, at least:
> 3 streams are missing: Flexmux steeams, SL steams & MP4 file streams
>A few comments :
>1) I don't think that SL stream is an additional stream type; in my view SL
>streams are included in the stream types that use MPEG-4 systems.
==> In my understanding, the streamtype identify the content of the bitstream. If you want to identify the format of the stream it is useful to differentiate SL stream, FlexMux stream and ES of each media streams and etc. But if you want to differentiate the content of the streams, then there is no need to differentiate SL streams from others. What is your intention for stream type discussions?
>2) The wording MP4 file stream is confusing; I don't think there is a need
>to "stream" an MP4 file; instead only a mime type is needed for file
>transfer.
==> I agree with this. In any case there is no MP4 stream.
>A question : for which streams we need to specify use of RTP and SDP ? In my
>understanding for :
>- VisualStream (with and without use of MPEG-4 Systems)
>- AudioStream (with and without use of MPEG-4 Systems)
>- FlexMux stream
>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream
In general we are not allowed to assume this condition because there is no normative limitation to this even though practically it is valid. Do you want to put a normative usage of FlexMux for those cases? And I think IPMP streams should not be multiplexed with other streams.
Sincerely,
Young.
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To:
Cc: , <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Types of MPEG-4 streams over IP
Message-ID: <0056890014590674000002L942*@MHS>
Date: Wed, 4 Oct 2000 16:26:02 +0200
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Dear Young,
You wrote :
>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream
In general we are not allowed to assume this condition because there is no
normative limitation to this even though practically it is valid. Do you
want to put a normative usage of FlexMux for those cases? And I think IPMP
streams should not be multiplexed with other streams.
I am asking just for getting opinions; I am not sure how practical it is to
have a separate session for each of the above streams. You may be right that
IPMP streams should not be multiplexed with other streams, but for that case
FlexMux could offer a useful multiplex solution for multiple IPMP messages
within an RTP payload.
Kind regards,
Jan
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Date: Wed, 04 Oct 2000 17:05:26 +0200
From: Krister Svanbro
Organization: Ericsson Research
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To: Qiaobing Xie ,
Magnus Westerlund
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Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
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Qiaobing, Magnus and others,
I am Krister Svanbro and work mainly in the ROHC WG.
I noticed that the ROHC WG name was mentioned below and
thought I perhaps could bring some information from a ROHC point of
view, which might help somewhat in this issue.
The ROHC WG has spent year 2000 on mainly developing a robust header
compression scheme for RTP/UDP/IP. We have also created a framework
based on compression profiles to enable development of other
compression profiles such as compression of TCP/IP the coming year.
The term robust origins from that fact we realised that we need to
design header compression schemes more robust to make them cope with
the relative large amount of packet loss that occur in e.g. cellular
systems. That does not, however, preclude in any way the usage of
the ROHC scheme in other types of (possible error prone)
environments.
We have to this date not looked upon compression of RTP
payloadformats and we do not plan to that either. I also think that
there should not be any need for compression of payloadformats. They
should be as efficient as possible from the beginning. AMR is
certainly a highly interesting codec in usage together with ROHC
to enable VoIP to a mobile, hence it is important that the
AMR format is efficient and does not waste any bits.
It becomes kind of strange if we in the ROHC WG maximize compression
efficiency with rather sofisticated methods to enable to go down
to a compressed header size of 1 byte for the IP/UDP/RTP header, and
then is 1 byte extra used for the RTP payloadformat only.
Hence, I would like to encourage you to keep efficiency in mind
when developing this payload format. I look forward to see the
results of this work.
Cheers,
/ Krister
Qiaobing Xie wrote:
>
> Hi, Magnus,
>
> See my comments below....
>
> Magnus Westerlund wrote:
> >
> > Hello,
> >
> > The payload format is designed with IP networks in mind. The problem we see is
> > that IP is transported over a wide range of different links with different
> > behavior, e.g. different radio and wireline links. Radio and wire do not have
> > the same error characteristics. AMR will be widly used over wireless links and
> > on these links bandwidth efficiency is extremely important. This lead to the
> > bitwise sorting. Octet sorting needs octet alignement for each block of the
> > payload, i.e. padding at the end of each block. Which we considered to be to
> > expensive.
>
> This further convinces me that the current design presented in the AMR
> draft is a radio-link-centric design, which is wrong in my opinion,
> because:
>
> 1) bandwidth efficiency over wireless links is more of a concern for
> ROHC working group. Architecturally, it should be delt with below IP by
> header compression/stripping, as being worked upon in rohc. It doesn't
> belong to RTP payload layer (unless it is something facilitating the
> underlying header compression/stripping).
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To:
Subject: Seminar: Catch the Vision of IP Multicast
Date: Wed, 4 Oct 2000 10:07:05 -0500
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From: "Bogdan Ghita"
To:
Subject: Routing for UDP/real-time vs TCP
Date: Wed, 4 Oct 2000 16:36:05 +0100
Organization: NRG
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Dear all
I have a question related to general routing issues:
I am looking into measuring RTT for a specific path using 2 different =
methods: TCP acknowledgements and RTCP NTP/LSR/DLSR fields.
Is there any specification regarding different handling by the servers =
of packets that carry TCP segments vs packets carrying UDP datagrams? I =
had several discussions lately about how some/all routers tend to drop =
first ICMP traffic, then TCP traffic and, at the end, UDP traffic, as =
the first category is not critical and the second can recover. If so, it =
would mean that there are several different transport performance rates =
for the same path, depending on the type of traffic being sent. Are the =
assumption/conclusion correct? Could you point me to some materials that =
detail this matter?
Thank you very much
Best regards
Bogdan Ghita
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Dear all
I have a question related to general =
routing=20
issues:
I am looking into measuring RTT for a =
specific path=20
using 2 different methods: TCP acknowledgements and RTCP NTP/LSR/DLSR=20
fields.
Is there any =
specification regarding different=20
handling by the servers of packets that carry TCP segments vs packets =
carrying=20
UDP datagrams? I had several discussions lately about how some/all =
routers tend=20
to drop first ICMP traffic, then TCP traffic and, at the end, UDP =
traffic, as=20
the first category is not critical and the second can recover. If so, it =
would=20
mean that there are several different transport performance rates for =
the same=20
path, depending on the type of traffic being sent. Are the =
assumption/conclusion=20
correct? Could you point me to some materials that detail this=20
matter?
Thank you very much
Best regards
Bogdan Ghita
------=_NextPart_000_0049_01C02E21.30CFFF30--
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Date: Wed, 04 Oct 2000 17:39:19 +0200
To: "Yoshihiro Kikuchi"
From: Stephan Wenger
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
Cc: "Colin Perkins" , "IETF AVT" ,
"Yoshihiro Kikuchi"
In-Reply-To: <067f01c02df4$61442560$cc51c485@eel.rdc.toshiba.co.jp>
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Dear Yoshihiro, Group,
To my knowledge the only reason why RFC2190 is not yet OBSOLETE is
that some early H.323 products still use it. Interoperability with such
products is probably not your main concern. So I would suggest to select
RFC2429 only.
Stephan
At 08:15 PM 10/4/2000 +0900, Yoshihiro Kikuchi wrote:
>Dear Colin, all,
>
> > >> When the short video header mode is used, RTP payload format for
>H.263
> > >< (e.g., RFC 2190 or RFC 2429) SHOULD be used.
> >
> > You should specify which of RFC 2190 or RFC 2429 is intended.
> >
>Can I ask suggestions from ITU-T H.263 experts on this selection of RFCs?
>Or, if ITU-T does not have any preference, I'm not sure if it is really
>necessary to change the text to decide this in "MPEG-4" RTP payload spec.
>
>Best regards,
>Yoshihiro
>
>----
> Yoshihiro Kikuchi
>
>E-mail: kiku@eel.rdc.toshiba.co.jp
>Corporate Research and Development Center
>TOSHIBA Corporation
>TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
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Date: Wed, 4 Oct 2000 09:17:04 -0700
To: jan.vandermeer@philips.com
From: Dave Singer
Subject: RE: Types of MPEG-4 streams over IP
Cc: dominique.curet@rd.francetelecom.fr, rem-conf@es.net,
4on2andIP-sys@advent.ee.columbia.edu
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>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream
I would be strongly opposed to mandating the use of flexmux for any
stream. What is wrong with Carsten et al.'s sync layer mapping into
RTP?
--
David Singer
Apple Computer/QuickTime
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Date: Wed, 4 Oct 2000 09:15:55 -0700
To: CURET Dominique FTRD/DMI/REN
From: Dave Singer
Subject: RE: Types of MPEG-4 streams over IP
Cc: rem-conf ,
4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
CURET Dominique FTRD/DMI ,
jan.vandermeer@philips.com,
"'Olivier Avaro'"
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>*******so, to my understanding, at least:
> 3 streams are missing: Flexmux steeams, SL steams & MP4 file streams
SL is not a stream type. It's the layer that carries streams. MP4
files are not directly streamed; they may be downloaded, but that
requires a MIME type, not a stream type.
>******about the muxcode table in band/out of band:
>Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
>**it certainly needs to be delivered.
>
>Pierre showed how it is possible to deliver it, accurately in time, by
>out of band means
>
>**But is 'out of band' compatible with a multicast or broadcast scenario.
>**So, I think agree with Jan that an inband means is necessary.
of course it is. my framework proposal showed how. in a multicast
or broadcast, there is always some information that is given to the
terminal to enable it to open the streams. in IETF world, that is an
SDP file. I showed how to put the IOD and the flexmux info into the
SDP file.
--
David Singer
Apple Computer/QuickTime
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From: CURET Dominique FTRD/DMI/REN
To: "'Dave Singer'" , jan.vandermeer@philips.com
Cc: CURET Dominique FTRD/DMI/REN ,
rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP
Date: Wed, 4 Oct 2000 19:15:11 +0200
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Dear all, dear Dave,
Dave said:
"I would be strongly opposed to mandating the use of flexmux for any
stream."
answer:
The applications that you have in mind don't seem to need multiplexing
at the application level. Fine with me.
When some other applications find this approach interesting, our job
is to give solutions to them. There should not be any technical
fundamentalism.
Dominique
_________________________________________
> Dominique Curet
> France Télécom R&D - DMI/PIA
> % + 33 (0)2 99 12 40 05 Fax : + 33 (0)2 99 12 40 98
> ) CCETT B.P. 59 4, rue du Clos Courtel
> 35512 Cesson-Sevigne Cedex 09 FRANCE
> mailto:dominique.curet@rd.francetelecom.fr
_________________________________________
-----Message d'origine-----
De: Dave Singer [mailto:singer@apple.com]
Date: mercredi 4 octobre 2000 18:17
À: jan.vandermeer@philips.com
Cc: dominique.curet@rd.francetelecom.fr; rem-conf@es.net;
4on2andIP-sys@advent.ee.columbia.edu
Objet: RE: Types of MPEG-4 streams over IP
>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream
I would be strongly opposed to mandating the use of flexmux for any
stream. What is wrong with Carsten et al.'s sync layer mapping into
RTP?
--
David Singer
Apple Computer/QuickTime
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Date: Wed, 04 Oct 2000 13:28:24 -0400
From: "Narasimhan, Sam (SD-EX)"
Subject: RE: Types of MPEG-4 streams over IP
To: "'Dave Singer'" , jan.vandermeer@philips.com
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4on2andIP-sys@advent.ee.columbia.edu
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Dave,
The MPEG requirements document asks for a payload
mapping scheme to carry flexmux streams. This is possible in
MPEG-2 transport and must be possible in RTP. We need to
define a mapping scheme which will be normative for those
applications that want to transmit flexmux streams over
RTP. I will strongly object to a mapping scheme that
excludes or precludes the carriage of flexmux streams
over RTP.
Regards
Sam Narasimhan
Motorola
-----Original Message-----
From: Dave Singer [mailto:singer@apple.com]
Sent: Wednesday, October 04, 2000 9:17 AM
To: jan.vandermeer@philips.com
Cc: dominique.curet@rd.francetelecom.fr; rem-conf@es.net;
4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP
>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream
I would be strongly opposed to mandating the use of flexmux for any
stream. What is wrong with Carsten et al.'s sync layer mapping into
RTP?
--
David Singer
Apple Computer/QuickTime
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Date: Wed, 04 Oct 2000 23:23:19 +0200
To: Krister Svanbro
From: Mikael Degermark
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
Cc: Qiaobing Xie ,
Magnus Westerlund , rem-conf@es.net
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ROHC WG co-chair speaking.
I have a couple of comments to this letter:
>Qiaobing Xie wrote:
>>
>> Hi, Magnus,
>>
>> See my comments below....
>>
>> Magnus Westerlund wrote:
>> >
>> > Hello,
>> >
>> > The payload format is designed with IP networks in mind. The problem we see is
>> > that IP is transported over a wide range of different links with different
>> > behavior, e.g. different radio and wireline links. Radio and wire do not have
>> > the same error characteristics. AMR will be widly used over wireless links and
>> > on these links bandwidth efficiency is extremely important. This lead to the
>> > bitwise sorting. Octet sorting needs octet alignement for each block of the
>> > payload, i.e. padding at the end of each block. Which we considered to be to
>> > expensive.
>>
>> This further convinces me that the current design presented in the AMR
>> draft is a radio-link-centric design, which is wrong in my opinion,
>> because:
>>
>> 1) bandwidth efficiency over wireless links is more of a concern for
>> ROHC working group.
That statement is too general. ROHC deals with the efficient transmission of
the RTP/UDP/IP header. The payload format is NOT a concern of the ROHC WG.
If a payload format is to be efficient over a wireless link, it must be made so by
others than ROHC.
>> Architecturally, it should be delt with below IP by
>> header compression/stripping, as being worked upon in rohc. It doesn't
>> belong to RTP payload layer (unless it is something facilitating the
>> underlying header compression/stripping).
Architecturally speaking (and here I remove my ROHC chair hat), bandwidth
efficiency over wireless links will best be obtained by
a) having an efficiently compressed payload, which might be somewhat resilient to
errors introduced by the wireless link.
b) compressing the RTP/UDP/IP header robustly and efficiently.
ROHC deals with b). Others, perhaps you people, need to deal with a).
The best result will be obtained if a) is done close to the application,
i.e., above the IP layer. a) is certainly not the task of ROHC, and neither should
it be.
The IETF should not worry ONLY about wireless, of course, but it certainly
seems appropriate to ensure that things work over wireless as well as over
the rest of the currently largely wired Internet.
Cheers,
Mikael Degermark
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From: Qiaobing Xie
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Mikael and Krister (and other ROHCers on the list):
Ok, I know I was getting myself in trouble when I mentioned the
"dreadful" word ROHC :-) :-)
Mikael Degermark wrote:
...snip...
> >> Architecturally, it should be delt with below IP by
> >> header compression/stripping, as being worked upon in rohc. It doesn't
> >> belong to RTP payload layer (unless it is something facilitating the
> >> underlying header compression/stripping).
>
> Architecturally speaking (and here I remove my ROHC chair hat), bandwidth
> efficiency over wireless links will best be obtained by
>
> a) having an efficiently compressed payload, which might be somewhat resilient to
> errors introduced by the wireless link.
>
> b) compressing the RTP/UDP/IP header robustly and efficiently.
>
> ROHC deals with b). Others, perhaps you people, need to deal with a).
> The best result will be obtained if a) is done close to the application,
> i.e., above the IP layer. a) is certainly not the task of ROHC, and neither should
> it be.
I think we are in "violent" agreement here :)
The point I was trying to make is, when you are looking at optimizing
the bandwidth efficiency over a *link* with a unique error
characteristics, the framework/approach used by ROHC is more
appropriate, which is to look for header compression/stripping solutions
below the IP layer. This matches your b) above.
But if you are designing an RTP payload format, which is meant to be
sent end-to-end over a routable IP connection, it does not make much
sense to optimize the payload format design according to the error
characteristics of any one type of the link which, after all, may or may
not be present in the connection.
I have not problem with a) and I think it is in deed the job of AVT WG.
No doubt that it is universally desirable to design any protocol message
as compact and as compressed as possible (as Krister said: not to waste
a bit) But in reality, you often have to make trade-off between saving
a bit in the message and the associated processing cost, especially when
the processing is done in a higher stack layer. The error resilience
part (or the lack of it) in the current AMR draft is exactly one of my
big concerns as I talked about in my first email.
Cheers,
-Qiaobing
>
> The IETF should not worry ONLY about wireless, of course, but it certainly
> seems appropriate to ensure that things work over wireless as well as over
> the rest of the currently largely wired Internet.
>
> Cheers,
>
> Mikael Degermark
From rem-conf-request@es.net Wed Oct 4 19:44:32 2000
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Date: Wed, 4 Oct 2000 16:36:23 -0700
To: CURET Dominique FTRD/DMI/REN
From: Dave Singer
Subject: RE: Types of MPEG-4 streams over IP
Cc: jan.vandermeer@philips.com,
CURET Dominique FTRD/DMI/REN ,
rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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At 7:15 PM +0200 10/4/00, CURET Dominique FTRD/DMI/REN wrote:
>Dear all, dear Dave,
>
>Dave said:
>"I would be strongly opposed to mandating the use of flexmux for any
>stream."
>
>answer:
>The applications that you have in mind don't seem to need multiplexing
>at the application level. Fine with me.
>When some other applications find this approach interesting, our job
>is to give solutions to them. There should not be any technical
>fundamentalism.
and Sam said:
> The MPEG requirements document asks for a payload
>mapping scheme to carry flexmux streams. This is possible in
>MPEG-2 transport and must be possible in RTP. We need to
>define a mapping scheme which will be normative for those
>applications that want to transmit flexmux streams over
>RTP. I will strongly object to a mapping scheme that
>excludes or precludes the carriage of flexmux streams
>over RTP.
I am not opposed to allowing the use of flexmux. The proposal was
that certain stream types (e.g. MPEG-7) could *only* be carried in
flexmux. Flexmux is not always suitable; it should be possible to
carry a single MPEG-4 stream in a simple RTP stream, without
requiring flexmux.
Indeed, I am not being a fundamentalist; though I personally do not
see much need for flexmux, I am happy that it be allowed. I would
like a technical solution for MPEG-4 which does not *mandate* it,
that's all.
--
David Singer
Apple Computer/QuickTime
From rem-conf-request@es.net Wed Oct 4 21:57:06 2000
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From: SIGNES Julien / ENVIVIO
To: "'Y. Matsui'" , 4on2andIP-sys@advent.ee.columbia.edu
Cc: rem-conf@es.net
Subject: RE: Types of MPEG-4 streams over IP
Date: Wed, 4 Oct 2000 18:48:52 -0700
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Hello,
> -----Original Message-----
> From: Y. Matsui [mailto:matsui@drl.mei.co.jp]
> Sent: Wednesday, October 04, 2000 6:42 PM
> To: 4on2andIP-sys@advent.ee.columbia.edu
> Cc: rem-conf@es.net
> Subject: Re: Types of MPEG-4 streams over IP
>
>
> Dear Jan and all,
>
> >Can we assume that the following streams are probably most
> efficiently
> >contained in a FlexMux stream, and consequently that is no
> need to specify
> >use of RTP and SDP for those streams:
> >
> >- ObjectDescriptorStream
> >- ClockReferenceStream
> >- SceneDescriptionStream
> >- MPEG-7Stream
> >- IPMPStream
> >- ObjectContentInfoStream
> >- MPEGJStream
>
> In my opinion, those streams should be carried on a reliable channel
> rather than RTP.
>
I agree in general they should be fully reliable channels carrying those,as
well as BIFS comands. However, note that a lot of streaming those days is
actually done by RTP over TCP.
> What I am not sure is the scene description stream.
> There are two kind of the scene description stream, one is BIFS-update
> and another is BIFS-Anim. In my understanding, BIFS-Anim should
> be streamed like audio/visual streams.
I agree.
>
> Best regards,
> Yoshinori Matsui
>
Julien
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RE: Types of MPEG-4 streams over IP
Hello,
> -----Original Message-----
> From: Y. Matsui [mailto:matsui@drl.mei.co.jp]
> Sent: Wednesday, October 04, 2000 6:42 =
PM
> To: 4on2andIP-sys@advent.ee.columbia.edu
> Cc: rem-conf@es.net
> Subject: Re: Types of MPEG-4 streams over =
IP
>
>
> Dear Jan and all,
>
> >Can we assume that the following streams =
are probably most
> efficiently
> >contained in a FlexMux stream, and =
consequently that is no
> need to specify
> >use of RTP and SDP for those =
streams:
> >
> >- ObjectDescriptorStream
> >- ClockReferenceStream
> >- SceneDescriptionStream
> >- MPEG-7Stream
> >- IPMPStream
> >- ObjectContentInfoStream
> >- MPEGJStream
>
> In my opinion, those streams should be carried =
on a reliable channel
> rather than RTP.
>
I agree in general they should be fully reliable =
channels carrying those,as well as BIFS comands. However, note that a =
lot of streaming those days is actually done by RTP over =
TCP.
> What I am not sure is the scene description =
stream.
> There are two kind of the scene description =
stream, one is BIFS-update
> and another is BIFS-Anim. In my =
understanding, BIFS-Anim should
> be streamed like audio/visual streams.
I agree.
>
> Best regards,
> Yoshinori Matsui
>
Julien
------_=_NextPart_001_01C02E6E.69E69BD8--
From rem-conf-request@es.net Wed Oct 4 21:57:21 2000
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From: "Y. Matsui"
To: <4on2andIP-sys@advent.ee.columbia.edu>
Cc:
References: <0056890014577469000002L992*@MHS>
Subject: Re: Types of MPEG-4 streams over IP
Date: Thu, 5 Oct 2000 10:41:35 +0900
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Dear Jan and all,
>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream
In my opinion, those streams should be carried on a reliable channel
rather than RTP.
What I am not sure is the scene description stream.
There are two kind of the scene description stream, one is BIFS-update
and another is BIFS-Anim. In my understanding, BIFS-Anim should
be streamed like audio/visual streams.
Best regards,
Yoshinori Matsui
From rem-conf-request@es.net Wed Oct 4 22:33:13 2000
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Date: Thu, 05 Oct 2000 04:30:33 +0200
To: Qiaobing Xie
From: Mikael Degermark
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
Cc: Mikael Degermark ,
Krister Svanbro ,
Qiaobing Xie ,
Magnus Westerlund , rem-conf@es.net
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At 05:41 PM 10/4/00 -0500, Qiaobing Xie wrote:
(snip)
>The point I was trying to make is, when you are looking at optimizing
>the bandwidth efficiency over a *link* with a unique error
>characteristics, the framework/approach used by ROHC is more
>appropriate, which is to look for header compression/stripping solutions
>below the IP layer. This matches your b) above.
>But if you are designing an RTP payload format, which is meant to be
>sent end-to-end over a routable IP connection, it does not make much
>sense to optimize the payload format design according to the error
>characteristics of any one type of the link which, after all, may or may
>not be present in the connection.
If one expects that a common case will be that the payload is sent across
such links, it may make a lot of sense. But this is of course difficult to
predict.
>I have not problem with a) and I think it is in deed the job of AVT WG.
>No doubt that it is universally desirable to design any protocol message
>as compact and as compressed as possible (as Krister said: not to waste
>a bit) But in reality, you often have to make trade-off between saving
>a bit in the message and the associated processing cost, especially when
>the processing is done in a higher stack layer.
The tradeoff is between the cost of sending the bits across the link and
the processing cost at the sender/receiver. As the end user will ultimately
pay for both, one has to consider the relative costs.
When the amr format is used for voice, it seems that either of two
scenarios will occur:
1) users use some sort of mobile terminal.
2) users use stationary terminals.
For 1, the cost of bandwidth is very likely to be higher than the processing cost
if the mobile uses cellular radio.
For 2, the cost of bandwidth is very low, but on the other hand
the incremental processing cost is zero when something like a PC is used
as the stationary terminal.
So this very simple analysis does seem to favor a high degree of compression.
>The error resilience
>part (or the lack of it) in the current AMR draft is exactly one of my
>big concerns as I talked about in my first email.
>
>Cheers,
>-Qiaobing
>
>>
>> The IETF should not worry ONLY about wireless, of course, but it certainly
>> seems appropriate to ensure that things work over wireless as well as over
>> the rest of the currently largely wired Internet.
>>
>> Cheers,
>>
>> Mikael Degermark
>
From rem-conf-request@es.net Thu Oct 5 00:43:14 2000
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From: "Yoshihiro Kikuchi"
To: "Stephan Wenger"
Cc: "Colin Perkins" , "IETF AVT" ,
"4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>,
"Yoshihiro Kikuchi"
References: <200010040444.AAA09294@purple.east.isi.edu> <5.0.0.25.2.20001004173727.02f62990@mail.cs.tu-berlin.de>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
Date: Thu, 5 Oct 2000 13:36:52 +0900
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Dear Stephan, Kris and all,
Thank you very much for your comments.
Seeing a comment from Kris and a suggestion form Stephan about H.263 RTP
format, I think it is reasonable to keep the use of H.263 RTP format as
"SHOULD", specifying only RFC2429 as suggested, and not to add the detailed
timestamp definition of the short video header mode.
When the short video header mode is used, RTP payload format for H.263
(RFC 2429) SHOULD be used.
Best regards,
Yoshihiro
----- Original Message -----
From: Stephan Wenger
To: Yoshihiro Kikuchi
Cc: Colin Perkins ; IETF AVT ; Yoshihiro
Kikuchi
Sent: Thursday, October 05, 2000 12:39 AM
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
> Dear Yoshihiro, Group,
>
> To my knowledge the only reason why RFC2190 is not yet OBSOLETE is
> that some early H.323 products still use it. Interoperability with such
> products is probably not your main concern. So I would suggest to select
> RFC2429 only.
>
> Stephan
>
> At 08:15 PM 10/4/2000 +0900, Yoshihiro Kikuchi wrote:
> >Dear Colin, all,
> >
> > > >> When the short video header mode is used, RTP payload format for
> >H.263
> > > >< (e.g., RFC 2190 or RFC 2429) SHOULD be used.
> > >
> > > You should specify which of RFC 2190 or RFC 2429 is intended.
> > >
> >Can I ask suggestions from ITU-T H.263 experts on this selection of RFCs?
> >Or, if ITU-T does not have any preference, I'm not sure if it is really
> >necessary to change the text to decide this in "MPEG-4" RTP payload spec.
> >
> >Best regards,
> >Yoshihiro
> >
> >----
> > Yoshihiro Kikuchi
> >
> >E-mail: kiku@eel.rdc.toshiba.co.jp
> >Corporate Research and Development Center
> >TOSHIBA Corporation
> >TEL: +81 +44 549 2288 FAX: +81 +44 520 1267
>
>
>
>
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To: <4on2andIP-sys@advent.ee.columbia.edu>,
Subject: Re: Types of MPEG-4 streams over IP
Message-ID: <0056900012513789000002L092*@MHS>
Date: Thu, 5 Oct 2000 10:08:11 +0200
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matsui@drl.mei.co.jp wrote:
> Dear Jan and all,
>
> >Can we assume that the following streams are probably most efficiently
> >contained in a FlexMux stream, and consequently that is no need to specify
> >use of RTP and SDP for those streams:
> >
> >- ObjectDescriptorStream
> >- ClockReferenceStream
> >- SceneDescriptionStream
> >- MPEG-7Stream
> >- IPMPStream
> >- ObjectContentInfoStream
> >- MPEGJStream
>
> In my opinion, those streams should be carried on a reliable channel
> rather than RTP.
I wish that was possible too, however ...
In a truly scalable distribution architecture
you must use connectionless protocols.
RTP on UDP multicast is the key candidate.
NB: Retransmission of lost data requires a return channel
that is not always present and if it were present
would probably be not very scalable so forget
about absolute reliability too, repetition (and FEC)
are what we have to make sure can work
Therefore scalable broadcast requires RTP transport
of ALL these streams as well as SDP in repeated SAP packets.
>
>
> What I am not sure is the scene description stream.
> There are two kind of the scene description stream, one is BIFS-update
> and another is BIFS-Anim. In my understanding, BIFS-Anim should
> be streamed like audio/visual streams.
>
Same thing for both, must be possible to broadcast them too.
Regards,
--
Philippe Gentric
Software Architect
PHILIPS DIGITAL NETWORKS
Broadcast & Internet Delivery Solutions
Laboratoires d'Electronique Philips
B.P. 15 - 22, Av. Descartes
94453 Limeil-Brevannes Cedex France
Phone : 33 (0)1 45 10 68 12
Fax : 33 (0)1 45 10 69 60
philippe.gentric@philips.com
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