From rem-conf Mon Feb 01 01:19:03 1999 
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
cc: Neal Schneider <nschneid@hotmail.com>, wchung@vmlabs.com, rem-conf@es.net
Subject: Re: Traffic Shaping
In-reply-to: Your message of "Thu, 28 Jan 1999 22:46:09 EST." <36B12F01.7D27E63A@dnrc.bell-labs.com>
Date: Mon, 01 Feb 1999 09:12:45 +0100
Message-ID: <806.917860365@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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In message <36B12F01.7D27E63A@dnrc.bell-labs.com>, Jonathan Rosenberg typed:

 >>> >Bolot, Garcia, "Control Mechanisms for Packet Audio in the Internet",
 >>> >Infocom 96
 
 >>ftp://ftp-sop.inria.fr/rodeo/bolot/96.Audio_ctl.ps.gz

there's some more recent work to appear in infocomm 99 by the same
folks
- see the umass report at http://gaia.cs.umass.edu/papers/
 J. Bolot, S. Fosse-Parisis, D. Towsley. ``Adaptive FEC-Based Error
Control for Interactive Audio in the Internet'', to appear in IEEE
Infocom' 99, (New York,
       USA, March 1999). 

the most important thing in this work is some neat stuff on utility
and that the subjective quality  is fixable with modest increasing FEC
overhead and is more important than objective measures....

 >>> >There is other work on emulating TCP behavior, but I don't have a
 >>> >specific reference (anyone?).

see the RMRG website and maillist....
 rm@mash.cs.berkeley.edu
http://www.east.isi.edu/rm/ 
 
 >>> Are you talking about providing some sort of acknowledgement-based
 >>> negotiation protocol over UDP

 >>No. I am talking about rate shaping in such a way that the rate used by
 >>the media stream is the same rate a TCP stream would use under the same
 >>loss conditions. 
 >>
 >> ... Does MGCP provide a mechanism for
 >>> modifying connections in progress?
 >>
 >>MGCP can modify its current connections. However, I don't know if rate
 >>adaptation is something the protocol is aiming to control. I'm also not
 >>sure rate adaptation is best done in the MGC, since its a pure media
 >>function it could be done entirely in the MG.
 >>
 >>-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
 >>

 cheers

   jon




From rem-conf Mon Feb 01 01:24:34 1999 
From rem-conf-request@es.net Mon Feb 01 01:24:33 1999
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To: rem-conf@es.net
Subject: New AVT charter posted
Date: Mon, 01 Feb 1999 09:22:25 +0100
Message-ID: <970.917860945@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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As some of you may have noticed, a new AVT charter has been posted on the
group's web site at http://www.ietf.org/html.charters/avt-charter.html
This is essentially the same as that presented in Orlando, with a few
clarifications based on IESG feedback and the milestones relaxed somewhat.

There are a large number of action items left over from the last meeting,
many of which have turned into milestones for February (drafts to submit
before the 26th February deadline for the Minneapolis meeting). These
action items include:

 - Post RTP implementation checklist draft
 - Working group last call on guidelines for payload format writers (BCP)
 - Post revised draft on PureVoice (qcelp) payload format to address WG 
   last call comments
 - Post payload format for MPEG-4 based on MPEG/IETF joint meetings
 - Working group last call on parity FEC draft (standards track)
 - Post revised RTP MIB and issue working group last call (stds track)
 - Post revised DTMF payload format draft, ready for WG last call
 - Post revised RTP membership (SSRC) sampling draft
 - Post revised Reed-Solomon draft
 - Post revised RTP spec and audio/video profile

As you can see, we have to get busy! 

Colin



From rem-conf Mon Feb 01 09:25:33 1999 
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From: "Nikos Laoutaris" <laoutaris@noc.uoa.gr>
To: <rem-conf@es.net>
Subject: RTP implementation for windows
Date: Mon, 1 Feb 1999 19:09:32 +0200
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Hello all at rem-conf   =20

    I am looking for RTP/RTCP implementations for windows. So far i have =
only found commercial products from Elemedia (at a rather high price =
tag) and RadVision. We are planning of integrating RTP/RTCP to our =
RSVP-MPEG-2 application, developed at the Communication and Networks =
laboratory at the university of Athens. Please inform me on the =
existance of non commercial RTP/RTCP protocol implementations. I know =
that vic and other Mbone tools that use RTP have been ported to windows =
but i have not found any source code for these ports.

Thanks in advance =20

Nikos Laoutaris
Network Operations Center
National and Kapodestrian University of Athens, Greece
e-mail : laoutaris@noc.uoa.gr

------=_NextPart_000_0010_01BE4E16.66608080
Content-Type: text/html;
	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD W3 HTML//EN">
<HTML>
<HEAD>

<META content=3Dtext/html;charset=3Diso-8859-1 =
http-equiv=3DContent-Type>
<META content=3D'"MSHTML 4.72.3110.7"' name=3DGENERATOR>
</HEAD>
<BODY bgColor=3D#c0c0c0>
<DIV><FONT color=3D#000000 size=3D2>Hello all at=20
rem-conf&nbsp;&nbsp;&nbsp;&nbsp;</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT><FONT color=3D#000000=20
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>&nbsp;&nbsp;&nbsp; I am looking for =
RTP/RTCP=20
implementations for windows. So far i have only found commercial =
products from=20
Elemedia (at a rather high price tag) and RadVision. We are planning of=20
integrating RTP/RTCP to our RSVP-MPEG-2 application, developed at the=20
Communication and Networks laboratory at the university of Athens. =
Please inform=20
me on the existance of non commercial RTP/RTCP protocol implementations. =
I know=20
that vic and other Mbone tools that use RTP have been ported to windows =
but i=20
have not found any source code for these ports.</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>Thanks in =
advance&nbsp;&nbsp;</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>Nikos Laoutaris<BR>Network =
Operations=20
Center<BR>National and Kapodestrian University of Athens, =
Greece<BR>e-mail : <A=20
href=3D"mailto:laoutaris@noc.uoa.gr">laoutaris@noc.uoa.gr</A></FONT></DIV=
></BODY></HTML>

------=_NextPart_000_0010_01BE4E16.66608080--




From rem-conf Mon Feb 01 09:32:11 1999 
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To: "Nikos Laoutaris" <laoutaris@noc.uoa.gr>
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Subject: Re: RTP implementation for windows 
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laoutaris@noc.uoa.gr said:
> I know that vic and other Mbone tools that use RTP have been ported to
> windows but i have not found any source code for these ports.

The initial port is actually contained within source code released. There 
shouldn't be many changes to the RTP part anyhow.

/P




From rem-conf Mon Feb 01 09:56:23 1999 
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To: Nikos Laoutaris <laoutaris@noc.uoa.gr>
cc: rem-conf <rem-conf@es.net>
Subject: Re: RTP implementation for windows
In-reply-to: Your message of "Mon, 01 Feb 1999 19:09:32 +0200." <001301be4e05$a3bb0ba0$255a86c3@Aragorn.noc.uoa.gr>
Date: Mon, 01 Feb 1999 17:53:20 +0100
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--> Nikos Laoutaris writes:
>    I am looking for RTP/RTCP implementations for windows. So far i have 
>only found commercial products from Elemedia (at a rather high price 
>tag) and RadVision. We are planning of integrating RTP/RTCP to our 
>RSVP-MPEG-2 application, developed at the Communication and Networks 
>laboratory at the university of Athens. Please inform me on the 
>existance of non commercial RTP/RTCP protocol implementations. I know 
>that vic and other Mbone tools that use RTP have been ported to windows 
>but i have not found any source code for these ports.

See http://www-mice.cs.ucl.ac.uk/multimedia/sofware/ for source code to vic
and rat (for unix and win32).

Colin



From rem-conf Mon Feb 01 20:01:10 1999 
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From:	"DR. DANNY TSANG" <eetsang@ee.ust.hk>
To:	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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	multicomm@cc.bellcore.com, activenets_all@ittc.ukansas.edu,
	fokus-users@fokus.gmd.de, arpanet-bboard@mc.lcs.mit.edu,
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Subject: IFIP BC'99
Cc:	eetsang@ee.ust.hk
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My apology if you receive multiple copies of this message.

The next IFIP Broadband Communications'99 will be held in Hong Kong 
on 10-12 November 1999. The deadline for paper submission is 28 February 1999
(4 weeks from today). Electronic submission is strongly encouraged. 
Please visit the conference web site at "www.ee.ust.hk/~bc99" for 
the different electronic submission formats.

Danny H.K. Tsang		Prof. Paul J. Kuehn
General Chair			Co-Chair

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


                              CALL FOR PAPERS

                                 IFIP TC 6
                      Fifth International Conference on

                        BROADBAND COMMUNICATIONS '99

                   Hong Kong Convention & Exhibition Centre
                                  Hong Kong

                             November 10-12, 1999


Scope of BC '99
---------------
Broadband communication networks form the essential infrastructure for 
new communication services. Network technologies will range from photonic
backbones, distributed networking, fast packet routing, IP networking and 
ATM switching transport networks to future mobile networks with seamless 
interworking. Middleware, network, service and application management software
are the enabling technologies to support a wide spectrum of conventional 
and new services.

BC '99 will focus on leading edge research arising from architectural, 
technological, operational, management and performance issues. Special 
interest will be given to the development in the Asian and Far East regions.

Conference topics of BC '99 include (but are not necessarily limited to):

Networking Infrastructures and Technologies
-------------------------------------------
IP Netwroks (Voice/Video over IP, DiffServ, MPLS)
ATM Networks (Voice/Video over ATM, QoS)
High Speed LANs
Wireless Networks 
Satellite Networks

Architectures and Protocols
---------------------------
Interworking of Fixed/Mobile Networks
Interworking of IP/ATM/SDH/WDM
Interworking of PSTN and IP Networks
Signalling, Multicast (IP/ATM)
Routing Protocols (mobile IP, QoS routing)
Addressing
Active Networks

Services and Applications
-------------------------
Multimedia Communications
Nomadic Communications and Computing
Virtual Private Networks
Service Integration
Enabling Technologies (middleware)

Service and Network Management
------------------------------
Configuration and Fault Management
Security
Accounting, Billing and Tariffing
Service and Application Management
Scaleable Service Provisioning

Performance, Quality of Service and Network Operations
------------------------------------------------------
Traffic Control and Resource Management
Traffic Modelling, Measurement, Analysis, Monitoring and Maintenance
Application QoS and Network QoS
Network Planning and Network Performance

SCIENTIFIC PROGRAMME COMMITTEE
------------------------------
Ajmone-Marsan, M. 	Politecnico di Torino, Italy
Albanese, A. 		International Computer Science Inst., U.S.A.
Bensaou, B. 		Centre for Wireless Commun., Singapore
Blondia, C. 		University of Antwerp, Belgium
Butscher, B. 		DeTeBerkom/GMD, Germany
Casaca, A. 		IST/INESC, Portugal
Casals, O. 		UPC, Spain
Chandran, S. 		Ericsson, Malaysia
Chao, J.		Polytechnic University, U.S.A.
Chen, W.T. 		National Tsing Hua University, Taiwan
Cheng, S. 		Southeast University, China
Chu, W. 		Open University of Hong Kong, Hong Kong
Costa, B.		CSELT, Italy
Cuthbert, L. 		QMW College London, UK
Denzel, W. 		IBM Rueschlikon, Switzerland
Drobnik, O. 		University of Frankfurt, Germany
Eberspaecher, J. 	Technical University of Munich, Germany
El-Zarki, M. 		University of Pennsylvania, U.S.A.
Fdida, S. 		LIP6 Paris, France
Gallassi, G. 		Italtel, Italy
Guerin, R. 		University of Pennsylvania, U.S.A.
Hébuterne, G. 		INT, France
Hubaux, J.-P. 		EPFL, Switzerland
Hui, J. 		Chinese University of Hong Kong, Hong Kong
Iversen, V.B. 		Technical University of Denmark, Denmark
Kawashima, K. 		NTT-AT, Japan
Killat, U. 		Technical University Hamburg-H., Germany
Koerner, U. 		University of Lund, Sweden
Kofman, D. 		Telecom Paris, France
Kuehn, P.J. 		University of Stuttgart, Germany
Leon-Garcia, A. 	University of Toronto, Canada
Leslie, I.		University of Cambridge, UK
Li, V.O.K. 		Hong Kong University, Hong Kong
Li, X.			Tsinghua University, China
Liang, X.J.		BUPT, China
Lin, X	.		Tsinghua University, China
Liu, E.Y.S. 		Cable and Wireless plc, UK
Low, S.			University of Melbourne, Australia
Mark, J. 		University of Waterloo, Canada
Miyaho, N. 		NTT, Japan
Niu, Z.			Tsinghua University, China
Nunes, M.S. 		IST/INESC, Portugal
Pettersen, H. 		Telenor R&D, Norway
Rathgeb, E. 		Siemens AG, Germany
Roberts, J.W. 		CNET, France
Rosenberg, C. 		Nortel Imperial College, UK
Ross, K. 		EURECOM, France
Saito, H. 		NTT, Japan
Stuettgen, H. 		NEC Europe Ltd., Germany
Spaniol, O. 		RWTH Aachen, Germany
Takahashi, Y. 		Nara Institute of Science and Tech., Japan
Tohmé, S. 		Telecom Paris, France
Tran-Gia, P. 		University of Wuerzburg, Germany
Tsang, D.H.K. 		Hong Kong University of Sc. & Tech., Hong Kong
van As, H. 		Vienna University of Technology, Austria
Van Landegem, T.	Alcatel, Belgium
Walke, B. 		RWTH Aachen, Germany
Wolisz, A. 		Technical U. of Berlin/GMD Fokus, Germany
Wong, J. 		University of Waterloo, Canada
Yang, T. 		Ascend Communications, U.S.A.
Zitterbart, M. 		Technical Univ. of Braunschweig, Germany


ORGANIZER
---------
International Federation on Information Processing
IFIP - WG 6.2: Broadband Communications

Department of Electrical & Electronic Engineering
Hong Kong University of Science & Technology


SUPPORTING ORGANIZATIONS:
------------------------
Ascend Communications Inc.
Cable & Wireless
Cisco Systems
Fore Systems
Global Technology Integrator
Hewlett Packard
Hongkong Telecom
Hongkong Telecom Institute of Information Technology
IEEE CAS/COM Hong Kong
Information Technology Division, Hong Kong Institution of Engineers


ORGANIZING COMMITTEE:
--------------------
Cheng, K.H. 		Hongkong Telecom, Hong Kong
Kuehn, P. J. 		University of Stuttgart, Germany
Liu, E. Y. S. 		Cable and Wireless plc, UK
Tsang, D. H. K. 	Hong Kong University of Science & Technology, Hong Kong


GENERAL CHAIR:
-------------
Tsang, D. H. K. 	Hong Kong University of Science & Technology, Hong Kong


CO-CHAIR:
--------
Kuehn, P. J. 		University of Stuttgart, Germany


INFORMATION FOR AUTHORS
-----------------------
Contributions will be refereed from full papers (in English). The submissions 
should not exceed 15 double spaced pages. The front page should contain the 
names of the authors, the corresponding author's affiliation, address, phone, 
fax and e-mail address, as well as an abstract. Authors of accepted papers
will be required to sign an IFIP copyright release form. The proceedings will 
be available at the conference, published in the IFIP series format by Kluwer 
Academic Publishers.

Submission through e-mail is strongly encouraged. Please send the compressed,
uuencoded postscript files to

  eetsang@ee.ust.hk

If e-mail submission is not possible, please send five copies of your 
manuscript by mail to

  Dr. Danny H. K. Tsang
  Department of Electrical and Electronic Engineering
  Hong Kong University of Science & Technology
  Clear Water Bay
  Kowloon, Hong Kong

For other submission formats, please consult our homepage 
http://www.ee.ust.hk/~bc99

IMPORTANT DATES:
---------------
Deadline for full paper submissions 		February 28th, 1999
Authors notification 				May 20th, 1999
Camera-ready version due 			July 15th, 1999


FURTHER INFORMATION:
-------------------
Home Page: www.ee.ust.hk/~bc99

Address:
Dr. Danny H. K. Tsang
Department of Electrical and Electronic Engineering
Hong Kong University of Science & Technology
Clear Water Bay, Kowloon, Hong Kong
Tel: (852) 2358 7045
Fax: (852) 2358 1485
E-mail: eetsang@ee.ust.hk



From rem-conf Tue Feb 02 15:05:36 1999 
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Subject: Berkeley Multimedia, Interfaces and Graphics Seminar 
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The Berkeley Multimedia, Interfaces and Graphics Seminar scheduled for
Wednesday, February 3, 1999 has been rescheduled to Wednesday, February 10,
1999.

There will not be a speaker on Wednesday, February 3, 1999.

I apologize for any inconvenience this might cause you.

Best Regards,
Florissa Colina


____________________________________________

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 Wed Feb 03 02:11:32 1999 
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From: "Gio tesista" <valchesi70@hotmail.com>
To: rem-conf@es.net
Subject:  multicast  S.O.S!!!
Date: Wed, 03 Feb 1999 01:47:57 PST
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I 'm italian student and I 'm making  some experiments on multicast. 
I have Linux configured 3 PC( kernel  2.0.36) with multicast support and 
one of these is a multicast router(mrouter-3.81).If we create a work 
session on our same  network,when we run "SD" or "SDR" I see us the 
session on other computers  and it's O.K..The problem is ONLY mrouter 
see external session (by  tunneling) out my network,  but the other 2 
computers no.What can I do  
to resolve this problem?

Thanks Giovanni

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



From rem-conf Wed Feb 03 02:36:53 1999 
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From: Uwe Jendricke <jendricke@telematik.iig.uni-freiburg.de>
Reply-To: jendricke@telematik.iig.uni-freiburg.de
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On Wed, 03 Feb 1999, Gio tesista wrote:
>I 'm italian student and I 'm making  some experiments on multicast. 
>I have Linux configured 3 PC( kernel  2.0.36) with multicast support and 
>one of these is a multicast router(mrouter-3.81).If we create a work 
>session on our same  network,when we run "SD" or "SDR" I see us the 
>session on other computers  and it's O.K..The problem is ONLY mrouter 
>see external session (by  tunneling) out my network,  but the other 2 
>computers no.What can I do  
>to resolve this problem?
>
>Thanks Giovanni

Hi Giovanni

for me it sounds like a multicast problem. Do all of your machines have
multicast-kernels? here is my kernel-config  (networking options) that works
with mbone:

    [*] Network firewalls                                                  x x
  x x    [*] Network aliasing                                                   x x
  x x    [*] TCP/IP networking                                                  x x
  x x    [*] IP: forwarding/gatewaying                                          x x
  x x    [*] IP: multicasting                                                   x x
  x x    [*] IP: syn cookies                                                    x x
  x x    [ ] IP: firewalling                                                    x x
  x x    [ ] IP: accounting                                                     x x
  x x    [ ] IP: optimize as router not host                                    x x
  x x    <*> IP: tunneling                                                      x x
  x x    [*] IP: multicast routing (EXPERIMENTAL)                               x x
  x x    < > IP: aliasing support                                               x x
  x x    --- (it is safe to leave these untouched)                              x x
  x

bye, 

Uwe
--
Uwe Jendricke                  <jendricke@iig.uni-freiburg.de>
Institut fuer Informatik und Gesellschaft, Abteilung Telematik
Universitaet Freiburg                Telefon: +49-761-203-4926
Friedrichstr. 50                         Fax: +49-761-203-4929
D-79098 Freiburg                       Sekr.: +49-761-203-4964
PGP-Fingerprint:     6AC7 9B7E 809A D4F5   2091 98D8 9044 D5DD



From rem-conf Wed Feb 03 02:55:50 1999 
From rem-conf-request@es.net Wed Feb 03 02:55:48 1999
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To: "Gio tesista" <valchesi70@hotmail.com>
cc: rem-conf@es.net
Subject: Re: multicast S.O.S!!! 
In-reply-to: Your message of "Wed, 03 Feb 1999 01:47:57 PST."
             <19990203094757.10217.qmail@hotmail.com> 
Date: Wed, 03 Feb 1999 19:53:32 +0900
From: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
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I'm not using Linux, but...

> I have Linux configured 3 PC( kernel  2.0.36) with multicast support and 
> one of these is a multicast router(mrouter-3.81).If we create a work 
> session on our same  network,when we run "SD" or "SDR" I see us the 
> session on other computers  and it's O.K..The problem is ONLY mrouter 
> see external session (by  tunneling) out my network,  but the other 2 
> computers no.What can I do  
> to resolve this problem?

Let's check step-by-step.

  1: does the mrouter machine have a kernel really configured
     as "multicast router"?

     I believe only "multicast support" kernel does not forward
     multicast packets. it simply is a multicast-clients.
     you must configure the kernel as multicast ROUTER.
     don't confuse this with "running mrouted".
     the kernel must function as MULTICAST ROUTER.

  2: mrouted's running well? no errors?
     what's the contents of mrouted.conf?
     tunneling is configured well?

     check the tunneling configuration of mrouted by mrinfo command.

  3: how mrouted's running;
     do "kill -USR1" and "kill -USR2" to examine the status of mrouted
     (and multicast routing table of the kernel).

  4: are there any multicast packets forwarded to your segment?
     or only SIP (SDP?) packets are blocked?
     multicast packets from outside have high enough TTL to be forwarded?

     use tcpdump command to see packets on your segment.

if you pass these all steps and still have any problem, then
post a message again with any more information
about your network configuration.

-- yozo.



From rem-conf Wed Feb 03 02:58:10 1999 
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From: "Angel L. Mateo" <angel.mateo@rediris.es>
To: "Gio tesista" <valchesi70@hotmail.com>
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Subject: Re: multicast S.O.S!!! 
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El d=EDa Wed, 03 Feb 1999 01:47:57 PST  "Gio tesista" escribi=F3:
> =

> I 'm italian student and I 'm making  some experiments on multicast. =

> I have Linux configured 3 PC( kernel  2.0.36) with multicast support an=
d =

> one of these is a multicast router(mrouter-3.81).If we create a work =

> session on our same  network,when we run "SD" or "SDR" I see us the =

> session on other computers  and it's O.K..The problem is ONLY mrouter =

> see external session (by  tunneling) out my network,  but the other 2 =

> computers no.What can I do  =

> to resolve this problem?
> =

	I replied you this message a few this ago, but my answer was not good fo=
r your situation.

	I think that I had this problem when my organization router was a SGI O2=
=2E In this situation, the only computer that could listen to multicast t=
rafic was this o2, but when this computer joint to a session, then all my=
 network could do it for the same session (I think that this is because a=
 SGI bug that make the o2 to ignore the IGMP join messages).

	Have you tried to join to session in the mrouter and then join to the sa=
me session on other computer?

	Have you checked that IGMP join messages reach the mrouted?

	Now, I have change the mrouted to a Linux box with kernel 2.0.36 and mro=
uted version 3.9beta3, and I have no problem.

	Why don't you try this version?


-----------------------------------------
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 Feb 03 12:44:18 1999 
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From: "Nick A. Fikouras" <nick@dcs.shef.ac.uk>
To: <rem-conf@es.net>
Subject: RTP and congestion control?
Date: Wed, 3 Feb 1999 20:36:48 -0000
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This is a multi-part message in MIME format.

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Hi,

I am a newbie with the RTP protocol so please excuse me if my question =
is not making sense.

It is presented in RFC 1889 that through the exchange of RTCP packets =
the sender can be made aware of jitter and packet loss. Both can be used =
to determine whether a network pipe is experiencing congestion. In that =
case are there any RTP considerations for congestion control that =
manipulation the session bandwidth or the payload size? If yes, where =
are they presented?

thanks in advance,

nick

------=_NextPart_000_00D8_01BE4FB4.EC329C80
Content-Type: text/html;
	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD W3 HTML//EN">
<HTML>
<HEAD>

<META content=3Dtext/html;charset=3Diso-8859-1 =
http-equiv=3DContent-Type>
<META content=3D'"MSHTML 4.72.3110.7"' name=3DGENERATOR>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT color=3D#000000 size=3D2>Hi,</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>I am a newbie with the RTP protocol =
so please=20
excuse me if my question is not making sense.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>It is presented in RFC 1889 that through the =
exchange of RTCP=20
packets the sender can be made aware of jitter and packet loss. Both can =
be used=20
to determine whether a network pipe is experiencing congestion. In that =
case are=20
there any RTP considerations for congestion control that manipulation =
the=20
session bandwidth or the payload size? If yes, where are they=20
presented?</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>thanks in advance,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>nick</FONT></DIV></BODY></HTML>

------=_NextPart_000_00D8_01BE4FB4.EC329C80--




From rem-conf Wed Feb 03 13:28:12 1999 
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Subject: Re: RTP and congestion control?
References: <00db01be4fb4$ec542e40$940ba78f@cds148.dcs.shef.ac.uk>
From: Jamshid Mahdavi <mahdavi@novell.com>
Date: 03 Feb 1999 13:23:38 -0800
In-Reply-To: "Nick A. Fikouras"'s message of "Wed, 3 Feb 1999 20:36:48 -0000"
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"Nick A. Fikouras" <nick@dcs.shef.ac.uk> writes:

> Hi,
> 
> I am a newbie with the RTP protocol so please excuse me if my question =
> is not making sense.
> 
> It is presented in RFC 1889 that through the exchange of RTCP packets =
> the sender can be made aware of jitter and packet loss. Both can be used =
> to determine whether a network pipe is experiencing congestion. In that =
> case are there any RTP considerations for congestion control that =
> manipulation the session bandwidth or the payload size? If yes, where =
> are they presented?
> 
> thanks in advance,
> 
> nick

You can find a number of relevant references at
	http://www.psc.edu/networking/tcp_friendly.html

These papers present a general scheme which attempts to mimic the
overall performance of TCP's congestion avoidance scheme.  Acceptance
for these ideas seems to be growing, and soon some standards will be
starting to appear which use these techniques.

--Jamshid



From rem-conf Wed Feb 03 19:31:33 1999 
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From: "Oh yeonhee" <yeonche@secsm.org>
To: <rem-conf@es.net>
Subject: about sdr..
Date: Wed, 4 Feb 1998 12:19:35 -0000
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hello
I'm Korean student.
I study about mbone, and test with sgi-IRIX 5.3 these days.
I met a problem.
I installed 'mrouted 3.8' successfully. 

~~~~~~~

comp23 31# mtrace 147.47.1.8
Mtrace from 147.47.1.8 to 147.46.113.54 via group 224.2.0.1
Querying full reverse path... 
  0  comp23.snu.ac.kr (147.46.113.54)
 -1  comp23.snu.ac.kr (147.46.113.54)  DVMRP  thresh^ 1  
 -2  nms.kren.ne.kr (147.47.1.8)  DVMRP  thresh^ 32  
 -3  nms.kren.ne.kr (147.47.1.8)
Round trip time 11 ms

Waiting to accumulate statistics... Results after 10 seconds:

  Source        Response Dest    Packet Statistics For     Only For Traffic
147.47.1.8      147.46.113.54    All Multicast Traffic     From 147.47.1.8
     v       __/  rtt    8 ms    Lost/Sent = Pct  Rate       To 224.2.0.1
147.47.1.8      nms.kren.ne.kr 
     v     ^      ttl   32        0/0    = --%   0 pps     0/0    = --%   0
pps
147.46.113.54   comp23.snu.ac.kr 
     v      \__   ttl   33   
147.46.113.54   147.46.113.54
  Receiver      Query Source


~~~~~~~
 
but, when I executed "sdr", there was no session.
I think there is some sessions at least... I think there is some problems.
but, I have no idea about that. give me some solutions. 
I have to configure other things?





From rem-conf Wed Feb 03 20:57:31 1999 
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From: Keiko Tanigawa <takahara@sdl.hitachi.co.jp>
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Hello,

At last IETF meeting in Orland , I heard that Cisco presented the RTP
multiplexing method for VoIP router,
but I cannot  find any drafts on the web.
Does anybody have any informations or know the pointer to the document?

Best regards,
Keiko.


--
Keiko Tanigawa [ takahara@sdl.hitachi.co.jp ]
4th Research Dept.
Systems Development Laboratory, Hitachi, Ltd.
292 Yoshida-cho, Totsuka-ku, Yokohama 244-0817 Japan
TEL:+81-45-881-1241(ext.4844)
Fax:+81-45-860-1675





From rem-conf Wed Feb 03 21:16:02 1999 
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I don't recall a Cisco proposal, but the current ones are:

http://www.ietf.org/internet-drafts/draft-ietf-avt-aggregation-00.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-mux-rtp-00.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-muxissues-00.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-germ-00.txt
http://search.ietf.org/internet-drafts/draft-tanigawa-rtp-multiplex-01.txt

-Jonathan R.

Keiko Tanigawa wrote:
> 
> Hello,
> 
> At last IETF meeting in Orland , I heard that Cisco presented the RTP
> multiplexing method for VoIP router,
> but I cannot  find any drafts on the web.
> Does anybody have any informations or know the pointer to the document?
> 
> Best regards,
> Keiko.
> 
> --
> Keiko Tanigawa [ takahara@sdl.hitachi.co.jp ]
> 4th Research Dept.
> Systems Development Laboratory, Hitachi, Ltd.
> 292 Yoshida-cho, Totsuka-ku, Yokohama 244-0817 Japan
> TEL:+81-45-881-1241(ext.4844)
> Fax:+81-45-860-1675

-- 
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 Feb 04 02:07:08 1999 
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To: "Nick A. Fikouras" <nick@dcs.shef.ac.uk>
cc: rem-conf@es.net
Subject: Re: RTP and congestion control? 
In-reply-to: Your message of "Wed, 03 Feb 1999 20:36:48 GMT."
             <00db01be4fb4$ec542e40$940ba78f@cds148.dcs.shef.ac.uk> 
Date: Thu, 04 Feb 1999 11:00:55 +0100
From: Dorgham Sisalem <sisalem@fokus.gmd.de>
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"Nick A. Fikouras" <nick@dcs.shef.ac.uk> writes:

> Hi,
> 
> I am a newbie with the RTP protocol so please excuse me if my question =
> is not making sense.
> 
> It is presented in RFC 1889 that through the exchange of RTCP packets =
> the sender can be made aware of jitter and packet loss. Both can be used =
> to determine whether a network pipe is experiencing congestion. In that =
> case are there any RTP considerations for congestion control that =
> manipulation the session bandwidth or the payload size? If yes, where =
> are they presented?
> 
> thanks in advance,
> 
> nick
there is also another paper by Henning Schulzrinne in Nossdav'98 not included
in the list but uses the RTCP messages for adapting the sender behavior in 
a TCP-friendly manner

"The Loss-Delay Based Adjustment Algorithm: A TCP-Friendly Adaptation Scheme"
ftp://ftp.fokus.gmd.de/pub/step/papers/Sisa9807:Loss.ps.gz

Regards
	Dorgham



From rem-conf Thu Feb 04 04:07:07 1999 
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
cc: Keiko Tanigawa <takahara@sdl.hitachi.co.jp>, rem-conf@es.net
Subject: Re: Question about Cisco's RTP (multiplexing) proposal
In-reply-to: Your message of "Thu, 04 Feb 1999 00:07:51 EST." <36B92B27.A9D94781@dnrc.bell-labs.com>
Date: Thu, 04 Feb 1999 11:59:58 +0100
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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There wasn't an internet-draft, but Dean Willis presented some ideas
discussed during the meeting (late one evening in the terminal room).
Essentially it was a brute-force UDP multiplexing scheme. I don't have
an electronic copy of the slides, unfortunately.

Colin

--> Jonathan Rosenberg writes:
>I don't recall a Cisco proposal, but the current ones are:
>
>http://www.ietf.org/internet-drafts/draft-ietf-avt-aggregation-00.txt
>http://www.ietf.org/internet-drafts/draft-ietf-avt-mux-rtp-00.txt
>http://www.ietf.org/internet-drafts/draft-ietf-avt-muxissues-00.txt
>http://www.ietf.org/internet-drafts/draft-ietf-avt-germ-00.txt
>http://search.ietf.org/internet-drafts/draft-tanigawa-rtp-multiplex-01.txt
>
>-Jonathan R.
>
>Keiko Tanigawa wrote:
>> 
>> Hello,
>> 
>> At last IETF meeting in Orland , I heard that Cisco presented the RTP
>> multiplexing method for VoIP router,
>> but I cannot  find any drafts on the web.
>> Does anybody have any informations or know the pointer to the document?
>> 
>> Best regards,
>> Keiko.
>> 
>> --
>> Keiko Tanigawa [ takahara@sdl.hitachi.co.jp ]
>> 4th Research Dept.
>> Systems Development Laboratory, Hitachi, Ltd.
>> 292 Yoshida-cho, Totsuka-ku, Yokohama 244-0817 Japan
>> TEL:+81-45-881-1241(ext.4844)
>> Fax:+81-45-860-1675
>
>-- 
>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 Feb 04 05:08:48 1999 
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Date: Thu, 04 Feb 1999 14:01:52 +0100
From: Schahram Dustdar <dustdar@ufg.ac.at>
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Where can I find a vat or rat version for Irix 6.5.
(vat for Irix 5.3 says wrong architecture).

Any pointers?

Thanks
schahram



From rem-conf Thu Feb 04 09:02:01 1999 
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From: "Gio tesista" <valchesi70@hotmail.com>
To: yozo@aohakobe.ipc.chiba-u.ac.jp
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Subject: Re: MULTICAST
Date: Thu, 04 Feb 1999 08:47:42 PST
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Sorry for my little clearness.
I try again.
The mrouter machine have a kernel configured as multicast router.
In mrouted.config I see:
  tunnel our_mrouter-host  end-tunnel-host  metric 1 thresold 24
I can't understand 3)-4):Kill-USR1 and Kill-USR2.If from mrouter I run 
ps -ax,I don't see these.Is there other method to esamine the status of 
mrouted and multicast routing table of the kernel?
By tcpdump,I can't see SIP,SDP;what?Can you help me to see if any 
pachets are blocked and if ttl value is high enough whit more details? 
I don't know if that I will write can be useful for you.  
 Now,I'm on a machine,"ing58",and it's not mrouter.
If I run ping 224.x.x.x,our mrouter answers as it's right.
I configured "ing89" as mrouter.
If from ing58(my not mrouter machine)run  mtrace ing89,I see

  Mtrace from  ing.89 to ing.58 via group 224.2.0.1
  Source & receiver are directly connected, no path to trace
 
Our mrouter(ing89)is connected by tunneling whit "ciscone",and "ciscone" 
whit "astra". 
If from my computer, I run  mtrace astra,I see

Mtrace from 192.135.23.2 to ing.58 via group 224.2.0.1
Querying full reverse path...
  0  ing58(my machine no mroute)
 -1  ing89(mrouter)             thresh^ 1
 -2  ciscone         Unknown protocol code 5  thresh^ 24
 -3  ? (192.135.23.2)  DVMRP  thresh^ 24
 -4  ? (192.135.23.2)
Round trip time 131 ms
Waiting to accumulate statistics... Results after 10 seconds:

Source      Respon Dest    Packet Statistics For     Only For Traffic
192.135.23.2    ing.58(mine)     All MultTraffic      From192.135.23.2 
     v       __/  rtt   79 ms    Lost/Sent=PctRate    To 224.2.0.1
192.135.23.2    ?
     v     ^      ttl   24      617/1271=49% 127pps    0/0= --% 0 pps
193.204.120.1
193.204.120.113 ciscone
     v     ^      ttl   25       -1/55= -1%  5 pps     0/0= --% 0 pps
     ing.89     ing89.unife.it
     v      \__   ttl   26          0        0 pps       0      0 pps
ing.58          ing.58(my host)
  Receiver      Query Source

Finally,if from my machine(ing58) do ifconfig,I see also the line

etho       .....................
           UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
I hope this can help you to help me!
Thanks
Giovanni

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



From rem-conf Thu Feb 04 09:36:56 1999 
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Date: Thu, 04 Feb 1999 09:24:03 -0800
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From: Florissa Colina <florissac@bmrc.berkeley.edu>
Subject: 2/10 Berkeley Multimedia, Interfaces and Graphics Seminar 
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The Berkeley Multimedia, Interfaces and Graphics Seminar scheduled for
Wednesday, February 3, 1999 has been rescheduled to Wednesday, February 10,
1999.

Berkeley Internet Broadcasting System 

(Wednesday February 10, 1999 1:00-2:30pm 405 Soda Hall) 

Lawrence A. Rowe 
EECS Department 
U.C. Berkeley 

For the past 4 years we have broadcast the Berkeley Multimedia, Graphics,
and Interfaces Seminar, a regularly scheduled seminar, world-wide on the
Internet using the Internet MBone tools. We began broadcasting live video
sources (e.g., broadcast television and cable networks) and other graduate
and undergraduate classes this past fall. These interactive programs are
archived for on-demand replay. The system is called the Berkeley Internet
Broadcasting System (BIBS). 

This talk will discuss the development and deployment of the BIBS system
including: 

     Broadcast infrastructure - the hardware and software used to produce
programs. 
     Studio classrooms - the analog audio/video and control systems used in
a classroom. 
     Software infrastructure - the software developed to produce programs
reliably and efficiently (e.g., broadcast manager, video effects,
     archive services, transcoding gateways, etc.) 

Several research problems suitable for masters and doctoral research will
be described. 

Our future plans include broadcasting more programs using various
technologies (e.g., Real Networks), improving the interaction with remote
participants. 

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 Thu Feb 04 18:04:22 1999 
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From: "Dean Willis" <Dean.Willis@MCI.COM>
To: "Colin Perkins" <C.Perkins@cs.ucl.ac.uk>,
        "Jonathan Rosenberg" <jdrosen@dnrc.bell-labs.com>
Cc: "Keiko Tanigawa" <takahara@sdl.hitachi.co.jp>, <rem-conf@es.net>
Subject: RE: Question about Cisco's RTP (multiplexing) proposal
Date: Thu, 4 Feb 1999 19:50:34 -0600
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Electronic copies of the slides we presented in Orlando can be retrieved
from:

http://www.greycouncil.com/ietf/trunkmuxing.pdf

Stephen Sprunk and I have been debating on whether this was worh writing
up an ID on. Current thoughts are probably not -- there are bigger fish
waiting to be fried.

I also suspect we may have reinvented a near-cousin of rfc1692 -- the
TMUX protocol.

Anybody else looked at RFC1692 lately?

--
Dean Willis

> -----Original Message-----
> From: Colin Perkins [mailto:C.Perkins@cs.ucl.ac.uk]
> Sent: Thursday, February 04, 1999 5:00 AM
> To: Jonathan Rosenberg
> Cc: Keiko Tanigawa; rem-conf@es.net
> Subject: Re: Question about Cisco's RTP (multiplexing) proposal
>
>
> There wasn't an internet-draft, but Dean Willis presented some ideas
> discussed during the meeting (late one evening in the terminal room).
> Essentially it was a brute-force UDP multiplexing scheme. I don't have
> an electronic copy of the slides, unfortunately.
>
> Colin
>
> --> Jonathan Rosenberg writes:
> >I don't recall a Cisco proposal, but the current ones are:
> >
> >http://www.ietf.org/internet-drafts/draft-ietf-avt-aggregation-00.txt
> >http://www.ietf.org/internet-drafts/draft-ietf-avt-mux-rtp-00.txt
> >http://www.ietf.org/internet-drafts/draft-ietf-avt-muxissues-00.txt
> >http://www.ietf.org/internet-drafts/draft-ietf-avt-germ-00.txt
> >http://search.ietf.org/internet-drafts/draft-tanigawa-rtp-mul
tiplex-01.txt
>
>-Jonathan R.
>
>Keiko Tanigawa wrote:
>>
>> Hello,
>>
>> At last IETF meeting in Orland , I heard that Cisco presented the RTP
>> multiplexing method for VoIP router,
>> but I cannot  find any drafts on the web.
>> Does anybody have any informations or know the pointer to the
document?
>>
>> Best regards,
>> Keiko.
>>
>> --
>> Keiko Tanigawa [ takahara@sdl.hitachi.co.jp ]
>> 4th Research Dept.
>> Systems Development Laboratory, Hitachi, Ltd.
>> 292 Yoshida-cho, Totsuka-ku, Yokohama 244-0817 Japan
>> TEL:+81-45-881-1241(ext.4844)
>> Fax:+81-45-860-1675
>
>--
>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 Feb 04 23:48:40 1999 
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Dean Willis wrote:
> 
> I also suspect we may have reinvented a near-cousin of rfc1692 -- the
> TMUX protocol.
> 
> Anybody else looked at RFC1692 lately?

I just gave it a quick read. It looks like it basically does what you
were proposing, except one layer higher, since you can mux UDP and TCP
packets together. 

-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 Feb 05 04:48:44 1999 
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Subject: RTP Payload types for miniature animatronic figures.
Date: Fri, 05 Feb 1999 12:40:31 +0100
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From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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there are now several protocols for  communicating and controlling
voice communication to/from automata
viz
http://www.homestead.com/hackfurby/
http://www.geekchic.com/~jpd/barney

we should be getting rtp payload types for these and adding appropriate
message types for them to the confbus work in mmusic too

surely? our children's future depends on this...

 cheers

   jon
sorry for the cross posting...



From rem-conf Fri Feb 05 05:28:49 1999 
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(Sorry for any cross-postings)

CALL FOR PAPERS 
Third IASTED/ISMM Conference on
Internet and Multimedia Systems and Applications (IMSA)
October 18-21, 1999 - Nassau, Grand Bahamas

http://www.iasted.com/conferences/1999/bahamas/imsa/c298.html


Schahram
--
Dr.Schahram Dustdar 
University of Art and Industrial Design
Center for Informatics Services (ZID)
Head of Department
Hauptplatz 8, A-4010 Linz-Austria/Europe
eMail:  dustdar@khsa.khs-linz.ac.at
Home:   http://khs-x6.zid.ufg.ac.at/~dustdar
WWW:    http://www.khs-linz.ac.at
Tel:    XX43-732-7898-260
Fax:    XX43-732-783508
--------------------------------------------
Austrian Multimedia National Support Centre 
for multimedia conferencing support



From rem-conf Fri Feb 05 11:28:51 1999 
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MAKE $1000 - $3000+  PER WEEK WORKING FROM HOME!!!

Want to get out of the rat race?
Do  you like to travel?

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From rem-conf Sun Feb 07 06:31:05 1999 
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Date: Sun, 7 Feb 1999 09:09:30 EST
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Subject: UK Videoconferencing Demo Numbers
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Dear Sir/Madam

The following auto-answer ISDN videoconferencing numbers are available
24hrs/day:

0906 55 25 501
0906 55 96 001
0906 55 96 026
0906 55 96 051

full details and lots of useful UK videoconferencing links are at:

http://members.aol.com/videoduck1/

Hope this is useful to you.



From rem-conf Sun Feb 07 12:10:10 1999 
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Date: Sun, 07 Feb 1999 11:56:24 -0800
From: Kamil Sarac <ksarac@cs.ucsb.edu>
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Dear Sir/Madam,

This mail is in regard of sdr-monitor effort at UCSB:

The original script that we sent out to participants had a recursive
call
which never returned back causing the sdr image to grow. We have written
a
second version of this script  which does not have such effects and will
not cause
the sdr image to grow.

This new script also enables you to not to send administratively scoped
sessions or
any encrypted sessions if you want. To do that you will need to modify
one
or two variable flags at the beginning of the script by following the
comments at the beginning of the code. But if you leave as it is, it
will
send administratively scoped sessions and will not send the "k " field
of
sdr entry (which contains information related to encryption ) of
encrypted
sessions.

Right now we are testing the new version and will be sending it out to
our
participants soon...

Sincerely,

Kamil Sarac.






From rem-conf Mon Feb 08 20:15:51 1999 
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From: "y.y. chang" <yuehyingchang@hotmail.com>
To: rem-conf@es.net
Subject: FreeBSD or Linux
Date: Mon, 08 Feb 1999 19:54:32 PST
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Dear all:

I am bringing up a conferencing system
using some Unix running on PCs.
As I have limited budget, I wonder which of the
low cost Unix's is better:  FreeBSD or Linux ?
I am in particular interested in the multicast
support.

Any pointers will be appreciated.

/yy

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



From rem-conf Tue Feb 09 02:19:34 1999 
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cc: rem-conf@es.net
Subject: Re: FreeBSD or Linux
In-Reply-To: <19990209035432.25270.qmail@hotmail.com>
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> I am bringing up a conferencing system
> using some Unix running on PCs.
> As I have limited budget, I wonder which of the
> low cost Unix's is better:  FreeBSD or Linux ?
> I am in particular interested in the multicast
> support.

I could't say which system is the better one.
Best way: try both, use both. Why ether FreeBSD or Linux?
You need about 1GB per system. With a bootmanager or from dos you can boot 
what you wont.

Some of my experience with FreeBSD and Linux (using for MBone conferencing):

FreeBSD:
+ I have less work for installation.
  (for base system, audio and video support I need 2 hours)
  http://www-mm.urz.tu-dresden.de/freebsd/ (sorry, only in german)
+ runs better with little RAM (32MB) than Linux
+ has many MBone-Tools as package (easy installation)

Linux:
+ is much more known and accepted than FreeBSD in Germany (don't know why)
+ V4L (video for linux)
+ the ALSA sound driver supports all mixer switches (analog monitor switch)
- has many different distibutions (I used debian and red hat)
- I have different stability problems on different versions 
  (audio or video hangs my system sometimes - should be solvable)
- there are some mrouted versions that would't prune
- I run in more pitfalls than on FreeBSD.

Rgds,
Christoph Fleck

,------------------------------------------------------------------.
| Referenzzentrum fuer multimediale Teledienste (MMRZ), TU Dresden |
|         Dr. Klaus Koehler, Christoph Fleck                       |
| e-mail: mmt-ref@tu-dresden.de             Tel.: 0351 / 463 5653  |
|    WWW: http://www-mm.urz.tu-dresden.de               (GERMANY)  |
`------------------------------------------------------------------'




From rem-conf Tue Feb 09 02:31:22 1999 
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Date: Tue, 9 Feb 1999 12:20:26 -0800
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Dear Responsible person,
=20
We  produce some surgical equipment, such as surgical operation lamps. =
But the reflectors that we use for our operation lamps are, not as good =
as we want to. If you have some reflectors that we can use for our =
operation lamps, please inform me, as soon as possible.If you wanna fax =
me;
=20
Fax:00(90) 212 635 12 10
=20
=20
Thanks for your interest,
=20
Tuba ALAYLI
Product Manager
=20
talayli@bicakcilar.com

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<DIV><FONT size=3D2>Dear Responsible person,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>We&nbsp; produce some surgical equipment, such as =
surgical=20
operation lamps. But the reflectors that we use for our operation lamps =
are, not=20
as good as we want to. If you have some reflectors that we can use for =
our=20
operation lamps, please inform me, as soon as possible.If you wanna fax=20
me;</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Fax:00(90) 212 635 12 10</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Thanks for your interest,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Tuba ALAYLI</FONT></DIV>
<DIV><FONT size=3D2>Product Manager</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><A=20
href=3D"mailto:talayli@bicakcilar.com">talayli@bicakcilar.com</A></FONT><=
/DIV></DIV></DIV></BODY></HTML>

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From rem-conf Tue Feb 09 09:45:29 1999 
From rem-conf-request@es.net Tue Feb 09 09:45:28 1999
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Date: Tue, 09 Feb 1999 09:38:33 -0800
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From: Florissa Colina <florissac@bmrc.berkeley.edu>
Subject: 2/10 Berkeley Multimedia, Interfaces and Graphics Seminar 
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The Berkeley Multimedia, Interfaces and Graphics Seminar scheduled for
Wednesday, February 3, 1999 has been rescheduled to Wednesday, February 10,
1999.

Berkeley Internet Broadcasting System 

(Wednesday February 10, 1999 1:00-2:30pm 405 Soda Hall) 

Lawrence A. Rowe 
EECS Department 
U.C. Berkeley 

For the past 4 years we have broadcast the Berkeley Multimedia, Graphics,
and Interfaces Seminar, a regularly scheduled seminar, world-wide on the
Internet using the Internet MBone tools. We began broadcasting live video
sources (e.g., broadcast television and cable networks) and other graduate
and undergraduate classes this past fall. These interactive programs are
archived for on-demand replay. The system is called the Berkeley Internet
Broadcasting System (BIBS). 

This talk will discuss the development and deployment of the BIBS system
including: 

     Broadcast infrastructure - the hardware and software used to produce
programs. 
     Studio classrooms - the analog audio/video and control systems used in
a classroom. 
     Software infrastructure - the software developed to produce programs
reliably and efficiently (e.g., broadcast manager, video effects,
     archive services, transcoding gateways, etc.) 

Several research problems suitable for masters and doctoral research will
be described. 

Our future plans include broadcasting more programs using various
technologies (e.g., Real Networks), improving the interaction with remote
participants. 

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 Feb 09 11:10:26 1999 
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Message-ID: <36C08605.3F4423D@BMRC.Berkeley.EDU>
Date: Tue, 09 Feb 1999 11:01:25 -0800
From: "Lawrence A. Rowe" <Rowe@bmrc.berkeley.edu>
Reply-To: Rowe@bmrc.berkeley.edu
Organization: U.C. Berkeley
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To: jgemmell@microsoft.com, ucb-mbone@cs.berkeley.edu,
        lindahl@ack.Berkeley.EDU, casner@cisco.com, berc@pa.dec.com,
        srodawa@oakland.edu, johnb@pal.xerox.com, mbone@ISI.EDU,
        sumisu@nttlabs.com, rem-conf@es.net, kevin@cc.gatech.edu,
        aswan@cs.berkeley.edu, mccanne@CS.Berkeley.EDU, delaroca@ucla.edu,
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Subject: Berkeley Seminar Wed 2/10 1-2.30 PDT
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Hi -

I want to call your attention to the seminar I am giving tomorrow at
Berkeley MIG Seminar.  I will be describing the studio classroom and
broadcast configuration we use to produce this seminar which has been
going on for over 4 years.  Berkeley is not connected to the
vBNS/Internet2 so multicast ought to work better than it has the past 18
months.

During the seminar I will also describe software we have developed and
are developing to improve the quality and reduce the effort required to
produce a broadcast. We are going to try a number of things that are
likely to be "high-risk" but should point the direction for future
interactive broadcasting using the Internet MBone.  A highlight will be
the first live demonstration of video effects applied to RTP streams
using Ketan Meyer-Patel's parallel video effects system.

For more info on the seminar see
  http://bmrc.berkeley.edu/mig/w4
I have attached an sdp file for the seminar so you can just launch it.

Enjoy!
	Larry
-- 
Professor Lawrence A. Rowe          Internet:  Rowe@BMRC.Berkeley.EDU
Computer Science Division - EECS       Phone: 510-642-5117
University of California, Berkeley       Fax: 510-642-5615
Berkeley, CA 94720-1776	           URL: http://bmrc.berkeley.edu/~larry
--------------EEF89BB5008C45BEF84FD13A
Content-Type: application/x-sd;
 name="mig.sdp"
Content-Transfer-Encoding: base64
Content-Disposition: inline;
 filename="mig.sdp"

dj0wDQpvPXBiaHVhbmcgMSAxIElOIElQNCAxMjguMzIuMTMxLjE2OQ0Kcz1VQ0JlcmtlbGV5
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MzENCmM9SU4gSVA0IDIyNC4yLjE2My43LzEyNw==
--------------EEF89BB5008C45BEF84FD13A--




From rem-conf Tue Feb 09 13:20:52 1999 
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From: Shujie <sdong@Oakland.edu>
To: "Lawrence A. Rowe" <Rowe@bmrc.berkeley.edu>
cc: rem-conf@es.net
Subject: Re: Berkeley Seminar Wed 2/10 1-2.30 PDT
In-Reply-To: <36C08605.3F4423D@BMRC.Berkeley.EDU>
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Dear Sir,

We are very interested in this seminar. Could you please tell me what
kind of tools I need to view it and record it? Thank you.

Sincerely,
Shujie

On Tue, 9 Feb 1999, Lawrence A. Rowe wrote:

> Hi -
> 
> I want to call your attention to the seminar I am giving tomorrow at
> Berkeley MIG Seminar.  I will be describing the studio classroom and
> broadcast configuration we use to produce this seminar which has been
> going on for over 4 years.  Berkeley is not connected to the
> vBNS/Internet2 so multicast ought to work better than it has the past 18
> months.
> 
> During the seminar I will also describe software we have developed and
> are developing to improve the quality and reduce the effort required to
> produce a broadcast. We are going to try a number of things that are
> likely to be "high-risk" but should point the direction for future
> interactive broadcasting using the Internet MBone.  A highlight will be
> the first live demonstration of video effects applied to RTP streams
> using Ketan Meyer-Patel's parallel video effects system.
> 
> For more info on the seminar see
>   http://bmrc.berkeley.edu/mig/w4
> I have attached an sdp file for the seminar so you can just launch it.
> 
> Enjoy!
> 	Larry
> -- 
> Professor Lawrence A. Rowe          Internet:  Rowe@BMRC.Berkeley.EDU
> Computer Science Division - EECS       Phone: 510-642-5117
> University of California, Berkeley       Fax: 510-642-5615
> Berkeley, CA 94720-1776	           URL: http://bmrc.berkeley.edu/~larry




From rem-conf Wed Feb 10 03:18:50 1999 
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Subject: Translate Your Web-Site into: Spanish, German, French, Italian, and Portuguese.
Date: Wed, 10 Feb 1999 01:25:13
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From rem-conf Thu Feb 11 15:55:19 1999 
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From: "Edward W. Knightly" <knightly@ece.rice.edu>
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Date: Thu, 11 Feb 1999 17:35:44 -0600
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Call For Papers - Special Issue of IEEE Network Magazine
on Integrated and Differentiated Services for the Internet

Guest Editors
=============

Hui Zhang                              Edward Knightly                        
School of Computer Science  	       ECE Department
Carnegie Mellon University  	       Rice University                       
5000 Forbes Ave             	       6100 Main Street                      
Pittsburgh, PA 15213-3891   	       Houston, TX 77005-1892                
				                                             
Phone:  (412) 268-8945                 Phone:  (713) 737-5748          
Fax:    (412) 268-5574                 Fax:    (713) 524-5237          
Email:  hzhang@cs.cmu.edu              Email:  knightly@ece.rice.edu   

Several important trends have emerged following the Internet's
success.  First, the Internet has evolved into a global and commercial
communication infrastructure. Second, Internet technology has become
the technical basis for not only the Internet, but also for most data
communication networks, both public and private. Finally, new and
diverse applications are emerging, including distributed multimedia
applications, making the inadequacies of a one-size-fits-all service
increasingly apparent.


These new developments challenge the original design goals of TCP/IP's
best effort service model and end-system-only traffic management, and
indicate potential opportunities for alternate service models and
traffic management schemes.  Consequently, new service models,
protocols, and resource management architectures have been proposed to
cope with emerging applications, and new business and administrative
requirements.  However, a number of questions must be addressed before
enhanced network services can be deployed on the Internet: What are
the appropriate service models?  What are the architectures and
algorithms required for multiple services to co-exist?  What are the
fundamental tradeoffs in scalability and granularity of quality of
service control?  This special issue of IEEE Network seeks to shed
light on the state-of-the-art and the future of integrated services
and differentiated services on the Internet.

Topics of interest include:

-Testbeds and trials for intserv and diffserv 
-QoS and scalability
-QoS in heterogeneous networks
-Multi-class scheduling architectures and algorithms 
-Internet admission control
-Per-hop-behavior definitions and approaches
-End-to-end QoS and service differentiation
-QoS routing
-QoS and aggregate traffic flows
-Aggregate vs. per-flow QoS
-Service measurement and monitoring 
-Pricing models
-Future of QoS on the Internet

Authors are requested to send an email to the guest editor, Ed Knightly, 
(knightly@ece.rice.edu), giving a URL where a postscript or PDF file can be 
downloaded by the guest editors.

Important Dates
---------------
  Paper Submission Deadline: March 1
  Feedback to Authors: April 1
  Final Manuscripts to Publisher: July 1
  Publication of Special Issue: September 1999

http://www.comsoc.org/socstr/techcom/ntwrk/special/knightly_zhang.html



From rem-conf Sat Feb 13 05:17:12 1999 
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To: "rem-conf@es.net" <rem-conf@es.net>
Subject: Re: vat IRIX 6.5 
In-reply-to: Your message of "Thu, 04 Feb 1999 14:01:52 +0100."
             <36B99A40.3C92564E@ufg.ac.at> 
Date: Sat, 13 Feb 1999 22:02:09 +0900
From: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
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> Where can I find a vat or rat version for Irix 6.5.
> (vat for Irix 5.3 says wrong architecture).
> 
> Any pointers?

how about rat-3.0.31 on Irix6.2?
see <http://www-mice.cs.ucl.ac.uk/multimedia/software/>.

-- yozo.




From rem-conf Sat Feb 13 05:27:03 1999 
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To: "Oh yeonhee" <yeonche@secsm.org>
cc: rem-conf@es.net
Subject: Re: about sdr.. 
In-reply-to: Your message of "Wed, 04 Feb 1998 12:19:35 GMT."
             <199902040304.MAA03801@ns.secsm.org> 
Date: Sat, 13 Feb 1999 22:17:42 +0900
From: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
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> I study about mbone, and test with sgi-IRIX 5.3 these days.
> I met a problem.
> I installed 'mrouted 3.8' successfully. 

> comp23 31# mtrace 147.47.1.8
> Mtrace from 147.47.1.8 to 147.46.113.54 via group 224.2.0.1
> Querying full reverse path... 
>   0  comp23.snu.ac.kr (147.46.113.54)
>  -1  comp23.snu.ac.kr (147.46.113.54)  DVMRP  thresh^ 1  
>  -2  nms.kren.ne.kr (147.47.1.8)  DVMRP  thresh^ 32  
>  -3  nms.kren.ne.kr (147.47.1.8)
> Round trip time 11 ms

  [snip]

> but, when I executed "sdr", there was no session.
> I think there is some sessions at least... I think there is some problems.
> but, I have no idea about that. give me some solutions. 
> I have to configure other things?

are you on comp23.snu.ac.kr?
I did mrinfo to several hosts but cannot see
how your site are connected to MBone. 

can you see any sap packets on your machine?
try "tcpdump host sap.mcast.net."

I bet you can see any session info which you create.

-- yozo.




From rem-conf Sat Feb 13 06:36:41 1999 
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To: "Gio tesista" <valchesi70@hotmail.com>
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Subject: Re: MULTICAST 
In-reply-to: Your message of "Thu, 04 Feb 1999 08:47:42 PST."
             <19990204164755.20213.qmail@hotmail.com> 
Date: Sat, 13 Feb 1999 23:25:08 +0900
From: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
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Gio, you still have any problem?

> I can't understand 3)-4):Kill-USR1 and Kill-USR2.If from mrouter I run 

do "man mrouted" and read it.
you can use the command "kill" to send signals to any process.
when mrouted process receive the signal USR1 or USR2,
it dumps the information.
you can examine those files and check how mrouted's working.

netstat command may show multicast-related kernel status.

> Our mrouter(ing89)is connected by tunneling whit "ciscone",and "ciscone" 

I try mtrace to your host...

  aohakobe:yozo% mtrace -s ing58.unife.it. aohakobe sap.mcast.net.
  Mtrace from 192.167.215.58 to 133.82.241.137 via group 224.2.127.254
  Querying full reverse path... * switching to hop-by-hop:
    0  aohakobe.ipc.chiba-u.ac.jp (133.82.241.137)
   -1  suzuran.ipc.chiba-u.ac.jp (133.82.241.67)  DVMRP  thresh^ 1  
   -2  ayame.cix.chiba-u.ac.jp (133.82.254.224)  DVMRP  thresh^ 16  
   -3  nic.ric-tsukuba.ad.jp (192.70.179.247)  DVMRP  thresh^ 32  
   -4  * mbone.imnet.ad.jp (202.241.2.71)  DVMRP  thresh^ 32  
   -5  mbone.otemachi.wide.ad.jp (202.249.3.129)  DVMRP  thresh^ 32  
   -6  sw01.TokyoNet.AD.JP (203.183.255.106)  DVMRP  thresh^ 32  
   -7  * dec3800-2-fddi-0.SanFrancisco.cw.net (204.70.158.61)  DVMRP  thresh^ 64  
   -8  dec3800-1-fddi-1.Dallas.cw.net (204.70.114.45)  DVMRP  thresh^ 1  
   -9  dec3800-1-fddi-0.Washington.cw.net (204.70.2.13)  DVMRP  thresh^ 1  
  -10  de-ws.ten-34.net (193.203.254.34)  DVMRP  thresh^ 48  
  -11  atm.ws1.de.ten-155.net (212.1.193.233)  DVMRP  thresh^ 48  
  -12  * astra.infn.it (192.135.23.2)  DVMRP  thresh^ 48  
  -13  * * ciscone.cineca.it (193.204.120.1)  PIM/Special  thresh^ 24  
  -14  * * ing89.unife.it (192.167.215.89)  DVMRP  thresh^ 24  
  -15  ing58.unife.it (192.167.215.58)
  Round trip time 1202 ms; total ttl of 72 required.
  
  aohakobe:yozo% 

the mbone connection seems ok...

-- yozo.




From rem-conf Sat Feb 13 07:32:27 1999 
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Subject: humone growth ADVERTISMENT AD
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From rem-conf Sat Feb 13 07:33:58 1999 
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To: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: Re: vat IRIX 6.5
In-reply-to: Your message of "Sat, 13 Feb 1999 22:02:09 +0900." <199902131302.WAA14695@aohakobe.ipc.chiba-u.ac.jp>
Date: Sat, 13 Feb 1999 15:28:06 +0100
Message-ID: <712.918919686@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Yozo Toda writes:
>
>> Where can I find a vat or rat version for Irix 6.5.
>> (vat for Irix 5.3 says wrong architecture).
>> 
>> Any pointers?
>
>how about rat-3.0.31 on Irix6.2?
>see <http://www-mice.cs.ucl.ac.uk/multimedia/software/>.

For some reason, that binary doesn't work on irix 6.5, if anyone manages to
figure out why, I'd be interested to hear it (we don't have any 6.5 machines
to test on).

Colin



From rem-conf Sat Feb 13 07:52:10 1999 
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From: Oh yeonhee <yeonche@secsm.org>
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To: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>, rem-conf@es.net
Subject: Re: about sdr..
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Thanks for your reply.

Because there was no reply, it was very hard for me to solve those problems.
At last, I found out that tunneling between Korea and other country was broken.
(Naturally, there was no output for "tcpdump host sap.mcast.net".)
Furthermore there is little multicast conference in Korea. so,  it is hard for
me to see the session in sdr.
Even though it is poor situation, I want to succeed this experiment.
At last, I made a sesson for testing by myself. But, I couldn't still see any
session in sdr.
(I think that maybe I don't know about session open exactly. I just selected
'quick call', and 'vic' was excuted. Video camela in IRIX5.3 was OK.)

I also tested with 'tcpdump' like this.(this is not when i conducted 'quick
call' with sdr)
-------------------------------------------------------------------------------------

# tcpdump -af -c 10 igmp
tcpdump: listening on ec0
00:17:54.512412 comp23.snu.ac.kr > 224.0.0.4: igmp dvmrp Probe [ttl 1]
00:17:54.513451 comp23.snu.ac.kr > 147.47.1.8: igmp dvmrp Probe
00:17:54.513654 comp23.snu.ac.kr > 147.47.1.8: igmp dvmrp Probe
00:18:01.835820 147.47.1.8 > comp23.snu.ac.kr: igmp dvmrp Probe (DF)
00:18:03.514889 comp23.snu.ac.kr > 147.47.1.8: igmp dvmrp Report[|igmp]
00:18:03.515195 comp23.snu.ac.kr > 147.47.1.8: igmp dvmrp Report[|igmp]
00:18:04.515045 comp23.snu.ac.kr > 224.0.0.4: igmp dvmrp Probe [ttl 1]
00:18:04.515919 comp23.snu.ac.kr > 147.47.1.8: igmp dvmrp Probe
00:18:04.516129 comp23.snu.ac.kr > 147.47.1.8: igmp dvmrp Probe
00:18:10.841380 147.47.1.8 > comp23.snu.ac.kr: igmp dvmrp Report[|igmp] (DF)
------------------------------------------------------------------------------------

Anyway, I need your advises.
I want to open the video session with sgi-IRIX5.3. and make certain of the
result.

Current situation is :
1. sgi-IRIX5.3(comp23.snu.ac.kr - 147.46.113.54) : tunneling with 147.47.1.8
mrouted-3.8, sdr, vic-2.8

2. ALzzA Linux release 5.2  - Kernel 2.2.0 on an i586 (comp98.snu.ac.kr -
147.46.113.135)   :  same subnet with upper system. also multicasting possible.
sdr, vic(for qcam) - for confirm, and other testing.

Thanks for reading my writing.
Bye~




From rem-conf Sat Feb 13 16:32:59 1999 
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In-reply-to: Your message of "Sun, 14 Feb 1999 00:45:25 +0900."
             <36C59E14.484AFDB9@secsm.org> 
Date: Sun, 14 Feb 1999 09:23:21 +0900
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> (I think that maybe I don't know about session open exactly. I just selected
> 'quick call', and 'vic' was excuted. Video camela in IRIX5.3 was OK.)

use "Create advertised session", then
it start multicast the session info.

-- yozo.




From rem-conf Sun Feb 14 03:39:17 1999 
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Subject: 2nd CFP session on COORDINATION
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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
	     http://www.gdl.cinvestav.mx/others/call.htm

			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 Sun Feb 14 16:04:11 1999 
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Date: Sun, 14 Feb 1999 15:43:27
To: mbone@isi.edu, rem-conf@es.net
From: Ross Finlayson <finlayson@live.com>
Subject: Update re. multicast MP3 streaming tools
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FYI, a quick update on the multicast MP3 audio tools that I first announced
a couple of weeks ago:

"liveCaster" (the multicast MP3 audio source):
	<http://www.live.com/liveCaster/>
This can now take its MP3 input not only from ".mp3" files, but also from
stdin, or from a HTTP stream (such as a 'Shoutcast' server, or just from a
MP3 file on the web).  Also, it will now automatically make an SDP
announcements for its session (although you can choose not to do so).

"playRTPMPEG" (the MP3/RTP/UDP multicast receiver tool):
	<http://www.live.com/multikit/playRTPMPEG.html>
Unix versions (Linux, FreeBSD, Solaris/SPARC and IRIX) are now available.
These will write the received MP3 frames to 'stdout', so you can use this
tool to feed a separate MP3 player program (such as "X11Amp").  (The
Windows version feeds the "Winamp" MP3 player, but by using a hack where it
acts as a special-purpose HTTP server.)  This tool doesn't do any buffering
of the incoming MP3/RTP/UDP data; instead, it assumes that whatever
application it feeds (via 'stdout') will have sufficient buffering.

(I repeat my plea for someone to add MP3 decoding to an existing multicast
audio tool, such as "rat".  This would ultimately be a better solution.)

Finally, both tools now do RTCP, although currently only at a basic level.
(CNAME is the only item sent in a SDES, and the RRs are 'empty'.)

	Ross.




From rem-conf Sun Feb 14 18:13:02 1999 
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From:     Joe Barnes <stockcrawler@yahoo.com>
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Subject: Are you in a rural area?
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This is not spam. You are currently subscribed to one of our links or newsletters.
To be removed from future mailings reply to sender with a blank message and our 
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the rest.

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From rem-conf Mon Feb 15 07:38:37 1999 
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: Conformance Tests for RTP Scalability Algorithms
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-rtcptest-00.txt
	Pages		: 9
	Date		: 12-Feb-99
	
   This document defines a set of tests that can be run on an RTP imple-
   mentation to determine the level of conformance with the various sca-
   lability algorithms defined in the draft version of RTP. These tests
   do not require access to internal implementations, and use black box
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From rem-conf Mon Feb 15 11:11:52 1999 
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I was just perusing the above  draft and notice that
	the table in section 4.5 gives the PCMA and PCMU
	sampling rates as variable. I thought 8kHz is what
	the world uses. 

	Also, it states 20mS/frame. I see no reason for
	this other than historical reasons and issues of
	software performance limitations.

[\] Robert Wood   

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

mailto:robert_wood@mitel.com



From rem-conf Mon Feb 15 11:59:48 1999 
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At 2:05 PM -0500 2/15/99, Robert Wood wrote:
>I was just perusing the above  draft and notice that
>	the table in section 4.5 gives the PCMA and PCMU
>	sampling rates as variable. I thought 8kHz is what
>	the world uses.
>

Well, that's what telephones use.  But there is nothing intrinsic about
this format that limits it to 8kHz, so there is no reason to limit the
sampling frequency in RTP, either.
David Singer
Apple Computer/QuickTime



From rem-conf Wed Feb 17 23:30:50 1999 
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From rem-conf Thu Feb 18 00:26:42 1999 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: rem-conf@es.net
Subject: MGCP Negotiation
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Has there been a standard method proposed for re-negotiating codec's 
between MGCP gateways?  Since there is no TCP negotiation channel, 
changing codec's midstream seems to require a proprietary scheme.

Regards, 
Neal A. Schneider 

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



From rem-conf Thu Feb 18 09:43:25 1999 
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To: rem-conf@es.net
Subject: How to send large rtcp packets?
Date: Thu, 18 Feb 1999 17:37:39 +0100
Message-ID: <4015.919359459@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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An RTP session with a large number of active senders will result in an
application generating RTCP reception reports with a large number of 
report blocks. Clearly, these become large and may exceed the MTU of the 
network.

What is the correct behaviour for such an application? There seem to be
multiple options:
	1) Send a single large packet, which will be fragmented. This is
	   undesirable, due to the loss multipler effect of fragmentation.
	2) Split the single RTCP packet into multiple smaller packets, and
	   send these back-to-back.
	3) Report only on a subset of the active sources, which will fit 
	   into an MTU sized packet. Which subset? Does it matter?
I guess the correct answer is (3), but can't find confirmation of this in
the spec anywhere.

Comments?

Colin



From rem-conf Thu Feb 18 10:18:01 1999 
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From: David Bukovinsky <D.Bukovinsky@cablelabs.com>
To: rem-conf@es.net, "'Neal Schneider'" <nschneid@hotmail.com>
Subject: RE: MGCP Negotiation
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In the case of fax/modem detection on an MGCP based residential gateway, the
client is given a notification request for fax/modem tones, and when
detected the client notifies the call agent. The call agent can then
instruct the gateway to switch to G.711 on both sides.

My current thoughts are that this would work in a similar fashion if a
bandwidth monitor is involved - if the network becomes saturated, the BW
manager notifies the call agent, the call agent sends down a modify
connection command to both sides to switch to a lower bandwidth codec.

Obviously there are issues with timing [i.e. receiving packets in G.729 when
you've already switched to G.711]; however, in discussions with DSP experts
we aren't seeing anything that would be a show stopper [i.e. you may lose
30ms in a worst case race condition].

---dave

> ----------
> From: 	Neal Schneider[SMTP:nschneid@hotmail.com]
> Sent: 	Thursday, February 18, 1999 9:33 AM
> To: 	rem-conf@es.net
> Subject: 	MGCP Negotiation
> 
> Has there been a standard method proposed for re-negotiating codec's 
> between MGCP gateways?  Since there is no TCP negotiation channel, 
> changing codec's midstream seems to require a proprietary scheme.
> 
> Regards, 
> Neal A. Schneider 
> 
> ______________________________________________________
> Get Your Private, Free Email at http://www.hotmail.com
> 



From rem-conf Thu Feb 18 10:55:55 1999 
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To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Colin Perkins wrote:

Hi Colin;

When I was looking at a lot of rtcp data with MultiMON, I noticed that
your option 3) was the one that was implemented.

What I can't remember is (application specific??) how they cycle through
the subset, or whether they just report on the first so many senders...

That may not be the spec, but it is how the majority of code out there
seems to be written. 

John Stewart
CRC Canada.

> What is the correct behaviour for such an application? There seem to be
> multiple options:
>         1) Send a single large packet, which will be fragmented. This is
>            undesirable, due to the loss multipler effect of fragmentation.
>         2) Split the single RTCP packet into multiple smaller packets, and
>            send these back-to-back.
>         3) Report only on a subset of the active sources, which will fit
>            into an MTU sized packet. Which subset? Does it matter?
> I guess the correct answer is (3), but can't find confirmation of this in
> the spec anywhere.
> 
> Comments?
> 
> Colin



From rem-conf Thu Feb 18 12:31:26 1999 
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Your question is probably best addressed to the megaco list,
megaco@baynetworks.com.

Neal Schneider wrote:
> 
> Has there been a standard method proposed for re-negotiating codec's
> between MGCP gateways?  Since there is no TCP negotiation channel,
> changing codec's midstream seems to require a proprietary scheme.
> 
> Regards,
> Neal A. Schneider
> 
> ______________________________________________________
> Get Your Private, Free Email at http://www.hotmail.com

-- 
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 Feb 18 12:53:10 1999 
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Colin Perkins wrote:
> 
> An RTP session with a large number of active senders will result in an
> application generating RTCP reception reports with a large number of
> report blocks. Clearly, these become large and may exceed the MTU of the
> network.
> 
> What is the correct behaviour for such an application? There seem to be
> multiple options:
>         1) Send a single large packet, which will be fragmented. This is
>            undesirable, due to the loss multipler effect of fragmentation.
>         2) Split the single RTCP packet into multiple smaller packets, and
>            send these back-to-back.
>         3) Report only on a subset of the active sources, which will fit
>            into an MTU sized packet. Which subset? Does it matter?

And what is the path MTU? Unless you do some kind of MTU discovery, you
won't know this at the RTP sender. I guess we could use the minimum
allowed MTU (I think there is such a beast). 

-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 Feb 18 13:09:48 1999 
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Subject: Re: How to send large rtcp packets? 
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> And what is the path MTU? 

Many of the MBone tools today take MTU=1500 for granted. 

/P




From rem-conf Thu Feb 18 13:22:24 1999 
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Quoting from draft-ietf-avt-rtp-new-02.ps, page 17:

If the overall length of a compound (RTCP) packet would exceed MTU of
the network path, it SHOULD be segmented into multiple shorter compound
packets to be transmitted in separate packets of the underlying
protocol.

There is a "may" instead of SHOULD in RFC1889. I am not sure if this is
implemented by anybody, though.

---
Igor Slepchin


Colin Perkins wrote:
> 
> An RTP session with a large number of active senders will result in an
> application generating RTCP reception reports with a large number of
> report blocks. Clearly, these become large and may exceed the MTU of the
> network.
> 
> What is the correct behaviour for such an application? There seem to be
> multiple options:
>         1) Send a single large packet, which will be fragmented. This is
>            undesirable, due to the loss multipler effect of fragmentation.
>         2) Split the single RTCP packet into multiple smaller packets, and
>            send these back-to-back.
>         3) Report only on a subset of the active sources, which will fit
>            into an MTU sized packet. Which subset? Does it matter?
> I guess the correct answer is (3), but can't find confirmation of this in
> the spec anywhere.
> 
> Comments?
> 
> Colin



From rem-conf Thu Feb 18 14:57:27 1999 
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To: Igor Slepchin <igors@dnrc.bell-labs.com>
cc: rem-conf@es.net
Subject: Re: How to send large rtcp packets?
In-reply-to: Your message of "Thu, 18 Feb 1999 16:16:27 EST." <36CC832B.4385668F@dnrc.bell-labs.com>
Date: Thu, 18 Feb 1999 22:51:14 +0100
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--> Igor Slepchin writes:
>Quoting from draft-ietf-avt-rtp-new-02.ps, page 17:
>
>If the overall length of a compound (RTCP) packet would exceed MTU of
>the network path, it SHOULD be segmented into multiple shorter compound
>packets to be transmitted in separate packets of the underlying
>protocol.
>
>There is a "may" instead of SHOULD in RFC1889. I am not sure if this is
>implemented by anybody, though.

Missed that... :-(

Still, does it fully answer the question? If those multiple packets are
sent together (ie: option 2 below), the RTCP transmission interval becomes
incorrect, since it rely on the average packet size so that two small
packets don't have the same effect as one large one. 

This gets us back to sending smaller packets by only reporting on a subset
of sources each interval. I'm concerned what this does to the semantics of
the RRs though: before we either got a report on all sources a receiver was
receiving, or on none of them (if the report was lost). If only some of the
sources are reported on, we can't tell if it's a partial report or if that
receiver is not receiving some sources.

Or have I missed something else?

Colin


>Colin Perkins wrote:
>> 
>> An RTP session with a large number of active senders will result in an
>> application generating RTCP reception reports with a large number of
>> report blocks. Clearly, these become large and may exceed the MTU of the
>> network.
>> 
>> What is the correct behaviour for such an application? There seem to be
>> multiple options:
>>         1) Send a single large packet, which will be fragmented. This is
>>            undesirable, due to the loss multipler effect of fragmentation.
>>         2) Split the single RTCP packet into multiple smaller packets, and
>>            send these back-to-back.
>>         3) Report only on a subset of the active sources, which will fit
>>            into an MTU sized packet. Which subset? Does it matter?
>> I guess the correct answer is (3), but can't find confirmation of this in
>> the spec anywhere.
>> 
>> Comments?
>> 
>> Colin



From rem-conf Thu Feb 18 15:03:34 1999 
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From: Stephen Casner <casner@cisco.com>
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Subject: Re: How to send large rtcp packets?
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Colin,

> What is the correct behaviour for such an application? There seem to be
> multiple options:
> 	1) Send a single large packet, which will be fragmented. This is
> 	   undesirable, due to the loss multipler effect of fragmentation.
> 	2) Split the single RTCP packet into multiple smaller packets, and
> 	   send these back-to-back.
> 	3) Report only on a subset of the active sources, which will fit 
> 	   into an MTU sized packet. Which subset? Does it matter?
> I guess the correct answer is (3), but can't find confirmation of this in
> the spec anywhere.

What I implemented was a combination of 2) and 3).  I fill up an RTCP
compound packet with multiple RR packets up to the MTU, then I have a
knob to control how many such compound RTCP packets may be generated
at once (back-to-back).  I think I have the knob currently set to 2.
If not all sources are reported at one interval, I continue
round-robin at the next interval.

This should all balance out if the interval calculation is done
correctly.  However, the average RTCP packet size calculation computed
by each participant may not be correct if the knob is set greater than
1.  This suggests that your guess at 3) as the right answer would be
valid.

There may be a problem with RTCP monitors if they interpret the lack
of an RR block for a particular source as an indication that no
packets were received from that source in that interval.

I suppose we should say something about this in the spec somewhere.

							-- Steve




From rem-conf Thu Feb 18 18:26:24 1999 
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To: DAAgarwal@lbl.gov, Jason_Sylvester@3com.com, Ksiple@home.com,
        MPerry@lbl.gov, M_Howard-Jeweler@lbl.gov, Rowe@BMRC.Berkeley.edu,
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Subject: Multithreaded devserv for Windows
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Hi All,

Multithreaded versions of devserv are now available for
Windows 95/NT (at ftp://george.lbl.gov/pub/mbone/devserv
or via link to src/binaries at http://www-itg.lbl.gov/
mbone/devserv).  If you are downloading either the source
or binary .zip packages, please be sure to get version 
"a1.0".  

As with the UNIX multithreaded versions, you should
see faster device startup/shutdown and no polling for
network receives.  

Please feel free to email me if there are problems or
if you have feedback, questions...

--Marcia

_____________________________________________________________________
Marcia Perry                            MPerry@lbl.gov
Lawrence Berkeley National Laboratory   WORK#  (510) 486-6786
1 Cyclotron Road                        FAX#   (510) 486-6363
Berkeley, CA 94720                      MS: 50B-2239



From rem-conf Thu Feb 18 20:42:08 1999 
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CC: Igor Slepchin <igors@dnrc.bell-labs.com>, rem-conf@es.net
Subject: Re: How to send large rtcp packets?
References: <6619.919378274@cs.ucl.ac.uk>
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Colin Perkins wrote:
> 
> Still, does it fully answer the question? If those multiple packets are
> sent together (ie: option 2 below), the RTCP transmission interval becomes
> incorrect, since it rely on the average packet size so that two small
> packets don't have the same effect as one large one.

Agreed. You need to send only one packet.

> 
> This gets us back to sending smaller packets by only reporting on a subset
> of sources each interval. I'm concerned what this does to the semantics of
> the RRs though: before we either got a report on all sources a receiver was
> receiving, or on none of them (if the report was lost). If only some of the
> sources are reported on, we can't tell if it's a partial report or if that
> receiver is not receiving some sources.

I don't see this conflict. Not getting a receiver report for some sender
just means that the receiver report was not sent (or, perhaps was lost).
This does not mean that the participant isn't getting RTP from that
sender. If this were the case, there would be a receiver report with
high loss rates.

This might serve as a useful way to decide what RR's to include: highest
loss rate first.

-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 Feb 19 00:50:00 1999 
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To: Stephen Casner <casner@cisco.com>
cc: Colin Perkins <C.Perkins@cs.ucl.ac.uk>, rem-conf@es.net
Subject: Re: How to send large rtcp packets?
In-reply-to: Your message of "Thu, 18 Feb 1999 15:02:41 PST." <Pine.WNT.4.10.9902181451480.289-100000@casner-pc.cisco.com>
Date: Fri, 19 Feb 1999 08:37:44 +0100
Message-ID: <709.919413464@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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In message <Pine.WNT.4.10.9902181451480.289-100000@casner-pc.cisco.com>, Stephe
n Casner typed:

so the questions are
1/ respect the bandwidth restriction on rtcp traffic and do so over a
per packet interval, not just the whole....
2/ make sure no-one is deterministcailly left out of reports
	of course, if sources come and go, this is tricky, so maybe
you want to report most recent seen first with higher probability?

as to MTU, i thought only 576 octets was mandated? if you use 1500
you are really gonna break lots of wireless and legacy links...


 >>Colin,
 >>
 >>> What is the correct behaviour for such an application? There seem to be
 >>> multiple options:
 >>> 	1) Send a single large packet, which will be fragmented. This is
 >>> 	   undesirable, due to the loss multipler effect of fragmentation.
 >>> 	2) Split the single RTCP packet into multiple smaller packets, and
 >>> 	   send these back-to-back.
 >>> 	3) Report only on a subset of the active sources, which will fit 
 >>> 	   into an MTU sized packet. Which subset? Does it matter?
 >>> I guess the correct answer is (3), but can't find confirmation of this in
 >>> the spec anywhere.
 >>
 >>What I implemented was a combination of 2) and 3).  I fill up an RTCP
 >>compound packet with multiple RR packets up to the MTU, then I have a
 >>knob to control how many such compound RTCP packets may be generated
 >>at once (back-to-back).  I think I have the knob currently set to 2.
 >>If not all sources are reported at one interval, I continue
 >>round-robin at the next interval.
 >>
 >>This should all balance out if the interval calculation is done
 >>correctly.  However, the average RTCP packet size calculation computed
 >>by each participant may not be correct if the knob is set greater than
 >>1.  This suggests that your guess at 3) as the right answer would be
 >>valid.
 >>
 >>There may be a problem with RTCP monitors if they interpret the lack
 >>of an RR block for a particular source as an indication that no
 >>packets were received from that source in that interval.
 >>
 >>I suppose we should say something about this in the spec somewhere.
 >>
 >>							-- Steve
 >>
 >>

 cheers

   jon




From rem-conf Fri Feb 19 01:03:04 1999 
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Subject: Re: How to send large rtcp packets? 
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             <709.919413464@cs.ucl.ac.uk> 
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AI was wrong about the 1500. Vic/Vat take 1024+headers for granted though. 

%79 Peppar:> grep MTU vic-2.8/rtp.h
#define RTP_MTU 1024

Thats is the limit vic uses for breaking up video packets. 

/P




From rem-conf Fri Feb 19 06:55:01 1999 
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From: Malathi Veeraraghavan <mv@duke.poly.edu>
To: comsoc.tac@tab.ieee.org, sb.all@ieee.org, rem-conf@es.net
Subject: Infocom 99
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IEEE Infocom99 will be held in New York city
>from March 21-25, 99.

The deadline for early registration is coming up: March 1, 99. 
Please register.

Check the web site http://www.comm.utoronto.ca/~infocom/
for details.


**********************************************************
Malathi Veeraraghavan
Associate Professor		        Tel: 718-260-3493
Dept. of Elec. Engg. 			Fax: 718-260-3074
Polytechnic University			Email: mv@poly.edu
6 Metrotech Center			Room: LC219	
Brooklyn, NY 11201
**********************************************************




From rem-conf Fri Feb 19 08:33:40 1999 
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CERN is pleased to announce the MBONE broadcast of the

             ATLAS Plenary Session and Physics Meeting
             =========================================
                       25-26 February 1999

                   Place: CERN Main Auditorium

*** Times are UTC+1 ***

Thursday 25 Feb. AM : Physics Meeting
Thursday 25 Feb. PM : Plenary Session
Friday 26 Feb. AM/PM: Physics Meeting

(Please see detailed agenda below)

The broadcast is announced on MBone via SDR as "CERN ATLAS" with scope
"World"  (ttl 127).

In case of questions or problems please contact: multicast@noc.cern.ch.


================== Detailed agenda:

  Thursday morning
  ================
  25 February 1999

  I) System performance (summaries)
  ---------------------------------

  - Muons                                          15' +  5'
  - Hadronic calorimetry                           15' +  5'
  - Electromagnetic calorimetry                    15' +  5'
  - Inner Detector                                 15' +  5'

  II) Combined performance
  ------------------------

  - Electron/gamma                                 50' +  5'

    Expect presentations on: E/p, gamma/jet with combined ATRIG/ATRECON,
                             J/psi to e+e-, electron ID ...

  - Jets/ETmiss                                    50' +  5'

    Expect presentations on: jet calibration, FCAL performance, 
                             tau/ETmiss, ETmiss tails for Z+jets,
                             compositeness (?) ...

  - Trigger                                        50' +  5'

    Expect presentations on: B-physics at high L, pi/K decays,
                             LVL2 muons, LVL2 electrons in ID
                             versus noise/eff. of system ...


  Thursday Afternoon: Plenary Session
  ===================================
  25 February 1999

 14:00
                    Introduction and status
                    - LHCC matters 
                    - Council and Non-Member State matters 
                    - decisions of the ATLAS EB since the last Plenary 
                    - status of milestones 
                    - video of some activities in the collaboration 
 14:45
                    Progress on understanding of the 100 MeV background
 15:15
                    Conclusions from the computing review
 16:15
                                                 --- tea break ---
 16:45
                    Overview of the work programme for the coming months
 17:15
                    First report from the Advisory Group committee on ATLAS
                    leadership responsibilities and renewal
 18:00
                    Brief announcements


  Friday morning
  ==============
  26 February 1999

  II) Combined performance (continued)
  ------------------------------------

  - Muons                                           50' +  5'

    Expect presentations on: muon matching (combined resolution/eff.),
                             low-pT muon ID (end-cap Inner Station,
                             tile and HEC) ...
  - b-tagging                                       30' +  5'

    Expect presentations on: soft electrons, soft muons,
                             combined b-tagging performance ...

  III) Structure functions        J. Huston         20' +  5'      
  ------------------------

  IV) Mass scale                                    50' +  5'
  --------------
          -Z to e+e-              S. Simion         10'
          -E/p calibration        M. Taddel         10'
          -Jet scale              M. Bosman         20'
          -Muon scale             C. Guyot          10'      

  V) Luminosity                   K. Piotrzkowski   20' +  5'
  -------------


        ****************     Lunch break     *******************
        ****************   Restart at 13:30  *******************
  Friday afternoon
  ================
  26 February 1999

  VI) QCD                                           35' +  5' 
  -------
  Expect presentations on: diffraction, jets/multijets, b-production,
                           top production ...

  VII) Precision physics                            30' +  5'
  ----------------------
          -Triple/quadruple 
           gauge couplings         J. B. Hansen     15'      
          -W mass measurement      F. Gianotti      10'
          -Sin2(thetaW)            S. Haywood        5'

  Expect presentations on: triple/quadruple gauge boson couplings,
                           W mass, sin2(thetaW) ...
                           
  VIII) Higgs physics                               45' +  5' 
  -------------------
          -Standard Model Higgs    K. Jakobs        15'
          -SUSY Higgs              E. Richter-Was   15'
          -Heavy Higgs             L. Poggioli      15'

  IX) Top physics                                   50' +  5'
  ---------------
  Expect presentations on: measurement of mass in top to jjb decays,
                           meas. of mass in multi-lepton decays,
                           rare decays, 4th family ...

  X) B-physics                                      40' +  5'
  ------------
  Expect presentations on: K/pi separation and impact on sin(2alpha),
                           new channels ...

  XI) SUSY physics and exotics                      50' +  5'  
  ----------------------------
          -SUSY summary                             15'
          -Technicolour            G. Azuelos       10'
          -Heavy gauge bosons      J. Collot        10'
          -Excited quarks          R. Mehdiyev       5'
          -Monopoles               V. Koreshev       5'
          -Compositeness           Z. Usubov         5'






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Here is where you can see the 1,800 page
Y2K report that you probably heard about
on the news. It's free.

Double Click On This Link: http://209.60.152.131





 
 
 
 
 
 



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From rem-conf Mon Feb 22 12:42:45 1999 
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Hi all,

I'm trying to run mrouted-3.9-beta3 with on my Linux box. =


On my old 2.0.36 kernel there was no problem but now I'm using 2.2.0 I ge=
t the =

following error message:

 warning - setsockopt MRT_ADD_MFC: Invalid argument

Does anybody now what could be wrong?

Thanks =


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





From rem-conf Tue Feb 23 01:45:57 1999 
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Subject: Re: Vic on video4linux? 
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El d=EDa Tue, 15 Dec 1998 12:57:46 +0900  Yozo Toda escribi=F3:
> =

> > 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 t=
ried.
> Patrick> =

> Patrick> http://user.cs.tu-berlin.de/~kraxel/linux/
> =

> try and tell us if it's usable or not (-:
>  =

	Yes, I have tried, and it's usable.


-----------------------------------------
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 Tue Feb 23 04:10:18 1999 
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From: GOULEAU Emmanuel CNET/DSM/REN
	 <emmanuel.gouleau@cnet.francetelecom.fr>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Sequence Number
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Dear AVT participants,

Having an RTP server streaming a pre-recorded file on an infinite loop, when
it switches from the end of the (n) session to the begining of the next
session (n+1), does it re-initialize the Sequence Number in the RTP header
with a random value? Or does it increment the Sequence Number by one?

Thank you in advance for your answer.

Emmanuel
**********************************************************************
 <<...>> 

Emmanuel GOULEAU
France Telecom CNET
"Broadcast and On-Line Services Networks"
Tel : +33 2 99 12 43 94  -  Fax : +33 2 99 12 40 98
mailto:emmanuel.gouleau@cnet.francetelecom.fr
**********************************************************************




From rem-conf Tue Feb 23 11:39:41 1999 
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From rem-conf Tue Feb 23 12:16:33 1999 
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Re: Subject: Get More Web Page Hits!

There is a piece of spam circuling with the preceding subject with poorly
forged stardust headers. It did not come from stardust and has nothing to
do with us.

Sigh.



Allen Smith, lazlor@stardust.com
IP Multicast.  Turn it on and tune-in to the future.



From rem-conf Tue Feb 23 13:04:54 1999 
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Actually it apparantly came from an unqualified address, thus "self
forging" to look like it came from my domain. Tricky.

Allen

Sorry for the traffic.



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Date: Tue, 23 Feb 1999 20:50:15 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: GOULEAU Emmanuel CNET/DSM/REN <emmanuel.gouleau@cnet.francetelecom.fr>
cc: "'rem-conf@es.net'" <rem-conf@es.net>
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> Having an RTP server streaming a pre-recorded file on an infinite loop, when
> it switches from the end of the (n) session to the begining of the next
> session (n+1), does it re-initialize the Sequence Number in the RTP header
> with a random value? Or does it increment the Sequence Number by one?

It should increment the sequence number by one, just as if it were one
long continuous transmission.  The same is true of the RTP timestamp:
it should be updated so that first sample of audio at the beginning of
the loop for the next play has the timestamp that would be the same as
the next packet of audio if the loop had one more than it really does,
and similarly for video frames.

I believe that the RTSP spec (RFC 2326) talks about this or a similar
scenario.
							-- Steve




From rem-conf Wed Feb 24 03:00:42 1999 
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Date: Wed, 24 Feb 1999 02:49:48
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From: Ross Finlayson <finlayson@live.com>
Subject: A new RTP payload format for MP3
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FYI, I have recently submitted an Internet-Draft describing a proposed new
RTP payload format for MP3 audio - one that should more tolerant of packet
loss than the existing standard (RFC 2250).  I've included the abstract below.

The I-D will probably appear as "draft-finlayson-rtp-mp3-00.txt"; for now,
you can find it online at <http://www.live.com/rtp-mp3.txt>.

I hope to be able to talk about this briefly at the Minneapolis IETF (if I
can afford to go to it :-(, including whether to adopt it as a product of
the AVT working group for possible standardization.

In the meantime I'd be interested in any comments (especially from experts
in MPEG audio).

	Ross.

---------------------
           A More Loss-Tolerant RTP Payload Format for MP3 Audio

                   <draft-finlayson-rtp-mp3-00.txt>

[...]

2. Abstract

While the RTP payload format defined in RFC 2250 is generally applicable
to all forms of MPEG audio or video, it is less suitable for MPEG 1 or 2,
layer III audio (commonly known as "MP3").  The reason for this is that
an MP3 frame is not a true "Application Data Unit" - it contains a
back-pointer to data in earlier frames, and so cannot be decoded
independently of these earlier frames.  Because RFC 2250 defines that packet
boundaries coincide with frame boundaries, it handles packet loss
inefficiently when carrying MP3 data.  The loss of an MP3 frame will render
some data in previous (or future) frames useless, even if they are received
without loss.

In this document we define a new RTP payload format for MP3 audio.  The new
format is essentially the same as that defined in RFC 2250, except for a
data-preserving rearrangement of the original MPEG frames, so that packet
boundaries now coincide with true MP3 "Application Data Units".  This new
format is therefore more data efficient in the face of packet loss.
---------------------




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(Please apologize, if you receive multiple copies of this CfP)


HUC99 - CALL FOR PAPERS
=======================

International Symposium on Handheld and Ubiquitous Computing
September 27-29, 1999
Karlsruhe, Germany
http://www.teco.edu/huc/
    
Organised by University of Karlsruhe, Telecooperation Office
Hosted by ZKM (Center for Art and Media), Karlsruhe
In cooperation with ACM SIGMOBILE, SIGAPP, SIGCHI and SIGGROUP


IMPORTANT DATES
---------------
  April 30, 1999  Deadline for submission of papers
  June 18, 1999   Author notification
  June 30, 1999   Deadline for submission of posters
  July 16, 1999   Final papers due


YOU ARE INVITED TO PARTICIPATE IN HUC99
---------------------------------------
Truly personal handheld and wearable technologies are small and
unobtrusive, usable most of the time and in most circumstances for
access to information and computing. Complimentary, environment-based
technologies make artefacts of our surrounding world computationally
accessible, and facilitate use of everyday things and environments as
ubiquitous computing interface.

The symposium will bring together researchers and practitioners from
different communities to explore links and synergies between personal
and environment-based technologies, in the context of innovative
applications with thematic focus on "Interaction with Environments",
"Information Appliances" and "Handheld CSCW".

The venue is Karlsruhe in the southwest of Germany, situated in the
Upper Rhine Valley bordering Black Forest, Alsace, and Palatine. The
town is within easy reach from international airports at Frankfurt and
Stuttgart. The symposium will be held at the Center for Art and Media
(ZKM), quite a new type of museum which involves the public with
interactive, hands-on installations.

CALL FOR CONTRIBUTIONS
----------------------
We solicit full papers (5000 words) and poster contributions (1500
words) related to the symposium themes. See below for a non-exhaustive
list of topics of interest. Note, that in addition to papers and
posters, also contributions to workshops are sought. For further
information see http://www.teco.edu/huc/.

TOPICS OF INTEREST INCLUDE:

- Handheld, wearable and environment-based computing
- Ultra-portable devices
- Personal information appliances
- Ubiquitous information appliances
- Ubiquitous computing environments
- Interactive environments
- Location- and context-awareness
- Ad hoc networking and collaboration
- Inherently mobile collaborative work
- Handhelds and wearables for groups and communities
- Awareness of collaboration and work environments
- Novel interfaces and usage metaphors
- Human factors in ubiquitous computing
- Unobtrusive human-computer interaction
- Linking virtual worlds and physical worlds
- Portable media, ubiquitous media, and ambient media
- Augmented reality
- Personal agents and ubiquitous agents
- Appliance networking and management
- Privacy and security concerns
- Mobile informatics
- Experimental school environments
- Applications in arts and entertainment
- New technologies for social interaction

MORE INFORMATION
----------------
For more information contact: 
        Hans-W. Gellersen
        Telecooperation Office (TecO)
        University of Karslruhe
        Vincenz-Priessnitz-Str. 1
        76131 Karlsruhe, Germany
        huc@teco.edu




From rem-conf Wed Feb 24 09:10:23 1999 
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CC: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: Sequence Number
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Stephen Casner wrote:
> 
> > Having an RTP server streaming a pre-recorded file on an infinite loop, when
> > it switches from the end of the (n) session to the begining of the next
> > session (n+1), does it re-initialize the Sequence Number in the RTP header
> > with a random value? Or does it increment the Sequence Number by one?
> 
> It should increment the sequence number by one, just as if it were one
> long continuous transmission.  The same is true of the RTP timestamp:
> it should be updated so that first sample of audio at the beginning of
> the loop for the next play has the timestamp that would be the same as
> the next packet of audio if the loop had one more than it really does,
> and similarly for video frames.
> 
> I believe that the RTSP spec (RFC 2326) talks about this or a similar
> scenario.
>                                                         -- Steve

Yes, in Appendix B.

-Anup.



From rem-conf Wed Feb 24 12:51:21 1999 
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Date: Wed, 24 Feb 1999 15:42:14 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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After an informal survey, a number of people have expressed interest in
participating in a first ``SIP Bake-Off''. The purpose of the bake-off
is to test for interoperability of finished and in in-progress SIP
implementations.

The bake-off will take place at Columbia University, New York, NY on
Thursday, April 8th and Friday, April 9th.

Details can be found at http://www.cs.columbia.edu/~hgs/sip/bakeoff.html

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



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Subject: wmATM'99 in Silicon Valley
References: <199811181053.LAA02134@pacman>
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<<CALL FOR PARTICIPATION>>


Dear Colleagues,


On behalf of the wmATM=9299 Committee, I am pleased to invite you to
participate and join the discussion in this important workshop.

Among over 220 papers submitted to wmATM=9299, 40 original papers are
accepted to be included into the Workshop programme and will be
published in the wmATM=9299 proceedings.

wmATM=9299 will be the significant international event focusing on the
exchange of experiences on wireless mobile ATM implementational issues
targeting 3G wireless communications among scientists, researchers,
engineers and government officers. Apart from the technical sessions,
most of the worldwide leaders in this field will share their expertise
with all participants in the workshop.

I expect this wmATM=9299 will reflect both technical and professional
excellence for this advanced technology, and you will absolutely benefit
well from this technical forum.

For further information about wmATM=9299, please browse the web site at:

http://www.delson.org/tf-wmatm/wmatm99/


Thank you for your time.

Yours truly,



Willie W. Lu, SrMIEEE
General Chair, wmATM=9299 Committee
1960 Linden Lane
Milpitas, CA 95035, U.S.A
E-mail: wwlu@ieee.org







From rem-conf Wed Feb 24 14:27:53 1999 
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload format for Interleaved Media
	Author(s)	: C. Perkins
	Filename	: draft-ietf-avt-interleaving-01.txt
	Pages		: 6
	Date		: 22-Feb-99
	
     This memo defines an interleaving scheme for RTP streams.
     This scheme is derived from the RTP payload format for redundant
     audio data [4] and hence is targetted primarily at streamed
     audio, although it may be of use in other scenarios.

A URL for this Internet-Draft is:
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Internet-Drafts are also available by anonymous FTP. Login with the username
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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From rem-conf Wed Feb 24 20:13:03 1999 
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From:     Hidden Treasures <mgr@Fiberia.com>
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Subject: ADV. 16 Successful years, 100% Guaranteed
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From rem-conf Thu Feb 25 12:31:49 1999 
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Berkeley Multimedia, Interfaces and Graphics Seminar

Parallel Software-Only Video Effects Processing

Wednesday, March 3, 1999, 1:00-2:30 p.m. 405 Soda Hall 

Ketan Meyer-Patel 
Computer Science Division - EECS, UC Berkeley 

Experience in the television and film industries shows that video effects
like fades, transitions, and titling improve the perceived quality of video
material. Traditional production methods for video effects, however, are
not well matched to the needs of Internet video streaming. In this talk, I
describe a parallel software-only video effects processing system designed
specifically for streaming Internet video. I will describe mechanisms for
supporting different types of parallelism and show how their design is
constrained by properties of streaming video formats and protocols. I will
also describe a novel control mechanism built on Scalable Reliable
Multicast (SRM). 

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 Thu Feb 25 20:59:39 1999 
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From:	"DR. DANNY TSANG" <eetsang@ee.ust.hk>
To:	commsoft@cc.belcore.com, multicomm@cc.bellcore.com,
	activenets_all@ittc.ukansas.edu, arpanet-bboard@mc.lcs.mit.edu,
	ActiveNets_Wire@ittc.ukans.edu, issll@mercury.lcs.mit.edu,
	xtp-relay@cs.concordia.ca, end2end-interest@isi.edu, enternet@bbn.com,
	hipparch@sophia.inria.fr, reres@laas.fr, rem-conf@es.net,
	sigmedia@bellcore.com, mobile-ip@tadpole.com, ieeetcpc@ccvm.sunysb.edu,
	cnon@maestro.bellcore.com, cabernet-events@ncl.ac.uk,
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	members@tmforum.org, apnoms-all@nile.postech.ac.kr,
	comsoc-gicb@ieee.org, giga@tele.pitt.edu, SIGMOB@ACM.ORG
Subject: IFIP BC'99 (Hong Kong) - new submission deadline
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Dear colleagues,

Sorry if you receive multiple copies of this message.

Due to the short announcement of the Call for Papers for IFIP Broadband
Communications '99, the paper submission for the original deadline, 
February 28, 1999, is slow. The new paper submission deadline for 
IFIP Broadband Communications'99 has now been extended to 22 MARCH 1999
as requested by some of the potential authors. Please circulate this message 
to your colleagues as wide as possible and to encourage them to submit a paper 
before the deadline.

Please also visit the conference web site for the latest annoucements
at http://www.ee.ust.hk/~bc99

Thank you in advance for your help and look forward to having your
participation in BC'99 in Hong Kong.

Sincerely yours,

Danny H.K. Tsang			Paul J. Kuehn
General Chair of BC'99			Co-chair of BC'99




From rem-conf Thu Feb 25 23:15:31 1999 
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Date: Fri, 26 Feb 1999 02:09:23 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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Folks,

I've submitted an updated FEC draft. Changes since last version:

* specification of usage with SDP
* more details on use with rfc2198

Until it appears in the archives, you can find a copy at:
http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-avt-fec-05.txt

Thanks,
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 Feb 26 04:57:40 1999 
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Date: Fri, 26 Mar 1999 13:58:08 +0100
From: =?iso-8859-1?Q?kav=E9?= Salamatian <salamat@lri.fr>
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Dear sir,

I have heard about a linux driver for the ATI all in wonder Pro video
grabber and of an update of vic to work with this grabber. Anybody can
get me a pointer.

Thanks

Kave





From rem-conf Fri Feb 26 04:57:40 1999 
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Date: Fri, 26 Feb 1999 13:52:59 +0100
From: pach@saturn.kt.agh.edu.pl (Andrzej Pach)
Message-Id: <199902261252.NAA19980@madzia.kt.agh.edu.pl>
Organization: University of Mining and Metallurgy
Address: Mickiewicza 30, 30-059 Krakow, POLAND
To: rem-conf@es.net
Subject: Broadband Access Conference '99 - CfP
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Apologies if you receive unintentional duplicates of this message.





		 Announcement and Call for Papers

		BROADBAND ACCESS CONFERENCE (BAC99)

		Cracow, Poland, October 11-13, 1999

		   http://BAC99.kt.agh.edu.pl/

Sponsored by IEEE Poland Section and Cracow Communications Society Chapter



Rapidly growing demand for Internet resources access and emerging broadband
interactive applications attracts telecom operators' and service providers'
attention to access technologies: copper (xDSL), fibre (FTTx) and wireles (WLL).

The conference addresses a full spectrum of issues related to a residential
access - from signal level, through network architectures and their life cycle
costs up to live field trial description.

The conference will help incumbent operators to determine their own technology
migration path to the broadband forefront that will match their current status
and prospected market niches.

The conference programme is organised in three streamed sessions:

	* Digital Subscriber Line

	* Fibre in the Loop

	* Wireless Solutions

Submitted extended abstracts will undergo three independent reviews
and accepted papers will appear in the conference proceedings. 



Digital Subscriber Line
=======================
The digital subscriber line technologies xDSL have received a great attention
>from the telecom operators in the past years. This stems from the fact that
telephone companies around the world are making decisions how to include
existing twisted-pair infrastructure in their next generation broadband access
networks. Telecom service users are more and more interested in high speed
access to Internet services. It seems that xDSL is the ideal solution for both
sides. However, there are some issues to be solved before these technologies
will become a commercial success. Original papers are invited on such topics as (but not limited to):
	* access network evolution from voice-band modems to broadband DSL
	  systems
	* signaling techniques
	* noise and crosstalk immunity
	* impact of xDSL access networks on the network equipment and cabling
	* status of standardization
	* deployment and testing guidelines
	* IP/ATM integrated services


Fibre in the Loop
=================
There is an common assumption that target solution for telecommunications
broadband access networks will be based on the fibre optic technology.
Deployment of fibre optic based access networks assures high channels
throughput, low signal delay and remarkable performance of the overall system.
The possibility of OA&M costs reduction and an extension of telecommunications
services provided to customers is very promising for telecommunication
operators. Despite of these advantages the process of FITL standardization is
still open. Original papers addressing relevant topics are hereby solicited:
	* Strategic planning of FTTx systems
	* FTTx standardization process 
	* Protocols for FTTx networks
	* Network management systems for FTTx networks
	* Optical devices and technologies for FTTx systems
	* Operation, Administration and Maintenance (OAM) of FTTx networks


Wireless Solutions
==================
The telecom operators are looking for wireless technology to replace a part 
of wire infrastructure in developing countries and big cities where laying
additional wire cables is costly. Wireless Local Loop WLL technology is
considered because the radio systems are deployed rapidly and easily extendible.
Today we have to set a longer term perspective of where WLL systems should be
implemented in the next few years. The aim is to provide new broadband
residential services and encourage competition for existing public telecom
operators with established copper plant. Efforts are now underway to the
development of Broadband Radio Access Networks BRAN. Original papers are invited
on such topics as:
	* WLL technology options covered: microwave point-to-multipoint,
	  cordless, cellular, satellite
	* high-bandwidth packet-oriented services in WLL
	* circuit-oriented services in WLL
	* International Mobile Telecommunications - 2000 IMT-2000 and LL
	* Radio spectrum allocation for the WLL
	* WLL traffic modelling, system capacity
	* standardization and technical options


Tutorials
=========
Proposals for tutorials are solicited. Evaluation of proposals will be based on
the expertise of the instructors, and on the subject matter. Potential
instructors are requested to submit a tutorial proposal of at most 5 pages,
including a biographical sketch, to the BAC99 chairs.



IMPORTANT DATES
===============

	* Submission of extended abstracts (max. 5 pages) due:	May 17, 1999

	* Authors notified:	June 30, 1999

	* Full paper camera ready due: September 10, 1999


SUBMISSION / REGISTRATION
=========================
Submit extended abstracts by e_mail to one of the BAC99 chairs.
For registration and general information contact a chair of the BAC99 
organising Committee, dr Artur Lason.

CONFERENCE COMMITTEE
====================

CONFERENCE CHAIRS

Prof. Andrzej R. PACH
General Chair of BAC'99

Department of Telecommunications
AGH Cracow University 
Al. Mickiewicza 30
30-059 Krakow, Poland
Email:	pach@kt.agh.edu.pl
Tel:	+48 12 634 5582
Fax:	+48 12 634 2372

Prof. Zdzislaw PAPIR
Program Chair of BAC'99

Department of Telecommunications
AGH Cracow University 
Al. Mickiewicza 30
30-059 Krakow, Poland
Email:	papir@kt.agh.edu.pl
Tel:	+48 12 634 5582
Fax:	+48 12 634 2372


TECHNICAL PROGRAM COMMITTEE
===========================
N.E. ANDERSEN, DSC Communications, Denmark
K. ASATANI, Univ. Kogakuin, Japan
A. AZCORRA, Univ. Politecnica de Madrid, Spain
D. BEM, Wroclaw Univ. Of Technology, Poland
W. Y. CHEN, Motorola, USA
A. CIZMAR, Univ. Kosice, Slovakia
A. DOBROGOWSKI, Poznan Univ. of Technology, Poland
A. DZIECH, AGH Cracow Univ., Poland
A.L. EKBLAD, Telia, Sweden
U. FERRERO, CSELT, Italy
K.-P. HO, Univ. Hong Kong
A. JAJSZCZYK, ITTI Ltd. And ATR, Poland
T. KESSLER, Deutsche Telecom, Germany
K. KIKUSHIMA, NTT, Japan
S. KUMAR, Ericsson, USA
B. ORTH, Deutsche Telekom, Germany
K. PAHLAVAN, Worcester Pol., USA
T. POLLET, Bell Alcatel, Belgium
M. POUSA, CET, Portugal
A. SIMMONDS, Sheffield Hallam Univ., UK
P. SPRUYT, Bell Alcatel, Belgium
R. VALADAS, Univ. Aveiro, Portugal
M. YANO, Fujitsu, Japan


ORGANISING COMMITTEE
====================
Chair: Dr. Artur LASON
Department of Telecommunications
AGH Cracow University 
Al. Mickiewicza 30
30-059 Krakow, Poland

Email: lason@kt.agh.edu.pl
Tel:	+48 12 634 5582
Fax:	+48 12 634 2372


Members:
R. Chodorek, AGH Cracow Univ., chodorek@kt.agh.edu.pl
K. Juszkiewicz, AGH Cracow Univ., juszkiew@kt.agh.edu.pl
W. Ludwin, AGH Cracow Univ., ludwin@kt.agh.edu.pl
M. Natkaniec, AGH Cracow Univ., natkanie@kt.agh.edu.pl
K. Wajda, AGH Cracow Univ., wajda@kt.agh.edu.pl
I. Wojdylo, AGH Cracow Univ.
M. Wroniewicz, AGH Cracow Univ. 
T. Zbiezek, AGH Cracow Univ., zbiezek@kt.agh.edu.pl


LOCAL ARRANGEMENTS
==================
Magdalena Lyko, Found. for Progress in Telecommunications, lyko@kt.agh.edu.pl
Krystyna Stefanek, AGH Cracow Univ., stefanek@kt.agh.edu.pl
Anna Kolek, AGH Cracow Univ.



From rem-conf Fri Feb 26 08:09:19 1999 
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Reply-To: "M. Reha Civanlar" <civanlar@research.att.com>
From: "M. Reha Civanlar" <civanlar@research.att.com>
To: "MPEG-4 on IP" <4onip-sys@fzi.de>, <rem-conf@es.net>
Subject: updated draft on MPEG4 payload format
Date: Fri, 26 Feb 1999 11:02:37 -0500
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I've submitted an updated draft describing "RTP payload format for MPEG4
data" to the Internet Drafts Administrator yesterday. Until it becomes
available in the archives, you can
download it from:

http://www.research.att.com/~mrc/draft-ietf-avt-rtp-mpeg4-01.txt (or .ps)

Thanks.

-----------------------------------------------------------------------
M. Reha Civanlar
Technology Leader, AT&T Labs - Research
100 Schultz Drive, Red Bank, NJ 07701, U.S.A
civanlar@research.att.com
Phone: +1 732 345 3305 Fax:+1 732 345 3033
http://www.research.att.com/info/mrc





From rem-conf Fri Feb 26 11:19:23 1999 
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From: Forrest Blair <fblair@s-vision.com>
To: rem-conf@es.net, h323implementors@imtc.org
Subject: Firewalls
Date: Fri, 26 Feb 1999 12:10:29 -0700
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Hello:

I'm looking for information and experience with respect to dealing with
Firewalls while implementing H.323.

Help on this issue is greatly appreciated.

Thanks,

Forrest Blair
Sorenson Vision



From rem-conf Fri Feb 26 12:02:29 1999 
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Hi,
 
 I'm looking for H.450.x ASN.1 source files. Any pointers will be
 appreciated.
 
 Thanks
 
 Raju



From rem-conf Fri Feb 26 12:15:31 1999 
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From: "El-Gebaly, Hani" <hani.el-gebaly@intel.com>
To: Forrest Blair <fblair@S-VISION.COM>, rem-conf@es.net,
        h323implementors@imtc.org
Subject: RE: Firewalls
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Forrest,

Check out http://support.intel.com/support/videophone/trail21/h323_wpr.htm

Hani ElGebaly,
Intel Corp.

> -----Original Message-----
> From:	Forrest Blair [SMTP:fblair@S-VISION.COM]
> Sent:	Friday, February 26, 1999 11:10 AM
> To:	rem-conf@es.net; h323implementors@imtc.org
> Subject:	Firewalls
> 
> Hello:
> 
> I'm looking for information and experience with respect to dealing with
> Firewalls while implementing H.323.
> 
> Help on this issue is greatly appreciated.
> 
> Thanks,
> 
> Forrest Blair
> Sorenson Vision
> -------------------------------------------------------------------------
> for list subscription instructions,
> send email to h323implementors-request@imtc.org with text "info"



From rem-conf Fri Feb 26 14:40:11 1999 
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for MPEG-4 Streams
	Author(s)	: R. Civanlar, V. Balabanian, A. Basso,
                          S. Casner, C. Herpel, C. Perkins
	Filename	: draft-ietf-avt-rtp-mpeg4-01.txt,.ps
	Pages		: 11
	Date		: 25-Feb-99
	
  This document describes a payload format for transporting MPEG-4
  encoded data using RTP. MPEG-4 is a recent standard from ISO/IEC for
  the coding of natural and synthetic audio-visual data. Several
  services provided by RTP are beneficial for MPEG-4 encoded data
  transport over the Internet. Additionally, the use of RTP makes it
  possible to synchronize MPEG-4 data with other real-time data types.
 
  This specification is a product of the Audio/Video Transport working
  group within the Internet Engineering Task Force and ISO/IEC MPEG-4 ad
  hoc group on MPEG-4 over Internet. Comments are solicited and should
  be addressed to the working group's mailing list at rem-conf@es.net
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A URL for this Internet-Draft is:
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Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
	"get draft-ietf-avt-rtp-mpeg4-01.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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Send a message to:
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NOTE:	The mail server at ietf.org can return the document in
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--OtherAccess--

--NextPart--





From rem-conf Fri Feb 26 22:55:11 1999 
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From: "HENG EE WEE" <hengew@hotmail.com>
To: rem-conf-request@es.net, vat-radio-request@elxr.jpl.nasa.gov,
  confctrl-request@isi.edu, int-serv-request@isi.edu, rem-conf@es.net,
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I'm a diploma student in Network Communication course,
currently i'm working on a research paper regarding
Mbone. After i have tried to get everything about 
Mbone over the Internet, i found that many of those 
articles and research papers just describe the general
information about Mbone, but what my lecturer 
required me to do is to find a specific topic on this
area(all about Mbone). Eventually, i have chosen 
a topic "Reliable techniques and protocols to improve 
video conferencing over the Mbone" so, could any
expert supply me all the information or links to
obtain those articles and techical or research 
papers regarding this area ? Thank you.

please reply me at -->hengew@hotmail.com 

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



From rem-conf Sun Feb 28 09:37:23 1999 
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To: mbone@ISI.EDU, rem-conf@es.net
cc: Ross Finlayson <rsf@CS.STANFORD.EDU>
Subject: Re: "liveCaster": MP3 streaming via multicast 
Subject: Re: [mbone-be] Update re. multicast MP3 streaming tools 
In-reply-to: Your message of "Sun, 14 Feb 1999 15:43:27."
             <3.0.5.16.19990214154327.20975616@shell7.ba.best.com> 
Date: Sun, 28 Feb 1999 18:21:40 +0100
From: Patrick De Muynck <Patrick.DeMuynck@belnet.be>
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I thought that RealPlayer G2 should be able to do the job, because:

- with the new so called "scalable multicast" support it's able to
  play an RTP/RCP multicasted stream and furthermore to parse the
  SDP announcement that describes the stream

- it has Codec support for playing MP3

It didn't work for me however.

Here's what I tried: I started broadcasting with the liveCaster,
sending SDP announcements.  I launched the RealPlayer by using Peter
Parnes' WEB-based multicast Session Directory
(http://www.cdt.luth.se/~peppar/progs/mSD/) and clicking on the
liveCaster announcement (alternatively you can just save the SDP
announcement on disk as a file with extension ".sdp"
and open that file in RealPlayer).  Note: if you install RealPlayer G2
on a PC it becomes the default program for the MIME type "application/sdp"
MIME type, file extension = .sdp .

The result was that it receives the stream but can't play it: actually
it goes to the RealNetworks WEB site to find an "unknown codec"
plugin.  Then it reports that no appropriate plugin was found :-(

That's rather strange because the RealPlayer streamed the MP3 file
>from disk all right...  So why?

FYI: using RealPlayer G2 / scalable multicast this way works for
playback of one-way MBone sessions which use H261/DVI/PCM...

I have a related question, which is if there exists a generic program
which can handle the "application/sdp" MIME type, i.e. it should be
able to accept an .sdp file, parse the SDP announcement, and then
startup the proper audio/video tools like MBone or RealPlayer G2.  So
something similar as mLaunch (http://www.cdt.luth.se/~peppar/progs/mlaunch/)
but it should also support Windows and it should be more configurable
concerning the choice of the tools (not only MBone).  Would "multikit"
be able to do this?

Patrick De Muynck

BELNET, the Belgian National Research Network

Patrick.DeMuynck@belnet.be  |	DWTC - BELNET
Tel : +32/2/238.36.76	    |	Wetenschapsstraat, 4
Fax : +32/2/513.57.30	    |	B-1000 Brussel


In message <3.0.5.16.19990214154327.20975616@shell7.ba.best.com>,
Ross Finlayson writes:
>FYI, a quick update on the multicast MP3 audio tools that I first announced
>a couple of weeks ago:
>
>"liveCaster" (the multicast MP3 audio source):
>	<http://www.live.com/liveCaster/>
>This can now take its MP3 input not only from ".mp3" files, but also from
>stdin, or from a HTTP stream (such as a 'Shoutcast' server, or just from a
>MP3 file on the web).  Also, it will now automatically make an SDP
>announcements for its session (although you can choose not to do so).
>
>"playRTPMPEG" (the MP3/RTP/UDP multicast receiver tool):
>	<http://www.live.com/multikit/playRTPMPEG.html>
>Unix versions (Linux, FreeBSD, Solaris/SPARC and IRIX) are now available.
>These will write the received MP3 frames to 'stdout', so you can use this
>tool to feed a separate MP3 player program (such as "X11Amp").  (The
>Windows version feeds the "Winamp" MP3 player, but by using a hack where it
>acts as a special-purpose HTTP server.)  This tool doesn't do any buffering
>of the incoming MP3/RTP/UDP data; instead, it assumes that whatever
>application it feeds (via 'stdout') will have sufficient buffering.
>
>(I repeat my plea for someone to add MP3 decoding to an existing multicast
>audio tool, such as "rat".  This would ultimately be a better solution.)
>
>Finally, both tools now do RTCP, although currently only at a basic level.
>(CNAME is the only item sent in a SDES, and the RRs are 'empty'.)
>
>	Ross.



From rem-conf Sun Feb 28 10:27:07 1999 
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Date: Sun, 28 Feb 1999 19:20:18 +0100 (MET)
From: Christian Isnard - CERN IT/IA <Christian.Isnard@cern.ch>
To: atlas-gen@atlas-lb.cern.ch, HRC@in2p3.fr, htasc@listbox.cern.ch,
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Subject: CERN Particle Physics MBone Announcement 
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CERN is pleased to announce the MBONE broadcast of the following Particle
Physics Seminar:


       The AMANDA South Pole neutrino telescope: first light 

       by Francis HALZEN / Univ. of Winsconsin, Madison, USA

             =========================================
               Tuesday 2 March 1999 at 16:30 (UTC+1)
                                    
                   Place: CERN Main Auditorium

The broadcast will be announced on MBone via SDR as "CERN Physics Seminar" with
scope "World" (ttl 127). 

In case of questions or problems please contact: multicast@noc.cern.ch.

========
ABSTRACT

We will discuss the performance of natural Antarctic ice between 1 and 2
kilometer depths as a particle detector.  We will present a preliminary
analysis of the first year of data from a neutrino telescope which uses large
volumes of ultra-transparent South Pole ice as a low-noise particle detector,
sensing the Cherenkov light from neutrino-induced muons and electrons. This
instrument is monitoring the sky for neutrinos from supernovae and gamma ray
bursts. We are already performing a first search for neutrino emission from the
most energetic cosmic processes involving pulsars, black holes, active galactic
nuclei and the like. The detector also has unique capabilities in searching for
neutrino mass and dark matter. We will argue however that a high energy
neutrino telescope should ultimately have an effective volume of order 1
kilometer cube and will present AMANDA's ongoing and future expansion. 





From rem-conf Sun Feb 28 10:35:07 1999 
From rem-conf-request@es.net Sun Feb 28 10:35:06 1999
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Date: Sun, 28 Feb 1999 10:23:59
To: Patrick De Muynck <Patrick.DeMuynck@belnet.be>
From: Ross Finlayson <finlayson@live.com>
Subject: Re: "liveCaster": MP3 streaming via multicast 
Cc: mbone@ISI.EDU, rem-conf@es.net
In-Reply-To: <199902281721.SAA22268@vivaldi.belnet.be>
References: <Your message of "Sun, 14 Feb 1999 15:43:27."             <3.0.5.16.19990214154327.20975616@shell7.ba.best.com>
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At 06:21 PM 2/28/99 +0100, Patrick De Muynck wrote:
>I thought that RealPlayer G2 should be able to do the job
[...]
>It didn't work for me however.
>
>Here's what I tried: I started broadcasting with the liveCaster,
[...]
>The result was that it receives the stream but can't play it: actually
>it goes to the RealNetworks WEB site to find an "unknown codec"
>plugin.  Then it reports that no appropriate plugin was found :-(
>
>That's rather strange because the RealPlayer streamed the MP3 file
>from disk all right...  So why?

My guess is that the "RealPlayer" doesn't implement RFC 2250 (MPEG/RTP
encapsulation), or perhaps doesn't implement it correctly.  (Maybe someone
>from RealNetworks could comment on whether they've gotten RealPlayer to
play MP3/RTP streams from "liveCaster", and if so, how?)

Anyway, for playing MP3/RTP streams on Windows machines, I suggest instead
using "playRTPMPEG" with "Winamp":
	<http://www.live.com/multikit/playRTPMPEG.html>
	<http://www.winamp.com/>

(If you have these two tools installed, then you can launch MP3/RTP
sessions directly from multikit.)


>I have a related question, which is if there exists a generic program
>which can handle the "application/sdp" MIME type, i.e. it should be
>able to accept an .sdp file, parse the SDP announcement, and then
>startup the proper audio/video tools
[...]
>Would "multikit" be able to do this?

Not at present, although it could be modified to do this (thanks for the
suggestion :-)  Right now multikit, like "sdr", can launch SDP sessions
>from its own, built-in directory GUI - but cannot launch SDP files that
come from someone else's directory GUI (e.g., from a separate web browser).

	Ross.





From rem-conf Sun Feb 28 23:12:22 1999 
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From: Ravi C <ravics@wipsys.soft.net>
To: "El-Gebaly, Hani" <hani.el-gebaly@intel.com>,
        Forrest Blair
	 <fblair@S-VISION.COM>, rem-conf@es.net,
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Subject: RE: Firewalls
Date: Mon, 1 Mar 1999 12:32:34 +0530 
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If I were a medical man, I should prescribe a holiday to any patient =
who
considered his work important.
*	Bertrand Russell


	-----Original Message-----
	From:	El-Gebaly, Hani [SMTP:hani.el-gebaly@intel.com]
	Sent:	samedi 27 f=E9vrier 1999 01:40
	To:	Forrest Blair; rem-conf@es.net; h323implementors@imtc.org
	Subject:	RE: Firewalls

	Forrest,

	Check out
http://support.intel.com/support/videophone/trail21/h323_wpr.htm


	Just tired.=20
	Got a error message that it was not found on the intel server.

	Hani ElGebaly,
	Intel Corp.

	> -----Original Message-----
	> From:	Forrest Blair [SMTP:fblair@S-VISION.COM]
	> Sent:	Friday, February 26, 1999 11:10 AM
	> To:	rem-conf@es.net; h323implementors@imtc.org
	> Subject:	Firewalls
	>=20
	> Hello:
	>=20
	> I'm looking for information and experience with respect to dealing
with
	> Firewalls while implementing H.323.
	>=20
	> Help on this issue is greatly appreciated.
	>=20
	> Thanks,
	>=20
	> Forrest Blair
	> Sorenson Vision
	>
------------------------------------------------------------------------=
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	> for list subscription instructions,
	> send email to h323implementors-request@imtc.org with text "info"



From rem-conf Sun Feb 28 23:30:54 1999 
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