From rem-conf Tue Dec 01 01:42:47 1998 
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From: Socrates Varakliotis <S.Varakliotis@cs.ucl.ac.uk>
To: Malcolm Caldwell <malcolm@it.ntu.edu.au>, rem-conf <rem-conf@es.net>
Cc: "s.varakliotis" <s.varakliotis@cs.ucl.ac.uk>
Subject: Re: Linux vat
Date: Tue, 1 Dec 1998 09:33:25 -0000
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-----Original Message-----
From: Malcolm Caldwell <malcolm@it.ntu.edu.au>
To: rem-conf@es.net <rem-conf@es.net>
Date: Tuesday, December 01, 1998 12:26 AM
Subject: Linux vat


>I am looking for a version of vat for linux that will run cards supported
by
>video for linux, or better video for linux 2.


You probably mean vic here.

>I have been on the list for a long time.  I have a lot of problems finding
>various versions of vic for different platforms.  I guess the problem is
that
>vic stopped being supported.  Heaps of people have made their own versions
>with various cards supported etc.


Have you seen the original vic web page at UCB/LBNL?
http://www-nrg.ee.lbl.gov/vic/

Isidor's page might be of interest, as well:
http://www.cs.ucl.ac.uk/staff/i.kouvelas/vic/

>What is needed is a real "vic" web page, with a list of all the different
>versions.


See also:
http://www.cs.brown.edu/~lsh/ooVIC/
http://kbs.cs.tu-berlin.de/payload/
and the MASH version:
http://www-mash.cs.berkeley.edu/mash/software/usage/vic.html

I hope this helps.

Kind regards,
Socrates Varakliotis
______________________________________________________________________
Department of Computer Science                Tel: +44 (0)171 419 3697
University College London                     Fax: +44 (0)171 387 1397
Gower Street, London, WC1E 6BT    www.cs.ucl.ac.uk/staff/S.Varakliotis
______________________________________________________________________




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From: "Scott Petrack" <Scott_Petrack@vocaltec.com>
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Subject: Re: A totally different (simpler?) approach to tones and errata
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Ok, I give up. I'm glad I submitted the first version.

Just so we're all on the same page, here would be the tone frequency
format,
after the few changes that we've mentioned:


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   modulation  |M|D|  volume   |          duration             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |R|R R R|       frequency       |R|R R R|       frequency       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |R|R R R|       frequency       |R|R R R|       frequency       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   ......

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |R|R R R|       frequency       |R|R R R|      frequency        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


where:

modulation is an integer F between 0 and 255  - gives the modulation
frequency in Hz, unless either M or D is 1.
M is the "mutliply by 2" bit. If 1 then the modulation frequency is
doubled.
D is the "divide by 3 bit.  If 1 then modulation frequency is divided by
three.
(so modulation frequency is  F * (M+1) / (3D+1))
(This is a very ad hoc scheme, but it allows all of the freqencies in the
ITU draft)


Volume and duration are as in the Named signal event ("dtmf") format.

R is reserved and must be 0 right now. (We might use it to signal some
phase reversals or individual modulations, etc if it's ever needed. BTW,
the ear can hear (say) a 440 Hz tone played with N  phase reversals per
second. The discontinuities in the derivative N times per second show up as
harmonics that can be heard or detected. My friend swears that this is the
difference between fax tones and modem tones, but I have not seen the
standard. Is it V.8?? But for now no one seems to need it.)


Frequency is a positive integer between 0 and 4096.

This basically allows one to play Hungarian dial tone, or indeed  the
Brahms Hungarian Dances which were perhaps its inspiration. I think that
the specification should note that telephone systems (outside Hungary) are
only REQUIRED to be able to synthesize two tones simultaneously.

BTW, if we include a separate modulation for each frequency, we will be
able to synthesize some speech.... (maybe hungarian will need two
modulation for each frequency).

Scott













Henning Schulzrinne <hgs@cs.columbia.edu> on 11/30/98 06:52:59 PM

To:   Scott Petrack
cc:   rem-conf@es.net
Subject:  Re: A totally different (simpler?) approach to tones and errata




Scott Petrack wrote:
>
> >I just don't see the advantage of a table instead of the simple
> >enumeration of frequencies.
>
> The simple enumeration of frequencies is what is done in the DTMF draft:
> after all, draft-ietf-avt-dtmf.txt  just gives a table where index 1
means
> "697 Hz  +   1209 Hz". I just thought that the advantages there also
apply
> here.

This is, I believe, somewhat different, since the set of DTMF signals
won't be changed unless you modify a large fraction of the world's 640
million telephones. Adding new central-office tones doesn't require
nearly that amount of change - one reason why DTMF is universal and CO
tones are not. I think of DTMF "tones" as much more like the signal
events like "Busy", except that they have a duration.

> My understanding is that the combination of frequencies that is used
> in the world's telephone networks is pretty fixed and stable. It isn't

The problem is the "pretty". A few changes here and there are enough to
make the table approach unwieldy at best. There may also be tones that
the table misses (such as CNG and other fax/modem/control tones). A
representation that can handle tones in the voiceband range seems much
safer and adds only marginal space overhead. From a receiver
perspective, it also seems easier, since I don't have to store the
table.

Plus, I can use it to play advertising jingles :-)

This is not a major issue, but anything that doesn't require yet another
IANA registry or 'vintages' on specifications is a win in my book...







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Berkeley Multimedia, Interfaces and Graphics Seminar

Towards a Human-Centered Interaction Architecture

Wednesday December 2, 1998 12:30-2:00 pm 405 Soda Hall

Terry Winograd
Computer Science Department, Stanford University 

We are developing an integrated multi-device workspace, based on a high
level architecture for organizing multi-person multi-modal interactions
with computer systems. The architecture provides several mechanisms for
coping with fundamental properties of human interaction: object-based
perception, context-dependent interpretation, and action-perception
coupling. The talk will describe the underlying principles and the ways
that they are being applied in our planned interactive workspace. 

The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 

Sponsored by the Berkeley Multimedia Research Center

____________________________________________

Florissa Colina
Berkeley Multimedia Research Center (BMRC)
http://bmrc.berkeley.edu/
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: florissac@bmrc.berkeley.edu



From rem-conf Tue Dec 01 12:35:54 1998 
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To: malcolm@it.ntu.edu.au
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Subject: Re: Linux vat
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In lists.avt.rem-conf you write:

>I am looking for a version of vat for linux that will run cards supported by
>video for linux, or better video for linux 2.

diff and RedHat binary (i386) for a vic version with video4linux support
available at:
	http://www.in-berlin.de/User/kraxel/v4l

Tested with the bttv driver.

  Gerd

-- 
Beware "We should", extend a hand to "How do I"...
	-- Alan Cox on /.



From rem-conf Tue Dec 01 14:43:23 1998 
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From: akp4233@worldnet.att.net
To: "Hot list of candidates"@es.net
Date: Mon, 30 Nov 1998 16:12:04 -0800
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<color><param>0100,0100,0100</param><FontFamily><param>Times New Roman</pa=
ram>Hi,


I have a few strong candidates available for consulting work. There 
profile is provided for your reference.


Please contact me to pursue any one of these candidates. 


I would appreciate it very much if you please forward this profile to othe=
r 
hiring managers in your organization.


Gridcomp provides contract engineers  to fulfill your short term or long 
term employment requirements.


 


<bold>(1) Strong VC++ & NT candidate - available immediately. Has these 
skills:


<underline></bold>Experience:</underline>	4+ years of experience.  has wor=
ked for Oracle corp.

<underline>Languages:</underline>	Java, Microsoft Visual C++, C, Perl, SQL=
, HTML, CORBA 
IDL

<paraindent><param>out</param><underline>API=92s:</underline>	Windows Tele=
phony API (TAPI), OLE, ActiveX, 
Speech API (SAPI), CGI, Java Cryptography Extension 
(JCE 1.2)</paraindent>

<underline>OS:</underline>		Windows NT, 95, 3.1, MSDOS 6.x, UNIX (Solaris,=
 AIX, SunOS)

<underline>Education:</underline>	MS, Computer science, UCLA

		BS, summa Cum Laude, Computer science & Engineering, UCLA



<bold>(2) Murali:	Strong Java, SNMP, ATM, GUI candidate


SUMMARY :

</bold>Over 3+ years of experience with computer applications involving 
design, development, testing and Implementation.

Expertise in building Network Management System(NMS) using SNMP 
protocols.

Expertise in using JAVA, JDBC, RMI, Java Socket Programming, HTML, 
CGI, Java Script, C,C++, VC++, Install Shield-V5 ,Unix Shell Script, Java 
Workshop, Microsoft Frontpage-8,paint Shop Pro-3.


<bold>TECHNICAL SKILLS:

<underline></bold>Protocols Worked</underline>		:HTTP and SNMP.

<underline>Area of Specialization</underline>		:Network Management.

<paraindent><param>out</param><underline>Languages & Packages</underline>	=
:C,C++,VC++(V-5),  JAVA (JDK1.1) 
, HTML, CGI, JDBC,  JAVA Socket 
Programming,,   Java Script, , 
RMI(Remote Method Invokation), 
Unix Shell Programming, 
InstallShield-V5,  Java Workshop-V1, 
Symantec Caf=E9, Microsoft FrontPage-
8, PaintShop Pro-3 , Oracle -7,</paraindent>

<underline>HP-OpenView

Operating Systems</underline> 		:DOS, UNIX ( SUN OS ), Windows95, 
Windows NT.

<underline>GUI</underline>				:X-Windows/Motif, Forms-4.

<underline>Tools Used</underline>			:Net X-ray,  SNMP-C, Sniffer.


<bold>(3) Rajiv:	Powerbuilder, Oracel, SQL, Foxpro candidate:</bold>


<bold>SUMMARY :

<paraindent><param>out</param></bold>Over 4 years of experience in the IT =
Industry in 
various environments.</paraindent>

<paraindent><param>out</param>Development which includes coding and design=
ing 
user interface using DBMS and RDBMS concepts. </paraindent>

<paraindent><param>out</param>Developed packages using Software Environmen=
t like 
FOXPRO and    4GL TOOLS like <bold>POWER 
BUILDER 4.0/5.0/6.0</bold> with ORACLE 7.x  which is a 
client/server technology.</paraindent>

<paraindent><param>out</param>Good knowledge on Inventory systems and  Sto=
ck 
Management Systems.</paraindent>

<paraindent><param>out</param>Hands on experience in Banking systems.</par=
aindent>


<paraindent><param>right</param><bold>TECHNICAL SKILLS:</paraindent>

<underline></bold><FontFamily><param>Arial</param>OPERATING SYSTEMS:</unde=
rline>	MS-DOS, Windows 95/98,Windows NT 
4.0.

				Windows 3.11 for workgroups.

<underline>FRONT END TOOL</underline>	:	<italic>POWER BUILDER 4.0/5.0/6.0<=
/italic>

<underline>RDBMS</underline>:			Oracle 7.x,Sybase SQL Anywhere 5.0, 

 				Watcom SQL 4.0

<underline>DBMS	</underline>: 			FOXPRO

<underline>LANGUAGES</underline>:			C, C++,FORTRAN

<underline>HARDWARE</underline>	:		INTEL PENTIUMS, IBM COMPATIBLES.

<underline>FAMILIAR WITH:</underline>		DEVELOPER 2000, DELPHI 1.0  




<bold><FontFamily><param>Times New Roman</param>(4) TV Murugan:	Power Buil=
der, Sybase, SQL, GUI candidate


SUMMARY:

</bold>Three years IT experience in software design, development, testing,=
 
implementation

and support using Power Builder (6.0/5.0/4.0 with PFC Application 
Development), Sybase System 10, Microsoft SQL Server 6.5 and GUIs. 
Working experience in applications such as Financial Accounting & 
Inventory control systems, Structural Engineering Design systems, 
Banking systems and Payroll systems.

	

<bold>TECHNICAL SKILLS:

<underline>Operating Systems</underline>	:	</bold>UNIX, Windows NT, Window=
s 95, and DOS.

<bold><underline>RDBMS</underline>		:	</bold>Microsoft SQL Server 6.5, Syb=
ase System 10

<bold><underline>GUI</underline>			:	</bold>Power Builder 6.0/5.0/4.0 with=
 PFC


<bold>(5) P. Ganesh:	C, C++, VC++, MFC, COM/DCOM OOP, Crystal 
reports candidate


<FontFamily><param>Arial</param>SUMMARY:

</bold>Overall 5 =BD years of experience in multimedia projects, applicati=
on 
development in Client Server environment and Internet.


<paraindent><param>out</param>Proficient in C, C++, VC++ (MFC, ODBC & DAO)=
, 
COM/DCOM, MS-Access & OOPS concepts</paraindent>

<paraindent><param>out</param>Experience in Analysis, Design, Development =
& Testing</paraindent>

<paraindent><param>out</param>4 years experience in VC++</paraindent>

<paraindent><param>out</param>2 years experience in BTrieve Crystal Report=
s</paraindent>

<paraindent><param>out</param>Business Processes known =96 Retail Trading =
& Point of 
Sale(POS)</paraindent>


<bold><FontFamily><param>Times New Roman</param>TECHNICAL SKILLS:

<underline></bold><FontFamily><param>Arial</param>Languages</underline>		:=
 C, C++

<underline>Frond End Tools</underline>	: Visual C++, Visual Basic

<underline>Operating Systems</underline>	: MS-DOS, Win 3.1, Windows for Wo=
rkgroups, 
Win95, Win98/NT

<underline>Third Party Tools</underline>	: Objective Grid, Dragon Voice

<underline>Internet Tools</underline>		: COM/DCOM<FontFamily><param>Times =
New Roman</param>


Contact Anil Patel at { HYPERLINK mailto:anil@gridcomp.com }<underline><co=
lor><param>0000,0000,FF00</param>anil@gridcomp.com</underline><color><para=
m>0100,0100,0100</param> or (650)494-8407 to pursue 
these candidates.






From rem-conf Tue Dec 01 20:54:42 1998 
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Date: Tue, 01 Dec 1998 23:45:20 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Subject: Re: RTP Multiplexing and Sequence Numbers
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Neal Schneider wrote:
> 
> The problem with this, arises when/if individual calls do not arrive at
> consistant intervals to the multiplexing engine (i.e. over a packet
> switched network, rather than a TDM.)   Then the offset becomes very
> difficult to calculate appropriately, if at all.   The original
> information must be retained.

This is a very different kind of muxing. Now, you are talking about
muxing in the middle of an IP network and demuxing in the middle of an
IP network. As opposed to the gateway to gateway case, there will be no
correlation between any of the components of any of the headers of any
of the packets which are muxed together, since presumably they come from
different sources. In this case, you will need to reconstruct the entire
packet at the demux point, both timestamp and sequence number; that is,
unless you are arguing for throwing away the timestamp for RTP in
general. So, I don't see how offsets nor anything else help you here.
Perhaps you have a different model in mind?

-Jonathan R.


-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



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Hi,

	I've just subscribed to this group and am 
	having some problems getting the archives, so
	please bear with me if I am repeating previous
	discussions. I have the following Qs:

1.	The draft rtpproto-00 has expired. Does this
	mean RTP will not be elevated to a protocol.
	I hope so, but a pity.

2.	The draft rtp-multiplex-01 as a method of reducing
	bandwidth utilisation between VOIP gateways:

	a) Although the IP overhead figures given in
	table 1 make it look like an incredible waste, 
	if one looks at the % of bandwidth utilization for 
	a single channel, using 100base-T as a reference,
	it's pretty small. Need we be so obsessed by bandwidth
	utilisattion?

	b) Rather than this complicated scheme of multiplexing 
	RTP headed packets, why not just trunk a number of channels
	through in one stream. Thus, an RTP trunking packet may be 
	one sample from each of 32 channels. This keeps latency to 
	a minimum which is highly desired by telcos.

3.	I have also the following observations:

	a)With G.711, the number of 20mS for a frame packetisation
	period is arbitary and not ideal from the point of view 
	of latency. My figures show there is very little difference 
	in bandwidth utilization per channel, as a percentage, 
	between 32 sample and 160 sample packets.

	In discussing this around here, I find this number of	
	20mS is a historical artifact. This is a typical frame
	period for compression and, secondly, software cannot
	process enough packets per second to allow smaller
	frames.

	b)Telcos really do place a premium on low-latency.
	

[\] Robert Wood   

The St. Lawrence river - fresh, warm, visible diving.

mailto:robert_wood@mitel.com



From rem-conf Wed Dec 02 08:41:51 1998 
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Subject: extended deadline for ISCC'99
Cc: tantawi@us.ibm.com, Hossam.Afifi@enst-bretagne.fr
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My apologies if you receive multiple copies of this email.

Due to several requests, the submission deadline for ISCC'99
has been extended to December 15, 1998.

For further information, please visit the following web site:

http://www.rennes.enst-bretagne.fr/~afifi/iscc99.html




From rem-conf Wed Dec 02 09:33:41 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: nschneid@hotmail.com, jdrosen@dnrc.bell-labs.com
Cc: rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
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You are correct, in that offsets have no correlation with each other in 
this general arrival case.   Is there any reason why sequence numbers 
could not retain all of the necessary information for reconstructing the 
RTP packet, if the original sampling rate was constant?   Where the 
reconstructed timestamp would be equal to (sequence number * sampling 
interval).

Rgds, 
Neal A. Schneider

>different sources. In this case, you will need to reconstruct the 
entire
>packet at the demux point, both timestamp and sequence number; that is,
>unless you are arguing for throwing away the timestamp for RTP in
>general. So, I don't see how offsets nor anything else help you here.
>Perhaps you have a different model in mind?
>
>-Jonathan R.
>
>
>-- 
>Jonathan D. Rosenberg                       Lucent Technologies
>Member of Technical Staff                   101 Crawfords Corner Rd.
>High Speed Networks Research                Holmdel, NJ 07733
>FAX:   (732) 834-5379                       Rm. 4C-526
>EMAIL: jdrosen@bell-labs.com
>URL: http://www.cs.columbia.edu/~jdrosen


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Wed Dec 02 11:51:42 1998 
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From: Jarno Rajahalme <Jarno.Rajahalme@research.nokia.com>
Reply-To: "Jarno.Rajahalme@research.nokia.com"
	 <Jarno.Rajahalme@research.nokia.com>
To: "'nschneid@hotmail.com'" <nschneid@hotmail.com>,
        "'jdrosen@dnrc.bell-labs.com'" <jdrosen@dnrc.bell-labs.com>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: RTP Multiplexing and Sequence Numbers
Date: Wed, 2 Dec 1998 14:46:20 -0500
Organization: Nokia Research Center / Boston
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That's right. As long as the time duration of each packet is constant, the 
timestamp can be derived from the sequence number as you stated. The 
problem is that when something like silence detection and comfort noise is 
used the original assumption does not hold any more. Typically the duration 
of the comfort noise packet is longer than that of a regular voice packet 
(or even unspecified). Sometimes packets with just silence are simply not 
transmitted.

In these cases it is difficult to know the proper placement of the next 
actual voice packet in the playout buffer, if the timestamp is omitted. In 
cases where the network jitter is bounded, the arrival times of the few 
first packets of the talkspurt could provide enough information for the 
proper playback of the talkspurt, though.

As the bandwidth saving from the silence detection is quite remarkable, it 
would be reasonable to expect every solution to work also when silence is 
suppressed in a form or another.

	Jarno Rajahalme
--
Jarno Rajahalme <Jarno.Rajahalme@research.nokia.com>
Nokia Research Center / Boston
3 Burlington Woods Drive, Suite 260
Burlington, MA 01803, USA

On Wednesday, December 02, 1998 12:30 PM, EXT Neal Schneider 
[SMTP:nschneid@hotmail.com] wrote:
> You are correct, in that offsets have no correlation with each other in
> this general arrival case.   Is there any reason why sequence numbers
> could not retain all of the necessary information for reconstructing the
> RTP packet, if the original sampling rate was constant?   Where the
> reconstructed timestamp would be equal to (sequence number * sampling
> interval).
>
> Rgds,
> Neal A. Schneider
>
> >different sources. In this case, you will need to reconstruct the
> entire
> >packet at the demux point, both timestamp and sequence number; that is,
> >unless you are arguing for throwing away the timestamp for RTP in
> >general. So, I don't see how offsets nor anything else help you here.
> >Perhaps you have a different model in mind?
> >
> >-Jonathan R.
> >
> >
> >--
> >Jonathan D. Rosenberg                       Lucent Technologies
> >Member of Technical Staff                   101 Crawfords Corner Rd.
> >High Speed Networks Research                Holmdel, NJ 07733
> >FAX:   (732) 834-5379                       Rm. 4C-526
> >EMAIL: jdrosen@bell-labs.com
> >URL: http://www.cs.columbia.edu/~jdrosen
>
>
> ______________________________________________________
> Get Your Private, Free Email at http://www.hotmail.com
>




From rem-conf Wed Dec 02 12:05:21 1998 
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        "'rem-conf@es.net'" <rem-conf@es.net>
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Jarno Rajahalme wrote:
> 
> That's right. As long as the time duration of each packet is constant, the
> timestamp can be derived from the sequence number as you stated. The
> problem is that when something like silence detection and comfort noise is
> used the original assumption does not hold any more. Typically the duration
> of the comfort noise packet is longer than that of a regular voice packet
> (or even unspecified). Sometimes packets with just silence are simply not
> transmitted.
> 
> In these cases it is difficult to know the proper placement of the next
> actual voice packet in the playout buffer, if the timestamp is omitted. In
> cases where the network jitter is bounded, the arrival times of the few
> first packets of the talkspurt could provide enough information for the
> proper playback of the talkspurt, though.

There is a more important problem related to backwards compatibility. In
the case we are discussing here, that of mux and demux points outside of
gateways or end systems, the RTP end users, gateways for example, won't
know that the muxing is even going on. Now, say you design a mux
protocol which generates incorrect timestamps in the RTP packet, under
the assumption that you don't need them. The gateways will receive the
RTP packets with the incorrect timestamp, but won't know they are
incorrect since there is no indication that these have been muxed and
demuxed prior to reception. As there are certainly gateways and end
systems in production know which make use of the timestamp, you will
break compatibility with them.

I believe that *if* you want to do this non end-to-end muxing, you MUST
reproduce exactly at the demux point the packets which enter at the mux
point, regardless of whether you like sequence numbers, timestamps, or
the like. I emphasize *if* as I'm not yet sure its such a hot idea. Its
really not clear what kind of mux gains you'll get, as there is not much
correlation in the headers of the muxed packets. 

-Jonathan R.


-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Wed Dec 02 12:08:15 1998 
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From: Imran Chaudhri <imran@chaudhri.net>
To: "'Neal Schneider'" <nschneid@hotmail.com>,
        "jdrosen@dnrc.bell-labs.com"
	 <jdrosen@dnrc.bell-labs.com>
Cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: RE: RTP Multiplexing and Sequence Numbers
Date: Wed, 2 Dec 1998 15:05:40 -0500
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Assuming both source and destination have access to the same synchronous =
network clock you can do away with timestamps. However whenever you =
cross an administrative or country boundary, this is not the case.
Imran

Imran N. Chaudhri
imran@chaudhri.net
T:301-963-3852
F:301-963-3851
M:301-537-8444


-----Original Message-----
From:	Neal Schneider [SMTP:nschneid@hotmail.com]
Sent:	Wednesday, December 02, 1998 12:30 PM
To:	nschneid@hotmail.com; jdrosen@dnrc.bell-labs.com
Cc:	rem-conf@es.net
Subject:	Re: RTP Multiplexing and Sequence Numbers

You are correct, in that offsets have no correlation with each other in=20
this general arrival case.   Is there any reason why sequence numbers=20
could not retain all of the necessary information for reconstructing the =

RTP packet, if the original sampling rate was constant?   Where the=20
reconstructed timestamp would be equal to (sequence number * sampling=20
interval).

Rgds,=20
Neal A. Schneider

>different sources. In this case, you will need to reconstruct the=20
entire
>packet at the demux point, both timestamp and sequence number; that is,
>unless you are arguing for throwing away the timestamp for RTP in
>general. So, I don't see how offsets nor anything else help you here.
>Perhaps you have a different model in mind?
>
>-Jonathan R.
>
>
>--=20
>Jonathan D. Rosenberg                       Lucent Technologies
>Member of Technical Staff                   101 Crawfords Corner Rd.
>High Speed Networks Research                Holmdel, NJ 07733
>FAX:   (732) 834-5379                       Rm. 4C-526
>EMAIL: jdrosen@bell-labs.com
>URL: http://www.cs.columbia.edu/~jdrosen


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com




From rem-conf Wed Dec 02 12:59:26 1998 
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From: Jarno Rajahalme <Jarno.Rajahalme@research.nokia.com>
Reply-To: "Jarno.Rajahalme@research.nokia.com"
	 <Jarno.Rajahalme@research.nokia.com>
To: "'jdrosen@dnrc.bell-labs.com'" <jdrosen@dnrc.bell-labs.com>
Cc: "'nschneid@hotmail.com'" <nschneid@hotmail.com>,
        "'rem-conf@es.net'"
	 <rem-conf@es.net>
Subject: RE: RTP Multiplexing and Sequence Numbers
Date: Wed, 2 Dec 1998 15:55:57 -0500
Organization: Nokia Research Center / Boston
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What is written below is valid, assuming that the endpoints indeed are 
using non-muxed RTP, and are e.g. using timestamps to synchronize multiple 
streams, etc.

However, if the multiplex payload format is used to multiplex NON-RTP 
streams, there is no harm done (since there were no timestamps to begin 
with), and the header overhead can be reduced to two bytes per voice frame. 
Applicable example could be a network of IP telephony gateways 
interconnected with a network of "mux and demux nodes".

Also, for simple IP telephony (single voice stream etc.) conversion from 
non-muxed RTP to/from muxed RTP could work as well. Also, deployed base of 
endpoints that would fail because of "artificial' timestamps could be small 
compared to the IP telephony deployment in the future.

Finally, the terminals could use capability negotiation with their 
gateways/proxies to determine the use of multiplex payload format also for 
single stream voice or for multiplexed voice/video.

With regards,

	Jarno Rajahalme

On Wednesday, December 02, 1998 2:59 PM, EXT Jonathan Rosenberg 
[SMTP:jdrosen@dnrc.bell-labs.com] wrote:
> There is a more important problem related to backwards compatibility. In
> the case we are discussing here, that of mux and demux points outside of
> gateways or end systems, the RTP end users, gateways for example, won't
> know that the muxing is even going on. Now, say you design a mux
> protocol which generates incorrect timestamps in the RTP packet, under
> the assumption that you don't need them. The gateways will receive the
> RTP packets with the incorrect timestamp, but won't know they are
> incorrect since there is no indication that these have been muxed and
> demuxed prior to reception. As there are certainly gateways and end
> systems in production know which make use of the timestamp, you will
> break compatibility with them.
>
> I believe that *if* you want to do this non end-to-end muxing, you MUST
> reproduce exactly at the demux point the packets which enter at the mux
> point, regardless of whether you like sequence numbers, timestamps, or
> the like. I emphasize *if* as I'm not yet sure its such a hot idea. Its
> really not clear what kind of mux gains you'll get, as there is not much
> correlation in the headers of the muxed packets.
>
> -Jonathan R.
>
>
> --
> Jonathan D. Rosenberg                       Lucent Technologies
> Member of Technical Staff                   101 Crawfords Corner Rd.
> High Speed Networks Research                Holmdel, NJ 07733
> FAX:   (732) 834-5379                       Rm. 4C-526
> EMAIL: jdrosen@bell-labs.com
> URL: http://www.cs.columbia.edu/~jdrosen
>


--
Jarno Rajahalme <Jarno.Rajahalme@research.nokia.com>
Nokia Research Center / Boston
3 Burlington Woods Drive, Suite 260
Burlington, MA 01803, USA




From rem-conf Wed Dec 02 15:02:24 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: Jarno.Rajahalme@research.nokia.com, jdrosen@dnrc.bell-labs.com
Cc: nschneid@hotmail.com, rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
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Why would the muxed protocol violate legacy gateways?   As long as the 
sequence number was accurately accounted for (even through periods of 
silence), one should be able to recreate timestamps accurately.  This 
should be transparent to the end-to-end RTP session.  Possibly the 
timestamp will be shifted by a certain constant, but the relationship 
between subsequent timestamps should remain the same.  Am I correct in 
assuming that it is the difference in timestamps that really counts?

Regards,
Neal A. Schneider

>There is a more important problem related to backwards compatibility. 
In
>the case we are discussing here, that of mux and demux points outside 
of
>gateways or end systems, the RTP end users, gateways for example, won't
>know that the muxing is even going on. Now, say you design a mux
>protocol which generates incorrect timestamps in the RTP packet, under
>the assumption that you don't need them. The gateways will receive the
>RTP packets with the incorrect timestamp, but won't know they are
>incorrect since there is no indication that these have been muxed and
>demuxed prior to reception. As there are certainly gateways and end
>systems in production know which make use of the timestamp, you will
>break compatibility with them.
>
>I believe that *if* you want to do this non end-to-end muxing, you MUST
>reproduce exactly at the demux point the packets which enter at the mux
>point, regardless of whether you like sequence numbers, timestamps, or
>the like. I emphasize *if* as I'm not yet sure its such a hot idea. Its
>really not clear what kind of mux gains you'll get, as there is not 
much
>correlation in the headers of the muxed packets. 
>
>-Jonathan R.
>
>
>-- 
>Jonathan D. Rosenberg                       Lucent Technologies
>Member of Technical Staff                   101 Crawfords Corner Rd.
>High Speed Networks Research                Holmdel, NJ 07733
>FAX:   (732) 834-5379                       Rm. 4C-526
>EMAIL: jdrosen@bell-labs.com
>URL: http://www.cs.columbia.edu/~jdrosen
>


______________________________________________________
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From rem-conf Thu Dec 03 01:25:32 1998 
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    "jdrosen@dnrc.bell-labs.com" <jdrosen@dnrc.bell-labs.com>, 
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Subject: Re: RTP Multiplexing and Sequence Numbers
In-reply-to: Your message of "Wed, 02 Dec 1998 15:55:57 EST." <01BE1E0C.40F63270.Jarno.Rajahalme@research.nokia.com>
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The reason you want both sequence numbers and timestamps is the same reason 
you want them in RTP in the first place.  Three reasons being:

(a) without both tools cannot distinguish between packet loss and silence 
intervals.  Therefore they cannot determine when to invoke loss concealment 
mechanisms, retransmission requests, FEC recovery, etc...

(b) in the case where a talkspurt marker packet is lost you may lose 
legitimate opportunities to recalculate playout points.  With both timestamps 
and seqno's you can tell when this happens because of change of relationship 
between the two.

(c) you eliminate the opportunity for variable frame rate codecs.  I am not 
aware of any that have been standardised, but have seen papers describing 
research in this direction (although not recent work :-)

You do not have to send both in every packet, you just need a mechanism for 
reconstructing both.  GeRM does this with a compact representation, using 
single bits to indicate (dis)continuities in seqno and timestamp and then 
updated fields follow the header.

It is reasonable that the majority of streamed applications (voice, video, 
stock tickers, text messages, etc) need both irrespective of whether they are 
RTP-based or not.

regards
- Orion
-- 
Orion Hodson                                 [Research Student]
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
Networked Multimedia Research Group      Tel: ++(0)171 419 3704
Department of Computer Science           Fax: ++(0)171 419 1397
University College London, Gower Street
London, WC1E 6BT, UK.     
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

<01BE1E0C.40F63270.Jarno.Rajahalme@research.nokia.com>Jarno Rajahalme writes:
> What is written below is valid, assuming that the endpoints indeed are 
> using non-muxed RTP, and are e.g. using timestamps to synchronize multiple 
> streams, etc.
> 
> However, if the multiplex payload format is used to multiplex NON-RTP 
> streams, there is no harm done (since there were no timestamps to begin 
> with), and the header overhead can be reduced to two bytes per voice frame. 
> Applicable example could be a network of IP telephony gateways 
> interconnected with a network of "mux and demux nodes".
> 
> Also, for simple IP telephony (single voice stream etc.) conversion from 
> non-muxed RTP to/from muxed RTP could work as well. Also, deployed base of 
> endpoints that would fail because of "artificial' timestamps could be small 
> compared to the IP telephony deployment in the future.
> 
> Finally, the terminals could use capability negotiation with their 
> gateways/proxies to determine the use of multiplex payload format also for 
> single stream voice or for multiplexed voice/video.
> 
> With regards,
> 
> Jarno Rajahalme
> 
> On Wednesday, December 02, 1998 2:59 PM, EXT Jonathan Rosenberg 
> [SMTP:jdrosen@dnrc.bell-labs.com] wrote:
> > There is a more important problem related to backwards compatibility. In
> > the case we are discussing here, that of mux and demux points outside of
> > gateways or end systems, the RTP end users, gateways for example, won't
> > know that the muxing is even going on. Now, say you design a mux
> > protocol which generates incorrect timestamps in the RTP packet, under
> > the assumption that you don't need them. The gateways will receive the
> > RTP packets with the incorrect timestamp, but won't know they are
> > incorrect since there is no indication that these have been muxed and
> > demuxed prior to reception. As there are certainly gateways and end
> > systems in production know which make use of the timestamp, you will
> > break compatibility with them.
> >
> > I believe that *if* you want to do this non end-to-end muxing, you MUST
> > reproduce exactly at the demux point the packets which enter at the mux
> > point, regardless of whether you like sequence numbers, timestamps, or
> > the like. I emphasize *if* as I'm not yet sure its such a hot idea. Its
> > really not clear what kind of mux gains you'll get, as there is not much
> > correlation in the headers of the muxed packets.
> >
> > -Jonathan R.
> >
> >
> > --
> > Jonathan D. Rosenberg                       Lucent Technologies
> > Member of Technical Staff                   101 Crawfords Corner Rd.
> > High Speed Networks Research                Holmdel, NJ 07733
> > FAX:   (732) 834-5379                       Rm. 4C-526
> > EMAIL: jdrosen@bell-labs.com
> > URL: http://www.cs.columbia.edu/~jdrosen
> >
> 
> 
> --
> Jarno Rajahalme <Jarno.Rajahalme@research.nokia.com>
> Nokia Research Center / Boston
> 3 Burlington Woods Drive, Suite 260
> Burlington, MA 01803, USA
> 
> 







From rem-conf Thu Dec 03 06:08:59 1998 
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Subject: Re: RTP Multiplexing and Sequence Numbers
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Neal Schneider wrote:
> 
> Why would the muxed protocol violate legacy gateways?   As long as the
> sequence number was accurately accounted for (even through periods of
> silence), one should be able to recreate timestamps accurately.  This
> should be transparent to the end-to-end RTP session.  Possibly the
> timestamp will be shifted by a certain constant, but the relationship
> between subsequent timestamps should remain the same.  Am I correct in
> assuming that it is the difference in timestamps that really counts?

I assume that "sequence number was accurately accounted for during
silence" would mean artificially jumping it to match the timestamp gap
during silence periods. As Orion has pointed out, this will break
systems which differentiate between loss repair and silence comfort
noise insertion (and there is a difference), eliminate the possibility
of variable rate codecs, and make talkspurt detection harder. Its more
than that. RTCP loss statistics would be way off, reporting astronomical
losses. This would then break any sender adaptive schemes which use loss
as a guide for FEC insertion, codec selection, etc. I know of both
research and commercial products that use adaptive techniques at the
sender based on RTCP loss statistics.

Also, doing this kind of SN manipulation will require stateful mux and
demux to keep track of the SN. SHould it screw up in any way,
potentially all voice calls through it become garbled.

-Jonathan R.



-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu Dec 03 07:04:17 1998 
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From: subbiah@NASBPD01BS.ntc.nokia.com (Subbiah Baranitharan NRC/Bos)
To: robert_wood@Mitel.COM (Robert Wood)
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This is to answer your Q 2.a

You can consider a scenario where a new operator like to setup a IP =
telephony network. In this case, the operator has to lease the trunk from a =
=
Telco. If this is the case, an operator would like to have a scheme by which=
 =
every bit (he/she pays for) is used in an efficient way. I don't have to =
repeat the 9 cents and 10 cents difference between operator and what it =
means for customers.

The important think to consider is that we develop a compromised scheme tha=
t =
meets the service requirement of Telephony and at the same time improves the=
 =
bw efficiency.

Your comment on 2.b is not clear to me. How do we identify the channels on =
=
RTP trunking packet ?

>This is a typical frame period for compression and, secondly, software =
cannot
> process enough packets per second to allow smaller frames.

Are you aware of the 10 ms and lower frames.

regards,

Barani Subbiah
 ----------
From: Robert Wood
To: avt
Subject: Intro & Q
Date: Wednesday, December 02, 1998 11:11AM

Hi,

 I've just subscribed to this group and am
 having some problems getting the archives, so
 please bear with me if I am repeating previous
 discussions. I have the following Qs:

1. The draft rtpproto-00 has expired. Does this
 mean RTP will not be elevated to a protocol.
 I hope so, but a pity.

2. The draft rtp-multiplex-01 as a method of reducing
 bandwidth utilisation between VOIP gateways:

 a) Although the IP overhead figures given in
 table 1 make it look like an incredible waste,
 if one looks at the % of bandwidth utilization for
 a single channel, using 100base-T as a reference,
 it's pretty small. Need we be so obsessed by bandwidth
 utilisattion?

 b) Rather than this complicated scheme of multiplexing
 RTP headed packets, why not just trunk a number of channels
 through in one stream. Thus, an RTP trunking packet may be
 one sample from each of 32 channels. This keeps latency to
 a minimum which is highly desired by telcos.

3. I have also the following observations:

 a)With G.711, the number of 20mS for a frame packetisation
 period is arbitary and not ideal from the point of view
 of latency. My figures show there is very little difference
 in bandwidth utilization per channel, as a percentage,
 between 32 sample and 160 sample packets.

 In discussing this around here, I find this number of
 20mS is a historical artifact. This is a typical frame
 period for compression and, secondly, software cannot
 process enough packets per second to allow smaller
 frames.

 b)Telcos really do place a premium on low-latency.


[\] Robert Wood

The St. Lawrence river - fresh, warm, visible diving.

mailto:robert_wood@mitel.com





From rem-conf Thu Dec 03 10:12:54 1998 
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Subject: Re: RTP Multiplexing and Sequence Numbers
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I agree that manipulating sequence numbers (the basis for packet order, 
packet loss, and timing is there is no timestamp) is a jeopardous 
solution. 

In the case of multiplexing many voices, however, the benefits of 
completely dropping a channel in periods of silence are not substantial.  
Granted, if all voices were silent there would be an argument for 
holding back packets, but one could just take that as a infrequent loss.  
Could one continuously send packets with a 'comfort noise' payload 
format and incremental sequence numbers during periods of silence?  And 
merely achieve less efficiency when there are fewer voices being 
multiplexed? 

Regards,  
Neal A. Schneider

>I assume that "sequence number was accurately accounted for during
>silence" would mean artificially jumping it to match the timestamp gap
>during silence periods. As Orion has pointed out, this will break
>systems which differentiate between loss repair and silence comfort
>noise insertion (and there is a difference), eliminate the possibility
>of variable rate codecs, and make talkspurt detection harder. Its more
>than that. RTCP loss statistics would be way off, reporting 
astronomical
>losses. This would then break any sender adaptive schemes which use 
loss
>as a guide for FEC insertion, codec selection, etc. I know of both
>research and commercial products that use adaptive techniques at the
>sender based on RTCP loss statistics.
>
>Also, doing this kind of SN manipulation will require stateful mux and
>demux to keep track of the SN. SHould it screw up in any way,
>potentially all voice calls through it become garbled.
>
>-Jonathan R.
>
>
>
>-- 
>Jonathan D. Rosenberg                       Lucent Technologies
>Member of Technical Staff                   101 Crawfords Corner Rd.
>High Speed Networks Research                Holmdel, NJ 07733
>FAX:   (732) 834-5379                       Rm. 4C-526
>EMAIL: jdrosen@bell-labs.com
>URL: http://www.cs.columbia.edu/~jdrosen
>


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Thu Dec 03 11:19:27 1998 
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Neal Schneider wrote:
> 
> I agree that manipulating sequence numbers (the basis for packet order,
> packet loss, and timing is there is no timestamp) is a jeopardous
> solution.
> 
> In the case of multiplexing many voices, however, the benefits of
> completely dropping a channel in periods of silence are not substantial.
> Granted, if all voices were silent there would be an argument for
> holding back packets, but one could just take that as a infrequent loss.
> Could one continuously send packets with a 'comfort noise' payload
> format and incremental sequence numbers during periods of silence?  And
> merely achieve less efficiency when there are fewer voices being
> multiplexed?

In the case of mid-network multiplexing, you could try this, but I still
see problems. Firstly, the RTCP statistics will still be totally fouled
up (even worse, in fact), since the loss rates will most likely be
negative (receiver gets more than sender sends) since the mux point is
inserting these comfort noise packets. Highest SN sent statistic will
also be fouled up. Thus, adaptive schemes will still fail. Secondly, I
thought the whole point of this argument was that sending the timestamp
was too much overhead. Well, its 32 bits per user. Given a person is in
silence roughly 50% of the time, you want to send a comfort noise
filler, which is likely more than 64 bits, instead of sending nothing. I
believe you will actually end up sending more bits than if you just
leave the timestamp in place, and let it work as it was originally
designed to.

You could have the mux point hijack the RTCP also, and then modify it on
the fly based on new statistics. But, you'll lose the end to end
statistics provided by RTCP, in addition to now requiring some really
serious processing.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu Dec 03 11:30:05 1998 
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Subbiah Baranitharan NRC/Bos wrote:
> 
> Your comment on 2.b is not clear to me. How do we identify the channels on RTP trunking packet ?

I think I understand the comment - are you (Robert) arguing for the
equivalent of ATM's circuit emulation in IP? In other words, there are N
64 kbps channels. Take one byte from each every 125us, and make a packet
of N bytes, and then send. There are certain cases where this does make
sense, but it has the following limitations:

1. channel identification is based on positioning within the packet (ie,
3 bytes in is channel 3). Silence detection pretty much throws this for
a loop. See our muxissues draft, where we discuss the use of bitmasks to
indicate which packets are in silence. This approach also has
implications on signaling, synchronization, and reliability.

2. With compression, the codecs generate an entire frame of data all at
once, every T ms (usually T=20 or 30). By sending a byte from each in a
packet, you will actually increase the latency by a factor of two (T
frame delay, plus T more ms to finish sending the packet, one byte per
packet)

3. You won't be able to support variable rate codecs, or use of
different codecs in a multiplex.

4. The packet overhead is higher, since you have only one byte per
packet from each user, as opposed to a frames worth per user.

5. Your packet rate jumps up. For 64kbps PCM, you'll have 1 packet every
125us, which is 8000 packets per second. With the kind of muxes we've
been discussing, you'll have a packet every 30ms (say), which is 33
packets per second. Big difference, especially since routers are
sensitive to packet rates.


So, IP CES has some problems. Using a more frame/packet oriented
multiplex eliminates these problems.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



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Date: Thu, 03 Dec 1998 14:44:14 -0500
From: Robert Wood <robert_wood@Mitel.COM>
Organization: Mitel Corporation
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Cc: Subbiah Baranitharan NRC/Bos <subbiah@NASBPD01BS.ntc.nokia.com>,
        "'rem-conf'" <rem-conf@es.net>
Subject: Re: Intro & Q
References: <1998Dec03.095300.1991.399158@rhino.ntc.nokia.com> <3666E534.88ABDD22@dnrc.bell-labs.com>
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Jonathan Rosenberg wrote:
> 
> Subbiah Baranitharan NRC/Bos wrote:
> >
> > Your comment on 2.b is not clear to me. How do we identify the channels on RTP trunking packet ?
> 
> I think I understand the comment - are you (Robert) arguing for the
> equivalent of ATM's circuit emulation in IP? In other words, there are N
> 64 kbps channels. Take one byte from each every 125us, and make a packet
> of N bytes, and then send. There are certain cases where this does make
> sense, but it has the following limitations:
> 
	Yes, where N may be between 32 and 256.

	I personally like small packets for low latency. 32 to 64 samples.

> 1. channel identification is based on positioning within the packet (ie,
> 3 bytes in is channel 3). 

	I don't see this as a problem. If the trunking is static,
	we are always sending the same link(s) down one packet's stream.
	If the trunking is dynamic, the voice channel is allocated a
	position in the packet at call setup time.

> Silence detection pretty much throws this for
> a loop. 

	There is no silence suppression or voice compression
	implemented. This is intended for simple, raw trunking.

> See our muxissues draft, where we discuss the use of bitmasks to
> indicate which packets are in silence. This approach also has
> implications on signaling, synchronization, and reliability.
> 
> 2. With compression, the codecs generate an entire frame of data all at
> once, every T ms (usually T=20 or 30). By sending a byte from each in a
> packet, you will actually increase the latency by a factor of two (T
> frame delay, plus T more ms to finish sending the packet, one byte per
> packet)

	The RTP packet concatenations you are considering are appropriate
	for this situation.

> 4. The packet overhead is higher, since you have only one byte per
> packet from each user, as opposed to a frames worth per user.

	One could put in several bytes per channel and still keep
	latency low.

	Thanks for the feedback.

[\] Robert Wood   

The St. Lawrence river - fresh, warm, visible diving.

mailto:robert_wood@mitel.com



From rem-conf Thu Dec 03 11:48:16 1998 
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To: Neal Schneider <nschneid@hotmail.com>
cc: rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
In-reply-to: Your message of "Thu, 03 Dec 1998 10:04:45 PST." <19981203180447.25856.qmail@hotmail.com>
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<19981203180447.25856.qmail@hotmail.com>Neal Schneider writes:
> I agree that manipulating sequence numbers (the basis for packet order, 
> packet loss, and timing is there is no timestamp) is a jeopardous 
> solution. 
> 
> In the case of multiplexing many voices, however, the benefits of 
> completely dropping a channel in periods of silence are not substantial.  
> Granted, if all voices were silent there would be an argument for 
> holding back packets, but one could just take that as a infrequent loss.  
> Could one continuously send packets with a 'comfort noise' payload 
> format and incremental sequence numbers during periods of silence?  And 
> merely achieve less efficiency when there are fewer voices being 
> multiplexed? 

Yes, but consider:

(a) you probably lose the b/w saving you gain from not sending timestamps or 
not sending sequence numbers.  B/w lost between application and multiplexor.  
Or do you want the multiplexor to send comfort noise packets?  You still have 
the problem that the app must send comfort noise for the multiplexor to 
generate packets and multiplex may have to be comfort noise codec smart.
(b) you force every application to implement comfort noise generation and 
transmission (incompatibilities? bugs? backward compatibility?).
(c) comfort noise probably wants to be sent at less frequent intervals 
(probably variable intervals depending on how background noise evolves).
(d) not all codecs have a comfort noise scheme devised (ok there are generic 
schemes but you may find boundary effects at transition between codec and 
comfort noise).
(e) apps may have problems mixing multiple comfort noise sources especially at 
transitions between multiple speakers.

And that's before you worry about RTCP issues...

- Orion.





From rem-conf Thu Dec 03 11:53:36 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: nschneid@hotmail.com, jdrosen@dnrc.bell-labs.com
Cc: Jarno.Rajahalme@research.nokia.com, rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
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I wasn't intending to insert noise packets at the mux point.  I figured 
that the source would generate these packets.  Again, I believe that 
manipulation at the mux-demux engines should be minimal.  Yes, this 
defeats some of the intentions of efficient silence suppression, but 
can't a payload format be specified (small field), in which the actual 
noise payload be recreated on the demux or end-point based on a a noise 
level negotiated out of band?  

Also, One might assume that bandwidth utilization would not be much of 
an issue from the source to the mux point, considering that multiplexing 
is inserted at points of aggregation.  

I want a 32-bit mux header which contains all the necessary information 
for retaining a RTP session.   Non-dynamic information outside this 
header format would like to be negotiated out-of-band. 

Regards, 
Neal A. Schneider 

>In the case of mid-network multiplexing, you could try this, but I 
still
>see problems. Firstly, the RTCP statistics will still be totally fouled
>up (even worse, in fact), since the loss rates will most likely be
>negative (receiver gets more than sender sends) since the mux point is
>inserting these comfort noise packets. Highest SN sent statistic will
>also be fouled up. Thus, adaptive schemes will still fail. Secondly, I
>thought the whole point of this argument was that sending the timestamp
>was too much overhead. Well, its 32 bits per user. Given a person is in
>silence roughly 50% of the time, you want to send a comfort noise
>filler, which is likely more than 64 bits, instead of sending nothing. 
I
>believe you will actually end up sending more bits than if you just
>leave the timestamp in place, and let it work as it was originally
>designed to.
>
>You could have the mux point hijack the RTCP also, and then modify it 
on
>the fly based on new statistics. But, you'll lose the end to end
>statistics provided by RTCP, in addition to now requiring some really
>serious processing.
>
>-Jonathan R.
>
>
>-- 
>Jonathan D. Rosenberg                       Lucent Technologies
>Member of Technical Staff                   101 Crawfords Corner Rd.
>High Speed Networks Research                Holmdel, NJ 07733
>FAX:   (732) 834-5379                       Rm. 4C-526
>EMAIL: jdrosen@bell-labs.com
>URL: http://www.cs.columbia.edu/~jdrosen


______________________________________________________
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From rem-conf Thu Dec 03 12:40:07 1998 
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remove



From rem-conf Thu Dec 03 13:23:38 1998 
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To: rem-conf@es.net
Subject: 43rd IETF-ORLANDO, FL: Multicast Schedule
Date: Thu, 03 Dec 1998 16:19:17 -0500
From: Marcia Beaulieu <mbeaulie@ns.cnri.reston.va.us>
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	   Orlando IETF Multicast Guide (as of 12/03/98)


Following is the tentative schedule of plenary meetings and working
group sessions to be transmitted; to interpret the acronyms, see the
agenda at http://www.ietf.org/meetings/agenda_orlando.txt 
It is possible that this schedule will be modified.

MULTICAST GUIDE


Note that times are in Eastern Time. UTC (aka GMT) also provided.


MONDAY     0930-1130   1300-1500   1530-1730   1930-2200
    (UTC)  1430-1630   1800-2000   2030-2230   2430-0300
=========+===========+===========+===========+===========+
 CHAN 1  |   idr     |  mobileip |    pim    |  mboned   |
=========+===========+===========+===========+===========+
 CHAN 2  |   ruts    |  diffserv |  diffserv |           |
=========+===========+===========+===========+===========+


TUESDAY    0900-1000   1015-1115   1300-1400   1415-1515
    (UTC)  1400-1500   1515-1615   1800-1900   1915-2015
=========+===========+===========+===========+===========+
 CHAN 1  |   nfsv4   |   nfsv4   |  tcpsat   |  tcpsat   |
=========+===========+===========+===========+===========+
 CHAN 2  |   mmusic  |   mmusic  |  ngtrans  |  ngtrans  |
=========+===========+===========+===========+===========+

    TUESDAY   (CON'T)  1545-1645   1700-1800 
                (UTC)  2045-2145   2200-2300
            =========+===========+===========+
             CHAN 1  |  mboned   |  mboned   |
            =========+===========+===========+
             CHAN 2  |  policy   |  policy   |
            =========+===========+===========+    


WEDNESDAY  0900-1130   1300-1500   1530-1730   1930-2200
    (UTC)  1400-1630   1800-2000   2030-2230   2430-0300
=========+===========+===========+===========+===========+
 CHAN 1  |   pilc    |    idmr   |   issll   |  plenary  |
=========+===========+===========+===========+===========+
 CHAN 2  |   msdp    |     rap   |  tcpimpl  |  plenary  |
=========+===========+===========+===========+===========+


THURSDAY   0900-1130   1300-1500   1530-1730
    (UTC)  1400-1630   1800-2000   2030-2230
=========+===========+===========+===========+
 CHAN 1  |  malloc   |    avt    |  iptel    |
=========+===========+===========+===========+
 CHAN 2  |   ion     |   rsvp    |  ipngwg   |
=========+===========+===========+===========+

FRIDAY     0900-1130
    (UTC)  1400-1630  
=========+===========+
 CHAN 1  |    avt    |
=========+===========+
 CHAN 2  |   ipngwg  |
=========+===========+
                                                                  

Each day's program may also be replayed by tape delay from: 2330
						     (UTC): 0430

Advice for remote participants:

Please keep your microphones muted and your video transmissions
disabled during the plenaries and working group sessions, unless
or until invited to respond by the chair of the session.

Vat users can disable reception of accidental sources of audio
multicasts (such as people who forget to mute their mics) by
clicking in the box next to that source's name.



From rem-conf Fri Dec 04 06:44:40 1998 
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Date: Fri, 04 Dec 1998 09:26:03 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Subject: Re: RTP Multiplexing and Sequence Numbers
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Neal Schneider wrote:
> 
> I wasn't intending to insert noise packets at the mux point.  I figured
> that the source would generate these packets.  Again, I believe that
> manipulation at the mux-demux engines should be minimal.  Yes, this
> defeats some of the intentions of efficient silence suppression, but
> can't a payload format be specified (small field), in which the actual
> noise payload be recreated on the demux or end-point based on a a noise
> level negotiated out of band?

Well, this would then require the end systems to know about the mux.

> 
> Also, One might assume that bandwidth utilization would not be much of
> an issue from the source to the mux point, considering that multiplexing
> is inserted at points of aggregation.

I'm not so sure how often this assumption is true. It seems to me that
access bandwidth is quite frequently what is at a premium. In this case,
the utilization of the access links to the gateways may be important. In
any case, it still seems easier and less expensive to just send the
timestamp as currently defined.

> 
> I want a 32-bit mux header which contains all the necessary information
> for retaining a RTP session.   Non-dynamic information outside this
> header format would like to be negotiated out-of-band.

For mid-network mux, I don't see how this is possible. In my mind, the
model is that of gateways from many different providers/domains, feeding
packets into the mux. There is no special signaling between the gateways
and mux, or mux and demux points. The gateways an muxes will need to be
configured with each others addresses. The gateways will also need to
know which mux to send packets to for a given destination IP address.
One could design protocols to automate this, but then we'd be talking
about RTP routing protocols. If this model is not what you have in mind,
please say so.

Anyway, in this model, there is no correlation between the field values
in the headers across packets being muxed. How, then, can you expect to
compress to 32 bits per mux packet, and still retain all of the
information? Perhaps you are suggesting to modify everything - rewrite
timestamps, SN's, etc. to force them to be correlated? Sort of a
man-in-the-middle attack kind of thing. Even in such a scenario, its not
clear how much you will be able to change, and what the impacts will be
on existing gateways. 


-Jonathan R.


-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Fri Dec 04 10:53:19 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
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>> J Rosenberg:
>> in which the actual
>> noise payload be recreated on the demux or end-point based on a a 
noise
>> level negotiated out of band?
>
>Well, this would then require the end systems to know about the mux.
>

I am not sure if I follow.  Isn't there a standard comfort noise level 
payload format for the RTP?  As long as the demux engine can recreate a 
compatible RTP packet (retaining required information), then the 
multiplexing should be transparent to the end user.

>> 
>> Also, One might assume that bandwidth utilization would not be much 
of
>> an issue from the source to the mux point, considering that 
multiplexing
>> is inserted at points of aggregation.
>
>I'm not so sure how often this assumption is true. It seems to me that
>access bandwidth is quite frequently what is at a premium. In this 
case,
>the utilization of the access links to the gateways may be important. 
In
>any case, it still seems easier and less expensive to just send the
>timestamp as currently defined.
>

Service Access Points are most certainly bandwidth bottlenecks.  That is 
why you must multiplex at these points (trunk interfaces, etc.)

About the timestamps, please help me understand what the issues are with 
this philosophy:

Forget about dropping packets during silence for a minute, let's assume 
that can be dealt with in other ways.  If the sampling rate and  
generation of source RTP packets is at a _constant_ timing interval 
(same bitrate unless negotiated out of band), can not all the 
information necessary for performing all RTCP, jitter, and FEC 
calculations be performed correctly using solely the sequence numbers. I 
know I'm beating this issue into the ground, I just want to make certain 
that timestamps are necessary.  

>For mid-network mux, I don't see how this is possible. In my mind, the
>model is that of gateways from many different providers/domains, 
feeding
>packets into the mux. There is no special signaling between the 
gateways
>and mux, or mux and demux points. The gateways an muxes will need to 

I believe that MGCP begins to define inter-gateway signaling in this 
manner.

be
>configured with each others addresses. The gateways will also need to
>know which mux to send packets to for a given destination IP address.
>One could design protocols to automate this, but then we'd be talking
>about RTP routing protocols. If this model is not what you have in 
mind,
>please say so.
>

An RTP routing protocol sounds like a good idea.  

>Anyway, in this model, there is no correlation between the field values
>in the headers across packets being muxed. How, then, can you expect to

This is true.  In this model, there is no need for the RTP header, 
merely RTP muxed headers.  The timestamps/sequence numbers have no 
correlation whatsoever, so why retain an overall header for the group.  

>compress to 32 bits per mux packet, and still retain all of the
>information? Perhaps you are suggesting to modify everything - rewrite
>timestamps, SN's, etc. to force them to be correlated? Sort of a
>man-in-the-middle attack kind of thing. Even in such a scenario, its 
not
>clear how much you will be able to change, and what the impacts will be
>on existing gateways. 

I don't want to force any correlation.  I just want optimize a header 
format for a specific application of the RTP.

Regards, 
Neal A. Schneider

______________________________________________________
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From rem-conf Fri Dec 04 10:56:37 1998 
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Date: Fri, 4 Dec 1998 10:54:11 PST
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <199812041854.KAA06302@mango.parc.xerox.com>
To: fenner@parc.xerox.com
Subject: Announcing mtrace 5.2
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There aren't any spectacular new features; lots of bug fixes and some
fairly minor changes.  The most interesting fixes/features are:

- No more "Unknown protocol 8"!
- No more defaulting to 224.2.0.1; the default group is now 0.0.0.0 .
  Older versions of IOS can't handle this default so you may have to
  specify a group if you see a "No route" from an older IOS router in
  the middle of the path.

The README is attached.

  Bill


$Id: README.mtrace,v 5.2 1998/12/04 05:04:15 fenner Exp $

	Multicast Traceroute

	Release 5.2
	December 3, 1998

	available from ftp.parc.xerox.com,
	file pub/net-research/ipmulti/mtrace-5.2.tar.gz
	binaries pub/net-research/ipmulti/mtrace-5.2-sparc-sunos41x.tar.gz
	         pub/net-research/ipmulti/mtrace-5.2-sparc-solaris2.tar.gz
	         pub/net-research/ipmulti/mtrace-5.2-alpha-osf1.tar.gz
	         pub/net-research/ipmulti/mtrace-5.2-sgi-irix.tar.gz
	         pub/net-research/ipmulti/mtrace-5.2-i386-freebsd.tar.gz


The 5.2 release fixes the following bugs:

o 100% loss was reported as "?/nn = --"; switch it back to 100%

o TTL calculation didn't handle IOS's claim that packets with TTL 0 would
  be forwarded.

o Part of the byte-swapping heuristic was removed; it caused more problems
  than it solved.

o Hop names are now printed correctly in passive mode

o The interface list is properly retrieved from the kernel; selecting
  an interface now works correctly on multi-homed 4.3-Tahoe-and-later
  hosts.  It properly skips down interfaces, thanks to John Kay,
  and stops when it finds the right interface, thanks to Mark Foster.

o mtrace no longer depends on trying to figure out how the compiler is
  going to pack an 8-bit and 24-bit integer into a 32-bit value and
  instead explicitly extracts the values from a 32-bit value (tr_rttlqid).

o passive mtrace joins the group on the interface specified with -i
  and now always prints * * * for the query source since it's unknown
  (thanks to Dave Thaler)

The 5.2 release has the following new features:

o New option -L <n> to only print stats when loss is greater than n%

o New option -f <n> to start a hop-by-hop trace at hop n

o Workaround for IOS bug CSCdi68628 which causes incorrect routers to
  reply to 3rd-party mtrace requests

o Options RAW_INPUT_IS_RAW and RAW_OUTPUT_IS_RAW for compilation on
  Linux and OpenBSD

o New routing protocol codes

o Passive mode now ignores most duplicate responses (instead of treating
  them as individual responses and printing out multiple stats runs
  at once)

o Workaround for some routers returning integer seconds instead of
  in units of 1/65535th as the spec says

o The default group of 224.2.0.1 has been removed; the default is now
  a "weak" mtrace (i.e. group 0.0.0.0).  This can cause problems with
  older versions of IOS; if you see "No route" where it doesn't seem
  to make sense, try the trace again specifying a group.



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From rem-conf Fri Dec 04 23:33:14 1998 
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From rem-conf Sat Dec 05 06:57:30 1998 
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Date: Sat, 5 Dec 1998 06:49:01 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: Neal Schneider <nschneid@hotmail.com>
cc: rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
In-Reply-To: <19981204184405.11188.qmail@hotmail.com>
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On Fri, 4 Dec 1998, Neal Schneider wrote:
> 
> Forget about dropping packets during silence for a minute, let's assume 
> that can be dealt with in other ways.  If the sampling rate and  
> generation of source RTP packets is at a _constant_ timing interval 
> (same bitrate unless negotiated out of band), can not all the 
> information necessary for performing all RTCP, jitter, and FEC 
> calculations be performed correctly using solely the sequence numbers. I 
> know I'm beating this issue into the ground, I just want to make certain 
> that timestamps are necessary.  

Stop!  I can't take it anymore!  Yes, you are right, if the duration
of each packet is the same and a packet is sent in every interval,
then the sequence number and timestamp are linearly related (given
modulo arithmetic).  If you don't maintain state between the two ends
to allow reconstruction of the timestamp to the same value as went in,
then it won't be possible to use the RTCP synchronization mechanisms
to sync two streams, say audio and video, together.  But you probably
don't want to send RTCP at all anyway because that's just going to
waste more bandwidth.

It's not a big surprise that if you discard many of the assumptions
underlying the design of RTP that it no longer seems like a good fit.
What you really want is a circuit-switched TDM hierarchy, and guess
what, it's already available!

> An RTP routing protocol sounds like a good idea.  

No, it is a TERRIBLE idea!  Why should we redesign and reconstruct at
the application level all of what has been built at the IP layer over
many years?

It seems to me that you are missing the whole idea of packet
switching.  It allows traffic to be bursty without wasting channel
bandwidth.  Voice is one example of bursty traffic, often estimated as
40% of the (full duplex) channel bandwidth utilization.  The use of
silence suppression is how we get back the efficiency to offset the
header overhead.  And packet switching provides the straightforward
multiplexing of all kinds of traffic at significantly different rates
without having to construct separate mechanisms for each kind of
traffic.

In the not-to-distant future when the voice traffic is just a blip
next to the data traffic, none of this standing on one's head
(e.g. multiplexing and routing at the application level) is going to
matter.  One can only hope that we haven't cemented in place too much
crap during the transition.
							-- Steve





From rem-conf Sun Dec 06 03:03:14 1998 
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Date: Sun, 6 Dec 1998 12:30:46 +0200 (EET)
From: Khaled Elsayed <kelsayed@alpha1-eng.cairo.eun.eg>
Reply-To: Khaled Elsayed <khaled@ieee.org>
To: tccc@ieee.org, a.jajszczyk@ieee.org,
        Laurent Toutain <Laurent.Toutain@enst-bretagne.fr>
Cc: atm97authors@inesc.pt, atm96@inesc.pt, ActiveNets_Wire@ittc.ukans.edu,
        issll@mercury.lcs.mit.edu, xtp-relay@cs.concordia.ca,
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        ieee.announce@bellcore.com, atm@sun.com
Subject: Call for Papers and Reviewers: IEEE Comm. Magazine
In-Reply-To: <199812021624.RAA04156@pacman>
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Sorry if you get multiple copies. Please distribute to your colleagues.
Thanks,

Khaled

--
Dr. Khaled Fuad Elsayed
Department of Electronics and Communications Engineering
Faculty of Engineering, Cairo University
Giza, Egypt 12613
E-mail: khaled@ieee.org
Fax: +202-572-3486
Phone: +202-567-8850

----------------------------- Cut Here ---------------------------------- 

                      Call for Papers and Reviewers
                       IEEE Communications Magazine
                         Internet Technology Series
Series Editors: Laurent Toutain (ENST Bretagne) and
                Khaled Elsayed (Cairo University)


The IEEE Communications Magazine invites contributions from authors with 
academic and industrial affiliation for the Internet Technology Series. 
The Focus of the Internet Technology Series is to bring the latest advances 
in state of the art techniques in the Internet to the large readers 
group of the Communications Magazine. Preference will be given to papers 
addressing technical issues that are encountered in implementing current or
future Internet products or solutions.

We are currently looking for articles in the following areas:
- Managing the flows and QoS support in the Internet
- Web Caching
- Terabit Routing
- Internet Telephony and VOIP

Article Style Information
=========================
Articles should be tutorial in nature and should be written in a style
comprehensible to readers outside the specialty of the article.

Articles may be edited for content, and will be copyedited for
compliance with the magazine's style guidelines. Page proofs will be
sent to the contact author for final 
review prior to publication.

Mathematical equations should not be used unless they are vital to the
presentation. Even then, they should be kept to a minimum. If the
article has numerous equations, please contact the editor handling the
manuscript.

References should be included only to guide readers to more information
on the topic; the reference list should not include every available
source (a limit of ten references is recommended). Use footnotes only 
where necessary.  

Articles should not exceed 4500 words. 

Figures and tables should be limited to a combined total of six. If the
article exceeds these recommended limits, please contact the editor
handling the manuscript.

Also check:
http://207.127.135.8/ci1 (author guidelines)
and
http://207.127.135.8/ci1/public/1998/may/cimess.html


Since this is an Internet related series, we would like to handle all the submissions and the reviewing process via the Internet. Send your submission in postscript or PDF (acrobat) format to the series editors. 

Khaled Elsayed: Khaled@ieee.org

Laurent Toutain: Laurent.Toutain@enst-bretagne.fr

In the mean time we would to ask for the help in evaluating the submitted
articles. If you are willing to provide high-quality timely reviews,
kindly send your name, affiliation, e-mail, fax, mailing address and
research interests to both the series editors. Reviewers will be
acknowledged by listing their names in the December issue (starting 1999)
and the IEEE Communications Magazine Home Page. Please note that we intend
to handle the whole review process electronically so you need to be able
to handle postscript and pdf e-mail attachments. 







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From rem-conf Mon Dec 07 01:04:44 1998 
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Date: Mon, 07 Dec 1998 01:00:54 -0800
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Organization: EcoSystems
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remove

Sorry to bug everyon. I tried to cancel the standard way twice and it
didn't work for me. It's been interesting reviewing what you all are
doing.  Keep up the good work!

Ron Swenson



From rem-conf Mon Dec 07 07:09:24 1998 
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From rem-conf Mon Dec 07 09:39:12 1998 
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To: end2end-interest@isi.edu, ir-l%uccvma.bitnet@cunyvm.cuny.edu,
        rem-conf-request@es.net, rem-conf@es.net, sound@PASCAL.acm.org,
        tccc@cs.umass.edu
Subject: New NSF Research Focus
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Folks: I would like to draw your attention to a new research focus on
the topic of UNIVERSAL ACCESS jointly sponsored by the HCI and KCS
programs within the Information and Intelligent Systems (IIS) Division
of CISE.

The word "access" implies the ability to find, manipulate and use
information in an efficient and comprehensive manner. A primary objective
of the HCI/KCS research focus on universal access is to empower people
with disabilities so that they are able to participate as first class
citizens in the emerging information society. But more than that, the
research focus will benefit the nation as a whole, by advancing computer
technology so that all people can possess the skills needed to fully
harness the power of computing to enrich and make their lives more
productive within a tightly knit "national family" whose members
communicate naturally and painlessly through the sharing of (multimodal)
information.

For more information, please point your web browser at

	http://www.cise.nsf.gov/iis/index.html

then click on "IIS Programs" and from there go to either the HCI or KCS
page, where you will find a link to Universal Access.

Please feel free to contact me with any questions!


===============================================================
Ephraim P. Glinert, Program Director
Knowledge and Cognitive Systems / HCI / Universal Access
Division of Information and Intelligent Systems
National Science Foundation
4201 Wilson Blvd., Suite 1115
Arlington, VA  22230

Phone: (703) 306 1926
Fax:      (703) 306 0599
E-mail:  eglinert@nsf.gov
===============================================================



From rem-conf Mon Dec 07 10:49:32 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: nschneid@hotmail.com, casner@cisco.com
Cc: rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
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>It's not a big surprise that if you discard many of the assumptions
>underlying the design of RTP that it no longer seems like a good fit.
>What you really want is a circuit-switched TDM hierarchy, and guess
>what, it's already available!
>

For convergence purposes, I am just saying is that RTP could be better 
tailored to handle voice.

>> An RTP routing protocol sounds like a good idea.  
>
>No, it is a TERRIBLE idea!  Why should we redesign and reconstruct at
>the application level all of what has been built at the IP layer over
>many years?
>

Point well taken.  If voice is to be treated the same as all other data 
on the network, there is no reason to supersede layer 3 routing.  My 
point being, once a call is set up there is consistent bursts of packets 
for a relatively long connection period.   Setting up virtual paths for 
these sessions does not seem totally out the question.

>In the not-to-distant future when the voice traffic is just a blip
>next to the data traffic, none of this standing on one's head
>(e.g. multiplexing and routing at the application level) is going to
>matter.  One can only hope that we haven't cemented in place too much
>crap during the transition.
>							-- Steve

64kbps is comparatively even by today's standards.  Bandwidth will 
continue to be less and less of an issue, but consider the case of a 
network service provider who is now handling all of the voice traffic 
for the area.  You don't think that higher efficiency matters?  You 
think Cellular Phone services today aren't standing on their heads to 
find a more efficient means? 

Forget bandwidth.  There has to be necessary means to reduce latency for 
voice to sound decent.  Granted, some of that will come from  routing 
priorities, increased bandwidth, etc, but there has to be additional 
methods taken at the application/gateways in terms of sampling sizes, 
buffering, header manipulation, etc, to make a significant reduction.   
A latency across the country of 30 ms doesn't mean much when there is 
150 ms added from the application, buffering, etc, for voice in order to 
force some efficiency in the current RTP structure.

Regards, 
Neal A. Schneider

______________________________________________________
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From rem-conf Mon Dec 07 12:22:48 1998 
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Date: Mon, 7 Dec 1998 12:17:50 -0800 (PST)
From: Stephen Casner <casner@cisco.com>
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Subject: Revised AVT agenda
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To accommodate the shift of AVT's second session to Friday morning,
we've rearranged the agenda a bit because there were conflicts with
the MEGACO BOF.  The revised agenda follows.
						-- Steve & Colin



                            Audio/Video Transport WG

                                 A G E N D A

Thursday, December 10 at 1300-1500
==================================

 - Introduction                                                          5

 - Documents published since last meeting
        RFC2429 - H.263+ payload format
        RFC2431 - BT.656 payload format
        RFC2435 - JPEG video payload format
	RFC2448 - AT&T's Error Resilient Video Transmission Technique

 - Transport of DTMF tones                              (Petrack)       20
        draft-ietf-avt-tones-00.txt
        draft-ietf-avt-dtmf-01.txt

 - RTP multiplexing proposals
        draft-ietf-avt-muxissues-00.txt                 (Rosenberg)     15
        draft-ietf-avt-mux-rtp-00.txt                   (Subbiah)       10
        draft-tanigawa-rtp-multiplex-01.txt             (Hoshi)          5
        draft-ietf-avt-germ-00.txt                      (Handley)        5
   Discussion                                                           20

 - Transport of MPEG4 streams                           (Civanlar)      20

 - Generic payload format
        draft-guillemot-genrtp-00.txt                   (Guillemot)     20


Friday, December 11 at 0900-1130
================================

 - Update to RTP
        draft-ietf-avt-rtp-new-02.txt, .ps              (Casner)        10
        draft-ietf-avt-rtpsample-01.txt                 (Rosenberg)      5
        draft-parnes-rtp-sdes-00.txt                    (Casner)         5

 - Update to RTP audio/video profile
        draft-ietf-avt-profile-new-04.txt, .ps          (Casner)        20

 - Documents to act upon
        draft-ietf-avt-crtp-05.txt
        draft-ietf-avt-rtp-mib-02.txt                   (Baugher)       10
        draft-ietf-avt-rtp-format-guidelines-01.txt     (Handley)        5

 - Payload formats for loss tolerant media
        draft-ietf-avt-fec-04.txt                       (Rosenberg)      5
        draft-ietf-avt-reedsolomon-00.txt               (Rosenberg)      5
        draft-ietf-avt-interleaving-00.txt              (Perkins)        5

 - RTP Payload Format for PureVoice(tm) Audio
        draft-mckay-qcelp-01.txt

 - Charter update                                       (Perkins)       15




From rem-conf Mon Dec 07 16:32:54 1998 
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To: Neal Schneider <nschneid@hotmail.com>
cc: casner@cisco.com, rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
In-reply-to: Your message of "Mon, 07 Dec 1998 10:43:32 PST." <19981207184333.3677.qmail@hotmail.com>
Date: Tue, 08 Dec 1998 00:25:37 +0100
Message-ID: <9851.913076737@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Neal Schneider writes:
>64kbps is comparatively even by today's standards.  Bandwidth will 
>continue to be less and less of an issue, but consider the case of a 
>network service provider who is now handling all of the voice traffic 
>for the area.  You don't think that higher efficiency matters?  You 
>think Cellular Phone services today aren't standing on their heads to 
>find a more efficient means? 
>
>Forget bandwidth.  There has to be necessary means to reduce latency for 
>voice to sound decent.  Granted, some of that will come from  routing 
>priorities, increased bandwidth, etc, but there has to be additional 
>methods taken at the application/gateways in terms of sampling sizes, 
>buffering, header manipulation, etc, to make a significant reduction.   
>A latency across the country of 30 ms doesn't mean much when there is 
>150 ms added from the application, buffering, etc, for voice in order to 
>force some efficiency in the current RTP structure.

But the 150ms latency at the application comes from the requirement to
smooth jitter introduced by the network, not because of the overhead of
RTP. If you have a quality IP network, the application will adapt its
playout buffers to be small, and one can achieve very low latencies.

There are issues with low-efficiency when sending many RTP streams, but I
don't believe that these contribute to latency (in fact multiplexing to
increase efficiency is likely to _increase_ both latency and jitter).

-- 
Colin Perkins                   Email: c.perkins@cs.ucl.ac.uk
Department of Computer Science  Phone: +44 171 419 3666
University College London       WWW  : http://www.cs.ucl.ac.uk/staff/c.perkins/



From rem-conf Mon Dec 07 17:06:50 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: nschneid@hotmail.com, C.Perkins@cs.ucl.ac.uk
Cc: casner@cisco.com, rem-conf@es.net
Subject: Re: RTP Multiplexing and Sequence Numbers
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>-- 
>Colin Perkins    
>
>But the 150ms latency at the application comes from the requirement to
>smooth jitter introduced by the network, not because of the overhead of
>RTP. If you have a quality IP network, the application will adapt its
>playout buffers to be small, and one can achieve very low latencies.
>
>There are issues with low-efficiency when sending many RTP streams, but 
I
>don't believe that these contribute to latency (in fact multiplexing to
>increase efficiency is likely to _increase_ both latency and jitter).
>
          
Adaptive Jitter buffers are important in reducing latency.  Jitter will 
become less of an issue when the internet is better adapted for 
real-time traffic (i.e. "better than best effort" service), and is less 
of an issue today with intranets, controlled networks. 

The latencies which one cannot avoid with gateways are algorithmic and 
packetization delays.  Granted if one really wants compression, the 
algorithmic delay is unavoidable, but in the case of uncompressed G.711, 
the packetization delay can be substantially reduced. 

If one were to send 4ms samples of PCM, instead of the typical 30ms 
sample size, the latency would automatically drop substantially.  
Unfortunately, header overhead becomes a large problem with small sample 
sizes therefore this approach is not taken.  But given a standard, 
powerful multiplexing scheme, small sample sizes may become a reality, 
decreasing the end-to-end delay.

Regards, 
Neal A. Schneider

>But the 150ms latency at the application comes from the requirement to
>smooth jitter introduced by the network, not because of the overhead of
>RTP. If you have a quality IP network, the application will adapt its
>playout buffers to be small, and one can achieve very low latencies.
>
>There are issues with low-efficiency when sending many RTP streams, but 
I
>don't believe that these contribute to latency (in fact multiplexing to
>increase efficiency is likely to _increase_ both latency and jitter).
>
>-- 
>Colin Perkins                   Email: c.perkins@cs.ucl.ac.uk
>Department of Computer Science  Phone: +44 171 419 3666
>University College London       WWW  : 
http://www.cs.ucl.ac.uk/staff/c.perkins/


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remove

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From rem-conf Tue Dec 08 12:57:02 1998 
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From rem-conf Tue Dec 08 18:48:15 1998 
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From rem-conf Tue Dec 08 21:28:13 1998 
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the usenix lisa (system administration) conference in boston this week,
9:00am wednesday dec 9 thru 5:00pm friday dec 11, will broadcast selected
sessions.  the aproximate program follows:

	9:00am wed, dec 9	keynote, eric allman, sendmail inc.
	11:00			security session, 3 papers
	2:00pm			scaling a network, brent chapman, covad comm.
	4:00pm			storage performance, 3 papers

	9:00am thurs, dec 10	an adult web site, dan klein, cybertainment
	2:00pm			mailer wars (sendmail, qmail, exim)
	4:00pm			sysadmin certification debate

	9:00am friday, dec 11	LISA/NT conference overview
	11:00am			new thoughts and evolution, 3 papers
	2:00pm			palm pilot magic, barb dijker, labyrinth
	4:00pm			LISA quiz show closing session, rob kolstad

-evi



From rem-conf Thu Dec 10 05:11:59 1998 
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DATE: 10 Dec 98 3:43:09 AM
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SUBJECT: Maximize Your Website's Traffic. Increase Your Search Engine Rank!
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From rem-conf Thu Dec 10 11:36:47 1998 
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Sirs,

Does anyone know how NetMeeting computes is
(H.225) ConferenceIdentifier of ASN.1 type conferenceID
in the Setup message ?

I have been writng my own H.323 application which is
supposed to connect to NetMeeting.

Best Regards
Norbert Rossello





From rem-conf Fri Dec 11 04:21:32 1998 
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To: casner@cisco.com, rem-conf@es.net
Message-Id: <422566D7.00431C79.00@hrlsmtp.haifa.ibm.com>
Date: Fri, 11 Dec 1998 14:10:28 +0200
Subject: RTP Proposal - comparison
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X-Loop: rem-conf@es.net
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Following the avt discussion earlier today, we were requested to list
scenarios for using RTP-multiplexing.

My main interest is audio IP-Telephony, but I think that the RTP
multiplexing applies in general to implementations that require real-time
(or at least quick enough) response on one hand, and have a relatively
short payload on the other hand (and thus the headers, IP/UDP and possibly
the RTP, become a significant overhead).
Scott's signalling proposal (if adopted) could become a candidate for using
this mechanism too since it falls into the category above.

I will use the G.723.1 codec for comparison of the different proposals.
For G.723 I assume in the calculation 24 bytes per payload (generated every
30 msec).
The results may change a little based on the payload size for other codecs,
but I do not think that it will result in a dramatic change of the picture.

Before I go on with the numbers, I would suggest that a higher weight
should be given to scenarios that use a relatively short payload, rather
than those who use large ones (which may be the case with Video and Data),
since the main advantage in RTP multiplexing is for applications that have
short payload and thus the header overhead is their main concern.

I will assume X simultaneous calls, all using the same codec and payload
length (just for simplicity of the calculation) - I think that this
assumption is reasonable, and should not distort the general picture we get
>from the comparison.

PFS stands hereby for Payload Frame Size, and eventually will be given the
value 24 (per the above codec assumption).

Hitachi proposal:
--------------------------
Advantages:  Very simple to implement, do not loose any information
originally in the
                 RTP header (since it is multiplex as is).
Bandwidth Requirements:
     Without RTP Multiplexing = (40+PFS)*X
     With RTP Multiplexing       = 28 + (12+PFS)*X      (28 stands for
IP+UDP headers).

Asymptotic Bandwidth reduction (large X): 1- (12+PFS)/(40+PFS)
Assuming PFS=24:
     Asymptotic Bandwidth reduction: 1- (36)/(64)= 43 %


GeRM Proposal:
_______________

Advantage: All the information in the original RTP header can be
reconstructed, High compression ratio.

Bandwidth Requirements:
     Without RTP Multiplexing = (40+PFS)*X
     With RTP Multiplexing       = 28+12+ (1+ 1+PFS)*X               ( The
first '1' stands for the GeRM 1-byte Header,
                                   the second '1' stands for payload type
for the first block,
                                   and for the LSB of the SSRC which may be
needed due to                                 silence   characteristics for
the other blocks (we could even                                   use 0.5
byte on   average as a worse case) ).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)= 59 %


Bell Proposal:
____________
Advantage: High compression ratio

Bandwidth Requirements:
     Without RTP Multiplexing = (40+PFS)*X
     With RTP Multiplexing       = 28+12+ (2+PFS)*X     (The '2' stands for
the minimal mini-header per block,
                                   i.e  no need for 16-bit length).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)= 59 %

Since the CC field is not in the mini-header it may imply that no CSRC
information is transferred - which may be a disadvantage in some
centralized conferencing scenarios where we may have several contributor
sources to the same payload (and wish to identify them in the application).
Also timestamp information may be of importance, but do not fit into the
mini header.

Nokia Proposal
______________
Advantage: High compression ratio

Bandwidth Requirements:
     Without RTP Multiplexing = (40+PFS)*X
     With RTP Multiplexing       = 28+ (2+PFS)*X   (The '2' stands for the
minimal mini-header per block,
                                   i.e  no need for 16-bit length).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)= 59 %

Again, since the CC field is not in the mini-header it may imply that no
CSRC information is transferred, which may be a disadvantage in some
centralized conferencing scenarios where we may have several contributions
to the same payload.
Also timestamp information may be of importance, but do not fit into the
mini header. This proposal limits the length field is to only 5 bits (which
may be a limitation for payload scenarios that are slightly longer than 31
bytes, and could still benefit from multiplexing).

=========================

My Perspective:
_______________

I think that all the four proposals are basically good ones, but we need to
go forward.
It seem that three of the proposals fit into the 'high bandwidth reduction'
category, while the Hitachi fits into a 'medium bandwidth reduction
category'.
Two of the 'high bandwidth reduction' proposals are lossy in respect to the
RTP header information, and thus may limit their use in comparison to the
GeRM and Hitachi's proposal that are 'lossless' in that respect.

I would suggest to proceed with two of the proposals: GeRM and Hitachi,
since they both don't loose RTP header information, and represent the two
ends of simplicity vs. high compression.
Once the GeRM proposal is completed (like ID negotiation issue), we may
decide which of the two is preferred.


As for the RTCP issue. I think that the information is still valid in
respect to the 'global' packet, and can be associated
with the timestamp of the first block in each multiplexed packet.

Best Regards,
-- Gal.


-------------------------------------------------------
Gal Ashour
Audio/Video Group
IBM Research - Haifa Research Lab.
ISRAEL

ashour@haifa.vnet.ibm.com
-------------------------------------------------------





From rem-conf Fri Dec 11 04:48:26 1998 
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To: rem-conf@es.net
Message-Id: <422566D7.004614CF.00@hrlsmtp.haifa.ibm.com>
Date: Fri, 11 Dec 1998 14:42:55 +0200
Subject: RTP Proposal - comparison
X-Mailing-List: <rem-conf@es.net> 
X-Loop: rem-conf@es.net
Precedence: list




Following the avt discussion earlier today, we were requested to list
scenarios for using RTP-multiplexing.

My main interest is audio IP-Telephony, but I think that the RTP
multiplexing applies in general to implementations that require real-time
(or at least quick enough) response on one hand, and have a relatively
short payload on the other hand (and thus the headers, IP/UDP and possibly
the RTP, become a significant overhead).
Scott's signalling proposal (if adopted) could become a candidate for using
this mechanism too since it falls into the category above.

I will use the G.723.1 codec for comparison of the different proposals.
For G.723 I assume in the calculation 24 bytes per payload (generated every
30 msec).
The results may change a little based on the payload size for other codecs,
but I do not think that it will result in a dramatic change of the picture.

Before I go on with the numbers, I would suggest that a higher weight
should be given to scenarios that use a relatively short payload, rather
than those who use large ones (which may be the case with Video and Data),
since the main advantage in RTP multiplexing is for applications that have
short payload and thus the header overhead is their main concern.

I will assume X simultaneous calls, all using the same codec and payload
length (just for simplicity of the calculation) - I think that this
assumption is reasonable, and should not distort the general picture we get
>from the comparison.

PFS stands hereby for Payload Frame Size, and eventually will be given the
value 24 (per the above codec assumption).

Hitachi proposal:
--------------------------
Advantages:  Very simple to implement, do not loose any information
originally in the
                 RTP header (since it is multiplex as is).
Bandwidth Requirements:
     Without RTP Multiplexing = (40+PFS)*X
     With RTP Multiplexing       = 28 + (12+PFS)*X      (28 stands for
IP+UDP headers).

Asymptotic Bandwidth reduction (large X): 1- (12+PFS)/(40+PFS)
Assuming PFS=24:
     Asymptotic Bandwidth reduction: 1- (36)/(64)= 43 %


GeRM Proposal:
_______________

Advantage: All the information in the original RTP header can be
reconstructed, High compression ratio.

Bandwidth Requirements:
     Without RTP Multiplexing = (40+PFS)*X
     With RTP Multiplexing       = 28+12+ (1+ 1+PFS)*X               ( The
first '1' stands for the GeRM 1-byte Header,
                                   the second '1' stands for payload type
for the first block,
                                   and for the LSB of the SSRC which may be
needed due to                                 silence   characteristics for
the other blocks (we could even                                   use 0.5
byte on   average as a worse case) ).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)= 59 %


Bell Proposal:
____________
Advantage: High compression ratio

Bandwidth Requirements:
     Without RTP Multiplexing = (40+PFS)*X
     With RTP Multiplexing       = 28+12+ (2+PFS)*X     (The '2' stands for
the minimal mini-header per block,
                                   i.e  no need for 16-bit length).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)= 59 %

Since the CC field is not in the mini-header it may imply that no CSRC
information is transferred - which may be a disadvantage in some
centralized conferencing scenarios where we may have several contributor
sources to the same payload (and wish to identify them in the application).
Also timestamp information may be of importance, but do not fit into the
mini header.

Nokia Proposal
______________
Advantage: High compression ratio

Bandwidth Requirements:
     Without RTP Multiplexing = (40+PFS)*X
     With RTP Multiplexing       = 28+ (2+PFS)*X   (The '2' stands for the
minimal mini-header per block,
                                   i.e  no need for 16-bit length).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)= 59 %

Again, since the CC field is not in the mini-header it may imply that no
CSRC information is transferred, which may be a disadvantage in some
centralized conferencing scenarios where we may have several contributions
to the same payload.
Also timestamp information may be of importance, but do not fit into the
mini header. This proposal limits the length field is to only 5 bits (which
may be a limitation for payload scenarios that are slightly longer than 31
bytes, and could still benefit from multiplexing).

=========================

My Perspective:
_______________

I think that all the four proposals are basically good ones, but we need to
go forward.
It seem that three of the proposals fit into the 'high bandwidth reduction'
category, while the Hitachi fits into a 'medium bandwidth reduction
category'.
Two of the 'high bandwidth reduction' proposals are lossy in respect to the
RTP header information, and thus may limit their use in comparison to the
GeRM and Hitachi's proposal that are 'lossless' in that respect.

I would suggest to proceed with two of the proposals: GeRM and Hitachi,
since they both don't loose RTP header information, and represent the two
ends of simplicity vs. high compression.
Once the GeRM proposal is completed (like ID negotiation issue), we may
decide which of the two is preferred.


As for the RTCP issue. I think that the information is still valid in
respect to the 'global' packet, and can be associated
with the timestamp of the first block in each multiplexed packet.

Best Regards,
-- Gal.


-------------------------------------------------------
Gal Ashour
Audio/Video Group
IBM Research - Haifa Research Lab.
ISRAEL

ashour@haifa.vnet.ibm.com
-------------------------------------------------------







From rem-conf Fri Dec 11 05:30:05 1998 
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remove



From rem-conf Fri Dec 11 09:05:38 1998 
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Date: Fri, 11 Dec 1998 12:00:51 -0500 (EST)
From: Chinmay Padhye <padhye@eng.usf.edu>
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To: rem-conf@es.net
Subject: Congestion Control.
Message-ID: <Pine.GSO.4.05.9812111150470.22313-100000@sunblast>
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Hello:

	I am a graduate student at the University Of South Florida,
Tampa, FL. I am interested in the area of transporting voice over IP. I
have looked at a few papers on congestion control. However these papers
mainly deal with congestion control in streaming applications and do not
look at congestion control in interactive applications. 

	Can anyone point to me to a few papers on congestion control in
interactive voice applications. 

	
Thanks in advance...


Chinmay V. P.
Graduate Student.
Department of Computer Science and Engineering.
University Of South Florida.




From rem-conf Fri Dec 11 09:05:48 1998 
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Hello my community, 

Hello my friends both nearby and overseas; Coworkers both old and new.  I also greet my colleagues and leaders in this fast, vibrant industry. Through working and playing with all of you I have felt the admiration, inspiration and motivation needed to kee
p me going. 

For three and a half years I've had the privilege to serve as the Java Evangelist for Sun Microsystems. This role has taught me what it means to run fast and "run everywhere". The early Java team showed me what kind of incredible creative energy, talent, 
intelligence, persistence, humor, and insight it takes to bring a great new technology into the world. I'm grateful for that time. This is why I include a measure of sadness as well as excitement and hope among the feelings as I close this chapter of my c
areer. 

I am leaving Sun but of course I am not leaving Java. I am launching a Silicon Valley Java startup tentatively called BizTone-a subsidiary of Madura.net (http://www.madura.net). They are innovators in their use of the latest Java technologies including Ji
ni and JFC/Swing. And of course they have the killer app. Using Java to deliver the power of Fortune 500 Enterprise Resource Management to "the rest of us" (small and mid-sized businesses) over network or even phone lines.

I feel confident leaving the official duties of Java Evangelism especially at a time when America OnLine has announced its commitment to Java client browsers and hardware. Coupled with the announcement of the release of the Java 2 class libraries and the 
preliminary injunction against Microsoft, I can retire knowing that tens of millions of compatible clients are assured. Having evangelized some 100,000 people I can rest with the knowledge that I contributed in some way to the Java revolution. I am not cl
aiming that the Java story is complete, merely that these trends represent
the end of an important stage in Java's evolution. Finally, the
announcement of community source models for JDK suggest a natural
evolution of Java towards a open but compatible conclusion.

The richest, most meaningful events in one's life are those which are reflected in our relationships with others. I've had the good fortune to share some of myself with each of you and in turn appreciate what you have offered me. Without the mirror of my 
communities, I would be much the poorer in the knowledge I have of myself. The context of a community is further deepened by a sense of history and continuity. I hope that you will stay in contact with me through this and other life transitions. (Further 
information is provided below.)

Sincerely,
Miko Matsumura
Former Java Evangelist
President, US operations
VP of strategy
Madura.net


How often do you want to hear from Miko?
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At will: If you ever want to check in on me, feel free to email me at miko@miko.com or check my web page at www.miko.com







From rem-conf Fri Dec 11 11:13:50 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: rem-conf@es.net, ASHOUR@HAIFA.VNET.IBM.COM
Subject: Re: RTP Proposal - comparison
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>-- Gal.
>I would suggest to proceed with two of the proposals: GeRM and Hitachi,
>since they both don't loose RTP header information, and represent the 
two
>ends of simplicity vs. high compression.
>Once the GeRM proposal is completed (like ID negotiation issue), we may
>decide which of the two is preferred.

Both simplicity and efficiency are important factors in determining a 
useful multiplexing scheme. A standard solution should take them both 
into consideration and effectively compromise the GeRM and Hitachi 
proposals.  A GeRM header is greatly efficient, but its complexity may 
become a limiting factor in high-capacity gateways.   The Hitachi 
proposal has the least complexity involved, but does not achieve the 
efficiency numbers that make multiplexing a true benifit.  There are a 
few easy modifications to this (Hitachi) simple muxing scheme that would 
assist in efficiency (e.g. replacing the SSRC with user ID's, out of 
band payload identification, etc.)

Regards, 
Neal A. Schneider



>
>
>-------------------------------------------------------
>Gal Ashour
>Audio/Video Group
>IBM Research - Haifa Research Lab.
>ISRAEL
>
>ashour@haifa.vnet.ibm.com
>-------------------------------------------------------
>
>
>
>
>
>
>


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Fri Dec 11 15:37:07 1998 
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Date: Sat, 12 Dec 1998 01:30:19 +0200
Subject: Re: RTP (multiplexing) Proposals - comparison
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Hi Neal,
   I couldn't agree with you more.

Both simplicity and efficiency are important, which is the exact reason
that I suggested to concentrate on the two proposals that represent each of
those two categories (with a possibility of a later merger).
They both also share a common important characteristic, which is not losing
information from the original RTP header.

The Hitachi proposal may well be enhanced in the future,  but it has merit
even as is. The difference between 59% bandwidth reduction and 43% (as is
the case for G.723 under the assumption in my previous posting) is not
negligible, but yet it isn't a huge difference neither.

It may not be such a bad trade-off for its simplicity - but I guess that
this can be decided later on, after GeRM defines the needed means for its
ID and payload determination. Until a robust and reliable out-of-band
mechanism is provided for payload and ID determination, the Hitachi
proposal provides a simple solution that can be already implemented (based
on some well defined preliminary H.323 negotiation).

You may be well right, that once GeRM provides those mechanism, a third
alternative may be to adopt the Hitachi proposal, after incorporating into
it the ID and Payload negotiation schemes developed for GeRM.


Regards,
-- Gal.





"Neal Schneider" <nschneid@hotmail.com> on 11/12/98 21:09:33

To:   rem-conf@es.net, ASHOUR@HAIFA.VNET.IBM.COM
cc:    (bcc: Gal Ashour/IBMHAIFA)
Subject:  Re: RTP Proposal - comparison




>-- Gal.
>I would suggest to proceed with two of the proposals: GeRM and Hitachi,
>since they both don't loose RTP header information, and represent the
two
>ends of simplicity vs. high compression.
>Once the GeRM proposal is completed (like ID negotiation issue), we may
>decide which of the two is preferred.

Both simplicity and efficiency are important factors in determining a
useful multiplexing scheme. A standard solution should take them both
into consideration and effectively compromise the GeRM and Hitachi
proposals.  A GeRM header is greatly efficient, but its complexity may
become a limiting factor in high-capacity gateways.   The Hitachi
proposal has the least complexity involved, but does not achieve the
efficiency numbers that make multiplexing a true benifit.  There are a
few easy modifications to this (Hitachi) simple muxing scheme that would
assist in efficiency (e.g. replacing the SSRC with user ID's, out of
band payload identification, etc.)

Regards,
Neal A. Schneider



>
>
>-------------------------------------------------------
>Gal Ashour
>Audio/Video Group
>IBM Research - Haifa Research Lab.
>ISRAEL
>
>ashour@haifa.vnet.ibm.com
>-------------------------------------------------------
>
>
>
>
>
>
>


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com







From rem-conf Sat Dec 12 22:14:39 1998 
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To: "'rem-conf@es.net'" <rem-conf@es.net>, "'4onip-sys@fzi.de'"
	 <4onip-sys@fzi.de>
Subject: Comments on Generic RTP Payload Format proposal
Date: Sat, 12 Dec 1998 21:57:34 -0800
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I have a couple of minor comments on the Generic RTP Payload Format
proposal, draft-guillemot-genpayload-00.txt, that was presented at the IETF
AVT working group meeting.
 
The payload format supports grouping and fragmentation, of higher level
"application units".  It is also possible to carry arbitrary extension data.
It is through this extension mechanism that the payload format can be made
to support FEC with parity codes, FEC with Reed-Solomon codes, redundancy,
etc.
 
The correct interpretation of the extension data is given by the XT
(Extension Type) header field.  I think the main concern that I heard from
several people at the IETF meeting, was that this introduces an extra
multiplexing point.  To figure out how to interpret an RTP packet, one must
first look at the RTP Payload Type field, and then look at the XT field in
the generic payload format.
 
However, after having looked at the Internet-Draft again, I now think this
might not be such a big issue after all. The reason is that the
Internet-Draft does not specify any static assignments for the XT field.
Rather, it says that the values to be used for the XT field should be
communicated out-of-band, using SDP or similar mechanisms.
 
I would think that in most cases, the extension data field would not be
used, or it would only be used in one single way.  For instance, I think it
is unlikely that some packets in an RTP session would have extension data
for parity FEC, while other packets in the same session would use
Reed-Solomon FEC.
 
So, when SDP indicates that there are 0-1 valid values for the XT field, the
RTP receiver can effectively ignore the field.  Thus, the problem with the
additional multiplexing point becomes a non-issue in the typical case.
 
Given this, one could argue that the XT field should be removed, although
that would give up some flexibility that could be useful in the future.  In
any case, it seems clear that 8 bits is an excessively large size for this
field.
 
I would also like to second Jonathan Rosenberg's comment, that it is better
to specify the format of the extension data field by referring to the
relevant documents, rather than including the text explicitly in this
Internet-Draft.  Of course, there might be cases where no relevant document
exists.  In those cases, it might still be better to publish the format of
the extension data field in a different document, which would be separate
>from the document that specifies the Generic Payload Format.
 
I also have some comments regarding the alignment of the various fields in
the payload format.  The primary concern of any payload format that
implements "grouping" or "multiplexing", is to reduce the number of packets.
The second concern would be to reduce the number of bytes in the payload
format header.  This is quite important when very small application units
are grouped together.  To achieve this second goal, I think most people
would accept giving up word alignment of header fields, as long as the
header fields are still aligned to byte boundaries.
 
In this payload format, however, individual header fields are not byte
aligned.  Padding is inserted at the end of the last header field to ensure
that the payload begins at a byte boundary.  I know that byte alignment is
not an issue in the MPEG world (Just look at the MPEG-4 SL-Packet header
definition and you will be either horrified or delighted, depending on where
you come from) but in the IETF header fields are typically aligned.  
 
I have two different proposals for how the header syntax can be modified.
Both proposals ensure that header fields are byte aligned, but they work a
little bit differently.  The proposals all show that byte alignment can be
achieved at a maximum cost of one byte, sometimes even less.
 
Proposal 1:
 
* Reduce the XT field from 8 bits to 6 bits.  As discussed earlier, it is
hard to conceive of a situation where someone would want more than 64
different kinds of Extension Data in the same RTP session. 

* If E=0, insert a new field with 5 bits, directly following the F field.
The 5 bits are reserved and should always be set to 0.

* The size of the LENGTH field is increased from 13 to 16 bits.
 
This causes all header fields to be aligned with byte boundaries.  No
padding field is necessary.  This comes at a cost, however.  If you look at
example #3 in draft-guillemot-genpayload-00.txt, you might see that the
second header, which consists only of flags and a LENGTH field, will
increase from two bytes to three bytes.  And this is a fairly common case,
because it happens whenever the packet consists of extension data and an
application unit that is not being fragmented. 
 
Proposal 2:
 
* Reduce the XT field from 8 bits to 6 bits.  See comment above.

* If E=1, increase the size of the LENGTH field from 13 to 16 bits.

* If E=0, the size of the LENGTH field remains at 13 bits.

* If G=0 and E=0, the size of TSOFFSET is reduced from 16 to 13 bits.

* If G=1 or E=1, the size of TSOFFSET remains at 16 bits.

* If the FOFFSET field would directly follow the flag fields (because
neither LENGTH nor TSOFFSET is present), insert 5 reserved bits in front of
FOFFSET, instead of adding 5 bits of padding at the end of FOFFSET.
 
This causes all header fields to be aligned with byte boundaries.  And this
comes at no extra cost in terms of the header size.  As a matter of fact,
this scheme saves one entire byte when the payload header only contains the
TSOFFSET field.  (See example #1, in draft-guillemot-genpayload-00.txt.)  Of
course, you may already have noticed that there is another kind of cost
associated with this proposal: complexity.
 
Anders



From rem-conf Sun Dec 13 06:24:43 1998 
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From rem-conf Sun Dec 13 15:28:29 1998 
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Please find enclosed herewith the CFP for NOSSDAV 99. My apologies
in advance if you receive multiple copies of the same.

Pawan Goyal

======================================================================


	The 9th International Workshop on Network and Operating Systems
             Support for Digital Audio and Video (NOSSDAV 99)

 	        http://www.research.att.com/conf/nossdav99/

                           23-25 June, 1999
                              Hosted By:
                       	       AT&T Labs
	   Networking and Distributed Systems Research Laboratory

                          CALL FOR PAPERS

Objectives

The 9th International Workshop on Network and Operating Systems
Support for Digital Audio and Video (NOSSDAV 99) is the international
workshop concerned with state of the art technology in networking and
operating system support for multimedia systems.  For eight years,
NOSSDAV has proven to be an outstanding forum for researchers involved
in building innovative multimedia systems, networks and applications
in both industry and academia.

A key aspect of the workshop is that it provides extensive discussion
periods during which attendees can informally discuss their current
work and future research directions. Traditionally, NOSSDAV has
emphasized high quality experimental research based on prototype or
real systems.  NOSSDAV99 will continue this tradition.

Submissions

Submissions are sought in any area related to Network and Operating
Systems Support for Digital Audio and Video. Two types of submissions
are solicited:  position statements or work in progress reports and full
papers.  These will be reviewed in entirely different ways and have
different deadlines.

Full papers will be subject to a normal review process and are
restricted to fifteen pages.  Accepted papers will be presented at the
Workshop and will appear in the published proceedings.

Position statements and work in progress reports of no more than three
pages, will be reviewed by the program committee only.  Papers will be
chosen on the basis of their potential to provoke discussion.  Roughly
half of the workshop presentations will be allocated to this category.

Submissions, should be sent by electronic mail to nossdav99@research.att.com.
Please include

   1. The paper in PostScript or Microsoft Word (Word 6.0 or Word 97) form

   2. The title of the paper, the list of authors with complete contact
	information in the form of email address and phone number,
	and, in the case of full papers, an abstract summarizing the
	paper in PLAIN TEXT.

The paper abstracts will be circulated among the NOSSDAV program
committee members to solicit reviewers.

Please note that the proceedings of the workshop will be published as
a book and the best papers will be forwarded to selected journals for
publication.

Important Dates

Submission Deadline for Full Papers:            	1 March 1999
Acceptance Notification for Full Papers:        	30 April 1999
Final Full Papers Due:                          	30 May 1999

Submission Deadline for Position Statements:    	30 March 1999
Acceptance Notification for Position Statements:	30 April 1999

Workshop:                                       	23-25 June  1999

Program Chair:

Chuck Kalmanek
AT&T Labs
Shannon Laboratory
Room A113
180 Park Avenue
Florham Park NJ 07032
crk@research.att.com

Other Correspondence:

nossdav99@research.att.com

Program Committee

     Ian Leslie, Cambridge University, UK
     Charles Kalmanek, AT&T Labs Research
     Jim Kurose, University of Masachusetts
     Tom Little, Boston University
     Derek McAuley, Microsoft Research, UK
     Henning Schulzrinne, Columbia University
     Hide Tokuda, Keio University
     Domenico Ferrari, Universita Cattolica, Italy
     Kevin Jeffay, University of North Carolina
     Mike Jones, Microsoft
     Doug Shepherd, Lancaster University, UK
     Richard Black, University of Glasgow, UK
     Jason Nieh, Columbia University
     Partho Mishra, AT&T Labs Research
     S. Keshav, Cornell University
     H. Zhang, Carnegie Mellon University
     Raj Yavatkar, Intel
     Duane Northcutt, Sun Microsystems

LOCATION

AT&T Learning Center, Basking Ridge NJ




From rem-conf Mon Dec 14 03:36:38 1998 
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Subject: RE: Comments on Generic RTP Payload Format proposal
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> ....
>
> However, after having looked at the Internet-Draft again, I now think this
> might not be such a big issue after all. The reason is that the
> Internet-Draft does not specify any static assignments for the XT field.
> Rather, it says that the values to be used for the XT field should be
> communicated out-of-band, using SDP or similar mechanisms.
>
right.

> I would think that in most cases, the extension data field would not be
> used, or it would only be used in one single way.  For instance,
> I think it
> is unlikely that some packets in an RTP session would have extension data
> for parity FEC, while other packets in the same session would use
> Reed-Solomon FEC.
>
> So, when SDP indicates that there are 0-1 valid values for the XT
> field, the
> RTP receiver can effectively ignore the field.  Thus, the problem with the
> additional multiplexing point becomes a non-issue in the typical case.
>
I agree that the number of eXTension types are limited to a small number.
Typical "use cases" for the XT field can be

1. Providing the FEC information
2. In-band signaling of sth. (e.g. change of time-resolution, change the XOR
mask, ... )

For us the typical case is that some PDUs are protected with FEC ("High
Prioroty data") and some are not. Or some are more protected than others, so
we want to use Unequal Error Protection (UEP) depending on the media
importance. We assume for that it there is some mechanism above the "payload
creation layer" that provides AccessUnits together with a priority similar
to the Elementary Stream Interface in MPEG4.


> Given this, one could argue that the XT field should be removed, although
> that would give up some flexibility that could be useful in the
> future.  In
> any case, it seems clear that 8 bits is an excessively large size for this
> field.
>
the sizes of the fields are open for discussion...

> I would also like to second Jonathan Rosenberg's comment, that it
> is better
> to specify the format of the extension data field by referring to the
> relevant documents, rather than including the text explicitly in this
> Internet-Draft.  Of course, there might be cases where no
> relevant document
> exists.  In those cases, it might still be better to publish the format of
> the extension data field in a different document, which would be separate
> from the document that specifies the Generic Payload Format.
>
We just wanted to give some examples of the use of the XT type to make it
more clear. It is just more convenient to read the data inside the document
rather than specifying a reference with section number. There are also some
slight changes to the format from Rosenberg and there will be possibly some
more. As mentioned above I think the mask in the FEC payload changes maybe
not so often or never and can be transmitted separately with another XT
number.

> I also have some comments regarding the alignment of the various fields in
> the payload format.  The primary concern of any payload format that
> implements "grouping" or "multiplexing", is to reduce the number
> of packets.
> The second concern would be to reduce the number of bytes in the payload
> format header.  This is quite important when very small application units
> are grouped together.  To achieve this second goal, I think most people
> would accept giving up word alignment of header fields, as long as the
> header fields are still aligned to byte boundaries.
>
Agreed.

> In this payload format, however, individual header fields are not byte
> aligned.  Padding is inserted at the end of the last header field
> to ensure
> that the payload begins at a byte boundary.  I know that byte alignment is
> not an issue in the MPEG world (Just look at the MPEG-4 SL-Packet header
> definition and you will be either horrified or delighted,
> depending on where
> you come from) but in the IETF header fields are typically aligned.
>
I insisted on byte alignment of the payload because I have to implement
it...

> I have two different proposals for how the header syntax can be modified.
> Both proposals ensure that header fields are byte aligned, but they work a
> little bit differently.  The proposals all show that byte alignment can be
> achieved at a maximum cost of one byte, sometimes even less.
>
We haven't looked in that detail on the fields (problems with the
deadline...) so we appreciate any comments on that.

> Proposal 1:
>
> * Reduce the XT field from 8 bits to 6 bits.  As discussed earlier, it is
> hard to conceive of a situation where someone would want more than 64
> different kinds of Extension Data in the same RTP session.
>
> * If E=0, insert a new field with 5 bits, directly following the F field.
> The 5 bits are reserved and should always be set to 0.
>
> * The size of the LENGTH field is increased from 13 to 16 bits.
>
> This causes all header fields to be aligned with byte boundaries.  No
> padding field is necessary.  This comes at a cost, however.  If
> you look at
> example #3 in draft-guillemot-genpayload-00.txt, you might see that the
> second header, which consists only of flags and a LENGTH field, will
> increase from two bytes to three bytes.  And this is a fairly common case,
> because it happens whenever the packet consists of extension data and an
> application unit that is not being fragmented.
>
We optimised for that case (it is assumed to be the most common one) but we
still feel not satisfied with the header fields sizes. So thank you for you
comment.

> Proposal 2:
>
> * Reduce the XT field from 8 bits to 6 bits.  See comment above.
>
> * If E=1, increase the size of the LENGTH field from 13 to 16 bits.
>
> * If E=0, the size of the LENGTH field remains at 13 bits.
>
> * If G=0 and E=0, the size of TSOFFSET is reduced from 16 to 13 bits.
>
> * If G=1 or E=1, the size of TSOFFSET remains at 16 bits.
>
> * If the FOFFSET field would directly follow the flag fields (because
> neither LENGTH nor TSOFFSET is present), insert 5 reserved bits
> in front of
> FOFFSET, instead of adding 5 bits of padding at the end of FOFFSET.
>
> This causes all header fields to be aligned with byte boundaries.
>  And this
> comes at no extra cost in terms of the header size.  As a matter of fact,
> this scheme saves one entire byte when the payload header only
> contains the
> TSOFFSET field.  (See example #1, in
> draft-guillemot-genpayload-00.txt.)  Of
> course, you may already have noticed that there is another kind of cost
> associated with this proposal: complexity.
>
For me it seems a bit too complex.

Again thank you for your comments.

> Anders
>
>
Stefan.

--
                         Computer centre University of Stuttgart
Stefan  Wesner            http://www-ks.rus.uni-stuttgart.de/PKB
Allmandring 3a                mailto:wesner@rus.uni-stuttgart.de
70550 Stuttgart  Tel.: +49 711 685 4275   Fax.: +49 711 678 8363




From rem-conf Mon Dec 14 09:54:47 1998 
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Hi all,

Does anybody know where can I find vic on video4linux?

Thanks,


--------------------------------------------------------------------
Pedro Miguel Ruiz Martinez       <pedrom@dif.um.es>
Laboratorio de Redes
Facultad de Informatica
Universidad de Murcia
Tlf +34 968364644
Spain
--------------------------------------------------------------------





From rem-conf Mon Dec 14 11:00:17 1998 
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Date: Mon, 14 Dec 1998 13:53:22 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Subject: Re: RTP (multiplexing) Proposals - comparison
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Gal Ashour wrote:
> 
> Both simplicity and efficiency are important, which is the exact reason
> that I suggested to concentrate on the two proposals that represent each of
> those two categories (with a possibility of a later merger).
> They both also share a common important characteristic, which is not losing
> information from the original RTP header.

Implicit in this statement is the assumption that we are not designing a
gateway to gateway protocol. Not that I'm saying we shouldn't design a
more general purpose one (as we want one for MPEG4, for example); what I
am saying is that for G2G, you can get better efficiencies and less
complexity than a more general purpose one. You should also note that
its not so clear how much efficiency germ will provide. The SSRC and
timestamps both won't compress well when used with silence suppression,
I suspect. 

As a result, it was decided that we need to run some simulations to
understand real performance under varying conditions. Germ in particular
has performance very much dependent on dynamic effects, less so for the
others. But, depending on your assumptions (such as changing of lengths
or encodings), the various proposals will have quite a variance in
performance. I welcome thoughts on the models/assumptions for the
simulations.

> 
> The Hitachi proposal may well be enhanced in the future,  but it has merit
> even as is. The difference between 59% bandwidth reduction and 43% (as is
> the case for G.723 under the assumption in my previous posting) is not
> negligible, but yet it isn't a huge difference neither.

I'm not sure the Hitachi proposal is within scope. Its not really an RTP
mux protocol so much as a UDP mux protocol, and its missing some stuff
for that anyway (port numbers, length fields). This was discussed during
the meeting on Friday. Dean Willis presented a mux protocol that filled
in the gaps, in fact. It was also mentioned that there may already be
UDP mux protocols....


-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Mon Dec 14 11:56:30 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: ASHOUR@HAIFA.VNET.IBM.COM, jdrosen@dnrc.bell-labs.com
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>Implicit in this statement is the assumption that we are not designing 
a
>gateway to gateway protocol. Not that I'm saying we shouldn't design a
>more general purpose one (as we want one for MPEG4, for example); what 
I
>am saying is that for G2G, you can get better efficiencies and less
>complexity than a more general purpose one. You should also note that
>its not so clear how much efficiency germ will provide. The SSRC and
>timestamps both won't compress well when used with silence suppression,
>I suspect. 

Perhaps two multiplexing formats should become standard for RTP.  One 
specific to the G2G configuration; the other more general purpose. 
Clearly there are efficiency advantages to the gateway proposals which 
should not be ignored (utilizing overall timestamps, sequence numbers, 
etc.)  A more general purpose proposal would have to be more flexible in 
order to handle less correlated channels, which will cause less 
efficiency.  Separating the two (G2G and general) might be a good idea 
at this time, since G2G currently commands the largest need for 
multiplexing.

>in the gaps, in fact. It was also mentioned that there may already be
>UDP mux protocols....
>

This would be a good solution for the general case, retaining all of the 
necessary RTP information.

Regards, 
Neal A. Schneider

______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Mon Dec 14 14:22:22 1998 
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Jonathan Rosenberg wrote:
>
>Implicit in this statement is the assumption that we are not designing
>a gateway to gateway protocol. Not that I'm saying we shouldn't design a
>more general purpose one (as we want one for MPEG4, for example); what
>I am saying is that for G2G, you can get better efficiencies and less
>complexity than a more general purpose one. You should also note that
>its not so clear how much efficiency germ will provide. The SSRC and
>timestamps both won't compress well when used with silence suppression,
>I suspect.

Even if you focus on a G2G scenario, I think that when supporting
conferencing you may well consider the need of CSRC fields in order to
identify several sources streamed out from an MCU in a single payload.
Identifying those sources in the endpoint clients seems an important
feature to me (or at least one that I wouldn't automatically consider
'redundant'). If I didn't miss something, both the Nokia and Bell proposals
lack the CC field in their mini-header, and thus there is no easy way to
incorporate the CCRC information in each potential block.
This is just one example that comes to my mind, while there may well be
others (even when limiting ourselves to a G2G scenario).

My main focal point is G2G too, but my perspective is a little different.
What I would like to see is a one cross-solution RTP-mux format. When used
for G2G though, the multiplexer in the sending Gateway may impose
additional restrictions on its input streams, while producing the output
multiplexed stream - as it sees fit. Thus, if changes in timestamps are
'minor enough' for the specific solution to ignore them - it can do so even
when using GeRM's scheme.
What I'm basically saying is that it should be the application's (or
solution's) decision, rather than an apriori lack of support in the format
itself.

If we finally decide to adopt the GeRM decision, we may achieve two goals:
1. Providing one generic and compatible format for transferring RTP
payloads with
   a low bandwidth overhead (while the content could be MPEG4, G2G
IP-Telephony, or
   anything else that you would consider using above RTP in the first
place).
2. Each solution (G2G or other), may decide under which conditions some of
the
   RTP header information is not "very important", and may thus be ignored
even
   when using the GeRM difference scheme. This way we achieve efficiency at
least as
   good as with the Nokia and Bell proposals (and possibly even a slightly
better one).
Using the above approach you get the desired efficiency, and yet do not
compromise on flexibility.

Besides, I didn't ignore silence suppression effects when doing the
bandwidth reduction calculation for GeRM, which is basically why I put an
average of two bytes as GeRM overhead rather than just one (GeRM minimal
overhead is a single byte, while the Nokia and Bell proposal require two
bytes as the minimum overhead).
According to the GeRM scheme the SSRC is divided to two portions (the LSB
byte, and the rest 3 MSB). It is likely to assume changes only in the LSB
of the SSRC, and thus even when counting for 50% silence, we should expect
at worse case an average of half a byte due to the difference coding when
using silence suppression. To be on the safe side, I rounded the half byte
in one calculation to one byte, and adding it to the GeRM header byte I got
an overhead of two bytes (which is similar to the best case mini-header
overhead when using either the Nokia or Bell proposals).
Besides, from your excellent (and I do mean it) work regarding the mux
issues, I recall that you mentioned that even during silence periods,
coders may choose to force some information in order to produce a natural
background noise on the other end. In this scenario silence suppression
counts even less in respect to the GeRM proposal (since packets are
produced even during silence).

>
> I'm not sure the Hitachi proposal is within scope. Its not really an RTP
> mux protocol so much as a UDP mux protocol, and its missing some stuff
> for that anyway (port numbers, length fields).
>
I am new to the avt WG, but thought that its charter is providing means for
transporting audio and video streams. I'm not sure if I care if the
proposal that is eventually adopted by the group falls into the UDP-mux
category or the RTP-mux category. The target, as I understand it, is not to
provide 'RTP-mux' format per se, but rather a format that will reduce
bandwidth overhead when transmitting 'many' RTP packets. I wish that the
chosen format will limit flexibility as little as possible (if at all), and
thus be useful even for implementations that we don't think of today - but
may need tomorrow.

If indeed there are other UDP-mux protocols, then  we do need to evaluate
them. Still, I'm saying that my own opinion is that out of the 4 original
proposals we should now focus on GeRM and the Hitachi proposals (in
addition to new proposals that we didn't evaluate yet).
The Nokia and Bell proposals incorporate built-in decisions regarding what
is necessary and what is not in the original RTP header. If we leave this
decision dynamic for the specific solution to decide we get at least as
good results using GeRM (if not even slightly better ones).

As a last comment, from the calculation that I did for the G.723.1 case
(posted earlier to this mailing list), it seems that the typical difference
between the efficient schemes of GeRM/Bell/Nokia and the somewhat less
efficient scheme of Hitachi is not that dramatic, and may not be such a bad
trade-off for its simplicity. Yet, a merged GeRM/Hitachi scheme may
eventually result in a powerful format in terms of simplicity, efficiency
and (yes) flexibility. By merging I mainly refer to a method for reducing
the SSRC field to one byte, which will make the enhanced Hitachi format a
little more similar to the other proposal that have an ID field in the
order of one byte.

Best Regards,
-- Gal A.

------------------------------------
Gal Ashour
Audio/Video Technologies Group
IBM Research - Haifa Research Lab.
ISRAEL

ashour@haifa.vnet.ibm.com
------------------------------------






From rem-conf Mon Dec 14 14:58:49 1998 
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From: subbiah@NASBPD01BS.ntc.nokia.com (Subbiah Baranitharan NRC/Bos)
To: ASHOUR@HAIFA.VNET.IBM.COM (Gal Ashour), casner@cisco.com (casner),
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Organization: Nokia Telecommunications
Date: Mon, 14 Dec 1998 17:52:24 -0500
Subject: RE: RTP Proposal - comparison
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Hello,

I appreciate your effort in presenting arguments with numbers. At the same =
=
time I have to say that efficiency you have indicated is not correct. =
Efficiency varies under different conditions with each proposal. You can =
look closely the GerM draft and see how it changes when Gateways co-operate =
=
and when they don't cooperate.

>From my point of view, we need a solution that suits G2G when the G =
interconnect clouds of PSTN or GSM users (non RTP sources). At this moment, =
=
IP is being deployed in the infrastructure and what we need is a solution =
that meets the requirements of such scenario. In this case, many fields in =
the RTP headers can be considered as common for all sources multiplexed =
within a packet. For example, RTP SN is useful for insequence delivery of =
packets between gateways. Multiplexing in UDP is possible if we have larger =
=
header (most prob. subset of RTP header). We chose RTP since we do not want =
=
to reinvent the wheel. An optimized solution that satisfies the large market=
 =
should be considered.

I have to second Steve Casner comment on an earlier mail that what we are =
trying to do is an interim solution. When we have full blown IP model and =
voice traffic becomes tiny compared to data traffic, RTP multiplexing will =
not make any sense. However, until we reach that point we need to encourage =
=
vendors and operators to deploy IP based infrastructure in the network.

Also, I have to say that when we have RTP sources, it is better to use =
compression from the source rather than at some aggregation point. In =
general, bottleneck is in the access side of the network.

regards,

Barani Subbiah
Nokia Research

 ----------
From: Gal Ashour
To: casner; rem-conf
Subject: RTP Proposal - comparison
Date: Friday, December 11, 1998 2:10PM




Following the avt discussion earlier today, we were requested to list
scenarios for using RTP-multiplexing.

My main interest is audio IP-Telephony, but I think that the RTP
multiplexing applies in general to implementations that require real-time
(or at least quick enough) response on one hand, and have a relatively
short payload on the other hand (and thus the headers, IP/UDP and possibly
the RTP, become a significant overhead).
Scott's signalling proposal (if adopted) could become a candidate for using=

this mechanism too since it falls into the category above.

I will use the G.723.1 codec for comparison of the different proposals.
For G.723 I assume in the calculation 24 bytes per payload (generated every=

30 msec).
The results may change a little based on the payload size for other codecs,=

but I do not think that it will result in a dramatic change of the picture.=


Before I go on with the numbers, I would suggest that a higher weight
should be given to scenarios that use a relatively short payload, rather
than those who use large ones (which may be the case with Video and Data),
since the main advantage in RTP multiplexing is for applications that have
short payload and thus the header overhead is their main concern.

I will assume X simultaneous calls, all using the same codec and payload
length (just for simplicity of the calculation) - I think that this
assumption is reasonable, and should not distort the general picture we get=

>from the comparison.

PFS stands hereby for Payload Frame Size, and eventually will be given the
value 24 (per the above codec assumption).

Hitachi proposal:
 --------------------------
Advantages:  Very simple to implement, do not loose any information
originally in the
                 RTP header (since it is multiplex as is).
Bandwidth Requirements:
     Without RTP Multiplexing =3D (40+PFS)*X
     With RTP Multiplexing       =3D 28 + (12+PFS)*X      (28 stands for
IP+UDP headers).

Asymptotic Bandwidth reduction (large X): 1- (12+PFS)/(40+PFS)
Assuming PFS=3D24:
     Asymptotic Bandwidth reduction: 1- (36)/(64)=3D 43 %


GeRM Proposal:
_______________

Advantage: All the information in the original RTP header can be
reconstructed, High compression ratio.

Bandwidth Requirements:
     Without RTP Multiplexing =3D (40+PFS)*X
     With RTP Multiplexing       =3D 28+12+ (1+ 1+PFS)*X               ( The
first '1' stands for the GeRM 1-byte Header,
                                   the second '1' stands for payload type
for the first block,
                                   and for the LSB of the SSRC which may be=

needed due to                                 silence   characteristics for=

the other blocks (we could even                                   use 0.5
byte on   average as a worse case) ).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=3D24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)=3D 59 %


Bell Proposal:
____________
Advantage: High compression ratio

Bandwidth Requirements:
     Without RTP Multiplexing =3D (40+PFS)*X
     With RTP Multiplexing       =3D 28+12+ (2+PFS)*X     (The '2' stands f=
or
the minimal mini-header per block,
                                   i.e  no need for 16-bit length).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=3D24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)=3D 59 %

Since the CC field is not in the mini-header it may imply that no CSRC
information is transferred - which may be a disadvantage in some
centralized conferencing scenarios where we may have several contributor
sources to the same payload (and wish to identify them in the application).=

Also timestamp information may be of importance, but do not fit into the
mini header.

Nokia Proposal
______________
Advantage: High compression ratio

Bandwidth Requirements:
     Without RTP Multiplexing =3D (40+PFS)*X
     With RTP Multiplexing       =3D 28+ (2+PFS)*X   (The '2' stands for th=
e
minimal mini-header per block,
                                   i.e  no need for 16-bit length).

Asymptotic Bandwidth reduction (large X): 1- (2+PFS)/(40+PFS)
Assuming PFS=3D24:
     Asymptotic Bandwidth reduction: 1- (26)/(64)=3D 59 %

Again, since the CC field is not in the mini-header it may imply that no
CSRC information is transferred, which may be a disadvantage in some
centralized conferencing scenarios where we may have several contributions
to the same payload.
Also timestamp information may be of importance, but do not fit into the
mini header. This proposal limits the length field is to only 5 bits (which=

may be a limitation for payload scenarios that are slightly longer than 31
bytes, and could still benefit from multiplexing).

=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

My Perspective:
_______________

I think that all the four proposals are basically good ones, but we need to=

go forward.
It seem that three of the proposals fit into the 'high bandwidth reduction'=

category, while the Hitachi fits into a 'medium bandwidth reduction
category'.
Two of the 'high bandwidth reduction' proposals are lossy in respect to the=

RTP header information, and thus may limit their use in comparison to the
GeRM and Hitachi's proposal that are 'lossless' in that respect.

I would suggest to proceed with two of the proposals: GeRM and Hitachi,
since they both don't loose RTP header information, and represent the two
ends of simplicity vs. high compression.
Once the GeRM proposal is completed (like ID negotiation issue), we may
decide which of the two is preferred.


As for the RTCP issue. I think that the information is still valid in
respect to the 'global' packet, and can be associated
with the timestamp of the first block in each multiplexed packet.

Best Regards,
 -- Gal.


 -------------------------------------------------------
Gal Ashour
Audio/Video Group
IBM Research - Haifa Research Lab.
ISRAEL

ashour@haifa.vnet.ibm.com
 -------------------------------------------------------







From rem-conf Mon Dec 14 15:46:33 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: rem-conf@es.net, subbiah@NASBPD01BS.ntc.nokia.com
Subject: RE: RTP Proposal - comparison
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>Also, I have to say that when we have RTP sources, it is better to use 
=
>compression from the source rather than at some aggregation point. In =
>general, bottleneck is in the access side of the network.
>
>regards,
>
>Barani Subbiah
>Nokia Research

What if the aggregation point is the access side of the network (i.e. IP 
phones hanging off of a Lan).  It wouldn't make sense to put compression 
inside 250 IP phones if you only needed it for a few outside lines.   

--Neal

______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Mon Dec 14 16:07:04 1998 
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Date: Mon, 14 Dec 1998 18:59:32 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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To: Gal Ashour <ASHOUR@HAIFA.VNET.IBM.COM>
CC: Neal Schneider <nschneid@hotmail.com>, rem-conf@es.net
Subject: Re: RTP (multiplexing) Proposals - comparison
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Gal Ashour wrote:
> 
> Jonathan Rosenberg wrote:
> >
> >Implicit in this statement is the assumption that we are not designing
> >a gateway to gateway protocol. Not that I'm saying we shouldn't design a
> >more general purpose one (as we want one for MPEG4, for example); what
> >I am saying is that for G2G, you can get better efficiencies and less
> >complexity than a more general purpose one. You should also note that
> >its not so clear how much efficiency germ will provide. The SSRC and
> >timestamps both won't compress well when used with silence suppression,
> >I suspect.
> 
> Even if you focus on a G2G scenario, I think that when supporting
> conferencing you may well consider the need of CSRC fields in order to
> identify several sources streamed out from an MCU in a single payload.
> Identifying those sources in the endpoint clients seems an important
> feature to me (or at least one that I wouldn't automatically consider
> 'redundant'). If I didn't miss something, both the Nokia and Bell proposals
> lack the CC field in their mini-header, and thus there is no easy way to
> incorporate the CCRC information in each potential block.
> This is just one example that comes to my mind, while there may well be
> others (even when limiting ourselves to a G2G scenario).

I'm not sure I see how this works. Where is the MCU, and where is the
gateway? Are you saying an RTP mixer sends RTP to a gateway, which then
sends it on an RTP trunk to another gateway? I'm not clear on how the
mixer and gateway are interacting here.


> Besides, I didn't ignore silence suppression effects when doing the
> bandwidth reduction calculation for GeRM, which is basically why I put an
> average of two bytes as GeRM overhead rather than just one (GeRM minimal
> overhead is a single byte, while the Nokia and Bell proposal require two
> bytes as the minimum overhead).
> According to the GeRM scheme the SSRC is divided to two portions (the LSB
> byte, and the rest 3 MSB). It is likely to assume changes only in the LSB
> of the SSRC, and thus even when counting for 50% silence, we should expect
> at worse case an average of half a byte due to the difference coding when
> using silence suppression.

Yes, but its not just the SSRC. The SN is also affected. It will quickly
diverge (even if all users start with the same SN) and I think will be
completely uncompressable. Thats another 16 bits for every muxed user.
We could possibly do something about this by imroving the compression. I
wonder if we actually tried real compression if this would help. In
other words, treat the RTP headers as a piece of data, and run Lempel
Ziv on it. I wonder if it would be better or worse than germ under
various scenarios. Or, one could use a generic UDP mux and then apply IP
payload compression....

-Jonathan R.



-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Mon Dec 14 20:03:24 1998 
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To: rem-conf@es.net
Subject: Re: Vic on video4linux? 
In-reply-to: Your message of "Mon, 14 Dec 1998 18:46:36 +0100."
             <199812141746.SAA04623@labredes14.dif.um.es> 
Content-ID: <25013.913693957.0@aohakobe.ipc.chiba-u.ac.jp>
Date: Tue, 15 Dec 1998 12:57:46 +0900
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> Does anybody know where can I find vic on video4linux?

Patrick Vielle said in the message to MBONE mailing list (Sep.22, 1998)

Patrick> Also there is a VIC compiled for video4linux which I have'nt tried.
Patrick> 
Patrick> http://user.cs.tu-berlin.de/~kraxel/linux/

try and tell us if it's usable or not (-:
 
-- yozo.




From rem-conf Tue Dec 15 06:48:33 1998 
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 IF YOU ARE UNDER 21 DELETE NOW!
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From rem-conf Tue Dec 15 09:29:25 1998 
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Jonathan Rosenberg wrote:
>
> I'm not sure I see how this works. Where is the MCU, and where is the
> gateway? Are you saying an RTP mixer sends RTP to a gateway, which then
> sends it on an RTP trunk to another gateway? I'm not clear on how the
> mixer and gateway are interacting here.
>
The example referred to a scenario where one of the Gateways (say Gateway
A) is also acting as an MCU for the conference (with a possibility of also
several IP and PSTN clients attached to it).
In this scenario Gateway A could have several inputs (from its IP clients
and from its PSTN clients), it would mix them together and provide the
mixed stream to the various clients (one of them could be Gateway B). There
is an advantage in using RTP-mux for packets sent between Gateway A and
Gateway B. Having valid CSRC fields would allow Gateway B to identify the
sources who contributed to a specific payload and pass this information to
its own clients. Since both Nokia and Bell's proposal do not carry the CC
field in their proposed mini header, there would be no vehicle for carrying
this information - a result of a non-flexible format.
Actually, the example is not that important, but rather the principle as
restated in my second comment below.


> Yes, but its not just the SSRC. The SN is also affected. It will quickly
> diverge (even if all users start with the same SN) and I think will be
> completely uncompressable. Thats another 16 bits for every muxed user.
> We could possibly do something about this by imroving the compression. I
> wonder if we actually tried real compression if this would help. In
> other words, treat the RTP headers as a piece of data, and run Lempel
> Ziv on it. I wonder if it would be better or worse than germ under
> various scenarios. Or, one could use a generic UDP mux and then apply IP
> payload compression....
>
Sure, but then you can reflect to comments regarding timestamps in my
previous posting. If
SN, timestamps (or any other field for this matter) is not important in a
specific context, the SPECIFIC implementation could "mask" the appropriate
bit in the GeRM header, and just not send this "not needed" information -
this is much better than Lempel-Ziv.
In this case the output format from this 'customized GeRM Implementation'
will be a 100% compatible with the GeRM format, and still extremely
efficient (or at least as efficient as the implementation cares to be).
My main point was and is, let us leave the decision of what is "necessary"
and what we can "live without" to the specific implementation (which no
doubt knows best).
The specific implementation can decide (when using GeRM) if a specific
portion of a header is worth coding into its output multiplexed stream or
not. This will serve us better in the future when using the adopted format
in other implementation, each with its own flavor of which segment of the
RTP header is important and which is not.
I think that it is a wrong approach to make the application's life "easy"
by limiting the format...



Barani Subbiah Wrote:
>
> At this moment, IP is being deployed in the infrastructure and what we
need is a solution that
> meets the requirements of such scenario. In this case, many fields in the
RTP headers can be
> considered as common for all sources multiplexed within a packet. For
example, RTP SN is useful
> for insequence delivery of packets between gateways.
>
See my comment above. If a field is common or its change is not of 'real
significance', no penalty in GeRM either - the implementation can easily
enough ignore the change and still produce a perfectly legal output as much
as the (GeRM) format is concerned.

> I have to second Steve Casner comment on an earlier mail that what we are
trying to do is an
> interim solution. When we have full blown IP model and voice traffic
becomes tiny compared to
> data traffic, RTP multiplexing will not make any sense. However, until we
reach that point we
> need to encourage vendors and operators to deploy IP based infrastructure
in the network.
> Also, I have to say that when we have RTP sources, it is better to use
compression from the
> source rather than at some aggregation point. In general, bottleneck is
in the access side of
> the network.
>
If we wouldn't have GeRM on the table, you may have got my vote. I do not
believe in interim solutions if we can get at about the same time a
solution that will serve us better and longer. Both GeRM and Hitachi can be
available as soon as we want to (or at least as quickly as the other
proposals). The difference between the Hitachi and the other three
proposals in terms of efficiency, could become quite small once enhanced a
little.
Even as is, the Hitachi proposal provides a significant reduction in
bandwidth requirements. It may be (but then maybe not) that it is worth to
go into some complexities in order to squeeze the percentage even more -
this is by all means an open question that in my opinion needs to be
addressed by the group.
If we decide we need to squeeze every byte, using the GeRM format will
probably serve us at least as well as the Nokia and Bell's proposals
(taking into account the implementation's ability to mask unimportant
differences).
If on the other hand we decide that we can live with a simpler (and less
efficient) format, we can get it from the Hitachi's proposal (or possibly
other UDP-mux formats - if available).

Best Regards,
-- Gal.        (ashour@haifa.vnet.ibm.com)

------------------------------------
Gal Ashour
Audio/Video Technologies Group
IBM Research - Haifa Research Lab.
ISRAEL

ashour@haifa.vnet.ibm.com
------------------------------------





From rem-conf Tue Dec 15 10:43:00 1998 
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-------- On 14/12/98 10:52 PM GMT  subbiah@NASBPD01BS.ntc.nokia.com
(Subbiah Baranitharan NRC/Bos)   wrote:

>Also, I have to say that when we have RTP sources, it is better to use
compression >from the source rather than at some aggregation point. In
general, bottleneck is in >the access side of the network.
>
>regards,
>
>Barani Subbiah
>Nokia Research

Just by removing the IP and UDP headers, you already save major overhead as
far as
bandwidth is concerned.

The more important thing to save is, I think, the number of IP packets in
order to
reduce the amount of forwarding decisions done in the router.

Does anybody know of a good IP simulation device/software that can be used
to test
various mux scenarios?

-- Elad.






From rem-conf Tue Dec 15 10:56:28 1998 
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From: subbiah@NASBPD01BS.ntc.nokia.com (Subbiah Baranitharan NRC/Bos)
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I assume you are not referring to IP phones on LAN to PSTN.

 In case of IP phones connected on LANs via some IP trunk then we are =
looking at different functionalities. Basically, you strip of the RTP packet=
 =
coming to your ingress aggregation point and then try to multiplex. At the =
egress point you need reconstruct the full RTP header.

There are few things to note. With this scheme

1. Time stamps will more likely different since generated at source.
2. SN will be different. and we need to regenerate the SN at the egress =
point.
3. In terms of H.323 signaling, we need 3 signaling procedure to setup a =
connection 1)Source - Ingress point  - 2) Ingress - egress 3) egress to =
destination

My argument is that you get more efficiency using compression at the source=
 =
than at some where in the middle.

regards,

Barani Subbiah
Nokia Research
 ----------
From: Neal Schneider
To: Subbiah Baranitharan (NRC/Bos); rem-conf
Subject: RE: RTP Proposal - comparison
Date: Monday, December 14, 1998 3:40PM

>Also, I have to say that when we have RTP sources, it is better to use
=3D
>compression from the source rather than at some aggregation point. In =3D
>general, bottleneck is in the access side of the network.
>
>regards,
>
>Barani Subbiah
>Nokia Research

What if the aggregation point is the access side of the network (i.e. IP
phones hanging off of a Lan).  It wouldn't make sense to put compression
inside 250 IP phones if you only needed it for a few outside lines.

 --Neal

______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com





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To: rem-conf@es.net
Subject: Slides from Orlando AVT meeting
Date: Tue, 15 Dec 1998 20:42:06 +0100
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Most of the slides presented at the AVT meeting in Orlando are now
available from http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/.
The remainder will become available as soon as I receive copies.

Colin



From rem-conf Tue Dec 15 13:11:31 1998 
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From rem-conf Tue Dec 15 13:14:19 1998 
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     We have been requested to do an M-Bone multicast tomorrw for an 
     all-day seminar on the Next Generation Internet Forum. We will be 
     broadcasting from 8:45 am until 4:00 pm Central Standard Time with a 
     long lunch break. 





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     We have been requested to do an M-Bone multicast tomorrw for an 
     all-day seminar on the Next Generation Internet Forum. We will be 
     broadcasting from 8:45 am until 4:00 pm Central Standard Time with a 
     long lunch break.
     
     Jim Cullen
     jcullen@anl.gov
     Argonne National Laboratory
     Argonne, IL 60439
     630-252-5600 





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From rem-conf Tue Dec 15 13:23:58 1998 
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Sender: "Chris Fulmer" <Chris.Fulmer.chrisf@nt.com>
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To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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May I suggest that in the future, it would be helpful for slides to be put
into a less-proprietary format than powerpoint?  Not only are not all
subscribers to this list on PCs, but it's possible for various Microsoft
document formats to have viruses.  (Such as MS Word macro viruses...  I
have no idea about Powerpoint.)  This is one of the reasons Adobe invented
PDF.

I'd offer to translate these to pdf or postscript or just plain JPGs, but I
can't do it without Powerpoint!


Colin Perkins wrote:

> Most of the slides presented at the AVT meeting in Orlando are now
> available from http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/.
> The remainder will become available as soon as I receive copies.
>
> Colin

--
Chris Fulmer  Nortel, Ltd     RTP, NC
These opinions do not necessarily reflect those of my employer.






From rem-conf Tue Dec 15 14:06:51 1998 
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From rem-conf Tue Dec 15 14:15:09 1998 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: nschneid@hotmail.com, rem-conf@es.net, subbiah@NASBPD01BS.ntc.nokia.com
Subject: RE: RTP Proposal - comparison
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I am thinking of essentially ethernet phones, where the bandwidth 
efficiency is not an issue due to small utilization on an ethernet.   
That is why compression at the Trunk Interface, e.g. DS1, would make 
more sense, rather than at every phone on a 100baseT. 

In terms of multiplexing, the sequence numbers would not be correlated 
at all between calls, and would therefore have to be retained.  The 
timestamps may not differ much from the source to gateway, since the 
transfer latency over an ethernet is very small.  So the timestamp from 
the original source may not need be maintained for each user (except for 
offsets.)  A full RTP header would have to be reconstructed, so all 
required RTP information must be maintained in-band or signaled 
out-of-band.  Maintaining RTP Signaling is a big issue with this 
scenario, however.

Regards, 
Neal A. Schneider

> In case of IP phones connected on LANs via some IP trunk then we are =
>looking at different functionalities. Basically, you strip of the RTP 
packet=
> =
>coming to your ingress aggregation point and then try to multiplex. At 
the =
>egress point you need reconstruct the full RTP header.
>
>There are few things to note. With this scheme
>
>1. Time stamps will more likely different since generated at source.
>2. SN will be different. and we need to regenerate the SN at the egress 
=
>point.
>3. In terms of H.323 signaling, we need 3 signaling procedure to setup 
a =
>connection 1)Source - Ingress point  - 2) Ingress - egress 3) egress to 
=
>destination
>
>My argument is that you get more efficiency using compression at the 
source=
> =
>than at some where in the middle.
>
>regards,
>
>Barani Subbiah
>Nokia Research
> ----------


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Tue Dec 15 14:21:47 1998 
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Powerpoint is preferred by the IETF
(http://www.ietf.org/instructions/slides.html).
PDF is not mentioned in the instructions.
I'll post postscript if asked to do so.

Mark

> -----Original Message-----
> From: Chris Fulmer [mailto:Chris.Fulmer.chrisf@nt.com]
> Sent: Tuesday, December 15, 1998 1:21 PM
> To: Colin Perkins
> Cc: rem-conf@es.net
> Subject: Re: Slides from Orlando AVT meeting
>
>
> May I suggest that in the future, it would be helpful for
> slides to be put
> into a less-proprietary format than powerpoint?  Not only are not all
> subscribers to this list on PCs, but it's possible for
> various Microsoft
> document formats to have viruses.  (Such as MS Word macro
> viruses...  I
> have no idea about Powerpoint.)  This is one of the reasons
> Adobe invented
> PDF.
>
> I'd offer to translate these to pdf or postscript or just
> plain JPGs, but I
> can't do it without Powerpoint!
>
>
> Colin Perkins wrote:
>
> > Most of the slides presented at the AVT meeting in Orlando are now
> > available from http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/.
> > The remainder will become available as soon as I receive copies.
> >
> > Colin
>
> --
> Chris Fulmer  Nortel, Ltd     RTP, NC
> These opinions do not necessarily reflect those of my employer.
>
>
>
>





From rem-conf Tue Dec 15 17:13:16 1998 
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NOUVEAUTES DERNIER TRIMESTRE 98


Plusieurs documents (c=E9d=E9roms - logiciels - cassettes vid=E9o) ont re=
joint le
catalogue du Service du Film de Recherche Scientifique, consultable sur
notre site web http://www.sfrs.fr :
23 c=E9d=E9roms et logiciels traitant d'ornithologie, d'astronomie,
d'arch=E9ologie, d'histoire, de politique, de p=E9dagogie, de physique, d=
e
chimie, de biologie, d'=E9cologie, d'=E9lectronique et de techniques
industrielles
4 films : trois sur les algues (biologie v=E9g=E9tale) et un sur les crev=
ettes
primitives (biologie animale)
Ces documents sont destin=E9s principalement aux =E9tudiants et =E0 leurs
professeurs, certains d'entre eux sont toutefois accessibles =E0 un plus =
large
public, concern=E9 par la culture scientifique. Une pr=E9sentation d=E9ta=
ill=E9e
vous est propos=E9e en annexe.

Par ailleurs, nous avons le plaisir de vous annoncer que le SFRS a re=E7u
plusieurs distinctions :
Le film =93Dans l'ombre du tiroir, =E9co-=E9pid=E9miologie de la maladie =
de Chagas=94
s'est vu d=E9cerner 3 prix :
- le prix =E9pidaure - cat=E9gorie promotion et communication - de la rec=
herche
en m=E9decine et =E9cologie
- le premier prix - cat=E9gorie maladies infectieuses et tropicales - du
Festival du Film de M=E9decine (FILMED) =E0 Amiens
- le prix du minist=E8re de la sant=E9 du festival Ekotopfilm en Slovaqui=
e
Le c=E9d=E9rom =93Mati=E8re molle, physique des objets de tous les jours =
par
Pierre-Gilles de Gennes=94 s'est vu d=E9cerner le Faust d'or du Forum des=
 Arts
de l'Univers Scientifique et Technologique =E0 Toulouse. Il a =E9galement=
 =E9t=E9
nomin=E9 au Prix Roberval 98.

Ci-joint, vous trouverez en pi=E8ce jointe une annexe des CD-Roms PC,
logiciels et films VHS.



St=E9phanie BERTHIER

Service du Film de Recherche Scientifique
96, Bld Raspail - 75272 Paris cedex 06

T=E9l. : 01.42.22.46.44    Fax : 01.42.22.93.50

berthier@sfrs.fr
http://www.sfrs.fr


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------=_NextPart_000_0033_01BE2849.2C2F8CF0--




From rem-conf Wed Dec 16 02:48:58 1998 
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From: "Angel L. Mateo" <angel.mateo@rediris.es>
To: Ross Finlayson <finlayson@live.com>
Cc: To: ;
Subject: Re: A "sdr" source-code patch for directory sessions 
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El d=EDa Fri, 13 Nov 1998 03:24:50  Ross Finlayson escribi=F3:
> FYI to those of you who use "sdr" as your SDP session browser:
> =

> I have developed a patch to the "sdr" source code that will allow "sdr"=
 to
> properly handle 'directory' SDP sessions (for instance, the "Test
> Sessions", "Music", "Lectures and Seminars" directories that are curren=
tly
> being announced).
> =

> In particular, when rebuilt with this patch, "sdr" can now be used to:
> - launch a directory session, with each directory's contents appearing =
in a
> new window
> - create new session announcements (including directory session
> announcements) within any directory
> =

> You can find this patch online at
> 	<http://www.live.com/sdrpatch.html>
> =

Hello,

	I have tried the patch but I can do it works. I have done the following:=


1. I have download the patch (I save with the sdrpatch.diff name), the di=
rectory icon, the plugins,...
2. The patch refers to sdr.old and sdr.new directorys, so I have rename t=
he sdr/src directory to sdr/src.old and I have copied sdr/src.old to sdr/=
sdr.new.
3. Then I have run: "patch < sdrpatch.diff. This reports me only the erro=
r that I don't have any parsed_plugins.tcl file.
4. A copy the plugins that I have downloaded to the sdr/src.new/plugins d=
irectory and the directory.xbm to the sdr/src.new
5. A rename sdr/src.new to sdr/src to correct compilation
6. I have gone the sgi directory (I have got an SGI O2 with IRIX6.3) and =
run ./configure.
7. I run make in the sgi directory.
8. After compilation I have a sdr executable file, so I run "make install=
" to install it to /usr/local/bin.
9. Run sdr.

	When I run sdr, it works with the same results than the normal sdr, but =
I have some problems. First, when it starts I have got the warning messag=
es:
Warning: icon directory overridden by icon directory for media directory
Warning: text Directory overridden by text Directory for media directory

	Then I have the list of sessions, but directory sessions don't have the =
related icon, I have the unknown icon. Then when I want to enter in any o=
f this sessions I have got an core file. With the other sessions I have n=
o problem.

	I have the tcl and tk versions of UCL and I have the next directory stru=
cture in /usr/local/src:

	sdr
	tcl-8.0
	tk-8.0


-----------------------------------------
Angel L. Mateo Martinez
Aplicaciones Telem=E1ticas de Trabajo Cooperativo
RedIRIS/CSIC
C/ Serrano, 142
28006  Madrid (Spain)
Tlfo: +34-91-5855145
Fax:  +34-91-5855146





From rem-conf Wed Dec 16 06:02:45 1998 
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          Wed, 16 Dec 1998 07:53:37 -0600
From: "Henry Sinnreich" <henry.sinnreich@mci.com>
To: "Mark Baugher" <mbaugher@intel.com>,
        "'Chris Fulmer'" <Chris.Fulmer.chrisf@nt.com>,
        "Colin Perkins" <C.Perkins@cs.ucl.ac.uk>
Cc: <rem-conf@es.net>
Subject: RE: Slides from Orlando AVT meeting
Date: Wed, 16 Dec 1998 07:53:21 -0500
Message-ID: <NBBBIIJFOKPMFOOILMBKIELNDFAA.henry.sinnreich@mci.com>
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PowerPoint and Word files can be saved very nicely as HTML. Nothing wrong
with HTML and GIF's?

BTW: Any fundamental criteria for which is more proprietary: PDF or
PPT/Word?  ;-)

Thanks, Henry

> -----Original Message-----
> From: Mark Baugher [mailto:mbaugher@intel.com]
> Sent: Tuesday, December 15, 1998 5:21 PM
> To: 'Chris Fulmer'; Colin Perkins
> Cc: rem-conf@es.net
> Subject: RE: Slides from Orlando AVT meeting
>
>
> Powerpoint is preferred by the IETF
> (http://www.ietf.org/instructions/slides.html).
> PDF is not mentioned in the instructions.
> I'll post postscript if asked to do so.
>
> Mark
>
> > -----Original Message-----
> > From: Chris Fulmer [mailto:Chris.Fulmer.chrisf@nt.com]
> > Sent: Tuesday, December 15, 1998 1:21 PM
> > To: Colin Perkins
> > Cc: rem-conf@es.net
> > Subject: Re: Slides from Orlando AVT meeting
> >
> >
> > May I suggest that in the future, it would be helpful for
> > slides to be put
> > into a less-proprietary format than powerpoint?  Not only are not all
> > subscribers to this list on PCs, but it's possible for
> > various Microsoft
> > document formats to have viruses.  (Such as MS Word macro
> > viruses...  I
> > have no idea about Powerpoint.)  This is one of the reasons
> > Adobe invented
> > PDF.
> >
> > I'd offer to translate these to pdf or postscript or just
> > plain JPGs, but I
> > can't do it without Powerpoint!
> >
> >
> > Colin Perkins wrote:
> >
> > > Most of the slides presented at the AVT meeting in Orlando are now
> > > available from http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/.
> > > The remainder will become available as soon as I receive copies.
> > >
> > > Colin
> >
> > --
> > Chris Fulmer  Nortel, Ltd     RTP, NC
> > These opinions do not necessarily reflect those of my employer.
> >
> >
> >
> >
>
>
>




From rem-conf Wed Dec 16 06:55:10 1998 
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To: Henry Sinnreich <henry.sinnreich@mci.com>
cc: rem-conf <rem-conf@es.net>
Subject: Re: Slides from Orlando AVT meeting 
In-reply-to: Your message of "Wed, 16 Dec 1998 07:53:21 EST."
             <NBBBIIJFOKPMFOOILMBKIELNDFAA.henry.sinnreich@mci.com> 
Date: Wed, 16 Dec 1998 09:47:03 -0500
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>BTW: Any fundamental criteria for which is more proprietary: PDF or
>PPT/Word?  ;-)

Not really on the subject of this list, but you can get the PDF spec
for free from:
  http://www.adobe.com/supportservice/devrelations/technotes.html
and it's not likely to become completely incompatible from one year to
the next.

Can you point me at the word and powerpoint specs?

Cheers,
	Mark



From rem-conf Wed Dec 16 06:55:13 1998 
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CC: Mark Baugher <mbaugher@intel.com>,
        "'Chris Fulmer'" <Chris.Fulmer.chrisf@nt.com>,
        Colin Perkins <C.Perkins@cs.ucl.ac.uk>, rem-conf@es.net
Subject: Re: Slides from Orlando AVT meeting
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Henry Sinnreich wrote:
> 
> PowerPoint and Word files can be saved very nicely as HTML. Nothing wrong
> with HTML and GIF's?
> 
> BTW: Any fundamental criteria for which is more proprietary: PDF or
> PPT/Word?  ;-)

At least PDF is documented in publically available specifications (I
don't think any of the MS formats are) and is a direct derivative of
PostScript. That said, there is now a useful tool for Solaris:
PCFileViewer, which allows one to look at the standard PC file types
like PowerPoint and Word. It can be configured as a MIME viewer, too.
It's freely available on the Sun web site. I don't know (but would like
to...) whether this exists for other Unix platforms.



From rem-conf Wed Dec 16 06:59:02 1998 
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From: =?ISO-8859-1?Q?Tobias_=D6brink?= <nv91-tob@nada.kth.se>
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Subject: RE: Slides from Orlando AVT meeting
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On Wed, 16 Dec 1998, Henry Sinnreich wrote:

> PowerPoint and Word files can be saved very nicely as HTML. Nothing wro=
ng
> with HTML and GIF's?
>=20
> BTW: Any fundamental criteria for which is more proprietary: PDF or
> PPT/Word?  ;-)

Is there any free PPT/Word reader available?

/Tobias
.........................................................................=
...
Tobias =D6brink                 Grad. Stud. Dept. of Teleinformatics, KTH
Phone numbers:				Paper mail address:
	+46 8 752 14 75 (IT, work)		KTH-ELECTRUM/204
        +46 8 751 17 93 (Fax)			S-164 40 Kista
						Sweden
E-mail: tobias@it.kth.se	URL: http://www.it.kth.se/~nv91-tob/
See if I'm logged in: http://www.it.kth.se/~nv91-tob/notice.html




From rem-conf Wed Dec 16 07:34:59 1998 
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From: =?ISO-8859-1?Q?Tobias_=D6brink?= <nv91-tob@nada.kth.se>
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To: rem-conf@es.net
Subject: Re: Slides from Orlando AVT meeting
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On Wed, 16 Dec 1998 wolt@igd.fhg.de wrote:

>=20
> Hi,
>=20
> you wrote:
>=20
>    Is there any free PPT/Word reader available?
>=20
> Actually, there is. Called Deja Vu, it came out early this year and is =
now
> included in the Solaris 7 distribution (but should still be available
> separately at www.sun.com).=20
:
:
> (and, FWIW DejaVu is unable to print, only to display on-screen)
:
:
> --=20
> 	later,
> 	Stephen
>=20
> Fraunhofer-IGD                 | mailto:
> Stephen Wolthusen              | stephen@wolthusen.com
> Rundeturmstr. 6 	       | swolthusen@acm.org
> 64283 Darmstadt, GERMANY       | wolt@igd.fhg.de
>                                | wolt@capcom.de
> 			       |=20
> Tel +49 (0) 6151 155 539 (Bs.) | Fax: +49 (0) 6151 155 499 (Bs.)
>     +49 (0) 172 916 9883       |      +49 (0) 6245 905 366 (Pri.)
>     +49 (0) 6245 6952   (Pri.) |

Thanks Stephen

/Tobias
.........................................................................=
...
Tobias =D6brink                 Grad. Stud. Dept. of Teleinformatics, KTH
Phone numbers:				Paper mail address:
	+46 8 752 14 75 (IT, work)		KTH-ELECTRUM/204
        +46 8 751 17 93 (Fax)			S-164 40 Kista
						Sweden
E-mail: tobias@it.kth.se	URL: http://www.it.kth.se/~nv91-tob/
See if I'm logged in: http://www.it.kth.se/~nv91-tob/notice.html







From rem-conf Wed Dec 16 08:52:57 1998 
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To: mbaugher@intel.com, Chris.Fulmer.chrisf@nt.com, C.Perkins@cs.ucl.ac.uk,
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Subject: RE: Slides from Orlando AVT meeting
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> From rem-conf-request@es.net Wed Dec 16 09:17:44 1998
> From: "Henry Sinnreich" <henry.sinnreich@mci.com>
> 
> PowerPoint and Word files can be saved very nicely as HTML. Nothing wrong
> with HTML and GIF's?

Nothing at all -- this is the right way to go (more below)...
 
> BTW: Any fundamental criteria for which is more proprietary: PDF or
> PPT/Word?  ;-)

Yup -- just as with PostScript, the spec. for PDF is openly available,
and there are already excellent freely available implementations, such 
as the PS to PDF converter that comes with ghostscript.  Do you know of
any comparable packages for PPT?  I doubt very much that they exist.
Both PS and PDF may have been developed outside of an open deliberative
process, but they both certainly function as open standards now.

A related comment on the broader issue here.
It was said in an earlier posting by another party that "the IETF
favors PowerPoint."  Translation: "certain participants at IETF meetings
prefer to use PowerPoint."  I'm not aware of any official pronouncement
on this point, and suspect that I would be just one member of a rather
large and unruly mob if any such a pronouncement were to be made.
It would run counter to the deeply rooted vendor- and platform-
independent ethic of the IETF and its work.

I think we all ought to strive mightily to use vendor-independent
standards in the work we do both in and out of the IETF.  And I think
that this mitigates against the use of proprietary systems such as
PowerPoint.

> Thanks, Henry

Cheerio, Rick Rodgers



From rem-conf Wed Dec 16 10:57:14 1998 
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Date: Wed, 16 Dec 1998 10:50:57 -0800 (PST)
From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
Message-Id: <199812161850.KAA21742@jackson.cs.ucsb.edu>
To: IPMULTICAST@STARDUST.COM, end2end-interest@ISI.EDU, rem-conf@es.net,
        tccc@ieee.org
Subject: 43rd IETF Sessions on the IMJ
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(lists bcc'ed to prevent accidental multi-mailing list responses).

***

     Announcing Availability of the 43rd IETF Sessions on the IMJ

I'm ready to announce that most of the sessions from the 43rd IETF in 
Orlando, FL that were transmitted on the MBone are now available on the 
IMJ (http://imj.gatech.edu) for on-demand playback. You can schedule 
programs via the WWW page and start the MBone sessions via SDR (or the 
WWW page with a simple plug-in).  For additional information about the 
MBone see http://mbone.com and http://ipmulticast.com/.

Not much to report this time around about the IETF except that we offered
two high speed video channels, one of which had two video feeds.  We
are gaining experience digitally recording sessions and now have a
relatively quick turnaround on availability.

For the record, the plan is to have the last 3 sets of IETF meetings
archived on the MBone.  We've got the last three available now:
41st, 42nd, and 43rd.  Future plans in regard to content include
making talks from conferences available as well.  I'm working on some
angles here.

Also, A NEW SERVICE has been started for those who want access to the 
source files themselves.  The latest batch of IETF files are available
at ftp://ftp.cc.gatech.edu/pub/people/kevin/.  Please be gentle on the
ftp server since these files are around 100 MB for video and 40 MB for
audio.

And finally, one interesting characteristic of the IMJ that most of
you won't notice but is interesting is that we now have two servers
on the back end of the IMJ.  There is one server at Georgia Tech
and one at UCSB.  The exact location of playout will depend on the
request and where the content is located.  There is no duplication
of content, and this "distributed playout" system was implemented
as a first step of a new project and because we ran out of disk space
at Georgia Tech.

Finally, and as usual, I've included the original IMJ e-mail announcement 
for those who want additional information about the IMJ.

Any suggestions for improvement are welcome.

-Kevin

****
              Announcing the Interactive Multimedia Jukebox


   At one point on the MBone list there was a discussion of what on-demand
servers exist or are being developed.  Well, I'd like to announce our 
version called the Interactive Multimedia Jukebox (IMJ).

  The IMJ web page is a request and scheduling interface for the playout 
of content over the MBone.  CONTENT IS ENCODED SO THAT IT CAN BE RECEIVED
WITH THE LATEST VERSIONS OF VIC AND VAT (ANY PLATFORM).  Also check out
mcontrol (http://imj.ucsb.edu/mcontrol/) a new MBone playout control tool 
for real-time broadcasts written in Java.  

   The IMJ page is located at http://imj.gatech.edu.  Information about 
the IMJ including general info, a postscript version of a paper about the 
IMJ, how-to-use information, and how it was implemented can be found at 
http://www.cc.gatech.edu/computing/Telecomm/IMJ/.  (Recent note:  the
paper will be appearing in the WWW7 conference in Brisbane, Australia)

   Some quick additional information about the IMJ is included below:


IMJ Audio/Video
-----------------
   The IMJ uses the WWW to submit requests which are then scheduled
for playout on the MBone.  Right now we are offering three channels:
Channels 1 and 2 are being broadcast at a TTL of 127.  Channel 3 is
for internal testing and has a a TTL of 15.  Each channel provides 
DVI-2 audio and 128 Kbps H.261 video.  The encoding was done using 
Henning Schulzrinne's rtpplay and rtpdump.

Scheduling
----------
   Program scheduling uses a set of criteria to decide on which channel
to schedule a program.  The current set of criteria are listed just
below the interactive schedule.  We are exploring lots of different
options.

Content
-------
   We have been working on the jukebox paradigm with Turner Broadcasting,
and as a result of their interest they have agreed to let us broadcast
content on the MBone.  I am particularly excited about this aspect because
of their help and interest in investigating new applications on the Internet.

   The plan is to add a new set of content about every two weeks.  When this 
happens, the old content will be moved to the secondary request menu for two 
weeks.  The old-old stuff will be removed.  Hopefully I'll be able to 
advertise on the MBone list when the new stuff is available.

Development Platform
--------------------
   As is mentioned in the paper on the IMJ information page we are using
the IMJ as a platform for a variety of research issues including
providing interactivity, supporting multiple heterogeneous streams,
video server organization, tracking usage, program scheduling, and
pricing.

Please send any feedback or suggestions to almeroth@cs.ucsb.edu or
kevin@cc.gatech.edu.





From rem-conf Thu Dec 17 03:57:41 1998 
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Hi!
We would like generate compressed RTP-headers for evaluation purposes
but we need traces of compressed header information and have 
difficulties to generate them our-self.

Is any implementation RTP/UDP/IP header compression available on the net ?
Is it possible to get traces of uncompressed or compressed headers ?

Could someone help me ?

Regards Lars Westberg



From rem-conf Thu Dec 17 05:15:42 1998 
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From: "Henry Sinnreich" <henry.sinnreich@mci.com>
To: "R. P. Channing Rodgers, M.D." <rodgers@nlm.nih.gov>, <mbaugher@intel.com>,
        <Chris.Fulmer.chrisf@nt.com>, <C.Perkins@cs.ucl.ac.uk>
Cc: <rem-conf@es.net>, <rodgers@billings.nlm.nih.gov>
Subject: RE: Slides from Orlando AVT meeting
Date: Thu, 17 Dec 1998 07:08:51 -0500
Message-ID: <NBBBIIJFOKPMFOOILMBKKENBDFAA.henry.sinnreich@mci.com>
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Of all the vendor independent graphic formats, the PNG format should be on
top of the list.
Please see http://w3c.org/Graphics/PNG/

(Fortunately my PowerPoint package can save as PNG)
Would PNG be acceptable as a matter of principle?

Thanks, Henry


> -----Original Message-----
> From: R. P. Channing Rodgers, M.D. [mailto:rodgers@nlm.nih.gov]
> Sent: Wednesday, December 16, 1998 11:45 AM
> To: mbaugher@intel.com; Chris.Fulmer.chrisf@nt.com;
> C.Perkins@cs.ucl.ac.uk; henry.sinnreich@mci.com
> Cc: rem-conf@es.net; rodgers@billings.nlm.nih.gov
> Subject: RE: Slides from Orlando AVT meeting
>
>
>
> > From rem-conf-request@es.net Wed Dec 16 09:17:44 1998
> > From: "Henry Sinnreich" <henry.sinnreich@mci.com>
> >
> > PowerPoint and Word files can be saved very nicely as HTML.
> Nothing wrong
> > with HTML and GIF's?
>
> Nothing at all -- this is the right way to go (more below)...
>
> > BTW: Any fundamental criteria for which is more proprietary: PDF or
> > PPT/Word?  ;-)
>
> Yup -- just as with PostScript, the spec. for PDF is openly available,
> and there are already excellent freely available implementations, such
> as the PS to PDF converter that comes with ghostscript.  Do you know of
> any comparable packages for PPT?  I doubt very much that they exist.
> Both PS and PDF may have been developed outside of an open deliberative
> process, but they both certainly function as open standards now.
>
> A related comment on the broader issue here.
> It was said in an earlier posting by another party that "the IETF
> favors PowerPoint."  Translation: "certain participants at IETF meetings
> prefer to use PowerPoint."  I'm not aware of any official pronouncement
> on this point, and suspect that I would be just one member of a rather
> large and unruly mob if any such a pronouncement were to be made.
> It would run counter to the deeply rooted vendor- and platform-
> independent ethic of the IETF and its work.
>
> I think we all ought to strive mightily to use vendor-independent
> standards in the work we do both in and out of the IETF.  And I think
> that this mitigates against the use of proprietary systems such as
> PowerPoint.
>
> > Thanks, Henry
>
> Cheerio, Rick Rodgers
>




From rem-conf Thu Dec 17 05:55:14 1998 
From rem-conf-request@es.net Thu Dec 17 05:55:12 1998
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To: Henry Sinnreich <henry.sinnreich@mci.com>
cc: "R. P. Channing Rodgers, M.D." <rodgers@nlm.nih.gov>, 
    mbaugher <mbaugher@intel.com>, 
    "Chris.Fulmer.chrisf" <Chris.Fulmer.chrisf@nt.com>, 
    rem-conf <rem-conf@es.net>, rodgers <rodgers@billings.nlm.nih.gov>
Subject: Re: Slides from Orlando AVT meeting
In-reply-to: Your message of "Thu, 17 Dec 1998 07:08:51 EST." <NBBBIIJFOKPMFOOILMBKKENBDFAA.henry.sinnreich@mci.com>
Date: Thu, 17 Dec 1998 13:48:00 +0100
Message-ID: <1041.913902480@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Quoting from http://www.ietf.org/instructions/slides.html

	"The online slide submissions are critical to the preparation of
 	 both the on-line and hardcopy versions of the proceedings, and
	 should be submitted either as a PowerPoint (preferred) or
	 PostScript file, one slide per page .  The submissions can be sent
	 either as a MIME message component or via an emailed URL." 

If that's what the secretariat want, then that's what we should provide
since their job is difficult enough as it is. If this argument has to
continue, can we please take it to someplace other than rem-conf?

Colin


--> Henry Sinnreich writes:
>Of all the vendor independent graphic formats, the PNG format should be on
>top of the list.
>Please see http://w3c.org/Graphics/PNG/
>
>(Fortunately my PowerPoint package can save as PNG)
>Would PNG be acceptable as a matter of principle?
>
>Thanks, Henry
>
>
>> -----Original Message-----
>> From: R. P. Channing Rodgers, M.D. [mailto:rodgers@nlm.nih.gov]
>> Sent: Wednesday, December 16, 1998 11:45 AM
>> To: mbaugher@intel.com; Chris.Fulmer.chrisf@nt.com;
>> C.Perkins@cs.ucl.ac.uk; henry.sinnreich@mci.com
>> Cc: rem-conf@es.net; rodgers@billings.nlm.nih.gov
>> Subject: RE: Slides from Orlando AVT meeting
>>
>>
>>
>> > From rem-conf-request@es.net Wed Dec 16 09:17:44 1998
>> > From: "Henry Sinnreich" <henry.sinnreich@mci.com>
>> >
>> > PowerPoint and Word files can be saved very nicely as HTML.
>> Nothing wrong
>> > with HTML and GIF's?
>>
>> Nothing at all -- this is the right way to go (more below)...
>>
>> > BTW: Any fundamental criteria for which is more proprietary: PDF or
>> > PPT/Word?  ;-)
>>
>> Yup -- just as with PostScript, the spec. for PDF is openly available,
>> and there are already excellent freely available implementations, such
>> as the PS to PDF converter that comes with ghostscript.  Do you know of
>> any comparable packages for PPT?  I doubt very much that they exist.
>> Both PS and PDF may have been developed outside of an open deliberative
>> process, but they both certainly function as open standards now.
>>
>> A related comment on the broader issue here.
>> It was said in an earlier posting by another party that "the IETF
>> favors PowerPoint."  Translation: "certain participants at IETF meetings
>> prefer to use PowerPoint."  I'm not aware of any official pronouncement
>> on this point, and suspect that I would be just one member of a rather
>> large and unruly mob if any such a pronouncement were to be made.
>> It would run counter to the deeply rooted vendor- and platform-
>> independent ethic of the IETF and its work.
>>
>> I think we all ought to strive mightily to use vendor-independent
>> standards in the work we do both in and out of the IETF.  And I think
>> that this mitigates against the use of proprietary systems such as
>> PowerPoint.
>>
>> > Thanks, Henry
>>
>> Cheerio, Rick Rodgers
>>
>



From rem-conf Thu Dec 17 06:22:52 1998 
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CC: "R. P. Channing Rodgers, M.D." <rodgers@nlm.nih.gov>, mbaugher@intel.com,
        Chris.Fulmer.chrisf@nt.com, C.Perkins@cs.ucl.ac.uk, rem-conf@es.net,
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Subject: Re: Slides from Orlando AVT meeting
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Henry Sinnreich wrote:
> 
> Of all the vendor independent graphic formats, the PNG format should be on
> top of the list.
> Please see http://w3c.org/Graphics/PNG/
> 
> (Fortunately my PowerPoint package can save as PNG)
> Would PNG be acceptable as a matter of principle?

The problem with PNG (and GIF) is that it's only pixels. With PDF and
PPT and other formats, I can print the slides at print (600 to 1200 dpi)
resolution and cut-and-paste. GIF/PNG slides look fine on the screen,
but look very low-res when printed, since the typical resolution is only
75 to 100 dpi.


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Subject: Packet Video 99 Submission Deadline EXTENDED to JAN 7th 99 
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                   ******** DEADLINE EXTENDED *********

----------------------------------------------------------------------
Due to the large volume of requests we are extending the deadline to

                              JAN. 7th 1999
----------------------------------------------------------------------




            CALL FOR PAPERS

    Packet Video '99

  The 9th International Packet Video Workshop
    26-27 April 1999
    New York City, U.S.A

  http://www.research.att.com/~mrc/PacketVideo99.html
  In Europe: http://www.inria.fr/rodeo/turletti/pv99/



Sponsored by: AT&T, Columbia University, EURASIP,
IEEE Signal Processing Society IMDSP & MMSP Technical Committees

The ninth International Packet Video Workshop (PV' 99) will be held at the
Davis auditorium, of Columbia University in New York City. The workshop is
devoted to presenting technological advancements and innovations in video
transmission over packet networks, in particular, the Internet. Packet Video
Workshops have been unique in providing a common ground for people from
video coding and networking fields. Presentations on theory and practice,
standards activities, and business and consumer applications are encouraged.

We cordially invite you to take part in this workshop by submitting your
work and look forward to seeing you in New York City in April 1999 for what
will be a most rewarding and exciting experience!

PV'99 is scheduled to take place in the week following Picture Coding
Symposium 99 (PCS' 99)  which is being held in Portland Oregon, on 21-23
April 1999. For information on  PCS'99 please contact:
pcs@pcs.ece.orst.edu

TECHNICAL PROGRAM
The technical program of Packet Video '99 will consist of invited talks,
submitted paper presentations, poster sessions and a plenary session: "After
Internet Telephony, Internet Television..."

Topics of interest include, but are not limited to, the following:
* Video streaming over the Internet
* Network adaptive video coding and transport
* Packetized video for home LAN's
* Packetized video for wireless/mobile systems
* Layered coding for error resilience and heterogeneous networks
* Packet loss resilient coding and transport
* Terminal and server architectures for Internet TV
* Efficient transcoding for heterogeneous networks
* Congestion control
* Error concealment
* Statistical multiplexing for greater network and terminal utilization
* Traffic shaping for efficient network and terminal utilization
* Interstream synchronization for multiple video presentations
* Performance modeling and evaluation
* Rate control for VBR video
* International Standards: MPEG-4, MPEG-7, H.263+, RTP, RTSP, SIP, SDP
* Multicasting, MBONE applications
* Implementations and commercial applications

Best Paper Award: The author of the best paper will receive: A $250 cash
prize graciously donated by NEC USA, C&C Research Laboratories.
Manuscripts submitted to PV'99 will be considered for publication in the
special issue of EURASIP Journal Signal Processing: Image Communication,
REAL-TIME VIDEO OVER THE INTERNET also.

Submissions
Please submit an electronic manuscript written in HTML, not exceeding 7
Mbytes and 10 printed pages. We will produce a CD-ROM containing the
accepted papers. Electronic components accompanying your submission are
strongly encouraged.
Submit your work in ONE of the following forms:

1) a set of HTML files organized in a single directory and compressed with
zip or WinZip OR
2) as a URL (you must have the rights to all the material there, and
guarantee stability of the files until the conference).

Detailed submission instructions for authors and help with manuscript
preparation
with HTML can be found on the conference web page:

               http://www.research.att.com/~mrc/PacketVideo99.html


ALL SUBMISSIONS MUST BE RECEIVED no later than: Jan. 7th, 1999,
NOTIFICATION OF ACCEPTANCE: February 28, 1999
FINAL PAPERS DUE: March 15, 1999



GENERAL CHAIR     PUBLICATION & LOCAL ARRANGEMENTS
M. Reha Civanlar    Andrea Basso
AT&T Labs - Research   AT&T Labs - Research
100 Schultz Drive, 3-213                100 Schultz Drive, 3-219
Red Bank, NJ 07701                      Red Bank, NJ 07701
USA      USA
civanlar@research.att.com               basso@research.att.com


  PACKET VIDEO' 99 INTERNATIONAL STEERING COMMITTEE

John  Arnold,            University of New South Wales,   Australia
Andrea Basso,            AT&T Labs - Research,            USA
Stephen Casner,          Precept Software,                USA
Shih-Fu Chang,           Columbia University,             USA
Leonardo Chiariglione,   CSELT,                           Italy
M. Reha Civanlar,        AT&T Labs - Research,            USA
Jon Crowcroft,           University College of London,    UK
Mohammed  Ghanbari,      University of Essex                 UK
Barry G. Haskell         AT&T Labs - Research,                   USA
Steven McCanne,          U. C. Berkeley,                  USA
Geoff Morrison,          BT Labs,                         UK
Joerg Ott,               University of Bremen,            Germany
Sakae Okubo,             Graphics Communication Labs,     Japan
D. Raychaudhuri,         NEC Research,                    USA
Amy  Reibman,            AT&T Labs - Research,            USA
Henning Schulzrinne,     Columbia University,             USA
Gary  Sullivan,          PictureTel,                      USA
Toshitaka Tsuda,         Fujitsu,                         Japan
Thierry  Turletti,       INRIA,                           France
Stephan Wenger,          Technical University of Berlin,  Germany
Hiroshi Yasuda,          NTT,                             Japan





From rem-conf Thu Dec 17 08:58:15 1998 
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From: rodgers@nlm.nih.gov (R. P. Channing Rodgers, M.D.)
Message-Id: <199812171653.LAA03547@billings.csb>
To: henry.sinnreich@mci.com, C.Perkins@cs.ucl.ac.uk
Subject: Re: Slides from Orlando AVT meeting
Cc: rodgers@nlm.nih.gov, mbaugher@intel.com, Chris.Fulmer.chrisf@nt.com,
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I stand corrected -- thanks for quoting chapter and verse, Colin,
and yes, no more about this here  8^)

But I will grumble to the secretariat -- I think this is a terrible
policy...

Cheerio, Rick Rodgers

> From C.Perkins@cs.ucl.ac.uk Thu Dec 17 08:50:08 1998
> To: Henry Sinnreich <henry.sinnreich@mci.com>
> cc: "R. P. Channing Rodgers, M.D." <rodgers@nlm.nih.gov>,
>         mbaugher <mbaugher@intel.com>,
>         "Chris.Fulmer.chrisf" <Chris.Fulmer.chrisf@nt.com>,
>         rem-conf <rem-conf@es.net>, rodgers <rodgers@billings.nlm.nih.gov>
> Subject: Re: Slides from Orlando AVT meeting
> Date: Thu, 17 Dec 1998 13:48:00 +0100
> From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
> Content-Length: 2895
> 
> Quoting from http://www.ietf.org/instructions/slides.html
> 
> 	"The online slide submissions are critical to the preparation of
>  	 both the on-line and hardcopy versions of the proceedings, and
> 	 should be submitted either as a PowerPoint (preferred) or
> 	 PostScript file, one slide per page .  The submissions can be sent
> 	 either as a MIME message component or via an emailed URL." 
> 
> If that's what the secretariat want, then that's what we should provide
> since their job is difficult enough as it is. If this argument has to
> continue, can we please take it to someplace other than rem-conf?
> 
> Colin
> 
> 
> --> Henry Sinnreich writes:
> >Of all the vendor independent graphic formats, the PNG format should be on
> >top of the list.
> >Please see http://w3c.org/Graphics/PNG/
> >
> >(Fortunately my PowerPoint package can save as PNG)
> >Would PNG be acceptable as a matter of principle?
> >
> >Thanks, Henry
> >
> >
> >> -----Original Message-----
> >> From: R. P. Channing Rodgers, M.D. [mailto:rodgers@nlm.nih.gov]
> >> Sent: Wednesday, December 16, 1998 11:45 AM
> >> To: mbaugher@intel.com; Chris.Fulmer.chrisf@nt.com;
> >> C.Perkins@cs.ucl.ac.uk; henry.sinnreich@mci.com
> >> Cc: rem-conf@es.net; rodgers@billings.nlm.nih.gov
> >> Subject: RE: Slides from Orlando AVT meeting
> >>
> >>
> >>
> >> > From rem-conf-request@es.net Wed Dec 16 09:17:44 1998
> >> > From: "Henry Sinnreich" <henry.sinnreich@mci.com>
> >> >
> >> > PowerPoint and Word files can be saved very nicely as HTML.
> >> Nothing wrong
> >> > with HTML and GIF's?
> >>
> >> Nothing at all -- this is the right way to go (more below)...
> >>
> >> > BTW: Any fundamental criteria for which is more proprietary: PDF or
> >> > PPT/Word?  ;-)
> >>
> >> Yup -- just as with PostScript, the spec. for PDF is openly available,
> >> and there are already excellent freely available implementations, such
> >> as the PS to PDF converter that comes with ghostscript.  Do you know of
> >> any comparable packages for PPT?  I doubt very much that they exist.
> >> Both PS and PDF may have been developed outside of an open deliberative
> >> process, but they both certainly function as open standards now.
> >>
> >> A related comment on the broader issue here.
> >> It was said in an earlier posting by another party that "the IETF
> >> favors PowerPoint."  Translation: "certain participants at IETF meetings
> >> prefer to use PowerPoint."  I'm not aware of any official pronouncement
> >> on this point, and suspect that I would be just one member of a rather
> >> large and unruly mob if any such a pronouncement were to be made.
> >> It would run counter to the deeply rooted vendor- and platform-
> >> independent ethic of the IETF and its work.
> >>
> >> I think we all ought to strive mightily to use vendor-independent
> >> standards in the work we do both in and out of the IETF.  And I think
> >> that this mitigates against the use of proprietary systems such as
> >> PowerPoint.
> >>
> >> > Thanks, Henry
> >>
> >> Cheerio, Rick Rodgers
> >>
> >
> 



From rem-conf Thu Dec 17 16:11:16 1998 
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Date: Thu, 17 Dec 1998 16:50:24
To: "Angel L. Mateo" <angel.mateo@rediris.es>
From: Ross Finlayson <finlayson@live.com>
Subject: Re: A "sdr" source-code patch for directory sessions 
Cc: rem-conf@es.net
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At 11:37 AM 12/16/98 +0100, Angel L. Mateo wrote:
>	I have tried the patch but I can do it works. I have done the following:
>
>1. I have download the patch (I save with the sdrpatch.diff name), the
directory icon, the plugins,...
>2. The patch refers to sdr.old and sdr.new directorys, so I have rename
the sdr/src directory to sdr/src.old and I have copied sdr/src.old to
sdr/sdr.new.
>3. Then I have run: "patch < sdrpatch.diff. This reports me only the error
that I don't have any parsed_plugins.tcl file.
>4. A copy the plugins that I have downloaded to the sdr/src.new/plugins
directory and the directory.xbm to the sdr/src.new

Angel,
	At this point, if you haven't done so already, please try copying the
files "sdr2.plugin.S50.directory" and "sdr2.plugin.S51.directory.sap.sdp"
(available from <http://www.live.com/sdrpatch.html>) to your "plugins/"
directory, and try rebuilding a new "parsed_plugins.tcl" file from scratch.

(Please follow up by replying just to me, to avoid bothering the entire
"rem-conf" list.)

By the way, it would be good if someone (e.g., UCL?) were to incorporate
this patch into the 'official' "sdr" release (including the pre-compiled
binaries).

	Ross.




From rem-conf Sat Dec 19 03:46:40 1998 
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From: Khalil Drira <khalil@laas.fr>
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        g-troup@ccrc.wustl.edu, fokus-users@fokus.gmd.de,
        arpanet-bboard@mc.lcs.mit.edu, hipparch@sophia.inria.fr,
        ieee.announce@bellcore.com, osimcast@bbn.com,
        issll@mercury.lcs.mit.edu, heads@hpovua.org, members@hpovua.org,
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Subject: CFP Coordination session in PDPTA'99
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	[Our apologies for duplicate messages]
=======================================================================
			   Call for Papers

			  Special Session on

 Coordination in Parallel and Distributed Applications and Activities

		   http://www.laas.fr/~khalil/cpdaa

			A session in PDPTA'99
			June 28 - July 1, 1999
	      Monte Carlo Resort, Las Vegas, Nevada, USA
		http://www.cps.udayton.edu/~pan/pdpta/
				   

    This session is an approved technical session in the 5th
International Conference on Parallel and Distributed Processing
Techniques and Applications, PDPTA'99 (http://www.cps.udayton.edu/~pan/pdpta
and http//www.jhpc.org/pdpta).

Aims  and  scope
===============

    The emergence of  new work  activities, where remote  participants
-supported by   coordination and  collaboration tools-  collaborate to
achieve complex tasks, makes coordination  from, both the activity and
the software point of view, an important research issue.  Coordination
is,    by  nature,  an inter-disciplinary     research area involving
different    research    skills and  serving     different application
areas. This  includes process coordination in multi-threaded programs,
components   coordination   in    concurrent applications,    workflow
management in CSCW, transactions in distributed applications, etc.
    The purpose of this session is to  bring together researchers, and
practitioners     working   on  coordination     in   these  different
disciplinaries.  The  session serves as   a  forum enabling experience
exchange between academia and industry  as well as between researchers
working in the different coordination research branches.


Topics
======

Topics of interest include but are not limited to:


    o  multisites coordination

    o  dynamic architecture management for multi-component 
	applications

    o  coordination software architecture

    o  coordination models and applications for collaborative work 
	and CSCW

    o  group decision protocols

    o  coordination models for programming languages and application 
       to Java and CORBA.

    o  formal and semi-formal techniques for coordination description 
	and analysis

Accepted  papers  publishing
============================

    Papers describing original   research results  are  solicited  and
authors  are  encouraged to   address both  theoretical and  practical
aspects of coordination.  All submitted papers will be subject to peer
review  and papers  accepted  for  presentation  at PDPTA'99   will be
published   in  the conference  Proceedings.  Some technical sessions
will be considered for publication in relevant international journals.


Paper  submission
=================

    Prospective  authors are invited to submit   three copies of their
draft paper (about  4 pages) to  one of the session co-chairs (address
is given below) by the due  date. E-mail and  Fax submissions are also
acceptable.  The length of the  Camera-Ready papers (if accepted) will
be limited to 7 pages. Papers  must not have been previously published
or currently submitted  for publication elsewhere.   The first page of
the draft paper should include: title of the paper, name, affiliation,
postal  address, E-mail address,  telephone number, and Fax number for
each author. The first page should also include the name of the author
who will  be presenting the  paper  (if accepted)  and a  maximum of 5
keywords.



Important  dates
=================

March 1, 1999 (Monday):		Draft papers (about 4 pages) due
April 5, 1999 (Monday):		Notification of acceptance
May 1, 1999 (Monday):		Camera-Ready papers & Prereg. due
June 28 - July 1, 1999:		PDPTA'99 Conference



Session  co-chairs
=================

Khalil DRIRA                             Raul JACINTO MONTES
Communication Tools and Software Group   Distributed and Parallel Systems Group
LAAS-CNRS                                CINVESTAV
7, Av. Colonel Roche                     Prol. Av. Lopez Mateos Sur 590
31077 Toulouse Cedex 4                   c.p. 45090 Guadalajara Jalisco
FRANCE                                   MEXICO
e-mail: khalil@laas.fr                   e-mail: rjacinto@gdl.cinvestav.mx
Tel: +33 5 61 33 63 22                   Tel: +52 3 6.84.15.80
Fax: +33 5 61 33 64 11                   Fax: +52 3 6.84.17.08

WEB-ADRESSES : PDPTA'99: 
    http://www.cps.udayton.edu/~pan/pdpta
    and http//www.jhpc.org/pdpta




From rem-conf Mon Dec 21 07:03:01 1998 
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Date: Mon, 21 Dec 1998 15:46:04 +0100
From: Christine Guillemot <Christine.Guillemot@irisa.fr>
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To: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
CC: "'rem-conf@es.net'" <rem-conf@es.net>,
        "'4onip-sys@fzi.de'" <4onip-sys@fzi.de>
Subject: Re: Comments on Generic RTP Payload Format proposal
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Dear Anders,

Thanks a lot for the constructive comments.

Anders Klemets (Exchange) a =E9crit:

> I have a couple of minor comments on the Generic RTP Payload Format
> proposal, draft-guillemot-genpayload-00.txt, that was presented at the =
IETF
> AVT working group meeting.
>
> The payload format supports grouping and fragmentation, of higher level
> "application units".  It is also possible to carry arbitrary extension =
data.
> It is through this extension mechanism that the payload format can be m=
ade
> to support FEC with parity codes, FEC with Reed-Solomon codes, redundan=
cy,
> etc.
>
> The correct interpretation of the extension data is given by the XT
> (Extension Type) header field.  I think the main concern that I heard f=
rom
> several people at the IETF meeting, was that this introduces an extra
> multiplexing point.  To figure out how to interpret an RTP packet, one =
must
> first look at the RTP Payload Type field, and then look at the XT field=
 in
> the generic payload format.

May be I am missing something here, and sorry if I do,
but I do not see why this should be
'more an issue' than to parse and interpret a payload header as in the pa=
yload
formats for H.261, H.263, H.263+, MPEG1 etc, .... The payload header in
these formats may contain some redundant data (duplicated picture headers=
),
that has to be parsed and interpreted by a so-called 'network adaptation =
layer'
which is going to examine if the previous packet has been received correc=
tly
..etc
and if the duplicated data has to be passed to the decoder.

The idea was to apply the same scenario but allowing different types of
protections, like a 'flexible payload header'.
For example, looking at the  video syntax hierarchy, the duplicated data
may have a different structure depending on the AU it refers to, dependin=
g on
the
level of the hierarchy (VOsequence, VO, VOL, GOV, VOPs....).
The same mechanim could be used also for redundant data as audio re-encod=
ed at
a lower rate, or for other types of protection like parity codes.

>
>
> However, after having looked at the Internet-Draft again, I now think t=
his
> might not be such a big issue after all. The reason is that the
> Internet-Draft does not specify any static assignments for the XT field.
> Rather, it says that the values to be used for the XT field should be
> communicated out-of-band, using SDP or similar mechanisms.

Yes.

>
>
> I would think that in most cases, the extension data field would not be
> used, or it would only be used in one single way. For instance, I think=
 it

> is unlikely that some packets in an RTP session would have extension da=
ta
> for parity FEC, while other packets in the same session would use
> Reed-Solomon FEC.

Yes I agree it would be probably unlikely to mix parity FEC and R-S FEC, =
butone
could think of mixing protection in the sense of repeating vital headers
with some protection like duplicating audio data at a lower rate, or like
parity codes of less vital but still important video information...

>
>
> So, when SDP indicates that there are 0-1 valid values for the XT field=
, the
> RTP receiver can effectively ignore the field.  Thus, the problem with =
the
> additional multiplexing point becomes a non-issue in the typical case.
>

Yes, the XT field is not used when one does not need/want additional
protectionof the streams or segments of streams like decoder config param=
eters
or others...

> Given this, one could argue that the XT field should be removed, althou=
gh
> that would give up some flexibility that could be useful in the future.=
  In
> any case, it seems clear that 8 bits is an excessively large size for t=
his
> field.
>

Yes, the number of bits for this field can be reduced.

> I would also like to second Jonathan Rosenberg's comment, that it is be=
tter
> to specify the format of the extension data field by referring to the
> relevant documents, rather than including the text explicitly in this
> Internet-Draft.  Of course, there might be cases where no relevant docu=
ment
> exists.  In those cases, it might still be better to publish the format=
 of
> the extension data field in a different document, which would be separa=
te
> from the document that specifies the Generic Payload Format.

Actually the part in the I-D concerning the usage for R-S and parity code=
sneeds
definitely to be revised.
Actually what we were thinking of, was to use the XT to select a
FEC mechanism, including the pattern used
for the FEC, namely the mask from a pre-defined set instead of having the=
 mask
inserted in the extension data, this would save the size of the field for=
 the
mask
(24 bits).  This may limit the number of possible patterns (masks) but at=
 this
level, we may not need all the possibilities provided by
the 24 bits mask.

>
>
> I also have some comments regarding the alignment of the various fields=
 in
> the payload format.  The primary concern of any payload format that
> implements "grouping" or "multiplexing", is to reduce the number of pac=
kets.
> The second concern would be to reduce the number of bytes in the payloa=
d
> format header.  This is quite important when very small application uni=
ts
> are grouped together.  To achieve this second goal, I think most people
> would accept giving up word alignment of header fields, as long as the
> header fields are still aligned to byte boundaries.
>
> In this payload format, however, individual header fields are not byte
> aligned.  Padding is inserted at the end of the last header field to en=
sure
> that the payload begins at a byte boundary.  I know that byte alignment=
 is
> not an issue in the MPEG world (Just look at the MPEG-4 SL-Packet heade=
r
> definition and you will be either horrified or delighted, depending on =
where
> you come from) but in the IETF header fields are typically aligned.
>
> I have two different proposals for how the header syntax can be modifie=
d.
> Both proposals ensure that header fields are byte aligned, but they wor=
k a
> little bit differently.  The proposals all show that byte alignment can=
 be
> achieved at a maximum cost of one byte, sometimes even less.
>
> Proposal 1:
>
> * Reduce the XT field from 8 bits to 6 bits.  As discussed earlier, it =
is
> hard to conceive of a situation where someone would want more than 64
> different kinds of Extension Data in the same RTP session.
>

yes :-).

> * If E=3D0, insert a new field with 5 bits, directly following the F fi=
eld.
> The 5 bits are reserved and should always be set to 0.
>
> * The size of the LENGTH field is increased from 13 to 16 bits.
>
> This causes all header fields to be aligned with byte boundaries.  No
> padding field is necessary.  This comes at a cost, however.  If you loo=
k at
> example #3 in draft-guillemot-genpayload-00.txt, you might see that the
> second header, which consists only of flags and a LENGTH field, will
> increase from two bytes to three bytes.  And this is a fairly common ca=
se,
> because it happens whenever the packet consists of extension data and a=
n
> application unit that is not being fragmented.
>
> Proposal 2:
>
> * Reduce the XT field from 8 bits to 6 bits.  See comment above.
>
> * If E=3D1, increase the size of the LENGTH field from 13 to 16 bits.
>
> * If E=3D0, the size of the LENGTH field remains at 13 bits.
>
> * If G=3D0 and E=3D0, the size of TSOFFSET is reduced from 16 to 13 bit=
s.
>
> * If G=3D1 or E=3D1, the size of TSOFFSET remains at 16 bits.
>
> * If the FOFFSET field would directly follow the flag fields (because
> neither LENGTH nor TSOFFSET is present), insert 5 reserved bits in fron=
t of
> FOFFSET, instead of adding 5 bits of padding at the end of FOFFSET.
>
> This causes all header fields to be aligned with byte boundaries.  And =
this
> comes at no extra cost in terms of the header size.  As a matter of fac=
t,
> this scheme saves one entire byte when the payload header only contains=
 the
> TSOFFSET field.  (See example #1, in draft-guillemot-genpayload-00.txt.=
)

Thanks for these suggestions. The document is in a very preliminary versi=
on
and is open for amendments.

> Of
> course, you may already have noticed that there is another kind of cost
> associated with this proposal: complexity.
>

Let us imagine that we have an application with first an MPEG-4 video str=
eam,
for which wewant only to protect the vital information, let's say the dec=
oder
configuration parameters,
by repeating this information. Let us consider for this video stream
two levels of such config parameters:
1) the GOV header parameters needed for the random access point functiona=
lity
and
2) the VOP decoder config parameters.
Let us assume that this application also uses an audio
stream in which we want to use the redundant mechanism which consists in
repeating
the audio data transmitted in packet n-1 and in packet n-2 at a lower rat=
e.

Why this payload format would be more complicated than supporting two pay=
load
formats, namely:
1)- the redundant audio payload format for which you would need a process
to parse and interpret the payload header, extract the data of packet n-1=
 and
data of
packet n-2, see if these data are really needed (namely if the previous n=
-1 and
n-2 packets
have been received correctly) in order to parse the right data to the rig=
ht
decoders.
2)- an MPEG-4 video payload format, which would also parse its own header=
,
extract the duplicated headers, and pass them to the decoder if relevant =
(i.e.
if previous
packets containing these headers have been lost).

The complexity resides in the protection mechanism (computing  the FEC,
extracting,
interpreting, the redundant data, recovering loss packets from the redund=
ant
data...), which
we would have to support in both cases, either in the case of 2 payload f=
ormats,
one
format for each stream, or in the case of one payload for both streams.

I do not see why the fact of having factorized the grouping mechanism int=
o one
single format,
and to propose to signal the type of protection with the XT field would b=
ring
more complexity.

Now if we want to use this mechanism for applying FEC (parity codes or R-=
S)
on different segments of streams, then this may be more complex. But the
complexity
would come from the target of Unequal Error Protection, so again from the=
 FEC
computation
mechanism, more than from the payload encapsulation process itself. Right=
?

thanks again,
Best regards,

Christine


> Anders






From rem-conf Mon Dec 21 08:24:32 1998 
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Message-ID: <367E7470.7213E5D1@dnrc.bell-labs.com>
Date: Mon, 21 Dec 1998 11:16:48 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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To: Christine Guillemot <Christine.Guillemot@irisa.fr>
CC: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>,
        "'rem-conf@es.net'" <rem-conf@es.net>,
        "'4onip-sys@fzi.de'" <4onip-sys@fzi.de>
Subject: Re: Comments on Generic RTP Payload Format proposal
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Christine Guillemot wrote:
> 
> > So, when SDP indicates that there are 0-1 valid values for the XT field, the
> > RTP receiver can effectively ignore the field.  Thus, the problem with the
> > additional multiplexing point becomes a non-issue in the typical case.
> >
> 
> Yes, the XT field is not used when one does not need/want additional
> protectionof the streams or segments of streams like decoder config parameters
> or others...
> 
> > Given this, one could argue that the XT field should be removed, although
> > that would give up some flexibility that could be useful in the future.  In
> > any case, it seems clear that 8 bits is an excessively large size for this
> > field.
> >
> 
> Yes, the number of bits for this field can be reduced.

This observation can apply equally well to the payload type indicator in
the RTP header. With dynamic payload types, the number of bits in the
header need only cover the number of possible codecs/payload formats one
might observe during a particular session.


> 
> > I would also like to second Jonathan Rosenberg's comment, that it is better
> > to specify the format of the extension data field by referring to the
> > relevant documents, rather than including the text explicitly in this
> > Internet-Draft.  Of course, there might be cases where no relevant document
> > exists.  In those cases, it might still be better to publish the format of
> > the extension data field in a different document, which would be separate
> > from the document that specifies the Generic Payload Format.
> 
> Actually the part in the I-D concerning the usage for R-S and parity codesneeds
> definitely to be revised.
> Actually what we were thinking of, was to use the XT to select a
> FEC mechanism, including the pattern used
> for the FEC, namely the mask from a pre-defined set instead of having the mask
> inserted in the extension data, this would save the size of the field for the
> mask
> (24 bits).  This may limit the number of possible patterns (masks) but at this
> level, we may not need all the possibilities provided by
> the 24 bits mask.

This also is an argument that can be applied directly to the parity
document. The original proposal by Budge et. al. did explicitly that; it
enumerated each FEC mechanism and assigned it a number. The packet
contained just that number, effectively a payload type, plus some other
parameters, like position in the block (I think). The more generic
mechanism was preferred seems it allows a really wide range of
algorithms with a not-so-unreasonable overhead. This allows for vendor
differentiation in terms of choices of algorithms, and for the ability
to do adaptation over a broad operating range. I don't see why the
reasons for choosing the more generic mechanism apply there but not
here.

It seems we are falling into this particular tradeoff issue over and
over, both here and in the muxing work - efficiency vs. flexibility. For
any protocol, one can make all kinds of assumptions about its operating
range, and then optimize the header to work only under those
assumptions. Or, one can leave it flexible at the expense of more bits.
Which is the right direction to go often seems a matter of religion and
personal preference. I think its fair to say that in the past,
generality and flexibility seem to have won (RTP being fairly flexible
and applicable over a wide range of operation). I personally feel that
choosing generality over efficiency (within reason, of course) is
better. Time and time again, we have seen reuse and innovative
applications of general protocols. Optimizing protocols for particular
applications just means that when you assumptions change, you need to
design a new protocol for the new assumptions.

-Jonathan R.



-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Tue Dec 22 19:18:46 1998 
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Message-Id: <199812230309.MAA16939@aohakobe.ipc.chiba-u.ac.jp>
To: rem-conf@es.net
Subject: Re: vic-2.8 problem. 
In-reply-to: Your message of "Wed, 11 Nov 1998 06:07:18 +0900."
             <199811102107.GAA28741@aohakobe.ipc.chiba-u.ac.jp> 
Date: Wed, 23 Dec 1998 12:09:40 +0900
From: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
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Hi. I sent a message about "vic-2.8 problem" on 11th, November.
Hakan from cdt.luth.se responded to me that 

> Vic 2.8 has a memory leak. 13 bytes are wasted for each incomming
> RTCP addr.

and gave me the diff below. (thanks!)
yes, I'm also observing the process is growing bigger and bigger and
die at last.

with this patch I tried and succeeded to compile vic-2.8 with
tcl7.6 and tk4.2, but dumps core right after starting...
(segmentation fault at Tcl_GetFileInfo().)
I know the right version of tcl and tk for vic-2.8 is tcl7.5 and tk4.1,
and <http://www-nrg.ee.lbl.gov/vic/> does say nothing about newer version,
but I want to avoid installing older version of tcl&tk.

anyone trying to provide the binary merging the patch,
or release vic-2.8.1 with tcl&tk8?

-- yozo.

%%%% diff -c /usr/local/src/multicast/vic/vic-2.8/source.cc{.orig,} %%%%

*** /usr/local/src/multicast/vic/vic-2.8/source.cc.orig	Fri Apr  5 00:05:45 1996
--- /usr/local/src/multicast/vic/vic-2.8/source.cc	Wed Nov 11 23:33:39 1998
***************
*** 250,260 ****
  			tcl.result(wrk);
  			return (TCL_OK);
  		}
! 		if (strcmp(argv[1], "addr") == 0) {
! 			strcpy(wrk, InetNtoa(addr_));
! 			tcl.result(wrk);
! 			return (TCL_OK);
! 		}
  		if (strcmp(argv[1], "srcid") == 0) {
  			sprintf(wrk, "%lu", (u_long)ntohl(srcid_));
  			tcl.result(wrk);
--- 250,263 ----
  			tcl.result(wrk);
  			return (TCL_OK);
  		}
!                 if (strcmp(argv[1], "addr") == 0) {
!              /* HL 980415 Memory leak fix (hakanl@cdt.luth.se) */
!                  char *ptr= InetNtoa(addr_);
!                  strcpy(wrk, ptr);
!                  free(ptr);
!                  tcl.result(wrk);
!                  return (TCL_OK);
!                 }
  		if (strcmp(argv[1], "srcid") == 0) {
  			sprintf(wrk, "%lu", (u_long)ntohl(srcid_));
  			tcl.result(wrk);

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%



From rem-conf Wed Dec 23 11:55:13 1998 
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Date: Wed, 23 Dec 1998 11:37:06 -0800 (PST)
From: Steven Berson <berson@ISI.EDU>
To: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
Cc: rem-conf@es.net
Subject: Re: vic-2.8 problem. 
In-Reply-To: <199812230309.MAA16939@aohakobe.ipc.chiba-u.ac.jp>
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On Wed, 23 Dec 1998, Yozo Toda wrote:

> anyone trying to provide the binary merging the patch,
> or release vic-2.8.1 with tcl&tk8?

Yozo,

The version of vic with FreeBSD has patches for tcl&tk8.

Cheers,
Steve




From rem-conf Thu Dec 24 10:58:45 1998 
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Subject: Free Tip
Date: ไ, 24 ใ๖๎แ๘ 1998 16:56:39
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Dear Investor,

Free Tip Iceberg would like you to have this free tip on world currency:

Dollar vs. Japanese Yen is going to go up in the next few weeks between 
4% and 8%.
we hope you find it useful.

and remember, this is just the Tip of the Iceberg.

regards,
the Free Tip Team.

note: If you are not and investor or otherwise do not want to get a free 
tip, reply with subject line "remove".
 
 
 
 
 
 
 
 
 
 
 
 
 



From rem-conf Sat Dec 26 17:52:04 1998 
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Dear Investor / Business person,

Free Tip Iceberg would like you to have the following tip on world currency:

Dollar is going to go up vs. the Japanese Yen between 4% and 8% in the next few weeks.

We hope you find it useful.

And remember, this is just the Tip of the Iceberg.

regards,
Free Tip Iceberg team.

note: if you are not an investor or business person or otherwise do not want to get a 
free tip, please reply with subject "remove".
 
 
 
 
 
 



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transmitted and you may press the start button on your fax when you hear
the fax tone from our fax.
2.   no reply forms can be accepted by email....only via fax or smail.
3.   your form must be typewritten or printed out on your computer printer
before you fax it;  sorry, but *no* handwritten forms will be acknowledged.
If you can't find someone with a typewriter or a computer printer, we
apologize for not being able to reply to you.
4.   faxes with cover pages will be rejected.  You must send *only* the
reply form.
5.   forms not *completely* filled in will not be acknowledged.
6.   you will receive a reply within 1 business day directly from the
company making the offer via email.  Therefore you must have an email
address.  If you read this message, then you must have an email address, or
access to one, at least.   :-)
7.   your fax must not exceed 1 page in length.   Faxes of 2 or more pages
will be sensed, then auto-terminated and deleted.  Your fax goes directly
onto our 5.0 gigabyte hard drive and we must limit all incoming faxes to 1
page.
8.   all faxes must begin with:
*------------cut here/begin-------------------------------------------*
and must end with:
*------------cut here/end--------------------------------------------*
9. Any fax not conforming to this format will be sensed by our software,
then auto-terminated and deleted from the hard drive, before any human ever
gets to see it.
10. The type on your fax must be dark and legible.   If in doubt, please
print it out darker before faxing it in.  If we can't read it, we can't
reply to you or send you our FREE catalogue.     :-(
11.  If this all seems too complicated for faxing, just do it the old
fashioned way via smail!!!


WHO WE ARE:

Tempting Tear-Outs is an advertising company that brings potential new
customers to the companies they advertise for.

MORE ABOUT THE COMPANY MAKING THE FREE OFFER:

The company making the offer is a magazine subscription agency based in the
USA.  They have over 1,100 popular USA titles available to be shipped to
*any* country, including of course, to anywhere in the USA!    They offer a
FREE 1 yr. subscription to your choice of over 200 of the titles in their
catalogue to any new customer using them for the first time.       The
dollar value of the freebies, based on the subscription prices directly
>from the publishers, ranges from $6.97 all the way up to $50.00!

For new customers in the USA, there is no charge for FPH (foreign postage &
handling), so the freebie is 100% free!   For new customers living
overseas, the only charge on the freebie would be for the FPH (foreign
postage & handling).

Their president has been in the magazine subscription business since 1973
and they are very customer-service oriented.   They will even help you with
address changes on your magazines, even if you move from one country to
another country.   They have thousands of happy customers in over 59
countries.

Their price guarantee is very simple:       they guarantee that their
subscription prices are the lowest available and they will BEAT any
legitimate, verifiable offer before you pay them or match it afterwards, by
refunding you the difference in price PLUS the cost of the postage stamp
you would use sending in the special offer to them, even 6 months after you
pay them, as long as it was current at the time of your offer.    Does that
sound fair?       Wouldn't it be great if everything you bought came with
that price guarantee?

Sometimes they are less than half of the next best deal out there,
sometimes just a little cheaper, but always you get the lowest rates
without having to shop around.     With 1,100+ titles on their list, they
would like to think that they have also the best selection around!

Within the USA, for their USA customers, they are cheaper than all their
competitors and even the publishers themselves.  This is their price
guarantee.         The 1 yr. freebie that you get with your first order is
completely free!

Overseas, (even after you factor in the cost of the FPH (foreign postage &
handling) and the conversion from USA Dollars to your currency), on the
average, they are generally around one-fourth to one-half of what the
newsstands overseas charge locally for USA magazines.  On some titles they
are as little as one-tenth of what the newsstands charge.  They are also
the cheapest subscription source for delivery overseas, including directly
>from the publishers themselves!   Some publishers don't even offer
subscriptions overseas.........but overseas subscriptions are this
company's specialty!  They feel that magazines should not be a luxury
overseas.   In the USA, people buy magazines and then toss them after
reading them for just a few minutes or hours.  They are so cheap in the
USA!   Well, this company would like to make it the same way for their
overseas customers.  They are also cheaper than all their competitors in
the USA and overseas, including the publishers themselves!   It is also
*highly unlikely* you will find any of their USA competitors calling you
overseas, in order to offer that personal touch, just to sell you a couple
of magazines!  But that is what this company specializes in and loves
doing!     Around one-half their business comes from overseas, so they are
very patient with new customers who only speak limited English as a 2nd
language.    Subscription prices quoted for overseas consist of the
subscription price, plus the FPH.   You add the two together and that is
your total cost.   The exception is the 1 yr. freebie you get with your
first order.   On that title, you pay *only* the FPH for the 1 yr. term.

Their prices are so cheap because when you deal with them, you cut-out all
the middlemen.


HERE IS HOW YOU CAN GET MORE INFO AND GET STARTED WITH THEM:

Simply fax or smail back to us the reply form listed at the top of this
message.   We will then forward your form on to the subscription agency.
They will then email their "big and juicy" catalogue to you, in whichever
of the two formats you chose.   The catalogue is FREE and makes for hours
of fascinating reading, on its own. It includes the complete list of
freebies, a complete list of all the titles they sell, as well as detailed
descriptions on most of the titles, along with lists of titles by category
of interest and their terms of sale.

They will then give you a friendly, no-pressure, no obligation, 5-minute
call to go over how they work and to answer any questions that you might
have, as well as give you up-to-the minute price quotes on any titles you
might be considering.     They will call you in whatever country you live
in, taking the time difference into account.        As they like to
emphasize the personal touch they give to each new customer, all first-time
orders can only be done via phone, so they can answer all your questions
completely and personally.   Once you have placed your first order via
phone, you will be able to place future orders and make inquiries on your
account, get price quotes, etc., all via email, if that is most convenient
for you.

Within the USA, they accept payment via check over the phone, Mastercard,
Visa, American Express, Diner's Club and Carte Blanche.    Overseas, they
accept Mastercard, Visa, American Express, Diner's Club and Carte Blanche,
even if your credit card is a local one in local currency (that most
merchants in the USA would not normally be willing to accept).

That's our introduction of our client that we represent.   We hope that we
have piqued your interest and that you will take the next step to get their
free catalogue!   Thank you for your time and interest.

--
Tempting Tear-Outs.
For more info on advertising rates, please write us on your company
letterhead, w/business card, via smail to:   Tempting Tear-Outs, 3835
Richmond Ave. Suite #200, Staten Island NY  10312-3828, USA.





From rem-conf Mon Dec 28 07:44:42 1998 
From rem-conf-request@es.net Mon Dec 28 07:44:40 1998
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Message-Id: <199812282317.VII5055@gnmj.jvokt.ktkn.com>
Date: Mon, 28 Dec 1998 11:17:34 PM
Subject: 10 Asian SEX Girl Pictures a day !!
X-UIDL: 870483743.727
From: wwspwq@rnii.igcrr.rrfh.com
Bcc:
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 IF YOU ARE UNDER 21 DELETE NOW!
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

 10 Asian SEX Girl Pictures a day !!
 
    http://members.tripod.com/xcds1234/door.html

 All models are at least 18 years of age!




From rem-conf Mon Dec 28 11:26:21 1998 
From rem-conf-request@es.net Mon Dec 28 11:26:20 1998
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	Mon, 28 Dec 1998 14:14:42 -0500 (EST)
Message-ID: <3687D958.8963C7BE@TheOTG.com>
Date: Mon, 28 Dec 1998 14:17:44 -0500
From: "John Weiler, CTO and Founder" <john_weiler@TheOTG.com>
Organization: The OBJECTive Technology Group, an OMG Affiliate
X-Mailer: Mozilla 4.06 [en] (WinNT; I)
MIME-Version: 1.0
To: ckobryn@shl.com
CC: Taieb Znati <znati@cs.pitt.edu>, commsoft@cc.bellcore.com, odp@dstc.edu.au,
        dbworld@cs.wisc.edu, enternet@BBN.COM, JavaCORBA@luke.org,
        ISWORLD@Danann.hea.ie, comsoc-chapters@IEEE.ORG, comsoc-gicb@IEEE.ORG,
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        sc6wg4@ntd.comsat.com, sigmedia@bellcore.com, tccc@IEEE.ORG,
        xtp-relay@cs.concordia.ca
Subject: Free OMG Industry@OOTS Object/Internet Primer, Jan. 14
References: <199805211953.PAA15292@zeus.cs.pitt.edu>
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Chris,

Attached is the final Industry@OOTS agenda. Please pass this complimentar=
y invite
to your customers, business associates and IT staff so they may take adva=
ntage of
this unique opportunity.  It is designed for those who are pushing relate=
d
architecture initiatives and increase cooperation between industry advoca=
tes.
Many of the speakers are also engaged in the Interoperability Clearinghou=
se
initiative.

------------------- INDUSTRY@OOTS Object/Internet Primer ----------------=
--

You and your associates are invited! Register today.....

On Jan. 14th, 1999, the OMG in cooperation with The OBJECTive Technology =
Group
will be co-hosting the OMG Industry@OOTS Object/Internet Primer at the Cr=
ystal
Gateway Mariott.  This one day free event will provide an architectural v=
iew of
how objects, components and the internet are being successfully adopted a=
nd
implemented by industry.  Industries=92 top practitioners will gather to =
provide a
comprehensive set of tutorials on how this new set of standards based
internet-enabled components come together for a Networked Information Ent=
erprise.

This FREE Industry@OOTS primer will be held at the Crystal Gateway Marrio=
tt in
Arlington, VA concurrently with the OMG Technical Committee meeting.  The=
 theme
will be "Networked Information Enterprise: From Architectures to Reality"=
 and will
focus on defining those sets of internet-enabled interoperable frameworks=
 that
have been successfully developed and deployed.

Updated program information will be online at http://www.theotg.com, and
registration at http://www.omg.org/techprocess/meetings/ootprmr.html. You=
 may also
register by faxing your contact information (name, organization, title, p=
hone,
fax, email)  to 508-820-4303, attention =93OMG INDUSTRY@OOTS Registration=
) or email
the same to loughry@omg.org.  SEATING IS EXTREMELY LIMITED, so register t=
oday.

AM Program: 0800-1200
0800 Introductions and State of the Market =96 JOHN WEILER CTO, The OBJEC=
Tive
Technology Group
0815 Architectures for Interoperability - RICHARD SOLEY CEO/Chairman OMG
0900 Networked Information Infrastructure - DR. LINTON WELLS Principal DA=
SD, OSD
C3I
0945 Enterprise CORBA Beans =96 BILL ROTH, Product Line Manager, Sun Micr=
osystems
(or alt)
1015 break
1030 COM/CORBA Interoperability =96 JEFF BRUNELL Group Mgr, META Group (C=
hair),
Andre Yee, Senior Director, R&D for SAGA Software, PETER KRUPP Sr. Archit=
ect Iona
Technologies, Rick Rosen, Sr. Architect, BEA (or alt)
1115 Components Architectures Panel =96 Brian Buck Director of Architectu=
res
Platinum Technology, PATTY DOCK VP Corporate Product Strategy, BEA, KEVIN=
 KERNAN
VP Rational Software, DENNIS REEDY Sr. Java Architect Sun Microsystems

PM Program: 1330-1800
1330 Implementing Internet-enabled Frameworks =96 MICHAEL NELSON Program =
Director,
Internet Technology IBM
1400 Developing the Component Architectures w/ XML, UML, MOF and XMI - SR=
IDHAR
IYENGAR Unisys Fellow
1430 Modeling Distributed Component-based Architectures Panel - NEAL MARP=
LE SAIC
(Chair), WILL TRACZ Lockheed Martin, SIDNEY BAILIN President KEI, JOAQAUI=
N MILLER
Sr. Architect MCI Worldcom
1515 break
1530 Making IT Happen Panel - KEN NORLAND Ernst & Young (Chair), BRUCE AM=
BLER
Lucent, BOB MARCUS VP GM, RICK BENNETT Architecture Foundations AT&T
1630 Electronic Commerce...Myths and Reality, Patty Dock, VP Corporate Pr=
oduct
Strategy BEA 1730 Interoperability Initiatives update - JOHN WEILER CTO T=
he
OBJECTive Technology Group,  David Signori ISO Deputy Director, DARPA (or=
 alt)

* bolded names =3D confirmed participation

A set of white papers will be assembled with the sponsoring organizations=
 and
posted at www.TheOTG.com. You MUST register beforehand via TheOTG web sit=
e as we
have done in the past for other OOTS seminars.  Registration is online vi=
a TheOTG
web site or the following: http://www.omg.org/techprocess/meetings/ootspr=
mr.html

Details of these interoperable frameworks (specific technologies, product=
s, and
standards), will be entered into an online knowledge repository called th=
e
Interoperability Clearinghouse, and made available to sponsors, subscribe=
rs and
conference attendees. Information on this joint govt/industry initiative =
can be
found; http://www.omg.org/techprocess/meetings/ic.html or
http://www.infoworld.com/cgi-bin/displayStory.pl?981113.whsoft.htm

Those who are not OMG members and are interested in attending the OMG Tec=
hnical
Committee meetings being held the week of Jan. 11th, can do so as a paid =
guest of
TheOTG ($350.)  Fee includes all sessions, breaks and lunches for entire =
week.
For information regarding registration and fees, please go to;
http://www.omg.org/techprocess/meetings/meetingreg.html

********************************************
John A. Weiler
CTO & Founder, The OBJECTive Technology Group
OMG Government Liaison
Member OMG, IEEE, ACM, AFCEA
8011 Washington, Ave.
Alexandria, VA 22308
703-768-0400(v) 703-765-9295(f)
http://www.TheOTG.com (Industry@OOTS Forums)
john_weiler@TheOTG.com

"From Architectures to Reality"
*********************************************





From rem-conf Mon Dec 28 15:56:08 1998 
From rem-conf-request@es.net Mon Dec 28 15:56:07 1998
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	id 0zumSC-0005fR-00; Mon, 28 Dec 1998 15:52:04 -0800
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	by mail1.radix.net (8.9.1/8.9.1) with ESMTP id RAA28978;
	Mon, 28 Dec 1998 17:48:07 -0500 (EST)
Message-ID: <36880B5D.B65B38CE@TheOTG.com>
Date: Mon, 28 Dec 1998 17:51:09 -0500
From: "John Weiler, CTO and Founder" <john_weiler@TheOTG.com>
Organization: The OBJECTive Technology Group, an OMG Affiliate
X-Mailer: Mozilla 4.06 [en] (WinNT; I)
MIME-Version: 1.0
To: john_weiler@omg.org
Subject: Complimentary Industry@OOTS Object/Internet Primer, Jan. 14th.
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Dear Colleagues,

The following is a complimentary invite to the OMG Industry@OOTS
Object/Internet
Architecture Primer held in Arlington VA on Jan. 14th 1999, from 0800 to
1800 hrs.
Registration: http://www.omg.org/techprocess/meetings/ootsprmr.html
inquiries: john_weiler@omg.org

----------------- INDUSTRY@OOTS Object/Internet Primer------------------

You and your associates are invited!

On Jan. 14th, 1999, the OMG in cooperation with The OBJECTive Technology
Group will be co-hosting the OMG Industry@OOTS Object/Internet Primer at
the Crystal Gateway Mariott.  This one day free event will provide an
architectural view of how objects, components and the internet are being
successfully adopted and implemented by industry.  Industries=92 top
practitioners will gather to provide a comprehensive set of tutorials on
how this new set of standards based internet-enabled components come
together for a Networked Information Enterprise.

This FREE Industry@OOTS primer will be held at the Crystal Gateway
Marriott in Arlington, VA concurrently with the OMG Technical Committee
meeting.  The theme will be "Networked Information Enterprise: From
Architectures to Reality" and will focus on defining those sets of
internet-enabled interoperable frameworks that have been successfully
adopted, developed and deployed.

Updated program information will be online at http://www.theotg.com, and
registration at
http://www.omg.org/techprocess/meetings/ootsprmr.html
You may also register by faxing your contact information (name,
organization, title, phone, fax, email)  to 508-820-4303, attention =93OM=
G
INDUSTRY@OOTS Registration) or email the same to loughry@omg.org.
SEATING IS EXTREMELY LIMITED, so register today.

AM Program: 0800-1200
0800 Introductions and State of the Market =96 JOHN WEILER CTO, The
OBJECTive Technology Group
0815 Architectures for Interoperability - RICHARD SOLEY CEO/Chairman OMG

0900 Networked Information Infrastructure - DR. LINTON WELLS Principal
DASD, OSD C3I
0945 Enterprise CORBA Beans =96 BILL ROTH, Product Line Manager, Sun
Microsystems (or alt)
1015 break
1030 COM/CORBA Interoperability =96 JEFF BRUNELL Group Mgr, META Group
(Chair), Andre Yee, Senior Director, R&D for SAGA Software, PETER KRUPP
Sr. Architect Iona Technologies, Rick Rosen, Sr. Architect, BEA (or alt)

1115 Components Architectures Panel =96 Brian Buck Director of
Architectures Platinum Technology, PATTY DOCK VP Corporate Product
Strategy, BEA, KEVIN KERNAN VP Rational Software, DENNIS REEDY Sr. Java
Architect Sun Microsystems

PM Program: 1330-1800
1330 Implementing Internet-enabled Frameworks =96 MICHAEL NELSON Program
Director, Internet Technology IBM
1400 Developing the Component Architectures w/ XML, UML, MOF and XMI -
SRIDHAR IYENGAR Unisys Fellow
1430 Modeling Distributed Component-based Architectures Panel - NEAL
MARPLE SAIC (Chair), WILL TRACZ Lockheed Martin, SIDNEY BAILIN President
KEI, JOAQAUIN MILLER Sr. Architect MCI Worldcom
1515 break
1530 Making IT Happen Panel - KEN NORLAND Ernst & Young (Chair), BRUCE
AMBLER Lucent, BOB MARCUS VP GM, RICK BENNETT Architecture Foundations
AT&T
1630 Electronic Commerce...Myths and Reality, Patty Dock, VP Corporate
Product Strategy BEA 1730 Interoperability Initiatives update - JOHN
WEILER CTO The OBJECTive Technology Group,  David Signori ISO Deputy
Director, DARPA (or alt)

* bolded names =3D confirmed participation

A set of white papers will be assembled with the sponsoring
organizations and posted at www.TheOTG.com. You MUST register beforehand
via TheOTG web site as we have done in the past for other OOTS
seminars.  Registration is online via TheOTG web http://www.theotg.com
site or the OMG site:
http://www.omg.org/techprocess/meetings/ootsprmr.html

Details of these interoperable frameworks (specific technologies,
products, and standards), will be entered into an online knowledge
repository called the Interoperability Clearinghouse, and made available
to sponsors, subscribers and conference attendees. Information on this
joint govt/industry initiative can be found;
http://www.omg.org/techprocess/meetings/ic.html or
http://www.infoworld.com/cgi-bin/displayStory.pl?981113.whsoft.htm

Those who are not OMG members and are interested in attending the OMG
Technical Committee meetings being held the week of Jan. 11th, can do so
as a paid guest of TheOTG ($350.)  Fee includes all sessions, breaks and
lunches for entire week.   For information regarding registration and
fees, please go to;
http://www.omg.org/techprocess/meetings/meetingreg.html

The OBJECTive Technology Group is an IT Architecture Consulting firm and
one of the primary sponsors of the Interoperability Clearinghouse
initiative.   Current engagements are helping refine distributed
architectures for major enterprises in Defense, Healthcare, Finance and
Telecommunication industries.  Some of these success and lessons learned
will be portrayed in the annual Industry@OOTS conference series, and
documented in the Interoperability Clearinghouse knowledge repository.

********************************************
John A. Weiler
CTO & Founder, The OBJECTive Technology Group
OMG Government Liaison
Member OMG, IEEE, ACM, AFCEA
8011 Washington, Ave.
Alexandria, VA 22308
703-768-0400(v) 703-765-9295(f)
http://www.TheOTG.com (Industry@OOTS Forums)
john_weiler@TheOTG.com

"From Architectures to Reality"
*********************************************



From rem-conf Tue Dec 29 12:03:04 1998 
From rem-conf-request@es.net Tue Dec 29 12:03:04 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0zv58f-0005Tv-00; Tue, 29 Dec 1998 11:49:09 -0800
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	by mail1.es.net with smtp (Exim 1.81 #2)
	id 0zv58X-0005Tk-00; Tue, 29 Dec 1998 11:49:08 -0800
From:     SKYSHOP OFFICE SUPPLIES <skyshop@excite.com>
To:        <rem-conf@es.net>
Message-Id: <419.436158.61900648skyshop@excite.com>
Subject: Are you paying a to much for office supplies? Skyshop can save you money .....
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Are you paying to much for office supplies? Skyshop CAN save you money.  

If you would like to see our product line & website please email us back in the subject 
line of the below email address and place the letters (SP)
Thank you!                                                                                    skyshop@gte.net
Skyshop Office Supplies "We CAN save you MONEY & TIME" !!


To be removed from our list please place the word remove and no further email will 
be sent to you at this mailing skyshop@gte.net




From rem-conf Tue Dec 29 12:35:53 1998 
From rem-conf-request@es.net Tue Dec 29 12:35:52 1998
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	id 0zv5mo-0006QZ-00; Tue, 29 Dec 1998 12:30:38 -0800
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	by mail1.es.net with esmtp (Exim 1.81 #2)
	id 0zv5mn-0006Q6-00; Tue, 29 Dec 1998 12:30:37 -0800
Received: from cisco.com (dhcp-i-222-174.cisco.com [171.69.222.174]) by jasper.cisco.com (8.8.5-Cisco.2-SunOS.5.5.1.sun4/8.6.5) with ESMTP id MAA13054 for <rem-conf@es.net>; Tue, 29 Dec 1998 12:29:36 -0800 (PST)
Message-ID: <36891E38.476D151E@cisco.com>
Date: Tue, 29 Dec 1998 12:23:52 -0600
From: "Sanjay K. Agrawal" <sagrawal@cisco.com>
Organization: Cisco Systems
X-Mailer: Mozilla 4.5 [en] (WinNT; I)
X-Accept-Language: en
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To: rem-conf@es.net
Subject: Voice and Video Traces 
Content-Type: multipart/mixed;
 boundary="------------A5F7B3A216491EB6F4B5A37D"
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This is a multi-part message in MIME format.
--------------A5F7B3A216491EB6F4B5A37D
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Hi,

I am performing networking simulations on the voice and video traffic. I
was
wondering if anybody can point me to packetized (IP) voice and
 video traces (H.261) for the use in my simulations. 

Thanks very much,

-Sanjay
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From rem-conf Tue Dec 29 20:28:04 1998 
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Date: 29 Dec 98 11:30:23 PM
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Subject: Ecstasy
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From rem-conf Wed Dec 30 06:10:58 1998 
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From: The IESG <iesg-secretary@ietf.org>
Subject: Protocol Action: Compressing IP/UDP/RTP Headers for Low-Speed 
         Serial Links to Proposed Standard                                 
Date: Wed, 30 Dec 1998 09:02:17 -0500
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  The IESG has approved the Internet-Draft "Compressing IP/UDP/RTP Headers 
  for Low-Speed Serial Links" <draft-ietf-avt-crtp-05.txt> as a Proposed 
  Standard. This document is the product of the Audio/Video Transport 
  Working Group. The IESG contact persons are Scott Bradner and Vern Paxson.
 
 
Technical Summary
 
This documents defines a protocol for reducing the size of IP, UDP
and RTP packets sent over a serial link by compressing out header
information that is constant, changes rarely, or only changes by
small increments.  For small packets and slow links, the savings
in transmission time can be quite significant.

Working Group Summary

The proposed protocol received solid working group support for its
technical merits.  There was however controversy within the WG that
the protocol could not be deployed soon enough to meet the quickly
emerging needs of Internet Telephony and similar applications.  The
alternative was to specify an end-to-end compression of just the RTP header
that could be implemented in the applications themselves as an interim
solution until the link-level compression specified in this draft could be
deployed.  Considering that the gain of compressing RTP alone would be
relatively small and that it could not be standardized in the necessary
timeframe, the prevalent position was that AVT should not define such
an interim solution.


Protocol Quality

Vern Paxson reviewed the specification for the IESG. The protocol is
sound, thoroughly analyzed, and builds appropriately on the earlier
work of RFC 1144.



From rem-conf Wed Dec 30 12:25:10 1998 
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Date: Wed, 30 Dec 1998 12:18:18 -0800 (PST)
From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
Message-Id: <199812302018.MAA07029@jackson.cs.ucsb.edu>
To: rem-conf@es.net
Subject: Re:  Voice and Video Traces
Cc: sagrawal@cisco.com
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>>I am performing networking simulations on the voice and video traffic. I
>>was
>>wondering if anybody can point me to packetized (IP) voice and
>> video traces (H.261) for the use in my simulations. 

The IETF source files would probably work:

ftp://ftp.cc.gatech.edu/people/kevin/

-Kevin



