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From rem-conf Sun Apr 02 09:28:55 2000 
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Hi

I would like to get a clarification on how header extensions should be
handled. 

How should an application behave if it receives an RTP video packet (say
H.261) and that packet has an unknown header extension? Should the
receiving application ignore the whole packet or should it just jump
over the header extension part and continue handling the packet as if
there wasn't any extension at all? 

/P



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From rem-conf Mon Apr 03 01:30:46 2000 
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Date: Mon, 3 Apr 2000 01:40:35 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: Nam Kyu Kim <batman@hei.co.kr>
cc: rem-conf@es.net
Subject: Re: [Question..] In RTP internet draft.
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On Fri, 31 Mar 2000, Nam Kyu Kim wrote:

> There seem to be a mistake In last sentence of item 1, I think that
> may be changed to
> " The constant C is set to the average RTCP packet size divided by total RTCP 
> bandwidth and n is set to the total number of members."

You are right, the text in the draft is broken.  Thank you for bringing
it to our attention.  I will fix it in the next revision.

							-- Steve





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From: Stephen Casner <casner@cisco.com>
To: Peter Parnes <peppar@cdt.luth.se>
cc: rem-conf <rem-conf@es.net>
Subject: Re: Header Extensions
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Peter,

> How should an application behave if it receives an RTP video packet (say
> H.261) and that packet has an unknown header extension? Should the
> receiving application ignore the whole packet or should it just jump
> over the header extension part and continue handling the packet as if
> there wasn't any extension at all? 

That decision is up to the application; either choice is acceptable.
The length field is explicit in the header extension definition to
allow the header extension be ignorable.  That is the choice I would
normally make.

Was your question triggered by the comment during the recent AVT
meeting about the vat and vic applications having demonstrated
interoperability on the header extension feature by simply dropping
all packets with the X bit set?  As I said, that is an acceptable
response.  In this particular case, I believe it was an expression of
policy by the implementors to say that use of header extensions should
be minimized.
							-- Steve





From rem-conf Mon Apr 03 01:32:01 2000 
From rem-conf-request@es.net Mon Apr 03 01:32:01 2000
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Date: Mon, 3 Apr 2000 01:41:54 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: Nikki Cranley <94426082@eeng.dcu.ie>
cc: rem-conf@es.net
Subject: Re: Lost RTCP packets
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This question is more appropriate for the AVT working group mailing
list rather than the general IETF mailing list, so I have redirected
my response there.

The RTCP packets are designed so that loss of an individual packet is
not critical.  In particular, in the RR packet, cumulative counts are
used rather than deltas so that a difference between the counts can be
calculated for any pair of RTCP packets to determine the delta for the
period between those two packets.

I believe that using TCP would be a bad idea because it can
significantly perturb the timing of the RTCP packets.  Note that the
timestamps in the RTCP SR and RR packets may be used to calculate a
round-trip time.  If the transport for RTCP were different than that
for RTP, the RTT calculation would not be applicable to the RTP
stream.

Not sure what you mean by "adding QoS".  If you mean retransmission,
see the drafts that were discussed at the recent AVT meeting.

							-- Steve

On Fri, 31 Mar 2000, Nikki Cranley wrote:

> Just a thought, what is the procedure when RTCP packets are lost between
> the client and server. These are fairly important bits of information.
> I am experimenting with RTP/RTCP over UDP/IP with a view to adding QoS -
> however I just wondered there if perhaps a hybrid stack should be used
> for example :-
> RTP/UDP/IP & RTCP/TCP/IP.
> Any comments or ideas about this ...
> 
> Thanking you, rgds,
> Nikki





From rem-conf Mon Apr 03 03:30:14 2000 
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Unknown and even rejected by many vendors and operators only a few =
months ago, SIP is on the brink of making a definitive name for itself.=20
Visit the SIP 2000 International Conference programme:
http://www.upperside.fr/basip.htm



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href=3D"http://www.upperside.fr/basip.htm">http://www.upperside.fr/basip.=
htm</A></DIV>
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From rem-conf Mon Apr 03 11:08:42 2000 
From rem-conf-request@es.net Mon Apr 03 11:08:42 2000
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From: Bill Manning <bmanning@ISI.EDU>
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Subject: Re: available bandwidth
To: gamze@luxxon.com (Gamze Seckin)
Date: Mon, 3 Apr 2000 10:52:15 -0700 (PDT)
Cc: confctrl@ISI.EDU, rem-conf@es.net
In-Reply-To: <004101bf9d93$9fb732c0$a953fea9@eas.asu.edu> from "Gamze Seckin" at Apr 03, 2000 10:39:53 AM
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% Hi,
% 
% I want to calculate the instantenous available link bandwidth on an IP =
% network.
% Can you give me pointers to any IETF RFCs or drafts about any work on =
% this issue?
% 
% thank you very much,
% gamze
% 
% ------=_NextPart_000_003E_01BF9D58.F274FFA0--

for your traditional ethernet networks, the instantenous available
bandwidth is either all of it or none of it.

perhaps you would like to clarify your terminology a bit. :)


--bill



From rem-conf Mon Apr 03 11:09:21 2000 
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From: Peter Parnes <peppar@cdt.luth.se>
Organization: Centre for Distance-spanning =?iso-8859-1?Q?Technology=2FLule=E5?= 
	University of Technology, Marratech AB
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My question was triggered by VIC's behavior although I had completely
missed the comments on the last AVT meeting.  

Anyhow, the application for using the extension header is in a
application I call mRobot, a simple echo application for audio, video
(and the other tools that Marratech Pro provides like chat, Web, file
dist and WB). I use the ext header to identify that the data has been
echoed via a mRobot to prevent loops if more than one mRobot is started
on the same group-port. Just looking at the RTCP is not a solution as it
can be lost (especially if a loop was created and the mRobots start
eating up all the bandwidth :-). The usage of the mRobot is to easier
debug mcast connectivity for end users by having a number of mRobots in
the same session at different geographical locations. 

As UCL seem to be the happy developers of vic these days, do you have
any plans to change vic to just ignore the extension header and not the
whole packet? 

-P



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From: Gamze Seckin <gamze@luxxon.com>
Subject: available bandwidth
To: confctrl@ISI.EDU, rem-conf@es.net
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Hi,

I want to calculate the instantenous available link bandwidth on an IP =
network.
Can you give me pointers to any IETF RFCs or drafts about any work on =
this issue?

thank you very much,
gamze

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
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<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I want to calculate the instantenous =
available link=20
bandwidth on an IP network.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Can you give me pointers to any IETF =
RFCs or=20
drafts&nbsp;about any work on this issue?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>thank you very much,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>gamze</FONT></DIV></BODY></HTML>

------=_NextPart_000_003E_01BF9D58.F274FFA0--




From rem-conf Mon Apr 03 11:12:10 2000 
From rem-conf-request@es.net Mon Apr 03 11:12:10 2000
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Subject: Re: available bandwidth
To: gamze@luxxon.com (Gamze Seckin)
Date: Mon, 3 Apr 2000 11:06:12 -0700 (PDT)
Cc: bmanning@ISI.EDU (Bill Manning), confctrl@ISI.EDU, rem-conf@es.net
In-Reply-To: <006701bf9d96$c5b211e0$a953fea9@eas.asu.edu> from "Gamze Seckin" at Apr 03, 2000 11:02:26 AM
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 since the Internet is packet oriented, is what you want to do is check
the "instantaneous available bandwidth" between the source and 
destinations -on a per packet basis-?
 Yow!
You might want to look at trends between endpoints via pchar or
equivalent. These tend to take a while to build up a reasonable
basis for trend analysis.


% 
% Sorry for the abstract question..Here is my problem:
% I want to know the instantaneous available bw over Internet for resource
% management for a group of video conferencing hosts.
% I recently read RFC 2676 (QoS Routing Mechanisms and OSPF Extensions).
% it is mentioned there, that the available link can be computed via a
% bandwidth management method (I could not find anything about it in OSPF 2).
% I am thinking of a very simple method (e.g. periodic delay measurement etc.
% to find this parameter).
% I may be looking into wrong direction. Any help is appreciated.
% thank you,
% gamze
% ----- Original Message -----
% From: Bill Manning <bmanning@ISI.EDU>
% To: Gamze Seckin <gamze@luxxon.com>
% Cc: <confctrl@ISI.EDU>; <rem-conf@es.net>
% Sent: Monday, April 03, 2000 10:52 AM
% Subject: Re: available bandwidth
% 
% 
% > % Hi,
% > %
% > % I want to calculate the instantenous available link bandwidth on an IP =
% > % network.
% > % Can you give me pointers to any IETF RFCs or drafts about any work on =
% > % this issue?
% > %
% > % thank you very much,
% > % gamze
% > %
% > % ------=_NextPart_000_003E_01BF9D58.F274FFA0--
% >
% > for your traditional ethernet networks, the instantenous available
% > bandwidth is either all of it or none of it.
% >
% > perhaps you would like to clarify your terminology a bit. :)
% >
% >
% > --bill
% >
% 
% 


-- 
--bill



From rem-conf Mon Apr 03 11:34:58 2000 
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Date: Mon, 03 Apr 2000 11:02:26 -0700
From: Gamze Seckin <gamze@luxxon.com>
Subject: Re: available bandwidth
To: Bill Manning <bmanning@ISI.EDU>
Cc: confctrl@ISI.EDU, rem-conf@es.net
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Sorry for the abstract question..Here is my problem:
I want to know the instantaneous available bw over Internet for resource
management for a group of video conferencing hosts.
I recently read RFC 2676 (QoS Routing Mechanisms and OSPF Extensions).
it is mentioned there, that the available link can be computed via a
bandwidth management method (I could not find anything about it in OSPF 2).
I am thinking of a very simple method (e.g. periodic delay measurement etc.
to find this parameter).
I may be looking into wrong direction. Any help is appreciated.
thank you,
gamze
----- Original Message -----
From: Bill Manning <bmanning@ISI.EDU>
To: Gamze Seckin <gamze@luxxon.com>
Cc: <confctrl@ISI.EDU>; <rem-conf@es.net>
Sent: Monday, April 03, 2000 10:52 AM
Subject: Re: available bandwidth


> % Hi,
> %
> % I want to calculate the instantenous available link bandwidth on an IP =
> % network.
> % Can you give me pointers to any IETF RFCs or drafts about any work on =
> % this issue?
> %
> % thank you very much,
> % gamze
> %
> % ------=_NextPart_000_003E_01BF9D58.F274FFA0--
>
> for your traditional ethernet networks, the instantenous available
> bandwidth is either all of it or none of it.
>
> perhaps you would like to clarify your terminology a bit. :)
>
>
> --bill
>




From rem-conf Mon Apr 03 14:34:06 2000 
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Date: Mon, 3 Apr 2000 14:24:32 -0700 (PDT)
From: SJ Lee <sjlee@CS.UCLA.EDU>
To: "S.J. (Sung-Ju) Lee" <sjlee@CS.UCLA.EDU>
Subject: CFP - MOBIHOC : Deadline extended to April 16th!
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[Please accept our apologies if you receive duplicate messages]

NOTE: The submission deadline has been extended to April 16th!
=========================================================================


               Announcement and Call for Papers


                 THE FIRST ANNUAL WORKSHOP ON 
             MOBILE AD HOC NETWORKING & COMPUTING

               in conjunction with Mobicom 2000

                        August 11, 2000
                  Boston, Massachusetts, USA

         http://www.ece.gatech.edu/~cktoh/workshop.html





SCOPE
=====

We are interested in work-in-progress, visionary papers, experimental
and systems-related papers. Papers should describe original, previously
unpublished, and not currently under review by another conference,
workshop, or journal. Topics of interest include, but are not limited to:

 - Ad Hoc Routing Protocols
 - Ad Hoc Multicasting Protocols
 - Ad Hoc Transport Issues
 - Service Discovery Protocols
 - Media Access Techniques
 - Low Power Algorithms and Protocols
 - Ad Hoc Mobile Applications
 - Sensor & Data Fusion Ad Hoc Networks
 - Quality of Service Issues
 - Ad Hoc Mobile Computing Platforms
 - Secure Ad Hoc Services

Please consult the program chairs if you are uncertain whether your paper
falls within the theme of this workshop.


SUBMISSIONS
===========

Postscript copies of your submissions should be sent to 
C.-K. Toh (cktoh@ece.gatech.edu) and Nitin Vaidya (vaidya@cs.tamu.edu). 
All papers will be reviewed for technical merit. Submissions should not
exceed 20 double-spaced pages (including text and figures). 


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

Paper submission due:                  April 16th, 2000
Notification of acceptance:            May 15th, 2000
Camera ready version due:              June 1st, 2000


SPONSORSHIP
===========

    ACM SIGMOBILE and IEEE Communications Society 


ORGANIZING COMMITTEE
====================

General Chair:
    Charles E. Perkins (Nokia Research Center) 

Technical Co-Chairs:
    C.-K. Toh (Georgia Institute of Technology)
    Nitin H. Vaidya (Texas A&M University)

Advisory Committee:
    Leonard Kleinrock (UCLA/Nomadix)
    Victor O.K. Li (USC/Hong Kong University)

Local Arrangement Chair:
    Katia Obraczka (USC/ISI)

Publicity Chair:
    Sung-Ju Lee (University of California, Los Angeles)

Registration Chair:
    Elizabeth M. Royer (University of California, Santa Barbara)

Publication Co-Chairs:
    Stefano Basagni (University of Texas, Dallas)
    Violet R. Syrotiuk (University of Texas, Dallas)

Treasurer:
    Bruce Worthman (IEEE)

Steering Committee:
    Charles E. Perkins (Nokia Research Center) 
    C.-K. Toh (Georgia Institute of Technology)
    Nitin H. Vaidya (Texas A&M University)

Technical Program Committee:
    Samir Das                     University of Cincinnati
    Deborah Estrin                USC/ISI
    J.J. Garcia-Luna-Aceves       University of California, Santa Cruz
    Mario Gerla                   University of California, Los Angeles
    Zygmunt Haas                  Cornell University
    Parviz Kermani                IBM Research
    Katia Obraczka                USC/ISI
    Stephen Pink                  University of Arizona
    Ram Ramanathan                BBN Technologies
    Adam Wolisz                   Technische Universitat Berlin






From rem-conf Tue Apr 04 00:17:21 2000 
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To: Bill Manning <bmanning@ISI.EDU>
cc: gamze@luxxon.com (Gamze Seckin), confctrl@ISI.EDU, rem-conf@es.net
Subject: Re: available bandwidth
In-reply-to: Your message of "Mon, 03 Apr 2000 11:06:12 PDT." <200004031806.LAA15889@zed.isi.edu>
Date: Tue, 04 Apr 2000 08:07:19 +0100
Message-ID: <1029.954832039@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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In message <200004031806.LAA15889@zed.isi.edu>, Bill Manning typed:

 >> since the Internet is packet oriented, is what you want to do is check
 >>the "instantaneous available bandwidth" between the source and 
 >>destinations -on a per packet basis-?
 >> Yow!

yes indeed

 >>You might want to look at trends between endpoints via pchar or
 >>equivalent. These tend to take a while to build up a reasonable
 >>basis for trend analysis.

pathchar gives you what the instantaneous bandwidth may have been some time
ago.....it also has problems on fast short-to-medium distance
links which treat ICMP in slow path but forward packets in fast path - the
delay is lower on the next hop than the current one! at least i find this on
UCL<->ACIRI path where we appear traverses several such hops on superjanet
and abiline...
 >>
 >>
 >>% 
 >>% Sorry for the abstract question..Here is my problem:
 >>% I want to know the instantaneous available bw over Internet for resource
 >>% management for a group of video conferencing hosts.
 >>% I recently read RFC 2676 (QoS Routing Mechanisms and OSPF Extensions).
 >>% it is mentioned there, that the available link can be computed via a
 >>% bandwidth management method (I could not find anything about it in OSPF 2).
 >>% I am thinking of a very simple method (e.g. periodic delay measurement etc.
 >>% to find this parameter).
 >>% I may be looking into wrong direction. Any help is appreciated.
 >>% thank you,
 >>% gamze
 >>% ----- Original Message -----
 >>% From: Bill Manning <bmanning@ISI.EDU>
 >>% To: Gamze Seckin <gamze@luxxon.com>
 >>% Cc: <confctrl@ISI.EDU>; <rem-conf@es.net>
 >>% Sent: Monday, April 03, 2000 10:52 AM
 >>% Subject: Re: available bandwidth
 >>% 
 >>% 
 >>% > % Hi,
 >>% > %
 >>% > % I want to calculate the instantenous available link bandwidth on an IP =
 >>% > % network.
 >>% > % Can you give me pointers to any IETF RFCs or drafts about any work on =
 >>% > % this issue?
 >>% > %
 >>% > % thank you very much,
 >>% > % gamze
 >>% > %
 >>% > % ------=_NextPart_000_003E_01BF9D58.F274FFA0--
 >>% >
 >>% > for your traditional ethernet networks, the instantenous available
 >>% > bandwidth is either all of it or none of it.
 >>% >
 >>% > perhaps you would like to clarify your terminology a bit. :)
 >>% >
 >>% >
 >>% > --bill
 >>% >
 >>% 
 >>% 
 >>
 >>
 >>-- 
 >>--bill

 cheers

   jon




From rem-conf Tue Apr 04 09:38:58 2000 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 4/5 Building the 4th Generation, All-IP Cellular Network --
  James Kempf, Sun Microsystems
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Berkeley Multimedia, Graphics and Interfaces Seminar

Building the 4th Generation, All-IP Cellular Network

Wednesday April 5, 2000 
1:10 - 2:30 PDT
Fujitsu Seminar Room (405 Soda Hall) 

James Kempf 
Sun Microsystems 

Third generation cellular networks are being deployed in Europe (GPRS) and
are about to be deployed in the US (CDMA-2000). Third generation networks
are characterized by tunneling protocols that provide IP for data services
to handsets. Multimedia (and in particular voice) services continue to use
traditional protocols. 

This talk will discuss research topics and standardization work toward
designing 4th generation, all-IP networks, in which the cellular provider
becomes a wireless ISP and the handset an IP phone.
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Thu Apr 06 00:54:41 2000 
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Date: Thu, 06 Apr 2000 09:37:24 +0200
From: Peter Parnes <peppar@cdt.luth.se>
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	University of Technology, Marratech AB
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Hi 

I am trying to find a device for Windows 2000 that can act as a loop
back interface and support multicast. The MSLoopback thing shipped with
2K allows applications to join mcast groups but now data flows over the
device to other applications on the same machine. 

Anybody out there know off a loop back interface that do support mcast
under win2k? 

/P



From rem-conf Thu Apr 06 01:56:39 2000 
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To: 
Subject: TOOLS Europe 2000 - Call for participation
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[Apologies if you receive multiple copies of this message]
[Please forward this message to colleagues you think might be interested]
-------------------------------------------------------------------------

TOOLS Europe 2000
   "Enterprise Architecture - Patterns - Components"

Mont St Michel & St Malo, France
    June 5-8, 2000

http://www.tools.com/europe


CALL FOR PARTICIPATION

TOOLS is the major international conference series devoted to applications
of object-oriented technology. TOOLS Europe 2000 focuses on Enterprise
Architecture, Patterns, and Components.  It will be held in Le Mont St
Michel, at the borderline of Normandy and Brittany in the north-west of
France, which can easily and quickly be reached from Paris.

TOOLS Europe 2000 will continue the commitment to excellence of 
earlier TOOLS conferences in Europe, Australia, Asia and the USA 
with an outstanding program:

- Top keynotes by J. Coplien, B. Meyer, W. Pree, J. Daniels, I. Graham
- 18 tutorials by leading experts in Enterprise Architecture, Patterns,
       Components, UML, Java, Embeded Software, and more...
- 4 workshops, including J. Coplien's two day "Mastery of Patterns" workshop
- 7 technical sessions by selected industry and academic presenters
- a panel on "What is architecture?"
- an exiting social program, with a cocktail diner at "La Mere Poulard", a
visit of the of Mt St Michel Abbey, and a visit and banquet in St Malo.


Full programme and online registration available from

http://www.irisa.fr/TE2000

(Early registration rate applies until May 5, 2000)





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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 4/12 CueVideo: Automated Multimedia Indexing and Retrieval --
  Dulce Ponceleon and Savitha Srinivasan, IBM Almaden Research Center 
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Berkeley Multimedia, Graphics and Interfaces Seminar

CueVideo: Automated Multimedia Indexing and Retrieval

Wednesday April 12, 2000 
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

Dulce Ponceleon and Savitha Srinivasan
IBM Almaden Research Center  

We are witnessing a proliferation of digital media: image, video, audio.
Indexing and preparation of such media for effective search and browse,
represents the key bottleneck towards wider usage of digital media for
variety of applications. The CueVideo project at the IBM Almaden Research
center focuses on developing fully automated means of indexing, hyper
linking and retrieval of audiovisual media material for effective searching
and browsing by users. The core components of the system include: video
analysis and segmentation, visualization and summarization techniques,
spoken document retrieval and cross-modal indexing of audio/video, related
slides and text material. 

Video segmentation is typically used as a first step towards automated
video content analysis. Shot-boundary detection and transitions/effects
detection constitute one of the technologies used to create our video
summaries: storyboard, variants of moving storyboard , and content-based
fast play. These video summaries are new videos of compact size, hence
suitable for download, low-bandwidth streaming environments and quick
browsing. The spoken document retrieval component is based on the use of a
large vocabulary speech
recognition on audio tracks extracted from a video. The transcribed text is
indexed to enable retrieval of relevant audio/video segments. We partition
the problem of spoken document indexing into two distinct levels: The first
is to arrive at a Yahoo or Alta Vista model for the video collection, where
the unit of retrieval is a single audio/video clip. The second is searching
function for audio, where the unit of retrieval is a segment within the
audio/video which best satisfies the user's query. 

Finally, the CueVideo project is specially interested in enabling
applications that enhance the learning experience in education and
training. Towards this end, CueVideo addresses automatic indexing of
instructional material, automatic summarization of audio/video, indexing of
discussed topics and off-line integration with minimal capture constraints. 
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Fri Apr 07 07:01:22 2000 
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Date: Fri, 07 Apr 2000 18:54:11 +0530
From: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
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Subject: reg. length and padding in compound rtcp packet
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I have a doubt regarding the length calculation and padding in compound
rtcp packet
 The document says that
1. "padding should only be required on thelast individual packet because

the compound packet is encrypted as a whole(page 23)"
2. "The length of this RTCP packet in 32-bit words minus one including
the header and any padding"
       for each individual packets in a rtcp compound packet

If i am sending an "RTCP-BYE" packet as one of the packets in the
RTCP-compound packet.  I must do padding only in the bye packet.  But
what if one of the packets previous to the bye packet fall inbetween a
32-bit boundary. Padding must not be done to that packet according to
point 1.  Then  how to calcualte the length of that packet, since it
doesn't fall on a 32-bit boundary (length must be always in units of
32-bits) and padding must not be done.

Can anyone give an solution for this problem.




From rem-conf Fri Apr 07 13:35:30 2000 
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Date: Fri, 07 Apr 2000 13:27:51 -0700
From: Wenqing Jin <wjin@vovida.com>
Organization: Vovida Networks, Inc.
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To: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
CC: rem-conf@es.net
Subject: Re: reg. length and padding in compound rtcp packet
References: <38EDE17A.580E4C17@chiplogic.com>
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"T.V.Rami Reddy" wrote:
> 
> I have a doubt regarding the length calculation and padding in compound
> rtcp packet
>  The document says that
> 1. "padding should only be required on thelast individual packet because
> 
> the compound packet is encrypted as a whole(page 23)"
> 2. "The length of this RTCP packet in 32-bit words minus one including
> the header and any padding"
>        for each individual packets in a rtcp compound packet
> 
> If i am sending an "RTCP-BYE" packet as one of the packets in the
> RTCP-compound packet.  I must do padding only in the bye packet.  But
> what if one of the packets previous to the bye packet fall inbetween a
> 32-bit boundary. Padding must not be done to that packet according to
> point 1.  Then  how to calcualte the length of that packet, since it
> doesn't fall on a 32-bit boundary (length must be always in units of
> 32-bits) and padding must not be done.
> 
> Can anyone give an solution for this problem.

I think you are confused with the padding associate with the padding field
and the padding for alignment of 32-bit. The former is some additional paddings
probably for encryption, that's not mandatory. But padding for 32-bit boundary
is required in any case. So pad bytes to the 32-bit boundary first, and then 
if you have additional padding, add them in the last packet.  
-- 
Wenqing Jin
Senior Software Engineer
Vovida Networks, Inc.
(408) 383-1077



From rem-conf Sat Apr 08 04:14:57 2000 
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From rem-conf Sat Apr 08 04:50:18 2000 
From rem-conf-request@es.net Sat Apr 08 04:50:17 2000
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From: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
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I have some more doubts regarding padding in rtcp compound packet.
We are not using encryption, so padding for it is not required.
In padding intermediate packets (rtpSR, rtpRR, etc), in compound packet to 32-bit
boundary should we set the corresponding padding field TRUE with the last padding
byte containing the count of the total padding bytes added.
rami reddy t.v.

Wenqing Jin wrote:

> "T.V.Rami Reddy" wrote:
> >
> > I have a doubt regarding the length calculation and padding in compound
> > rtcp packet
> >  The document says that
> > 1. "padding should only be required on thelast individual packet because
> >
> > the compound packet is encrypted as a whole(page 23)"
> > 2. "The length of this RTCP packet in 32-bit words minus one including
> > the header and any padding"
> >        for each individual packets in a rtcp compound packet
> >
> > If i am sending an "RTCP-BYE" packet as one of the packets in the
> > RTCP-compound packet.  I must do padding only in the bye packet.  But
> > what if one of the packets previous to the bye packet fall inbetween a
> > 32-bit boundary. Padding must not be done to that packet according to
> > point 1.  Then  how to calcualte the length of that packet, since it
> > doesn't fall on a 32-bit boundary (length must be always in units of
> > 32-bits) and padding must not be done.
> >
> > Can anyone give an solution for this problem.
>
> I think you are confused with the padding associate with the padding field
> and the padding for alignment of 32-bit. The former is some additional paddings
> probably for encryption, that's not mandatory. But padding for 32-bit boundary
> is required in any case. So pad bytes to the 32-bit boundary first, and then
> if you have additional padding, add them in the last packet.
> --
> Wenqing Jin
> Senior Software Engineer
> Vovida Networks, Inc.
> (408) 383-1077




From rem-conf Sat Apr 08 14:40:28 2000 
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From: "R. P. Channing Rodgers, M.D." <rodgers@nlm.nih.gov>
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To: K.Hasler@cs.ucl.ac.uk, O.Hodson@cs.ucl.ac.uk, mbone@isi.edu,
        rem-conf@es.net
Cc: aronson@billings.nlm.nih.gov, rodgers@billings.nlm.nih.gov
Subject: SunVideoPlus 1.3, SunForum, and UCL vic
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Dear Colleagues,

>From the flow of postings to these groups, I suspect that many others
are as frustrated with the PCI-based SunVideoPlus card as we have been.
We achieved a certain amount of stabiilty for a time by using the Osprey-
supplied Sun-style packages (ftp://www.mmac.com/pub/osprey1k/solaris/vic):

   MMACo1kp Osprey-1500 Audio/Video Codec PCI Card Device Driver. 
   MMACo1ku Osprey-1x00 Audio/Video Codec Runtime Support Software
   MMACo1kx Osprey-1x00 Audio/Video Codec XIL 1.3 Runtime Support Software

instead of the SunVideoPlus drivers supplied by Sun.  Under this regime,
we were able to successfully run various versions of UCL vic.

However, we have been attempting to perform interoperability tests between
various open-standards-based teleconferencing tools (UCL tools, SunForum,
netmeeting, CUSeeMe) on various platforms (Sun, PC, Mac).  So we installed
SunForum (http://www.sun.com/desktop/products/software/sunforum/)
on a SPARC 60 under Solaris 2.6.  SunForum works, except for video;
the video-related icons and menu choices on the sunforum application are
always grayed out, as if the application does not detect the card.  The
Sun documentation claims that the SunVideo_Plus 1.3 software
(http://www.sun.com/desktop/products/graphics/sunvideoplus.html)
is required for SunForum.  So we uninstalled the MMAC* packages,
and installed SunVideo_Plus 1.3, comprising these packages:

   SUNWo1kpu   SunVideo Plus Runtime Support Software (sparc) 1.3,REV=1.3.1
   SUNWopi     OPI Runtime Library Package (sparc) 2.2,REV=2.2.0
   SUNWopi1k   SunVideoPlus OPI Runtime Package (sparc) 2.2,REV=2.2.0
   SUNWsvpab   SunVideo Plus Collection (all) 188.2.4
   SUNWvtsvp   SunVTS Sun Video Plus test module
   SUNWo1kpd.u SunVideo Plus PCI Card Dev Driver (sparc.sun4u) 1.3,REV=2.3.1
   SUNWo1kpx   SunVideo Plus XIL-1.3 Runtime Support Software

However, there is no change in the behavior of sunforum (video still not
working), and we are uncertain as to whether vic is going to work or not
(have not completed modification of our shell script wrapper).

Has anyone succeeded in getting all components of sunforum to work on a similar
platform, while also still successfully supporting the UCL version of vic?
Any ideas about what we are doing wrong here?

Thanks and Cheerio, Rick Rodgers



From rem-conf Sat Apr 08 21:52:30 2000 
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Subject: OASD C3I CONFERENCE:  "Distributed Networked Computing for a Secu
	re Defense" 
Date: Fri, 24 Mar 2000 10:49:30 -0500
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*************************************************************** 
"Distributed Networked Computing for a Secure Defense" 
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To Register for DEFENSE@E-BUSINESS online, visit www.theotg.com/OSD 
To Register via fax, simply return the form at the bottom of this message to
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*****************************************************************

Dear Colleague,

The Office of the Secretary of Defense (C3I) and the sponsors of
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Networked Computing for a Secure Defense" forum.  IT leadership from DoD,
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Learn how Information Assurance, Distributed Object Computing, and the
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  -  William Vass, Director of Technology, OSD C3I  
  -  Steve Vinoski, CTO, IONA Technologies  
  -  Dr. Jim Waldo, Distinguished Engineer, Sun Microsystems  
  -  Andrew Watson, CTO and Chief Architect, Object Management Group  
  -  John A. Weiler, Executive Director, Interoperability Clearinghouse  
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  -  Howard Winter, Project Manager, Unified Cryptological Architecture,
National Security Agency ...And a host of other industry dignitaries.

Register Online at http://www.theotg.com/OSD

COSTS:  
Early Bird Government Rate: $195  
Early Bird Commercial Rate: $295 (AFCEA, ICH and OMG members $270)
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(After April 14, $100 late fee will be applied, on-site Registration: $450,
Meal-passes: $50)

NOTE April 26, from 09:00 to 13:00, the Interoperability Clearinghouse will
be co-hosting an International Interoperability Working Group with NDIA, OSD
ATL, European Union, and the Software Productivity Consortium.  Attendance
is open. Cost is $35 to cover lunch.  Goal is to establish software
architecture validation models for COTS integration.  Call 703-768-4975 or
1-800-SEE-OOTS for assistance

Reserve your room at the Hilton by calling (703) 418-6800 and asking for the
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For Group rates or sponsorship information call 703-768-0400.

Government Chair: Mr. William Vass, OSD C3I, 703-602-2720 x102,
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From rem-conf Sun Apr 09 22:37:55 2000 
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>From john_weiler@e-interop.com  Wed Apr  5 13:54:05 2000
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From rem-conf Sun Apr 09 23:04:35 2000 
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Removal instructions below

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To be removed call: 888-800-6339 X1377






From rem-conf Mon Apr 10 00:26:48 2000 
From rem-conf-request@es.net Mon Apr 10 00:26:47 2000
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Date: Mon, 10 Apr 2000 00:33:03 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
cc: rem-conf@es.net
Subject: Re: reg. length and padding in compound rtcp packet
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On Sat, 8 Apr 2000, T.V.Rami Reddy wrote:

> I have some more doubts regarding padding in rtcp compound packet.
> We are not using encryption, so padding for it is not required.  In
> padding intermediate packets (rtpSR, rtpRR, etc), in compound packet
> to 32-bit boundary should we set the corresponding padding field
> TRUE with the last padding byte containing the count of the total
> padding bytes added.

Are you referencing RFC 1889 or the draft revisions of it?  See:

http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-07.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-07.ps

The RTCP SR and RR packets are naturally a multiple of 32 bits long,
so they don't need any padding to a 32-bit boundary.  Only the SDES
and BYE packets require padding to a 32-bit boundary.  For both of
those, the current text says:  "Note that this padding is separate
>from that indicated by the P bit in the RTCP header."  So, the answer
to your question is "no".  Is the text not clear enough?  If not,
please point out a specific location that needs to be changed.

							-- Steve




From rem-conf Mon Apr 10 01:08:34 2000 
From rem-conf-request@es.net Mon Apr 10 01:08:33 2000
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To: K.Hasler@cs.ucl.ac.uk, O.Hodson@cs.ucl.ac.uk, mbone@isi.edu,
	rem-conf@es.net, aronson@billings.nlm.nih.gov,
	rodgers@billings.nlm.nih.gov
Subject: Re: SunVideoPlus 1.3, SunForum, and UCL vic
In-reply-to: Your message of "Sat, 08 Apr 2000 17:25:42 -0400."
             <200004082125.RAA01713@billings.nlm.nih.gov>
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Date: Mon, 10 Apr 2000 17:33:01 +0930
From: Mark Prior <mrp@connect.com.au>
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     Has anyone succeeded in getting all components of sunforum to work on a similar
     platform, while also still successfully supporting the UCL version of vic?
     Any ideas about what we are doing wrong here?

I don't know about the SunForum stuff but at the IETF one of the
workstations we were using was a U5 running Solaris 7 with SunVideo
Plus 1.3 and that didn't have any problems with the UCL version of
vic.

Mark.



From rem-conf Mon Apr 10 02:57:09 2000 
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From: "Nam Kyu Kim" <batman@hei.co.kr>
To: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
Cc: <rem-conf@es.net>
References: <Pine.WNT.4.21.0004100021190.-310617@oak.cisco.com>
Subject: Re: reg. length and padding in compound rtcp packet
Date: Mon, 10 Apr 2000 18:52:15 +0900
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	charset="iso-8859-1"
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Dear Reddy and all.

Let me add more explanation for clarification.

> The RTCP SR and RR packets are naturally a multiple of 32 bits long,
> so they don't need any padding to a 32-bit boundary.  Only the SDES
> and BYE packets require padding to a 32-bit boundary.  For both of
> those, the current text says:  "Note that this padding is separate
> from that indicated by the P bit in the RTCP header."  So, the answer
> to your question is "no".  Is the text not clear enough?  If not,
> please point out a specific location that needs to be changed.

The RTCP SR and RR packets are naturally a multiple of 32 bits long.
So only the SDES and BYE packets may require padding to a 32-bit =
boundary.
But Text says only last packet can set P bit, and BYE packet must be =
last if it exist.
So the problem is what I do if SDES packet need to pad.
RTP internet draft says in page33=20
"The list of items in each chuck MUST be terminated by one or more null =
octets,=20
the first of which is interpreted as an item type of zero to denote the =
end of the list.
No length octet follows the null item type octet, but additional null =
octet MUST be=20
included if needed to pad until the next 32-bit boundary. Note that this =
padding=20
is separate from that indicated by the P bit in the RTCP header."

And there is another question regarding above sentences.
What shall I do, if there is no need to pad null octets until 32-bit =
boundry.
But text says "The list of items in each chuck MUST be terminated by one =
or more null octets".
I have to pad 4 bytes null octets?

-----------------------------------------------------
Nam Kyu Kim=20
R & D Team 9
Mobile Telecommunication Terminal Division
HYUNDAI ELECTRONICS INDUSTRIES Co., Ltd.

TEL : (82-2)580-5511       FAX : (82-2)580-5487
Home : (82-2)2282-3077   C.P. : 017-211-3077
E-mail : batman@good.to
-----------------------------------------------------


------=_NextPart_000_0023_01BFA31D.E355E6E0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2314.1000" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>Dear Reddy and =
all.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>Let me add more explanation =
for=20
clarification.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>&gt; The RTCP SR and RR =
packets are naturally a=20
multiple of 32 bits long,<BR>&gt; so they don't need any padding to a =
32-bit=20
boundary.&nbsp; Only the SDES<BR>&gt; and BYE packets require padding to =
a=20
32-bit boundary.&nbsp; For both of<BR>&gt; those, the current text =
says:&nbsp;=20
"Note that this padding is separate<BR>&gt; from that indicated by the P =
bit in=20
the RTCP header."&nbsp; So, the answer<BR>&gt; to your question is =
"no".&nbsp;=20
Is the text not clear enough?&nbsp; If not,<BR>&gt; please point out a =
specific=20
location that needs to be changed.<BR><BR>The RTCP SR and RR packets are =

naturally a multiple of 32 bits long.</FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>So only the SDES and BYE =
packets may require padding=20
to a 32-bit boundary.</FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>But Text says only last =
packet can set P bit, and BYE=20
packet must be last if it exist.</FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>So the problem is what I do =
if SDES packet need to=20
pad.</FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>RTP internet draft says in =
page33 </FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>"The list of items in each =
chuck MUST be terminated by=20
one or more null octets, </FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>the first of which is =
interpreted as an item type of=20
zero to denote the end of the list.</FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>No length octet follows the =
null item type octet, but=20
additional null octet MUST be </FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>included if needed to pad =
until the next 32-bit=20
boundary. <U>Note that this padding </U></FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2><U>is separate from that =
indicated by the P bit in the=20
RTCP header</U>."</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>And there is&nbsp;another =
question&nbsp;regarding=20
above sentences.</FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>What shall I do, if there is =
no need to pad null=20
octets until 32-bit boundry.</FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>But text says "The list of =
items in each chuck MUST be=20
terminated by one or more null octets".</FONT></DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>I have to pad 4 bytes null =
octets?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3D&#44404;&#47548;=20
size=3D2>-----------------------------------------------------<BR>Nam =
Kyu Kim=20
<BR>R &amp; D Team 9<BR>Mobile Telecommunication Terminal =
Division<BR>HYUNDAI=20
ELECTRONICS INDUSTRIES Co., Ltd.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>TEL :=20
(82-2)580-5511&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX : =
(82-2)580-5487<BR>Home=20
: (82-2)2282-3077&nbsp;&nbsp; C.P. : 017-211-3077<BR>E-mail : <A=20
href=3D"mailto:batman@good.to">batman@good.to</A><BR>--------------------=
---------------------------------</FONT></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0023_01BFA31D.E355E6E0--




From rem-conf Mon Apr 10 04:29:33 2000 
From rem-conf-request@es.net Mon Apr 10 04:29:32 2000
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To: rem-conf@es.net
Subject: AVT last meeting minutes
From: Lionel Vidal <lionel.vidal@bst.bsf.alcatel.fr>
Date: 10 Apr 2000 13:23:00 +0200
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Hello,

I am looking for an URL where I could find the slides presented in AVT
during last meeting in Adelaide.  Are such informations available
somewhere?

Regards,
Lionel.




From rem-conf Mon Apr 10 04:44:56 2000 
From rem-conf-request@es.net Mon Apr 10 04:44:56 2000
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To: Lionel Vidal <lionel.vidal@bst.bsf.alcatel.fr>
cc: rem-conf@es.net
Subject: Re: AVT last meeting minutes
In-reply-to: Your message of "10 Apr 2000 13:23:00 +0200." <u8zymi4iz.fsf@bst.bsf.alcatel.fr>
Date: Mon, 10 Apr 2000 12:41:48 +0100
Message-ID: <1585.955366908@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Lionel Vidal writes:
>I am looking for an URL where I could find the slides presented in AVT
>during last meeting in Adelaide.  Are such informations available
>somewhere?

I'm in the process of sorting them now, they should be available from
	http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/
later today. If anyone has not yet sent me their slides, please do so
and I'll ensure they're included in the minutes.

Colin



From rem-conf Mon Apr 10 08:09:45 2000 
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To: <rem-conf@es.net>
Subject: IP Policing 2000, CFP
Date: Mon, 10 Apr 2000 17:00:12 +0200
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IP Policing 2000, Paris, 12-15 September

=20
The need to police IP networks has grown in importance with the rise of =
a new generation of applications such as VPNs, VoIP or IPSec.
Approaches under consideration include COPS, DEN/LDAP and PIB, but their =
ultimate potential remains undecided.
=20
See:
http://www.upperside.fr/baippol.htm


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2314.1000" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT color=3Dblack face=3DARIAL size=3D2>IP Policing 2000, Paris, =
12-15=20
September<BR></FONT></DIV>
<DIV><FONT color=3Dblack face=3DARIAL size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3Dblack face=3DARIAL size=3D2>The need to police IP =
networks has=20
grown in importance with the rise of a new generation of applications =
such as=20
VPNs, VoIP or IPSec.<BR>Approaches under consideration include COPS, =
DEN/LDAP=20
and PIB, but their ultimate potential remains undecided.</FONT></DIV>
<DIV><FONT color=3Dblack face=3DARIAL size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3Dblack face=3DARIAL size=3D2>See:</FONT></DIV>
<DIV><FONT color=3Dblack face=3DARIAL size=3D2></FONT><FONT size=3D2><A=20
href=3D"http://www.upperside.fr/baippol.htm">http://www.upperside.fr/baip=
pol.htm</A></FONT></DIV>
<DIV>&nbsp;</DIV></FONT></DIV></FONT></DIV></BODY></HTML>

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From rem-conf Mon Apr 10 08:13:14 2000 
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From: "Le Masurier, Laura" <LLEMASURIER@COURTSERVICE.GSI.GOV.UK>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Videoconferencing
Date: Mon, 10 Apr 2000 16:11:08 +0100
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Hi,
 
  I work for the Court Service in London England.  We are interested in
video conferencing.  I have searched the net unsuccessfully.  All I want to
know is where there are centres that I could have a video conference in the
London, Chester or Leeds areas.  If I wanted a video conference, where could
I go.  
 
  Can you give me any information atall that may help us?
 
Many Thanks



Laura Lemasurier 
Laura Le Masurier 
Modernising the Civil Courts
0171 210 1620 



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From rem-conf Mon Apr 10 22:57:38 2000 
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Date: Tue, 11 Apr 2000 10:55:48 +0530
From: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
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To: Nam Kyu Kim <batman@hei.co.kr>
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Subject: Re: reg. length and padding in compound rtcp packet
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Hi Nam Kyu Kim,
Please see my comments in the lines.

with regards.
-rami reddy t.v.

Nam Kyu Kim wrote:

> Dear Reddy and all. Let me add more explanation for clarification. >
> The RTCP SR and RR packets are naturally a multiple of 32 bits long,
> > so they don't need any padding to a 32-bit boundary.  Only the SDES
> > and BYE packets require padding to a 32-bit boundary.  For both of
> > those, the current text says:  "Note that this padding is separate
> > from that indicated by the P bit in the RTCP header."  So, the
> answer
> > to your question is "no".  Is the text not clear enough?  If not,
> > please point out a specific location that needs to be changed.
>
> The RTCP SR and RR packets are naturally a multiple of 32 bits long.So
> only the SDES and BYE packets may require padding to a 32-bit
> boundary.But Text says only last packet can set P bit, and BYE packet
> must be last if it exist.So the problem is what I do if SDES packet
> need to pad.
>
> ****I guess we can add padding bytes at the end of SDES but not need
> to set
> padding bit.  I think we need to set padding bit only if encryption is
> used.  Otherwise all the component packets in a compound packet are
> aligned to 32bit boudary and padding bit  must not be set.
>
> RTP internet draft says in page33"The list of items in each chuck MUST
> be terminated by one or more null octets,the first of which is
> interpreted as an item type of zero to denote the end of the list.No
> length octet follows the null item type octet, but additional null
> octet MUST beincluded if needed to pad until the next 32-bit boundary.
> Note that this paddingis separate from that indicated by the P bit in
> the RTCP header." And there is another question regarding above
> sentences.What shall I do, if there is no need to pad null octets
> until 32-bit boundry.But text says "The list of items in each chuck
> MUST be terminated by one or more null octets".I have to pad 4 bytes
> null octets?**** I think yes.
>
>
>
>
>
>  -----------------------------------------------------
> Nam Kyu Kim
> R & D Team 9
> Mobile Telecommunication Terminal Division
> HYUNDAI ELECTRONICS INDUSTRIES Co., Ltd. TEL : (82-2)580-5511
> FAX : (82-2)580-5487
> Home : (82-2)2282-3077   C.P. : 017-211-3077
> E-mail : batman@good.to
> -----------------------------------------------------

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<html>
Hi Nam Kyu Kim,
<br>Please see my comments in the lines.
<p>with regards.
<br>-rami reddy t.v.
<p>Nam Kyu Kim wrote:
<blockquote TYPE=CITE><style></style>
<font face="??"><font size=-1>Dear
Reddy and all.</font></font>&nbsp;<font face="??"><font size=-1>Let me
add more explanation for clarification.</font></font>&nbsp;<font face="??"><font size=-1>>
The RTCP SR and RR packets are naturally a multiple of 32 bits long,</font></font>
<br><font face="??"><font size=-1>> so they don't need any padding to a
32-bit boundary.&nbsp; Only the SDES</font></font>
<br><font face="??"><font size=-1>> and BYE packets require padding to
a 32-bit boundary.&nbsp; For both of</font></font>
<br><font face="??"><font size=-1>> those, the current text says:&nbsp;
"Note that this padding is separate</font></font>
<br><font face="??"><font size=-1>> from that indicated by the P bit in
the RTCP header."&nbsp; So, the answer</font></font>
<br><font face="??"><font size=-1>> to your question is "no".&nbsp; Is
the text not clear enough?&nbsp; If not,</font></font>
<br><font face="??"><font size=-1>> please point out a specific location
that needs to be changed.</font></font>
<p><font face="??"><font size=-1>The RTCP SR and RR packets are naturally
a multiple of 32 bits long.</font></font><font face="??"><font size=-1>So
only the SDES and BYE packets may require padding to a 32-bit boundary.</font></font><font face="??"><font size=-1>But
Text says only last packet can set P bit, and BYE packet must be last if
it exist.</font></font><font face="??"><font size=-1>So the problem is
what I do if SDES packet need to pad.</font></font><font face="??"><font size=-1></font></font>
<p><font face="??"><font size=-1>****I guess we can add padding bytes at
the end of SDES but not need to set</font></font>
<br><font face="??"><font size=-1>padding bit.&nbsp; I think we need to
set padding bit only if encryption is used.&nbsp; Otherwise all the component
packets in a compound packet are aligned to 32bit boudary and padding bit&nbsp;
must not be set.</font></font><font face="??"><font size=-1></font></font>&nbsp;<font face="??"><font size=-1></font></font>
<p><font face="??"><font size=-1>RTP internet draft says in page33</font></font><font face="??"><font size=-1>"The
list of items in each chuck MUST be terminated by one or more null octets,</font></font><font face="??"><font size=-1>the
first of which is interpreted as an item type of zero to denote the end
of the list.</font></font><font face="??"><font size=-1>No length octet
follows the null item type octet, but additional null octet MUST be</font></font><font face="??"><font size=-1>included
if needed to pad until the next 32-bit boundary. <u>Note that this padding</u></font></font><font face="??"><font size=-1><u>is
separate from that indicated by the P bit in the RTCP header</u>."</font></font>&nbsp;<font face="??"><font size=-1>And
there is another question regarding above sentences.</font></font><font face="??"><font size=-1>What
shall I do, if there is no need to pad null octets until 32-bit boundry.</font></font><font face="??"><font size=-1>But
text says "The list of items in each chuck MUST be terminated by one or
more null octets".</font></font><font face="??"><font size=-1>I have to
pad 4 bytes null octets?</font></font>**** I think yes.
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;<font face="??"><font size=-1>-----------------------------------------------------</font></font>
<br><font face="??"><font size=-1>Nam Kyu Kim</font></font>
<br><font face="??"><font size=-1>R &amp; D Team 9</font></font>
<br><font face="??"><font size=-1>Mobile Telecommunication Terminal Division</font></font>
<br><font face="??"><font size=-1>HYUNDAI ELECTRONICS INDUSTRIES Co., Ltd.</font></font>&nbsp;<font face="??"><font size=-1>TEL
: (82-2)580-5511&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX : (82-2)580-5487</font></font>
<br><font face="??"><font size=-1>Home : (82-2)2282-3077&nbsp;&nbsp; C.P.
: 017-211-3077</font></font>
<br><font face="??"><font size=-1>E-mail : <a href="mailto:batman@good.to">batman@good.to</a></font></font>
<br><font face="??"><font size=-1>-----------------------------------------------------</font></font>&nbsp;</blockquote>
</html>

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From rem-conf Mon Apr 10 22:57:43 2000 
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Date: Mon, 10 Apr 2000 23:06:57 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: Nam Kyu Kim <batman@hei.co.kr>
cc: "T.V.Rami Reddy" <ramreddy@chiplogic.com>, rem-conf@es.net
Subject: Re: reg. length and padding in compound rtcp packet
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On Mon, 10 Apr 2000, Nam Kyu Kim wrote:

> The RTCP SR and RR packets are naturally a multiple of 32 bits long.
> So only the SDES and BYE packets may require padding to a 32-bit boundary.
> But Text says only last packet can set P bit, and BYE packet must be
> last if it exist.
> So the problem is what I do if SDES packet need to pad.

I will say it again, please read carefully: Only the SDES and BYE
packets require padding to a 32-bit boundary.  For both of those, the
current text says:  "Note that this padding is separate from that
indicated by the P bit in the RTCP header."

In other words, the padding of SDES and BYE packets to be a multiple
of 32 bits long HAS NOTHING TO DO WITH THE P BIT!!!  Every instance of
an SDES or BYE packet gets padded will nulls to be a multiple of 32
bits long.  Then after the desired collection of individual RTCP
packets are stacked into a compound RTCP packet, if the whole length
needs to be increased to be a multiple of an encryption block length
or for some other purpose, then the P bit can be turned on in the last
of the individual RTCP packets, the length field of that RTCP packet
is increased by the number of 32-bit words in the padding, and the
number of octets of padding is written into the last octet of the
padding, as described in the spec.  Is that clear?

> RTP internet draft says in page33 
> "The list of items in each chuck MUST be terminated by one or more
> null octets, the first of which is interpreted as an item type of
> zero to denote the end of the list.  No length octet follows the
> null item type octet, but additional null octet MUST be included if
> needed to pad until the next 32-bit boundary. Note that this padding
> is separate from that indicated by the P bit in the RTCP header."
> 
> And there is another question regarding above sentences.
> What shall I do, if there is no need to pad null octets until 32-bit boundry.
> But text says "The list of items in each chuck MUST be terminated by
> one or more null octets". 
> I have to pad 4 bytes null octets?

Yes, you have to add 4 null octets, but the first one is not padding.
The first one is a null item octet, indicating the end of the item
list.  The remaining three null octets are padding.

Please help me understand why the text you quoted above was not clear
on this point.
							-- Steve




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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 4/12 CueVideo: Automated Multimedia Indexing and Retrieval --
  Dulce Ponceleon and Savitha Srinivasan, IBM Almaden Research Center 
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Berkeley Multimedia, Graphics and Interfaces Seminar

CueVideo: Automated Multimedia Indexing and Retrieval

Wednesday April 12, 2000 
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

Dulce Ponceleon and Savitha Srinivasan
IBM Almaden Research Center  

We are witnessing a proliferation of digital media: image, video, audio.
Indexing and preparation of such media for effective search and browse,
represents the key bottleneck towards wider usage of digital media for
variety of applications. The CueVideo project at the IBM Almaden Research
center focuses on developing fully automated means of indexing, hyper
linking and retrieval of audiovisual media material for effective searching
and browsing by users. The core components of the system include: video
analysis and segmentation, visualization and summarization techniques,
spoken document retrieval and cross-modal indexing of audio/video, related
slides and text material. 

Video segmentation is typically used as a first step towards automated
video content analysis. Shot-boundary detection and transitions/effects
detection constitute one of the technologies used to create our video
summaries: storyboard, variants of moving storyboard , and content-based
fast play. These video summaries are new videos of compact size, hence
suitable for download, low-bandwidth streaming environments and quick
browsing. The spoken document retrieval component is based on the use of a
large vocabulary speech
recognition on audio tracks extracted from a video. The transcribed text is
indexed to enable retrieval of relevant audio/video segments. We partition
the problem of spoken document indexing into two distinct levels: The first
is to arrive at a Yahoo or Alta Vista model for the video collection, where
the unit of retrieval is a single audio/video clip. The second is searching
function for audio, where the unit of retrieval is a segment within the
audio/video which best satisfies the user's query. 

Finally, the CueVideo project is specially interested in enabling
applications that enhance the learning experience in education and
training. Towards this end, CueVideo addresses automatic indexing of
instructional material, automatic summarization of audio/video, indexing of
discussed topics and off-line integration with minimal capture constraints. 
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




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7th International Conference Cities & Ports
7e Conference Internationale Villes & Ports
6-9 november 2000
Marseille (France)
=3D=3D=3D=3D=3D
to rem-conf@es.net

Urban and port development authorities for coastal and river cities from
throughout the entire world will be meeting in Marseilles (France), from
the 6th to the 9th of November next, for the 7th Cities and Ports (IACP)
International Conference. More than 500 delegates representing the port
cities of more than 50 countries are expected in Marseilles to debate and
exchange views regarding the implementation of sustainable development in
port areas. This challenge is of interest to you ? This meeting concerns
you ? We propose, without any committment on your behalf, to keep you
informed free of charge of the programme development of this conference

> by sending a message to :
listegb.conference@aivp.net with SUBSCRIBE as subject
(mailto:listegb.conference@aivp.net?subject=3DSUBSCRIBE)

>or via our website
http://www.aivp.com/7conf/formgb.asp


=3D=3D=3D=3D=3D


Les responsables du d=E9veloppement urbain et portuaire des villes maritimes
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From rem-conf Wed Apr 12 11:19:43 2000 
From rem-conf-request@es.net Wed Apr 12 11:19:42 2000
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Date: Thu, 16 Mar 2000 09:45:42 +0900
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: rem-conf@es.net
CC: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Subject: RE: Result of MPEG-4 Visual RTP Interoperability Test
Message-ID: <015a01bf8ee0$f5833b20$cc51c485@kaiji.eel.rdc.toshiba.co.jp>
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------=_NextPart_000_0157_01BF8F2C.652F60C0
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Dear IETF AVT experts,

I'm attaching the contribution document to the MPEG meeting regarding
the MPEG-4 Visual RTP interoperability test for those who don't have
access to the MPEG ftp site.

Best regards,
Yoshihiro

-----Original Message-----
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>;
rem-conf@es.net <rem-conf@es.net>
Cc: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>; Yoshinori MATSUI
<matsui@drl.mei.co.jp>; Toshiyuki Nomura <t-nomura@ccm.cl.nec.co.jp>;
Toshiro Kawahara <kawahara@spg.yrp.nttdocomo.co.jp>; Shigeru Fukunaga
<fukunaga444@oki.co.jp>; Hidenobu Harasaki <harasaki@ccm.cl.nec.co.jp>;
Hideaki Kimata <kimata@nttvdt.hil.ntt.co.jp>; Yoshihiro Miyamoto
<miyamoto@dsp.cl.nec.co.jp>
Date: Wednesday, March 15, 2000 10:43 AM
Subject: Result of MPEG-4 Visual RTP Interoperability Test


>Dear MPEG&IETF RTP experts,
>
>I have upload the following contribution to the MPEG Noordwijkerhout
>meeting to the MPEG ftp site.
>
>[m5799]
>Title: Preliminary Result of MPEG-4 Visual RTP Interoperability Test
>Source: Toshiba, Matsushita, NEC, Oki
>
>In this contribution, we describe the result of the interoperability
>test for the MPEG-4 Audio/Visual RTP payload format.
>(draft-ietf-avt-rtp-mpeg4-es-00) The test was conducted by four
>companies (Matsushita, NEC, Oki and Toshiba) using MPEG-4 Visual Simple
>Profile as well as Core Profile (arbitrary shaped object). All steps of
>the interoperability test, including video and RTP stream exchanges and
>the final step with network connection, were finished successfully.
This
>verifies the maturity and interoperability of the MPEG-4 Visual RTP
>payload format.
>
>We would like to do some demonstrations in Noordwijkerhout (and
>Adelaide, if possible) to show this interoperability test result.
>
>Best regards,
>Yoshihiro
>
>----
>        Yoshihiro Kikuchi
>
>E-mail: kiku@eel.rdc.toshiba.co.jp
>Corporate Research and Development Center
>TOSHIBA Corporation
>TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
>
>
>
>
>


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From rem-conf Wed Apr 12 11:19:45 2000 
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Date: Fri, 17 Mar 2000 12:29:55 +0900
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: Paul Christ <christ@rus.uni-stuttgart.de>
CC: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>, rem-conf@es.net,
  Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Subject: RE: RE:RE: 2 Adelaide drafts ahead
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Dear Paul and all,

>>>   The decoding of
>>>   arbitrary shape VOP requires the dimensions of its bounding
>>>   rectangle, its horizontal and vertical spatial position, as well
as
>
>>>   the shape coding type.  This information is respectively provided
>>>   by the parameters "vop_width", "vop_height",
>>>   "vop_horizontal_mc_spatial_ref" and "vop_vertical_mc_spatial_ref",
>>>   "vop_shape_coding_type" in the VOP header.  The parameter
>>>   "change_conv_ratio_disable" is also needed to be able to decode
>>>   properly the video packet.  The "vop_constant_alpha parameter" as
>>>   well as the "vop_constant_alpha_value" (if vop_constant_alpha==1),
>>>  scaling factor applied to the decoded VOP before display need also
>>>   protection.
>
>> The issue on the protection of the VOP header fields for the
arbitrary
>
>> shaped objects was discussed one year ago(!) at the MPEG Seoul
>meeting.
>
>We have checked the output documents of the MPEG Seoul meeting (March
>99), and
>could not find any document addressing the above issues.
>
Which document did you check?
You can find the revised HEC syntax for the arbitrary shape at item 1 in
the WD1 of the corrigendum(N2693). This syntax has been adopted in the
ISO/IEC 14496-2/DCOR1(N2919). You can find the syntax in item 12 of the
DCOR1.

>We have been raising these points since november 98 (IETF in Orlando),
>we have raised them on the 4onIP list, many times during the year 1999,
>but these points have never been acknowledged.
>
>Yoshihiro, you were yourself, on the 6th of March 99,
>writing on the 4onIP list the following mail
>arguing in the same direction, and recognizing the
>limitations of the HEC mechanism for profiles other than the simple
>profile:
>

(snip)

Yes, I surely wrote this mail. If I remember correctly, there were
suggestions by Katto-san and Carsten to revised the HEC syntax to
support the shape coding.  I posted a contribution to the MPEG Seoul
meeting regarding this issue(m4480). The issue was discussed at the
meeting, and the conclusion was to revise the HEC syntax as suggested
and adopt it in the Corrigendum of MPEG-4 Visual so that this HEC
functionality can be used for all MPEG-4 Visual compliant codecs and all
kind of networks. So the problem I raised in my very old mail has been
solved. This is what I wrote in my previous mail.

>We have looked for the 14496-2/DCOR1 corrigendum and finally found this
>document as an output document of the MELBOURNE meeting in OCTOBER
>1999 (and not as an output from the Seoul meeting in march 99)..
>This corrigendum has apparently been prepared at the same time as the
>draft-jnb-mpeg4av-rtp-00.txt, and been discussed in October 1999 in the
>MPEG4 video group.
>
No. The corrigendum of 14496-2 was initiated at the March 1999 MPEG
meeting more than half year before Melbourne, and WD1 of the corrigendum
including the revised HEC syntax was published in March as I wrote
above. It has been integrated in the reference software (ISO/IEC
14496-5) before the MPEG July 1999 meeting. It is also before Melbourne.

>There were two ways for correcting these limitations,
>either by modifying the
>visual syntax, either by proposing to add protection in the
>encapsulation process.
>You apparently proposed the first solution, while we were and are still
>defending the second approach.  The second proposal being
>more flexible in the case of pre-encoded streams for
>streaming applications.
>
>Given all the discussions that had taken place between
>the MPEG-4 and the IETF
>on these questions between november 98 and october 99, we would have
>expected to post this corrigendum on the 4onIp reflector.

If I was not so kind enough to post the WD of the corrigendum, I'm sorry
for it. But I'm not sure if I, just as one member of MPEG, can post the
official output document from MPEG to the open reflector.

>At least, this would have allowed to discuss the pro's
>and con's on the two possible ways to solve the problem,
>and to find in a consensual way the
>best solution.
>
I understand there were discussion on the reflector before March as I
wrote above. And
there is a conclusion.

>The lack of notion of video packet and the lack of protection of the
>essential parameters for the enhancement layers in the
>context of scalable  coding is a problem.
>This means not being able to use the scalable features in
>a context of packet losses (since the enhancement layer is
>unlikely to be decodable, in case of losses),  although one
>of the interests of scalability is its amenability for
>congestion control on IP. It is especially essential for
>congestion and rate control when you have pre-encoded streams
>(streaming applications) or in multicast scenarios.
>
My understanding on the packet loss resiliency of the scalable coding is
that the scaleable coding itself has resiliency in the meaning that we
can use base layer for decoding even when enhancement layer is corrupt.
So, the base layer should be protected well. This is why MPEG-4 error
resiliency tools(video packet and others) are supported in the base
layer. But the resiliency required for the enhancement layer is
relatively low.
I remember that there were such discussions in MPEG before defining
Table G-1.

>So, with respect to this last point, it comes again to
>several possible solutions:
>One additional corrigendum to the video syntax, for the
>enhancement layers, or/and relying on an additional mechanism
>in the encapsulation process, which we again support.
>
I have to ask other MPEG-4 video experts, but is this corrigendum really
possible? It requires not only syntax changes, but also changes on the
video decoding process. All video coding operation using prediction from
the surrounding macroblocks, such as motion vector prediction and AC/DC
prediction, will be affected by this.

>> The configuration information is NOT restricted to a few bytes. It is
>> about 30 to 40 bytes for the typical case of the Simple Profile.
>
>30 to 40 bytes are a few bytes, compared to likely MTU sizes.
>Experiments made by people from Berkeley on congestion control
>assume MTU sizes of 1500 bytes.
>
Then, you can put the configuration information with a subsequent VOP
(or video packet(s) in the VOP) into a single RTP packet. The
fragmentation rules in section 3.2 of draft-ietf-avt-rtp-mpeg4-es-00 are
designed to be able to do so.
I hope you read carefully the specifications in the I/D.

Best regards,
Yoshihiro


----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267







From rem-conf Wed Apr 12 11:19:48 2000 
From rem-conf-request@es.net Wed Apr 12 11:19:47 2000
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Date: Thu, 16 Mar 2000 19:32:12 +0900
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: Paul Christ <christ@rus.uni-stuttgart.de>, rem-conf@es.net
CC: Angela Giannitrapani <giannitrapani@rus.uni-stuttgart.de>,
  Bilal EL ALI <Bilal.El_Ali@irisa.fr>,
  Christine Marie-Claude Guillemot <christine.guillemot@irisa.fr>,
  Stefan Wesner <wesner@rus.uni-stuttgart.de>,
  4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: 2 Adelaide drafts ahead
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Dear Paul and all,

>preparing the AVT group meeting in Adelaide and
>pursuing now our MPEG-4/RTP activities within
>the framework of the European IST project SoNG
>we would like to draw your attention to
>
>1- the fact that our draft
>
>"RTP Payload Format for MPEG-4 with
>Scaleable & Flexible Error Resiliency",
>(so-called 'GP' in short)
>
>is under revision.  Listening somewhat to the
>"complexity argument against us", we simplified a bit
>the existing draft (we dropped the fragmentation
>capability).
>
>2- the fact that a new draft
>
>- "RTP payload format for MPEG-4 Visual Advanced Profiles"
>is also being produced as a specialization exercise of
>our 'GP' towards visual and also as a "critique" of
>"draft-jnb-mpeg4av-rtp-00.txt ". Covering mostly the
>so-called simple visual profile and omitting a series of
>other important video profiles, we have to disagree
>with its arguments for a simple payload in general.
>

I read your Internet Drafts, but I cannot understand issues you are
trying to raise.

---  draft-gc-avt-mpeg4visual-00.txt ---

   The decoding of
   arbitrary shape VOP requires the dimensions of its bounding
   rectangle, its horizontal and vertical spatial position, as well as
   the shape coding type.  This information is respectively provided
   by the parameters "vop_width", "vop_height",
   "vop_horizontal_mc_spatial_ref" and "vop_vertical_mc_spatial_ref",
   "vop_shape_coding_type" in the VOP header.  The parameter
   "change_conv_ratio_disable" is also needed to be able to decode
   properly the video packet.  The "vop_constant_alpha parameter" as
   well as the "vop_constant_alpha_value" (if vop_constant_alpha==1),
   scaling factor applied to the decoded VOP before display need also
   protection.

The issue on the protection of the VOP header fields for the arbitrary
shaped objects was discussed one year ago(!) at the MPEG Seoul meeting.
The conclusion was to correct the HEC syntax to support the VOP header
for the arbitrary shape, and the corrected syntax has been adopted in
the
Corrigendum of MPEG-4 Visual(14496-2/DCOR1). All necessary information
in the VOP header for the decoding of the arbitrary shaped objects are
supported by the HEC.

   Depending  on  the  different  visual  profiles,  different  sets  of
   parameters will be present in the header of the VideoObjectPlane().
   In  the  simple  profiles,  essential  VOP  header  parameters  are:
   vop_coding_type,  modulo_time_base,  marker_bit,  vop_time_increment,
   fcodes when the VOP_coding_type is P or B, VOP_reduced_resolution if
   reduced_resolution_vop_enable is equal to 1.

You are correct in the meaning that all these information are supported
by HEC. But  VOP_reduced_resolution is NOT for the Simple Profile but
for the ARTS profile.

   Let us now consider the simple scalable profile supporting temporal
   and spatial scalability. Scalable or layered coding is very
   interesting for rate or congestion control on the Internet,
   especially in multicast. A key parameter in order to be able to
   decode a VOP in an enhancement layer is "ref_select_code" which
   signals the VOP that has been taken as a reference for the
   prediction.

   When scalability is applied on arbitrary shape objects, extra
   parameters need to be protected. For the shape decoding these
   parameters are "load_backward_shape", "backward_shape_width",
   "backward_shape_height", "backward_shape_horizontal_mc_spatial_ref",
   "backward_shape_vertical_mc_spatial_ref", "load_forward_shape",
   "forward_shape_width", "forward_shape_height",
   "forward_shape_horizontal_mc_spatial_ref",
   "forward_shape_vertical_mc_spatial_ref".  Another important
   parameter is "background_composition" which signals the usage of
   background composition in conjunction with scalability.

As you can see in Table G-2 in Annex G, any error resilience
tools(including video packet) are NOT supported for the enhancement
layer of the scalability. This is because the information above are not
supported in the HEC.
Even if the VOP header information above are protected by the RTP or
whatever, how you can used it for the error resilient decoding without
video packet? It is impossible to decode a middle part of a VOP without
video packet if the beginning part of the VOP is lost.


   The solution recommended in draft-jnb-mpeg4av-rtp-01.txt, when using
   the  combined  mode,  consists  in  transporting  this  configuration
   information in separate RTP packets, and in possibly repeating the
   corresponding RTP packets periodically if needed for protection
   purposes.  However, this vital information being restricted to a few
   bytes, to transport it in separate RTP packets leads to unnecessary
   overhead.

The configuration information is NOT restricted to a few bytes. It is
about 30 to 40 bytes for the typical case of the Simple Profile. When
the quantization matrix is transported by setting load_intra_quant_mat
and load_nonintra_quant_mat to 1 in Core, Main and other profiles, this
length will go up to about 200 bytes.

   More efficient transport can be reached by grouping this
   data with elementary stream data inside packets.  The same remark
   applies to the Group_of_VideoObjectPlane() entry point and to its
   corresponding header or configuration information.

It is possible to group the configuration information, GOV and the
subsequent VOP into one RTP packet by the fragmentation rule of the
MPEG-4 Visual RTP payload format. (See section 3.2 of
draft-ietf-avt-rtp-mpeg4-es-00.txt) This fragmentation rule is flexible,
and you can select either grouping or no grouping of the configuration
information or GOV adequately, depends on the bitrate, packet-loss rate,
etc.

   Depending on the visual profiles, different sets of key parameters
   will be present in the header of the VideoObjectPlane(), as
   described above.  Key parameters for the simple scalable, main and
   core profiles are not covered by the error resilience tools defined
   in MPEG-4.

I fully disagree with this as I wrote above.

   Five profiles have been defined for natural video content [3]:

Not only five profiles of MPEG-4 Visual Version 1 you raised, HEC
supports all Video coding tools having video packet for natural video
profiles in both Version 1 and Version 2. As you can see in the video
packet header syntax of Version 2 FDAM(14496-2/FDAM1), NEWPRED(ARTS
Profile), DRC(ARTS Profile), GMC(ACE Profile) are also supported by the
HEC.

I should disagree with the last sentence of bellow in
draft-ietf-avt-mpeg4streams-00.txt:

   A set of error resilience tools has been defined in the MPEG-4 vis-
   ual syntax in order to recover corrupted headers [5]. In particular,
   the VideoObjectPlane data is structured in video packets, the entry
   point being defined by the function video_packet_header(), and
   delimited by resync_markers. Basic configuration parameters can be
   inserted in the packet header.  However, this concerns only
   parameters used in the simple visual profile, many parameters
   essential in the simple scalable, main and core profiles are not
   covered by this mechanism [5].


Best regards,
Yoshihiro

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267








From rem-conf Wed Apr 12 11:19:49 2000 
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Date: Wed, 15 Mar 2000 10:38:13 +0900
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>, rem-conf@es.net
CC: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>,
  Yoshinori MATSUI <matsui@drl.mei.co.jp>,
  Toshiyuki Nomura <t-nomura@ccm.cl.nec.co.jp>,
  Toshiro Kawahara <kawahara@spg.yrp.nttdocomo.co.jp>,
  Shigeru Fukunaga <fukunaga444@oki.co.jp>,
  Hidenobu Harasaki <harasaki@ccm.cl.nec.co.jp>,
  Hideaki Kimata <kimata@nttvdt.hil.ntt.co.jp>,
  Yoshihiro Miyamoto <miyamoto@dsp.cl.nec.co.jp>
Subject: Result of MPEG-4 Visual RTP Interoperability Test
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Dear MPEG&IETF RTP experts,

I have upload the following contribution to the MPEG Noordwijkerhout
meeting to the MPEG ftp site.

[m5799]
Title: Preliminary Result of MPEG-4 Visual RTP Interoperability Test
Source: Toshiba, Matsushita, NEC, Oki

In this contribution, we describe the result of the interoperability
test for the MPEG-4 Audio/Visual RTP payload format.
(draft-ietf-avt-rtp-mpeg4-es-00) The test was conducted by four
companies (Matsushita, NEC, Oki and Toshiba) using MPEG-4 Visual Simple
Profile as well as Core Profile (arbitrary shaped object). All steps of
the interoperability test, including video and RTP stream exchanges and
the final step with network connection, were finished successfully. This
verifies the maturity and interoperability of the MPEG-4 Visual RTP
payload format.

We would like to do some demonstrations in Noordwijkerhout (and
Adelaide, if possible) to show this interoperability test result.

Best regards,
Yoshihiro

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267








From rem-conf Wed Apr 12 20:00:42 2000 
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To: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>, <rem-conf@es.net>
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Subject: Re: Result of MPEG-4 Visual RTP Interoperability Test
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Dear all,

I'm surprised to receive some mails I sent in March (!!) again today through
the reflector ;-)  I don't know why this happed, but please ignore these OLD
mails. For your reference, I'm attaching the header of the mail bellow. You
may find a route that the mail delivered accidentally.

----  start of mail header  ---
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Date: Wed, 15 Mar 2000 10:38:13 +0900
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>, rem-conf@es.net
CC: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>,
        Yoshinori MATSUI <matsui@drl.mei.co.jp>,
        Toshiyuki Nomura <t-nomura@ccm.cl.nec.co.jp>,
        Toshiro Kawahara <kawahara@spg.yrp.nttdocomo.co.jp>,
        Shigeru Fukunaga <fukunaga444@oki.co.jp>,
        Hidenobu Harasaki <harasaki@ccm.cl.nec.co.jp>,
        Hideaki Kimata <kimata@nttvdt.hil.ntt.co.jp>,
        Yoshihiro Miyamoto <miyamoto@dsp.cl.nec.co.jp>
Subject: Result of MPEG-4 Visual RTP Interoperability Test
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Best regards,
Yoshihiro


----- Original Message -----
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: <rem-conf@es.net>
Cc: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Sent: Thursday, March 16, 2000 9:45 AM
Subject: RE: Result of MPEG-4 Visual RTP Interoperability Test


> Dear IETF AVT experts,
>
> I'm attaching the contribution document to the MPEG meeting regarding
> the MPEG-4 Visual RTP interoperability test for those who don't have
> access to the MPEG ftp site.
>
> Best regards,
> Yoshihiro
>
> -----Original Message-----
> From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
> To: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>;
> rem-conf@es.net <rem-conf@es.net>
> Cc: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>; Yoshinori MATSUI
> <matsui@drl.mei.co.jp>; Toshiyuki Nomura <t-nomura@ccm.cl.nec.co.jp>;
> Toshiro Kawahara <kawahara@spg.yrp.nttdocomo.co.jp>; Shigeru Fukunaga
> <fukunaga444@oki.co.jp>; Hidenobu Harasaki <harasaki@ccm.cl.nec.co.jp>;
> Hideaki Kimata <kimata@nttvdt.hil.ntt.co.jp>; Yoshihiro Miyamoto
> <miyamoto@dsp.cl.nec.co.jp>
> Date: Wednesday, March 15, 2000 10:43 AM
> Subject: Result of MPEG-4 Visual RTP Interoperability Test
>
>
> >Dear MPEG&IETF RTP experts,
> >
> >I have upload the following contribution to the MPEG Noordwijkerhout
> >meeting to the MPEG ftp site.
> >
> >[m5799]
> >Title: Preliminary Result of MPEG-4 Visual RTP Interoperability Test
> >Source: Toshiba, Matsushita, NEC, Oki
> >
> >In this contribution, we describe the result of the interoperability
> >test for the MPEG-4 Audio/Visual RTP payload format.
> >(draft-ietf-avt-rtp-mpeg4-es-00) The test was conducted by four
> >companies (Matsushita, NEC, Oki and Toshiba) using MPEG-4 Visual Simple
> >Profile as well as Core Profile (arbitrary shaped object). All steps of
> >the interoperability test, including video and RTP stream exchanges and
> >the final step with network connection, were finished successfully.
> This
> >verifies the maturity and interoperability of the MPEG-4 Visual RTP
> >payload format.
> >
> >We would like to do some demonstrations in Noordwijkerhout (and
> >Adelaide, if possible) to show this interoperability test result.
> >
> >Best regards,
> >Yoshihiro
> >
> >----
> >        Yoshihiro Kikuchi
> >
> >E-mail: kiku@eel.rdc.toshiba.co.jp
> >Corporate Research and Development Center
> >TOSHIBA Corporation
> >TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
> >
> >
> >
> >
> >
>
>




From rem-conf Thu Apr 13 10:52:13 2000 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 4/19 Multicast Debugging -- Bill Fenner, AT&T Labs - Research
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Berkeley Multimedia, Graphics and Interfaces Seminar

Multicast Debugging

Wednesday April 19, 2000
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

Bill Fenner 
AT&T Labs - Research 

Network topologies can be complex, and the application and control traffic
can create a 'chemistry' of loads and flows that leads to unanticipated
network congestion and application failure. This session will provide
network engineers valuable information on debugging congestion,
misconfiguration, application failure to help keep networks and content
delivery reliable.
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Thu Apr 13 11:54:17 2000 
From rem-conf-request@es.net Thu Apr 13 11:54:16 2000
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Date: Thu, 13 Apr 2000 14:41:35 -0400 (EDT)
From: "R. P. Channing Rodgers, M.D." <rodgers@nlm.nih.gov>
Message-Id: <200004131841.OAA17976@billings.nlm.nih.gov>
To: mbone@isi.edu, rem-conf@es.net
Subject: Re: Solution to SunVideoPlus 1.3, SunForum, and UCL vic
Cc: aronson@billings.nlm.nih.gov, K.Hasler@cs.ucl.ac.uk, O.Hodson@cs.ucl.ac.uk,
        rodgers@billings.nlm.nih.gov
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Dear Colleagues,

This is a quick summary of the difficulties we had getting the new
SunVideo Plus 1.3 drivers from Sun to work, and the solution.

First off, a huge thanks to Michael Speers at Sun, who got in touch
with me based on my earlier postings to this list.  It's so refreshing
to see someone who really knows a product monitoring the right mail
lists, and participating in solving problems.

Background: we are using an UltraSPARC 60 under Solaris 2.6 with a
SunVideoPlus PCI card in it.  The card was in the third-from-bottommost
slot of the machine.  We were originally using device drivers from:
ftp://www.mmac.com/, as many colleagues on these lists had, like us,
experienced problems with Sun-supplied drivers.  With this setup we
successfully used UCL's vic.xil for video conferencing for ~18 months.
We recently installed sunforum 3, and found that it required Sun's
new drivers, SunvideoPlus 1.3, to do video conferencing.  We uninstalled
our non-Sun drivers, installed the SunvideoPlus 1.3 drivers, and found
that neither vic nor sunforum now worked.

Solution: we had to move the PCI card to the bottommost slot of the 4
PCI slots on the machine, disinstall the driver package (SUNWo1kpd),
remove all references to lines referencing devices *o1k* in
/etc/path_to_inst, do a boot -r, and re-install the SUNWo1kpd package.
The test program /opt/SUNWo1kp/bin/xil_display, vic, and sunforum are
all now working beautifully.

I append detailed installation notes for anyone who may find them
helpful.  I am amazed that I have not heard of others bumping into this
problem.  No where in any of the hardware or software literature I have
in print, or on the docs.sun.com site, do I find mention of the need
to install the card in a specific slot.

Cheerio, Rick Rodgers

----------
X-Sun-Data-Type: readme-file
X-Sun-Data-Description: readme-file
X-Sun-Data-Name: README.LOCAL
X-Sun-Charset: us-ascii
X-Sun-Content-Lines: 1107

LOCAL INSTALLATION NOTES

Package Name: SunVideoPlus
Version: 1.3
Release Date: 15 September 1999
Origin: http://www.sun.com/desktop/products/graphics/sunvideoplus_download.html
http://www.sun.com/desktop/products/graphics/SunVideoPlus_Download.html
Author: Sun Microsystems, Inc.
Language:
Software Description:
Software drivers and documentation for PCI-based SunVideoPlus video capture card
Installer: R. P. C. Rodgers
Installation Date: 6-13 April 2000
Installation Location: Bethesda MD
Installation Host: billings.nlm.nih.gov
Installation Hardware: SunOS billings 5.6 Generic_105181-15 sun4u sparc SUNW,Ultra-60
Compiler:
Installation OS: Solaris 2.6
Host-specific installation:
User-specific installation:
Software used by this package for operation:
Software used by this package for installation:
Software depending on this package for operation:
Software depending on this package for installation:
Environment variables:
Notes:
Received very helpful assistance from Mike Speer (Michael.Speer@Eng.Sun.COM,
(650)786-6368), as much of what one needed to do to get things working was,
as with earlier SunVideoPlus release, not documented with the materials
supplied by Sun.  IMPORTANT: we experienced serious problems geting the card
to work, because it was supposed to be in the bottommost slot on the Ultra 60
rather than in the third-from-bottom slot (even though we had the card working
for 18 months in that slot, using the non-Sun MMAC* drivers and vic.xil).
The Ultra-60 hardware guide leads one to think that all 4 PCI slots are
equivalent 33 MHz slots, except for the topmost which is 33/66 MHz.  There is
no mention of slot positions in the installation manual for SunVideoPlus 1.3,
which is not hardware-concerned.  Nor is it likely to be stated in the
original hardware booklet (which I can not find; I think our hardware people
failed to pass it along).  It is not mentioned in the online version of the
hardware manual at: http://docs.sun.com:80/ab2/coll.188.2/UGSUNVIDEOPLUS/@Ab2PageView/749?DwebQuery=sunvideo&Ab2Lang=C&Ab2Enc=iso-8859-1
--------------------------------------------------------------------------------

 1) Download following files:

       SunVideo_Plus_1.3_Install_Guide.ps
       SunVideo_Plus_1.3_Install_Guide.pdf
       svp_1.3_fcs.tar.gz

 2) Become user root.

 3) Unpack main distribution file:

       gunzip svp_1.3_fcs.tar.gz
       tar xvf svp_1.3_fcs.tar
       [creates many directories/files]
       rm svp_1.3_fcs.tar

 4) Convert installation guide (./SunVideo_Plus_1.3_Install_Guide.ps) to PDF
    (./SunVideo_Plus_1.3_Install_Guide.pdf).  Convert User's Guide
    (Docs/C/sunvideoplusUG.ps) to PDF (./sunvideoplusUG.pdf).

 5) Remove previous versions of SunVideoPlus software, if installed.  If one
    were using the Sun-supplied drivers, this would involve removing
    SUNWvtsvp, SUNWsvpab, SUNWo1kpu, and SUNWo1kpd (for SunVideoPlus 1.0), or
    SUNWvtsvp, SUNWsvpab, SUNWo1kpx, SUNWo1kpu, and SUNWo1kpd (for 1.1, 1.2).
    However, we used an alternative set of device drivers, supplied by Osprey
    (actual maker of the SunVideoPlus card) which had been found to work with
    the UCL MBONE videoconferencing tool, vic.xil:

       MMACo1kp Osprey-1500 Audio/Video Codec PCI Card Device Driver. 
       MMACo1ku Osprey-1x00 Audio/Video Codec Runtime Support Software
       MMACo1kx Osprey-1x00 Audio/Video Codec XIL 1.3 Runtime Support Software

    They are removed by:

       pkgrm MMACo1kp MMACo1ku MMACo1kx

    answering yes to all queries.
    [transcript appended]

 6) Install the common SunVideo_Plus 1.3 packages (SUNWo1kpu, SUNWopi,
    SUNWopi1k, SUNWsvpab, and SUNWvtsvp):

       cd  Common/Packages
       pkgadd -d .

    choosing "all" and then answering yes to all queries, and select "heavy"
    documentation options.  When the installation of package  begins, the
    installation process will ask you to specify the parent directory for the
    AnswerBook2 Collection for this package.  Specify the following path:

       /opt

    This installs:

       SUNWo1kpu  SunVideo Plus Runtime Support Software (sparc) 1.3,REV=1.3.1
       SUNWopi    OPI Runtime Library Package (sparc) 2.2,REV=2.2.0
       SUNWopi1k  SunVideoPlus OPI Runtime Package (sparc) 2.2,REV=2.2.0
       SUNWsvpab  SunVideo Plus Collection (all) 188.2.4
       SUNWvtsvp  SunVTS Sun Video Plus test module

    [transcript appended]

 7) Install the Solaris 2.6-specific SunVideo_Plus 1.3 packages (SUNWo1kpd.u,
    SUNWo1kpx):

       cd  ../../Solaris_2.6/Packages
       pkgadd -d .

    choosing "all" and then answering yes to all queries.
    This installs:

       SUNWo1kpd.u SunVideo Plus PCI Card Dev Driver (sparc.sun4u) 1.3,REV=2.3.1
       SUNWo1kpx   SunVideo Plus XIL-1.3 Runtime Support Software

    [transcript appended]

 8) Reboot the system:

       sync; sync
       reboot -- -r

 9) On depot server: create depot component of SunVideoPlus 1.3:

       mkdir /depot/package/SunVideoPlus_1.3
       mkdir /depot/package/SunVideoPlus_1.3/pdf
       cp ./SunVideo_Plus_1.3_Install_Guide.pdf \
          /depot/package/SunVideoPlus_1.3/pdf
       chown root:root \
         /depot/package/SunVideoPlus_1.3/pdf/SunVideo_Plus_1.3_Install_Guide.pdf
       chmod 644 \
         /depot/package/SunVideoPlus_1.3/pdf/SunVideo_Plus_1.3_Install_Guide.pdf
       cp ./sunvideoplusUG.pdf /depot/package/SunVideoPlus_1.3/pdf
       chown root:root /depot/package/SunVideoPlus_1.3/pdf/sunvideoplusUG.pdf
       chmod 644 /depot/package/SunVideoPlus_1.3/pdf/sunvideoplusUG.pdf
       cp ./README.LOCAL /depot/package/SunVideoPlus_1.3
       chown root:root /depot/package/SunVideoPlus_1.3/README.LOCAL
       chmod 644 /depot/package/SunVideoPlus_1.3/README.LOCAL

    [this depot package contains only documentation]

10) Complete depot installation on depot server:

       mv /depot/package/SunVideoPlus_1.3 /depot_server/sparc/SunOS5.6/package
       [or appropriate equivalent]
       remove older versions of SunVideoPlus from depot_server directory,
       or place them in the /depot/.exclude files on each client.

11) Note that the locally written shell wrapper for vic must be edited to
    support this new driver.  See file ./vic-2.8ucl3-xil-svpdriver.wrapper.

12) Complete depot installation on each client:

       depot_init -v (or depot_update_client -v or appropriate equivalent)

13) Test; first, note initial value of LD_LIBRARY_PATH:

       LD_LIBRARY_PATH=/usr/openwin/lib:/usr/dt/lib:/depot/lib

    also note that there is a file:

       /devices/pci@1f,4000/pci1061,3@5:o1k0

    now, set environment for SunVideo Plus and XIL:

       setenv O1KHOME /opt/SUNWo1kp
       setenv XILHOME /opt/SUNWits/Graphics-sw/xil
       setenv LD_LIBRARY_PATH /opt/SUNWits/Graphics-sw/xil/lib:/opt/SUNWo1kp/lib:$LD_LIBRARY_PATH

    Finally, connect a video source to the card and run the test program:

       /opt/SUNWo1kp/bin/xil_display

    which results in a window popping up, a brief display, then the message:

       99 frames in 3.26783 seconds, 30.2953 frames/second

    Test further by running vic and sunforum if available.

--------------------------------------------------------------------------------
Special addendum on installation problems:

Note that the above instructions apply to the situation that would have
obtained had the card been properly installed in the bottommost of 4 PCI slots
on the UltraSPARC 60.

Initially, when the SunVideo Plus PCI card was in the wrong slot, the following
occurred when the "/opt/SUNWo1kp/bin/xil_display" test was run:

   error opening /dev/o1k0 [0]
   XilDefaultErrorFunc:
      error category: XIL_ERROR_OTHER
        error string: MMACo1k-14
            error id: MMACo1k-14
          is warning: FALSE
      primary error detected in file O1kDevice.cc, line 299 in XIL
   XilDefaultErrorFunc:
      error category: XIL_ERROR_SYSTEM
        error string: MMACo1k-1
            error id: MMACo1k-1
          is warning: FALSE
      primary error detected in file XilDeviceIOo1k.cc, line 279 in XIL
   XilDefaultErrorFunc:
      error category: XIL_ERROR_RESOURCE
        error string: Out of memory
            error id: di-1
          is warning: FALSE
      primary error detected in file XilDeviceManagerIOo1k.cc, line 70 in XIL
   XilDefaultErrorFunc:
      error category: XIL_ERROR_SYSTEM
        error string: Could not create input/output device
            error id: di-149
          is warning: FALSE
      primary error detected in file XilSystemState.cc, line 1530 in XIL
   failed to open MMACo1k device

To remedy this situation, we had to to do the following (as root):

1) Power down machine and move card to bottommost slot

2) Reapply power, and when UNIX is again running, remove driver package:

      pkgrm SUNWo1kpd

3) Reboot, rebulding devices:

      sync; reboot -r

4) Examine /etc/path_to_inst for lines of form:

      "/pci@1f,4000/pci1061,3@2" 0 "o1k"
      "/pci@1f,4000/pci1061,3@5" 1 "o1k"

   If such lines are present (they were) remove them.
   sync; boot -r

5) Re-install the device driver:

      cd  ./Solaris_2.6/Packages
      pkgadd -d .  [adding only: SUNWo1kpd]

5) Examine /etc/path_to_inst; there should be a single entry of the form:

      "/pci@1f,4000/pci1061,3@5" 0 "o1k"

6) Tests of /opt/SUNWo1kp/bin/xil_display, vic, and sunforum should now work.


--------------------------------------------------------------------------------
Output from "pkgrm MMACo1kp MMACo1ku MMACo1kx" command:

bil[root]csh:73>pkgrm MMACo1kp MMACo1ku MMACo1kx

The following package is currently installed:
   MMACo1kp        Osprey-1500 Audio/Video Codec PCI Card Device Driver.
                   (sparc.sun4u) 1.2,REV=2.2.1

Do you want to remove this package? y

## Removing installed package instance <MMACo1kp>

This package contains scripts which will be executed with super-user
permission during the process of removing this package.

Do you want to continue with the removal of this package [y,n,?,q] y
## Verifying package dependencies.
## Processing package information.
## Removing pathnames in class <devlinktab>
## Removing pathnames in class <none>
/opt/MMACo1k/osprey/vscale.ini
/opt/MMACo1k/osprey <shared pathname not removed>
/opt/MMACo1k <shared pathname not removed>
/opt <shared pathname not removed>
/kernel/drv/o1k.conf
/kernel/drv/o1k
/kernel/drv <shared pathname not removed>
/kernel <shared pathname not removed>
/etc/rc2.d/S93o1k-config
/etc/rc2.d <shared pathname not removed>
/etc/opt/MMACo1k/dbm/o1kctl_t7
/etc/opt/MMACo1k/dbm/o1kctl_t6
/etc/opt/MMACo1k/dbm/o1kctl_t5
/etc/opt/MMACo1k/dbm/o1kctl_t4
/etc/opt/MMACo1k/dbm/o1kctl_t3
/etc/opt/MMACo1k/dbm/o1kctl_t2
/etc/opt/MMACo1k/dbm/o1kctl_t1
/etc/opt/MMACo1k/dbm/o1kctl_t0
/etc/opt/MMACo1k/dbm/o1kctl_7
/etc/opt/MMACo1k/dbm/o1kctl_6
/etc/opt/MMACo1k/dbm/o1kctl_5
/etc/opt/MMACo1k/dbm/o1kctl_4
/etc/opt/MMACo1k/dbm/o1kctl_3
/etc/opt/MMACo1k/dbm/o1kctl_2
/etc/opt/MMACo1k/dbm/o1kctl_1
/etc/opt/MMACo1k/dbm/o1kctl_0
/etc/opt/MMACo1k/dbm
/etc/opt/MMACo1k
/etc/opt <shared pathname not removed>
/etc/init.d/o1k-config
/etc/init.d <shared pathname not removed>
/etc <shared pathname not removed>
## Executing postremove script.
## Updating system information.

Removal of <MMACo1kp> was successful.

The following package is currently installed:
   MMACo1ku        Osprey-1x00 Audio/Video Codec Runtime Support Software
                   (sparc) 1.2,REV=1.2.1

Do you want to remove this package? y

## Removing installed package instance <MMACo1ku>

This package contains scripts which will be executed with super-user
permission during the process of removing this package.

Do you want to continue with the removal of this package [y,n,?,q] y
## Verifying package dependencies.
## Processing package information.
## Executing preremove script.
/dev/rtvc[0-9]: No such file or directory
## Removing pathnames in class <compute>
## Removing pathnames in class <none>
/opt/MMACo1k/osprey/yyremap
/opt/MMACo1k/osprey/vtest.ini
/opt/MMACo1k/osprey/vtest
/opt/MMACo1k/osprey/vscale
/opt/MMACo1k/osprey/vph261.010
/opt/MMACo1k/osprey/vpcodep.2
/opt/MMACo1k/osprey/vpcode.jpg
/opt/MMACo1k/osprey/vpcode.2
/opt/MMACo1k/osprey/prv.263
/opt/MMACo1k/osprey/mpegenc.sun
/opt/MMACo1k/osprey/mpeg.ini
/opt/MMACo1k/osprey/monitor
/opt/MMACo1k/osprey/jpeg.ini
/opt/MMACo1k/osprey/jpeg
/opt/MMACo1k/osprey/hello.ini
/opt/MMACo1k/osprey/hello
/opt/MMACo1k/osprey/h26x.ini
/opt/MMACo1k/osprey/h26x
/opt/MMACo1k/osprey/h263p.ini
/opt/MMACo1k/osprey/h263p
/opt/MMACo1k/osprey/h261.ini
/opt/MMACo1k/osprey/h261
/opt/MMACo1k/osprey/g2.ini
/opt/MMACo1k/osprey/fpassf
/opt/MMACo1k/osprey/fpass.ini
/opt/MMACo1k/osprey/fpass
/opt/MMACo1k/osprey/evpcode.2
/opt/MMACo1k/osprey/dsp723.exe
/opt/MMACo1k/osprey/dsp.exe
/opt/MMACo1k/osprey/diag.ini
/opt/MMACo1k/osprey/diag
/opt/MMACo1k/osprey/cellb.ini
/opt/MMACo1k/osprey/cellb
/opt/MMACo1k/osprey
/opt/MMACo1k/lib/libotiaudio.so.1
/opt/MMACo1k/lib/libotiaudio.so
/opt/MMACo1k/lib/libo1kusr.so.1
/opt/MMACo1k/lib/libo1kusr.so
/opt/MMACo1k/lib
/opt/MMACo1k/diag/o1k_trace
/opt/MMACo1k/diag/o1k_dram
/opt/MMACo1k/diag/o1k_diags
/opt/MMACo1k/diag/o1k_db_reset
/opt/MMACo1k/diag
/opt/MMACo1k/bin/xilh_video_receiver
/opt/MMACo1k/bin/xilh_video_broadcast
/opt/MMACo1k/bin/xil_video_receiver
/opt/MMACo1k/bin/xil_video_broadcast
/opt/MMACo1k/bin/xil_record
/opt/MMACo1k/bin/xil_play
/opt/MMACo1k/bin/xil_grab
/opt/MMACo1k/bin/xil_display
/opt/MMACo1k/bin/xil_decompress
/opt/MMACo1k/bin/xil_compress
/opt/MMACo1k/bin/svc_uninstall_xil1.3
/opt/MMACo1k/bin/svc_install_xil1.3
/opt/MMACo1k/bin/svc_devices
/opt/MMACo1k/bin/soundtool
/opt/MMACo1k/bin/o1k_ctl
/opt/MMACo1k/bin/o1k_conf
/opt/MMACo1k/bin/o1k_audrec
/opt/MMACo1k/bin/o1k_audplay
/opt/MMACo1k/bin/o1k_audloop
/opt/MMACo1k/bin/o1k_audhdr
/opt/MMACo1k/bin/SunVideoPlus
/opt/MMACo1k/bin
/opt/MMACo1k
## Updating system information.

Removal of <MMACo1ku> was successful.

The following package is currently installed:
   MMACo1kx        Osprey-1x00 Audio/Video Codec XIL 1.3 Runtime Support Software
                   (sparc) 5.6,REV=1.2.1

Do you want to remove this package? y

## Removing installed package instance <MMACo1kx>

This package contains scripts which will be executed with super-user
permission during the process of removing this package.

Do you want to continue with the removal of this package [y,n,?,q] y
## Verifying package dependencies.
## Processing package information.
## Executing preremove script.
## Removing pathnames in class <OWconfig>
## Removing pathnames in class <none>
/usr/openwin/lib/xil/locale/C/LC_MESSAGES/xilIO_MMACo1k.mo
/usr/openwin/lib/xil/locale/C/LC_MESSAGES <shared pathname not removed>
/usr/openwin/lib/xil/locale/C <shared pathname not removed>
/usr/openwin/lib/xil/locale <shared pathname not removed>
/usr/openwin/lib/xil/devhandlers/xilIO_MMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_UYVYMMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_Mpeg1MMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_JpegMMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_H261MMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_CellBMMACo1k.so.2
/usr/openwin/lib/xil/devhandlers <shared pathname not removed>
/usr/openwin/lib/xil <shared pathname not removed>
/usr/openwin/lib <shared pathname not removed>
/usr/openwin <shared pathname not removed>
## Executing postremove script.
## Updating system information.

Removal of <MMACo1kx> was successful.

--------------------------------------------------------------------------------
Output from "cd  Common/Packages; pkgadd -d ." command:

bil[root]csh:61>pkgadd -d .

The following packages are available:
  1  SUNWo1kpu     SunVideo Plus Runtime Support Software
                   (sparc) 1.3,REV=1.3.1
  2  SUNWopi       OPI Runtime Library Package
                   (sparc) 2.2,REV=2.2.0
  3  SUNWopi1k     SunVideoPlus OPI Runtime Package
                   (sparc) 2.2,REV=2.2.0
  4  SUNWsvpab     SunVideo Plus Collection
                   (all) 188.2.4
  5  SUNWvtsvp     SunVTS Sun Video Plus test module
                   (sparc) 2.1.1,REV=1999.09.17

Select package(s) you wish to process (or 'all' to process
all packages). (default: all) [?,??,q]: 

Processing package instance <SUNWo1kpu> from </home/rodgers/SunVideoPlus_1.3.dir/Common/Packages>

SunVideo Plus Runtime Support Software
(sparc) 1.3,REV=1.3.1
Copyright 1999 Sun Microsystems, Inc. All rights reserved.
Copyright (c) 1999 MultiMedia Access Corporation - Osprey Technologies Inc. 

All Rights Reserved.

This product and related documentation is protected by copyright laws and 
international treaties. No part of this product or related documentation  
may be reproduced, distributed, or decompiled in any form by any means 
without prior written authorization of MultiMedia Access Corporation. 
Unauthorized reproduction or distribution of this program, or any part of 
it, may result in severe civil and criminal penalties, and will be 
prosecuted to the maximum extent possible under the law.

MMAC reserves the right to change any products herein at any time and 
without notice. MMAC does not assume any liability arising from the 
application or use of this product, except as expressly agreed to in 
writing by MMAC. Use and purchase of this product does not convey a license
under any patent rights, copyrights, trademark rights, or any other 
intellectual property rights of MMAC, Sun Microsystems, or others. 

Sun, Sun Microsystems, Solaris, SunOS, OpenWindows, XView, VideoPix and
SunVideo are trademarks or registered trademarks of Sun Microsystems, Inc.
All other product names mentioned herein are the trademarks of their
respective owners.



Using </opt> as the package base directory.
## Processing package information.
## Processing system information.
## Verifying package dependencies.
## Verifying disk space requirements.
## Checking for conflicts with packages already installed.
## Checking for setuid/setgid programs.

Installing SunVideo Plus Runtime Support Software as <SUNWo1kpu>

## Installing part 1 of 1.
/opt/SUNWo1kp/bin/SunVideoPlus
/opt/SUNWo1kp/bin/o1k_audhdr
/opt/SUNWo1kp/bin/o1k_audloop
/opt/SUNWo1kp/bin/o1k_audplay
/opt/SUNWo1kp/bin/o1k_audrec
/opt/SUNWo1kp/bin/o1k_conf
/opt/SUNWo1kp/bin/o1k_ctl
/opt/SUNWo1kp/bin/soundtool
/opt/SUNWo1kp/bin/svc_devices
/opt/SUNWo1kp/bin/svc_install_xil1.3
/opt/SUNWo1kp/bin/svc_uninstall_xil1.3
/opt/SUNWo1kp/bin/xil_compress
/opt/SUNWo1kp/bin/xil_decompress
/opt/SUNWo1kp/bin/xil_display
/opt/SUNWo1kp/bin/xil_video_broadcast
/opt/SUNWo1kp/bin/xil_video_receiver
/opt/SUNWo1kp/bin/xilh_video_broadcast
/opt/SUNWo1kp/bin/xilh_video_receiver
/opt/SUNWo1kp/diag/o1k_db_reset
/opt/SUNWo1kp/diag/o1k_diags
/opt/SUNWo1kp/diag/o1k_dram
/opt/SUNWo1kp/diag/o1k_svts
/opt/SUNWo1kp/diag/o1k_trace
/opt/SUNWo1kp/examples/o1k_audloop/Makefile
/opt/SUNWo1kp/examples/o1k_audloop/audloop.c
/opt/SUNWo1kp/examples/o1k_audplay/Makefile
/opt/SUNWo1kp/examples/o1k_audplay/audplay.c
/opt/SUNWo1kp/examples/o1k_audrec/Makefile
/opt/SUNWo1kp/examples/o1k_audrec/audrec.c
/opt/SUNWo1kp/examples/o1k_conf/Makefile
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/address_button
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/audio_setting
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/avpanel_button
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/call_button
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/hangup_button
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/o1kconf.icon
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/o1kconf.mask
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/record_button
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/setup_button
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/video_setting
/opt/SUNWo1kp/examples/o1k_conf/bitmaps/view_button
/opt/SUNWo1kp/examples/o1k_conf/blank261.h
/opt/SUNWo1kp/examples/o1k_conf/cellb_util.cc
/opt/SUNWo1kp/examples/o1k_conf/cellb_util.h
/opt/SUNWo1kp/examples/o1k_conf/cmap.cc
/opt/SUNWo1kp/examples/o1k_conf/dither.cc
/opt/SUNWo1kp/examples/o1k_conf/dither.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_ab.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_ab.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_audio.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_audio.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_cntrl.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_conf.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_conf.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_conf_stubs.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_conf_ui.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_conf_ui.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_defines.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_misc.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_misc.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_network.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_network.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_thread.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_thread.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_timer.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_timer.h
/opt/SUNWo1kp/examples/o1k_conf/o1k_video.cc
/opt/SUNWo1kp/examples/o1k_conf/o1k_video.h
/opt/SUNWo1kp/examples/o1k_conf/packet.h
/opt/SUNWo1kp/examples/o1k_conf/receive.cc
/opt/SUNWo1kp/examples/o1k_conf/receive.h
/opt/SUNWo1kp/examples/o1k_conf/transmit.cc
/opt/SUNWo1kp/examples/o1k_conf/transmit.h
/opt/SUNWo1kp/examples/o1k_conf/window.cc
/opt/SUNWo1kp/examples/o1k_conf/window_state.h
/opt/SUNWo1kp/examples/soundtool/Makefile
/opt/SUNWo1kp/examples/soundtool/soundtool.c
/opt/SUNWo1kp/examples/soundtool/soundtool.icon
/opt/SUNWo1kp/examples/xil_broadcast/Makefile
/opt/SUNWo1kp/examples/xil_broadcast/cellb_util.c
/opt/SUNWo1kp/examples/xil_broadcast/cellb_util.h
/opt/SUNWo1kp/examples/xil_broadcast/cmap.c
/opt/SUNWo1kp/examples/xil_broadcast/dither.c
/opt/SUNWo1kp/examples/xil_broadcast/dither.h
/opt/SUNWo1kp/examples/xil_broadcast/packet.h
/opt/SUNWo1kp/examples/xil_broadcast/params.h
/opt/SUNWo1kp/examples/xil_broadcast/receive.c
/opt/SUNWo1kp/examples/xil_broadcast/receive.h
/opt/SUNWo1kp/examples/xil_broadcast/rparams.c
/opt/SUNWo1kp/examples/xil_broadcast/tparams.c
/opt/SUNWo1kp/examples/xil_broadcast/transmit.c
/opt/SUNWo1kp/examples/xil_broadcast/transmit.h
/opt/SUNWo1kp/examples/xil_broadcast/window.c
/opt/SUNWo1kp/examples/xil_broadcast/window_state.h
/opt/SUNWo1kp/examples/xil_broadcast/xilbroadcast.c
/opt/SUNWo1kp/examples/xil_broadcast/xilvc.c
/opt/SUNWo1kp/examples/xil_compress/Makefile
/opt/SUNWo1kp/examples/xil_compress/xil_compress.c
/opt/SUNWo1kp/examples/xil_decompress/Makefile
/opt/SUNWo1kp/examples/xil_decompress/memmap.c
/opt/SUNWo1kp/examples/xil_decompress/memmap.h
/opt/SUNWo1kp/examples/xil_decompress/xil_decompress.c
/opt/SUNWo1kp/examples/xil_decompress/xilcis_color.c
/opt/SUNWo1kp/examples/xil_display/Makefile
/opt/SUNWo1kp/examples/xil_display/dcmap.c
/opt/SUNWo1kp/examples/xil_display/ddither.c
/opt/SUNWo1kp/examples/xil_display/ddither.h
/opt/SUNWo1kp/examples/xil_display/display.c
/opt/SUNWo1kp/examples/xilh_broadcast/Makefile
/opt/SUNWo1kp/examples/xilh_broadcast/blank261.h
/opt/SUNWo1kp/examples/xilh_broadcast/cellb_util.c
/opt/SUNWo1kp/examples/xilh_broadcast/cellb_util.h
/opt/SUNWo1kp/examples/xilh_broadcast/cmap.c
/opt/SUNWo1kp/examples/xilh_broadcast/dither.c
/opt/SUNWo1kp/examples/xilh_broadcast/dither.h
/opt/SUNWo1kp/examples/xilh_broadcast/packet.h
/opt/SUNWo1kp/examples/xilh_broadcast/params.h
/opt/SUNWo1kp/examples/xilh_broadcast/receive.c
/opt/SUNWo1kp/examples/xilh_broadcast/receive.h
/opt/SUNWo1kp/examples/xilh_broadcast/rparams.c
/opt/SUNWo1kp/examples/xilh_broadcast/tparams.c
/opt/SUNWo1kp/examples/xilh_broadcast/transmit.c
/opt/SUNWo1kp/examples/xilh_broadcast/transmit.h
/opt/SUNWo1kp/examples/xilh_broadcast/window.c
/opt/SUNWo1kp/examples/xilh_broadcast/window_state.h
/opt/SUNWo1kp/examples/xilh_broadcast/xilbroadcast.c
/opt/SUNWo1kp/examples/xilh_broadcast/xilh261packet.c
/opt/SUNWo1kp/examples/xilh_broadcast/xilvc.c
/opt/SUNWo1kp/include/oti_audio_device.h
/opt/SUNWo1kp/include/oti_libaudio.h
/opt/SUNWo1kp/lib/libo1kusr.so <symbolic link>
/opt/SUNWo1kp/lib/libo1kusr.so.1
/opt/SUNWo1kp/lib/libotiaudio.so <symbolic link>
/opt/SUNWo1kp/lib/libotiaudio.so.1
/opt/SUNWo1kp/osprey/cellb
/opt/SUNWo1kp/osprey/cellb.ini
/opt/SUNWo1kp/osprey/diag
/opt/SUNWo1kp/osprey/diag.ini
/opt/SUNWo1kp/osprey/dsp.exe
/opt/SUNWo1kp/osprey/dsp723.exe
/opt/SUNWo1kp/osprey/evpcode.2
/opt/SUNWo1kp/osprey/fpass
/opt/SUNWo1kp/osprey/fpass.ini
/opt/SUNWo1kp/osprey/fpassf
/opt/SUNWo1kp/osprey/g2.ini
/opt/SUNWo1kp/osprey/h261
/opt/SUNWo1kp/osprey/h261.ini
/opt/SUNWo1kp/osprey/h263p
/opt/SUNWo1kp/osprey/h263p.ini
/opt/SUNWo1kp/osprey/h26x
/opt/SUNWo1kp/osprey/h26x.ini
/opt/SUNWo1kp/osprey/h323
/opt/SUNWo1kp/osprey/hello
/opt/SUNWo1kp/osprey/hello.ini
/opt/SUNWo1kp/osprey/jdec
/opt/SUNWo1kp/osprey/jdec.ini
/opt/SUNWo1kp/osprey/jpeg
/opt/SUNWo1kp/osprey/jpeg.ini
/opt/SUNWo1kp/osprey/monitor
/opt/SUNWo1kp/osprey/mpeg.ini
/opt/SUNWo1kp/osprey/mpegenc.sun
/opt/SUNWo1kp/osprey/prv.263
/opt/SUNWo1kp/osprey/vpcode.2
/opt/SUNWo1kp/osprey/vpcode.jpg
/opt/SUNWo1kp/osprey/vpcodep.2
/opt/SUNWo1kp/osprey/vph261.010
/opt/SUNWo1kp/osprey/vscale
/opt/SUNWo1kp/osprey/vscale.ini
/opt/SUNWo1kp/osprey/vtest
/opt/SUNWo1kp/osprey/vtest.ini
/opt/SUNWo1kp/osprey/yyremap
[ verifying class <none> ]

Installation of <SUNWo1kpu> was successful.

Processing package instance <SUNWopi> from </home/rodgers/SunVideoPlus_1.3.dir/Common/Packages>

OPI Runtime Library Package
(sparc) 2.2,REV=2.2.0
Copyright 1999 Sun Microsystems, Inc. All rights reserved.
Copyright (c) 1999 ViewCast.com - Osprey Technologies Inc. 

All Rights Reserved.

This product and related documentation is protected by copyright laws and 
international treaties. No part of this product or related documentation  
may be reproduced, distributed, or decompiled in any form by any means 
without prior written authorization of ViewCast.com.
Unauthorized reproduction or distribution of this program, or any part of 
it, may result in severe civil and criminal penalties, and will be 
prosecuted to the maximum extent possible under the law.

VCST reserves the right to change any products herein at any time and 
without notice. VCST does not assume any liability arising from the 
application or use of this product, except as expressly agreed to in 
writing by VCST. Use and purchase of this product does not convey a license
under any patent rights, copyrights, trademark rights, or any other 
intellectual property rights of VCST, Sun Microsystems, or others. 

Sun, Sun Microsystems, Solaris, SunOS, OpenWindows, XView, VideoPix and
SunVideo are trademarks or registered trademarks of Sun Microsystems, Inc.
All other product names mentioned herein are the trademarks of their
respective owners.



Using </opt> as the package base directory.
## Processing package information.
## Processing system information.
## Verifying package dependencies.
## Verifying disk space requirements.
## Checking for conflicts with packages already installed.
## Checking for setuid/setgid programs.

This package contains scripts which will be executed with super-user
permission during the process of installing this package.

Do you want to continue with the installation of <SUNWopi> [y,n,?] y

Installing OPI Runtime Library Package as <SUNWopi>

## Installing part 1 of 1.
/opt/SUNWopi/lib/libopi.so <symbolic link>
/opt/SUNWopi/lib/libopi.so.1
/opt/SUNWopi/lib/libopic.so <symbolic link>
/opt/SUNWopi/lib/libopic.so.1
/opt/SUNWopi/lib/libopisw.so <symbolic link>
/opt/SUNWopi/lib/libopisw.so.1
[ verifying class <none> ]
## Executing postinstall script.

Installation of <SUNWopi> was successful.

Processing package instance <SUNWopi1k> from </home/rodgers/SunVideoPlus_1.3.dir/Common/Packages>

SunVideoPlus OPI Runtime Package
(sparc) 2.2,REV=2.2.0
Copyright 1999 Sun Microsystems, Inc. All rights reserved.
Copyright (c) 1999 ViewCast.com - Osprey Technologies Inc. 

All Rights Reserved.

This product and related documentation is protected by copyright laws and 
international treaties. No part of this product or related documentation  
may be reproduced, distributed, or decompiled in any form by any means 
without prior written authorization of ViewCast.com.
Unauthorized reproduction or distribution of this program, or any part of 
it, may result in severe civil and criminal penalties, and will be 
prosecuted to the maximum extent possible under the law.

VCST reserves the right to change any products herein at any time and 
without notice. VCST does not assume any liability arising from the 
application or use of this product, except as expressly agreed to in 
writing by VCST. Use and purchase of this product does not convey a license
under any patent rights, copyrights, trademark rights, or any other 
intellectual property rights of VCST, Sun Microsystems, or others. 

Sun, Sun Microsystems, Solaris, SunOS, OpenWindows, XView, VideoPix and
SunVideo are trademarks or registered trademarks of Sun Microsystems, Inc.
All other product names mentioned herein are the trademarks of their
respective owners.



Using </opt> as the package base directory.
## Processing package information.
## Processing system information.
   3 package pathnames are already properly installed.
## Verifying package dependencies.
## Verifying disk space requirements.
## Checking for conflicts with packages already installed.
## Checking for setuid/setgid programs.

This package contains scripts which will be executed with super-user
permission during the process of installing this package.

Do you want to continue with the installation of <SUNWopi1k> [y,n,?] y

Installing SunVideoPlus OPI Runtime Package as <SUNWopi1k>

## Installing part 1 of 1.
/opt/SUNWopi/lib/libo1kdrw.so <symbolic link>
/opt/SUNWopi/lib/libo1kdrw.so.1
/opt/SUNWopi/lib/libopio1k.so <symbolic link>
/opt/SUNWopi/lib/libopio1k.so.1
[ verifying class <none> ]
## Executing postinstall script.
Performing merge for /opt/SUNWopi/config/opi.devices...

Installation of <SUNWopi1k> was successful.

Processing package instance <SUNWsvpab> from </home/rodgers/SunVideoPlus_1.3.dir/Common/Packages>

SunVideo Plus Collection
(all) 188.2.4
Copyright 1999 Sun Microsystems, Inc. All rights reserved.

The installation options are as follows: 
Option:	Description:
--------------------------------------------
1. nil:    leave data on cdrom, but update AB2 Catalog for cdrom access.
2. heavy:  add the complete package to the system and update the AB2 Catalog.

Enter the number of an installation option from the list above (1 or 2).


Select an installation option: 2
Installation option: heavy selected.

The following request for input asks you to specify
the parent directory of this AnswerBook2 Collection.


Specify the parent path of this AnswerBook2 Collection directory: /opt
Using </opt> as the package base directory.
## Processing package information.
## Processing system information.
## Verifying package dependencies.
## Verifying disk space requirements.
## Checking for conflicts with packages already installed.
## Checking for setuid/setgid programs.

This package contains scripts which will be executed with super-user
permission during the process of installing this package.

Do you want to continue with the installation of <SUNWsvpab> [y,n,?] y

Installing SunVideo Plus Collection as <SUNWsvpab>

## Installing part 1 of 1.
/opt/SUNWsvpab/booklist.txt
/opt/SUNWsvpab/books/SUNVIDEOUG/ebt/SUNVIDEOUG.dat
/opt/SUNWsvpab/books/SUNVIDEOUG/ebt/SUNVIDEOUG.edr
/opt/SUNWsvpab/books/SUNVIDEOUG/ebt/SUNVIDEOUG.tag
/opt/SUNWsvpab/books/SUNVIDEOUG/ebt/search.tdr <symbolic link>
/opt/SUNWsvpab/books/SUNVIDEOUG/ebt/toc.tdr
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/11337.eps.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/address.tif.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/beingCalled.tiff.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/controlPanel.tif.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/graphic1.tif.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/graphic2.tif.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/graphic3.tif.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/graphic4.tif.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/figures/graphic5.tif.gif
/opt/SUNWsvpab/books/SUNVIDEOUG/index/index.dat
/opt/SUNWsvpab/books/SUNVIDEOUG/index/vocab.dat
/opt/SUNWsvpab/books/SUNVIDEOUG/styles.ent
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/ebt/UGSUNVIDEOPLUS.dat
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/ebt/UGSUNVIDEOPLUS.edr
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/ebt/UGSUNVIDEOPLUS.tag
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/ebt/search.tdr <symbolic link>
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/ebt/toc.tdr
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic1.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic10.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic11.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic12.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic13.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic2.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic4.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic5.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic7.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic8.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/figures/graphic9.gif
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/index/index.dat
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/index/vocab.dat
/opt/SUNWsvpab/books/UGSUNVIDEOPLUS/styles.ent
/opt/SUNWsvpab/libidx/books.lst
/opt/SUNWsvpab/libidx/freqs/index.dat
/opt/SUNWsvpab/libidx/freqs/vocab.dat
/opt/SUNWsvpab/styles.ent
[ verifying class <Main> ]
/opt/SUNWsvpab/collinfo
/opt/SUNWsvpab/socat
[ verifying class <CONFIG> ]
## Executing postinstall script.
Adding AnswerBook Collections with command:
/usr/lib/ab2/bin/ab2admin -o install -n SUNWsvpab -d /opt/SUNWsvpab/collinfo 
Added  : SunVideo Plus  Collection
Restarting the server
/usr/lib/ab2/bin/ab2admin -o stop
/usr/lib/ab2/bin/ab2admin -o start

Installation of <SUNWsvpab> was successful.

Processing package instance <SUNWvtsvp> from </home/rodgers/SunVideoPlus_1.3.dir/Common/Packages>

SunVTS Sun Video Plus test module
(sparc) 2.1.1,REV=1999.09.17
Copyright 1999 Sun Microsystems, Inc. All rights reserved.
Using </> as the package base directory.
## Processing package information.
## Processing system information.
   3 package pathnames are already properly installed.
## Verifying package dependencies.
WARNING:
    The <SUNWo1kpd> package "SunVideo Plus PCI Card Device
    Driver." is a prerequisite package and should be
    installed.
WARNING:
    The <SUNWvts> package "Online Validation Test Suite" is
    a prerequisite package and should be installed.

Do you want to continue with the installation of <SUNWvtsvp> [y,n,?] y
## Verifying disk space requirements.
## Checking for conflicts with packages already installed.
## Checking for setuid/setgid programs.

This package contains scripts which will be executed with super-user
permission during the process of installing this package.

Do you want to continue with the installation of <SUNWvtsvp> [y,n,?] y

Installing SunVTS Sun Video Plus test module as <SUNWvtsvp>

## Installing part 1 of 1.
/opt/SUNWvts/bin/o1k_dma.ref
/opt/SUNWvts/bin/o1ktest
/opt/SUNWvts/bin/o1ktest_info.o
/opt/SUNWvts/bin/sparcv9/o1ktest_info.o
/opt/SUNWvts/bin/testadd
/opt/SUNWvts/bin/testrm
/opt/SUNWvts/lib/locale/C/LC_MESSAGES/o1ktest.msg
[ verifying class <none> ]
## Executing postinstall script.
/usr/ccs/bin/nm: /opt/SUNWvts/bin/vtsinfo.so: No such file or directory
o1ktest_info.o is in vtsinfo.a, and testprobe_o1ktest in vtsinfo.so.
o1ktest is added to SunVTS.

Installation of <SUNWvtsvp> was successful.

The following packages are available:
  1  SUNWo1kpu     SunVideo Plus Runtime Support Software
                   (sparc) 1.3,REV=1.3.1
  2  SUNWopi       OPI Runtime Library Package
                   (sparc) 2.2,REV=2.2.0
  3  SUNWopi1k     SunVideoPlus OPI Runtime Package
                   (sparc) 2.2,REV=2.2.0
  4  SUNWsvpab     SunVideo Plus Collection
                   (all) 188.2.4
  5  SUNWvtsvp     SunVTS Sun Video Plus test module
                   (sparc) 2.1.1,REV=1999.09.17

Select package(s) you wish to process (or 'all' to process
all packages). (default: all) [?,??,q]: q

--------------------------------------------------------------------------------
Output from "cd  ../../Solaris_2.6/Packages; pkgadd -d ." command:

bil[root]csh:63>pkgadd -d .

The following packages are available:
  1  SUNWo1kpd.u     SunVideo Plus PCI Card Device Driver.
                     (sparc.sun4u) 1.3,REV=2.3.1
  2  SUNWo1kpx       SunVideo Plus XIL-1.3 Runtime Support Software
                     (sparc) 5.6,REV=1.3.0

Select package(s) you wish to process (or 'all' to process
all packages). (default: all) [?,??,q]: 

Processing package instance <SUNWo1kpd.u> from </home/rodgers/SunVideoPlus_1.3.dir/Solaris_2.6/Packages>

SunVideo Plus PCI Card Device Driver.
(sparc.sun4u) 1.3,REV=2.3.1
Copyright 1999 Sun Microsystems, Inc. All rights reserved.
Copyright (c) 1999 MultiMedia Access Corporation - Osprey Technologies Inc. 

All Rights Reserved.

This product and related documentation is protected by copyright laws and 
international treaties. No part of this product or related documentation  
may be reproduced, distributed, or decompiled in any form by any means 
without prior written authorization of MultiMedia Access Corporation. 
Unauthorized reproduction or distribution of this program, or any part of 
it, may result in severe civil and criminal penalties, and will be 
prosecuted to the maximum extent possible under the law.

MMAC reserves the right to change any products herein at any time and 
without notice. MMAC does not assume any liability arising from the 
application or use of this product, except as expressly agreed to in 
writing by MMAC. Use and purchase of this product does not convey a license
under any patent rights, copyrights, trademark rights, or any other 
intellectual property rights of MMAC, Sun Microsystems, or others. 

Sun, Sun Microsystems, Solaris, SunOS, OpenWindows, XView, VideoPix and
SunVideo are trademarks or registered trademarks of Sun Microsystems, Inc.
All other product names mentioned herein are the trademarks of their
respective owners.



## Executing checkinstall script.
Using </> as the package base directory.
## Processing package information.
## Processing system information.
   6 package pathnames are already properly installed.
## Verifying package dependencies.
## Verifying disk space requirements.
## Checking for conflicts with packages already installed.
## Checking for setuid/setgid programs.

This package contains scripts which will be executed with super-user
permission during the process of installing this package.

Do you want to continue with the installation of <SUNWo1kpd> [y,n,?] y

Installing SunVideo Plus PCI Card Device Driver. as <SUNWo1kpd>

## Executing preinstall script.
removing old drivers
Looking for and Removing old device entries
## Installing part 1 of 1.
/etc/init.d/o1k-config
/etc/opt/SUNWo1kp/dbm/o1kctl_0
/etc/opt/SUNWo1kp/dbm/o1kctl_1
/etc/opt/SUNWo1kp/dbm/o1kctl_2
/etc/opt/SUNWo1kp/dbm/o1kctl_3
/etc/opt/SUNWo1kp/dbm/o1kctl_4
/etc/opt/SUNWo1kp/dbm/o1kctl_5
/etc/opt/SUNWo1kp/dbm/o1kctl_6
/etc/opt/SUNWo1kp/dbm/o1kctl_7
/etc/opt/SUNWo1kp/dbm/o1kctl_t0
/etc/opt/SUNWo1kp/dbm/o1kctl_t1
/etc/opt/SUNWo1kp/dbm/o1kctl_t2
/etc/opt/SUNWo1kp/dbm/o1kctl_t3
/etc/opt/SUNWo1kp/dbm/o1kctl_t4
/etc/opt/SUNWo1kp/dbm/o1kctl_t5
/etc/opt/SUNWo1kp/dbm/o1kctl_t6
/etc/opt/SUNWo1kp/dbm/o1kctl_t7
/kernel/drv/o1k
/kernel/drv/o1k.conf
[ verifying class <none> ]
/etc/rc2.d/S93o1k-config <linked pathname>
[ verifying class <devlinktab> ]
## Executing postinstall script.
            compatible: 'pci1061,3' + 'pciclass,048000'
Found o1k Based Hardware
Adding driver now...
Could not unlink /dev/dsk/c1t5d0s0 							because: No such file or directory

Installation of <SUNWo1kpd> was successful.

Processing package instance <SUNWo1kpx> from </home/rodgers/SunVideoPlus_1.3.dir/Solaris_2.6/Packages>

SunVideo Plus XIL-1.3 Runtime Support Software
(sparc) 5.6,REV=1.3.0
Copyright 1999 Sun Microsystems, Inc. All rights reserved.
Copyright (c) 1999 MultiMedia Access Corporation - Osprey Technologies Inc. 

All Rights Reserved.

This product and related documentation is protected by copyright laws and 
international treaties. No part of this product or related documentation  
may be reproduced, distributed, or decompiled in any form by any means 
without prior written authorization of MultiMedia Access Corporation. 
Unauthorized reproduction or distribution of this program, or any part of 
it, may result in severe civil and criminal penalties, and will be 
prosecuted to the maximum extent possible under the law.

MMAC reserves the right to change any products herein at any time and 
without notice. MMAC does not assume any liability arising from the 
application or use of this product, except as expressly agreed to in 
writing by MMAC. Use and purchase of this product does not convey a license
under any patent rights, copyrights, trademark rights, or any other 
intellectual property rights of MMAC, Sun Microsystems, or others. 

Sun, Sun Microsystems, Solaris, SunOS, OpenWindows, XView, VideoPix and
SunVideo are trademarks or registered trademarks of Sun Microsystems, Inc.
All other product names mentioned herein are the trademarks of their
respective owners.



## Executing checkinstall script.
XIL 1.3 is available on system...
Using </usr> as the package base directory.
## Processing package information.
## Processing system information.
   7 package pathnames are already properly installed.
## Verifying package dependencies.
## Verifying disk space requirements.
## Checking for conflicts with packages already installed.
## Checking for setuid/setgid programs.

This package contains scripts which will be executed with super-user
permission during the process of installing this package.

Do you want to continue with the installation of <SUNWo1kpx> [y,n,?] y

Installing SunVideo Plus XIL-1.3 Runtime Support Software as <SUNWo1kpx>

## Installing part 1 of 1.
/usr/openwin/lib/xil/devhandlers/xilCompute_CellBMMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_H261MMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_JpegMMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_Mpeg1MMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilCompute_UYVYMMACo1k.so.2
/usr/openwin/lib/xil/devhandlers/xilIO_MMACo1k.so.2
/usr/openwin/lib/xil/locale/C/LC_MESSAGES/xilIO_MMACo1k.mo
[ verifying class <none> ]
[ verifying class <OWconfig> ]

Installation of <SUNWo1kpx> was successful.

The following packages are available:
  1  SUNWo1kpd.u     SunVideo Plus PCI Card Device Driver.
                     (sparc.sun4u) 1.3,REV=2.3.1
  2  SUNWo1kpx       SunVideo Plus XIL-1.3 Runtime Support Software
                     (sparc) 5.6,REV=1.3.0

Select package(s) you wish to process (or 'all' to process
all packages). (default: all) [?,??,q]: q

--------------------------------------------------------------------------------



From rem-conf Thu Apr 13 12:07:14 2000 
From rem-conf-request@es.net Thu Apr 13 12:07:13 2000
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Reply-To: infos@aivp.net
From: "Infos - AIVP" <infos@aivp.net>
To: aivp@es.net
Subject: 7a Conferencia Internacional Ciudades y Puertos
Date: Thu, 13 Apr 2000 12:15:25 +0200
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7a Conferencia Internacional Ciudades y Puertos
7th International Conference Cities & Ports
6-9 november 2000
Marseille (France)
=====

Del 6 al 9 del próximo mes de noviembre, los responsables del desarrollo
urbano y portuario de las ciudades marítimas y fluviales de todo el mundo se
dan cita en Marsella (Francia) para la 7° Conferencia Internacional de las
Ciudades y Puertos organizada por la Asociación Internacional Ciudades y
Puertos. (AIVP). Se esperan en Marsella a más de 500 delegados
representantes de las ciudades portuarias de más de 50 países  para discutir
e intercambiar sus experiencias sobre la puesta en marcha de un desarrollo
sostenible de las plazas portuarias. ¿Le interesa este reto? ¿Le concierne
este encuentro? Sin ningún compromiso de su parte, le proponemos mantenerle
informado de forma gratuita sobre la evolución del programa de esta
conferencia

> enviando un mensaje a :
listees.conference@aivp.net con la palabra SUBSCRIBE en el objeto del
mensaje
(mailto:listees.conference@aivp.net?subject=SUBSCRIBE)

> via nuestro website
http://www.aivp.com/7conf/formes.asp


=====


Urban and port development authorities for coastal and river cities from
throughout the entire world will be meeting in Marseilles (France), from the
6th to the 9th of November next, for the 7th Cities and Ports (IACP)
International Conference. More than 500 delegates representing the port
cities of more than 50 countries are expected in Marseilles to debate and
exchange views regarding the implementation of sustainable development in
port areas. This challenge is of interest to you ? This meeting concerns you
? We propose, without any committment on your behalf, to keep you informed
free of charge of the programme development of this conference

> by sending a message to :
listegb.conference@aivp.net with SUBSCRIBE as subject
(mailto:listegb.conference@aivp.net?subject=SUBSCRIBE)

>or via our website
http://www.aivp.com/7conf/formgb.asp




From rem-conf Thu Apr 13 12:07:20 2000 
From rem-conf-request@es.net Thu Apr 13 12:07:20 2000
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Date: Thu, 13 Apr 2000 15:01:44 -0400 (EDT)
From: "R. P. Channing Rodgers, M.D." <rodgers@nlm.nih.gov>
Message-Id: <200004131901.PAA18007@billings.nlm.nih.gov>
To: mbone@isi.edu, rem-conf@es.net
Subject: sunforuym and "Directory server" software for conferencing?
Cc: aronson@billings.nlm.nih.gov, K.Hasler@cs.ucl.ac.uk, O.Hodson@cs.ucl.ac.uk
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Dear Colleagues,

A brief follow-on question.  Sunforum (and other) teleconferencing
software refer to "directory servers" such as the ones at:

   ils.microsoft.com
   ils1.microsoft.com, ..., ils9.microsoft.com
   ils.four11.com
   ils.business.four11.com
   ils.family.four11.com

I have never been able to successfully connect to any of these from
sunforum, but whether it is because they are too busy, incompatible,
require some kind of authentication, it's unclear.  I assume that these
servers are doing soe H.323-related activities such as port assignment,
as well as keeping you on a list of folks who are currently available to
conference with.

Does anyone know of (preferably freely available) server software, for
the creation of a local service of this type?

Thanks and Cheerio, Rick Rodgers




From rem-conf Fri Apr 14 00:41:21 2000 
From rem-conf-request@es.net Fri Apr 14 00:41:21 2000
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 14 Apr 2000 09:24:09 +0200 (MET DST)
Date: Fri, 14 Apr 2000 09:27:07 +0200
From: Carsten Burmeister <burmeister@panasonic.de>
Subject: Multiple selective retransmissions
To: rem-conf@es.net
Message-id: <38F6C84B.15F957AA@panasonic.de>
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Hi all,

we gave a short presentation in Adelaide about selective retransmissions
in RTP. We described mainly the content of the I-D: "RTP Payload Type
Format to Enable Selective Retransmissions", which can be found at:

http://www.pel.panasonic.de/ietf/docs/draft-miyazaki-avt-rtp-selret-00.txt
http://www.pel.panasonic.de/ietf/docs/draft-miyazaki-avt-rtp-selret-00.ps

In discussions after the meeting we got the impression that we have not
stated clear enough the need for the additional header field "SNHP". 

We think this kind of field is necessary to do multiple selective
retransmissions. We prepared some figures, which should explain this in
greater detail. This figures can be found under the following URL:

http://www.pel.panasonic.de/ietf/docs/Clarification_of_SNHP.ppt
http://www.pel.panasonic.de/ietf/docs/Clarification_of_SNHP.ps

The increased performance of video streaming applications over wireless
links, using this field, are shown by the means of simulations in the
I-D.

Any comments are welcome....

Carsten



From rem-conf Fri Apr 14 13:45:14 2000 
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From: Video Compression Workshop <videoc@ee.pdx.edu>
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Subject: Video Compression Short Course in Portland 8/7-12/2000
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Subject: Video Compression Short Course in Portland 8/7-12/2000

4 & 1/2 Days (August 7 - 12, 2000) in Portland Oregon

                                on

                IMAGE AND VIDEO COMPRESSION:
        FUNDAMENTALS, STANDARDS, AND APPLICATIONS

                        Featured:

 Majid Rabbani -                        Eastman Kodak
 M. Ibrahim Sezan -                     Sharp Labs. of America
 Thomas Gardos -                        Intel Corporation

Organized by Fu Li
Portland State University, Portland, Oregon, USA

                please visit

http://www.ee.pdx.edu/~compress

Contact Don Mueller at

Telephone: (503) 725-3806
Facsimile: (503) 725-3807
Email: donm@eas.pdx.edu

Comprehensive Coverage on JPEG-2000,
MPEG -4 and -7, and H.26x Standards!

With Latest Technology Demos

Seats limited. Please register today!




From rem-conf Mon Apr 17 01:09:11 2000 
From rem-conf-request@es.net Mon Apr 17 01:09:09 2000
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From: <jdarby001@earthlink.net>
Subject: We Have Answers for YOU
Date: Mon, 17 Apr 2000 00:11:17
Message-Id: <665.735308.759563@area51.japannet.com>
Bcc:
To: rem-conf@es.net
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Are  you interested in refinancing your existing mortgage, Paying 
off bills, Consolidating debt, Doing home improvements, or 
Purchasing a big ticket item?

Are you interested in Purchasing a new home or are you a First 
time homebuyer? If so, we have ANSWERS for YOU even if you have 
poor credit.

For more information please reply:jdarby002@earthlink.net



------------------------------------------------------------
Under Bill s.1618 TITLE III passed by the 105th U.S. Congress 
this letter cannot be considered "spam" as long as we include: 1) 
contact information (see above); and, 2) the way to be removed 
>from future mailings (see below). 

To be removed from this list, please mail to: 
jdarby001@earthlink.net with 'remove' in 
subject line and you will be removed from our list.
------------------------------------------------------------
 
 
 
 
 
 
 
 
 
 
 
 



From rem-conf Mon Apr 17 07:13:33 2000 
From rem-conf-request@es.net Mon Apr 17 07:13:31 2000
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Message-ID: <14587.6803.75296.499527@gargle.gargle.HOWL>
Date: Mon, 17 Apr 2000 16:07:15 +0200 (MET DST)
From: Matthias.Lang@ericsson.com
To: rem-conf@es.net
Subject: How to pack multiple GSM frames into one RTP packet
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Hi,

Apologies if this has come up before, I've searched and not found an answer.

How do you pack two GSM frames into one RTP packet? A GSM frame is
260 bits (32.5 octets) long.

Method #1: Since a single GSM frame is transmitted as 33 octets, i.e.
       the frame plus a nybble of padding, two GSM frames should be 
       transmitted in 66 octets. This is what RAT does.

Method #2: Since a single GSM frame is really 32.5 octets, two GSM
       frames should be transmitted as 65 octets. This is what another
       implementation I've seen does.

Maybe RFC1890 could say something like

	  RTP payloads with frames which do not end on octet
	  boundaries should (must?) be zero-padded to the next 
	  octet boundary. This applies even when several
	  frames are transmitted in one packet.

I note that this may cause problems for G.728. I don't know. Maybe it
should say the opposite, i.e. that no padding is allowed except at the
end of the packet. This has nothing to do with RTP padding. I can
think of efficient software and hardware codec implementations either
way.

Comments?

Matthias       




From rem-conf Mon Apr 17 09:03:09 2000 
From rem-conf-request@es.net Mon Apr 17 09:03:08 2000
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	Mon, 17 Apr 2000 18:00:42 +0200 (MET DST)
From: "Dr. Carsten Bormann" <cabo@tzi.org>
To: <Matthias.Lang@ericsson.com>, <rem-conf@es.net>
Subject: RE: How to pack multiple GSM frames into one RTP packet
Date: Mon, 17 Apr 2000 18:00:40 +0200
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> Method #1: Since a single GSM frame is transmitted as 33 octets, i.e.
>        the frame plus a nybble of padding, two GSM frames should be
>        transmitted in 66 octets. This is what RAT does.

This is also what the current draft standards document says:
http://search.ietf.org/internet-drafts/draft-ietf-avt-profile-new-08.txt,
section 4.5.8.1:

   In the GSM packing used by RTP, the bits SHALL be packed beginning
   from the most significant bit. Every 160 sample GSM frame is coded
   into one 33 octet (264 bit) buffer. Every such buffer begins with a 4
   bit signature (0xD), followed by the MSB encoding of the fields of
   the frame. The first octet thus contains 1101 in the 4 most
   significant bits (0-3) and the 4 most significant bits of F1 (0-3) in
   the 4 least significant bits (4-7). The second octet contains the 2
   least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so
   on.

Note that this is the document that is intended to supersede RFC1890.

Gruesse, Carsten

PS.: And I apologize for having my hand in designing it that way 8 years
ago...




From rem-conf Mon Apr 17 09:52:49 2000 
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Subject: Re: We Have Answers for YOU
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On Mon, 17 Apr 2000 jdarby001@earthlink.net wrote:

> Are  you interested in refinancing your existing mortgage, Paying 
> off bills, Consolidating debt, Doing home improvements, or 
> Purchasing a big ticket item?
> 
> Are you interested in Purchasing a new home or are you a First 
> time homebuyer? If so, we have ANSWERS for YOU even if you have 
> poor credit.
> 
> For more information please reply:jdarby002@earthlink.net
> 
> ------------------------------------------------------------
> Under Bill s.1618 TITLE III passed by the 105th U.S. Congress 
> this letter cannot be considered "spam" as long as we include: 1) 
> contact information (see above); and, 2) the way to be removed 
> from future mailings (see below). 
> 
> To be removed from this list, please mail to: 
> jdarby001@earthlink.net with 'remove' in 
> subject line and you will be removed from our list.
> ------------------------------------------------------------

They are apparently using some spamming software that tosses in the
``signature'' at the bottom.  Anyway, they are lying about s.1618 and
I include the relevant portion (Section 301) below.  BTW, I am not
sure if this has become law.

Regards,
Steve



                S.1618

Anti-slamming Amendments Act (Passed by the Senate)


             TITLE III--SPAMMING

SEC. 301. REQUIREMENTS RELATING TO TRANSMISSIONS OF UNSOLICITED
COMMERCIAL ELECTRONIC MAIL.

       (a) INFORMATION TO BE INCLUDED IN TRANSMISSIONS-

              (1) IN GENERAL- A person who transmits an unsolicited
                  commercial electronic mail message shall cause to
                  appear in each such electronic mail message the
                  information specified in paragraph (2).

              (2) COVERED INFORMATION- The following information shall
                  appear at the beginning of the body of an unsolicited
                  commercial electronic mail message under paragraph (1):

                    (A) The name, physical address, electronic mail
                        address, and telephone number of the
                        person who initiates transmission of the
                        message.

                    (B) The name, physical address, electronic mail
                        address, and telephone number of the
                        person who created the content of the message,
                        if different from the information under
                        subparagraph (A).

                    (C) A statement that further transmissions of
                        unsolicited commercial electronic mail to the
                        recipient by the person who initiates
                        transmission of the message may be stopped at no
                        cost to the recipient by sending a reply to
                        the originating electronic mail address with the
                        word `remove' in the subject line.

       (b) ROUTING INFORMATION- All Internet routing information
           contained within or accompanying an electronic mail message
           described in subsection (a) must be accurate, valid
           according to the prevailing standards for Internet
           protocols, and accurately reflect message routing.

       (c) EFFECTIVE DATE- The requirements in this section shall take
           effect 30 days after the date of enactment of this Act.





From rem-conf Tue Apr 18 02:28:12 2000 
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Message-ID: <38FC281F.C872FB9A@era.ericsson.se>
Date: Tue, 18 Apr 2000 11:17:20 +0200
From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Organization: Ericsson Research
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Hello!

I have two questions regarding RTP.

The first one concerns the initialization of the variable avg_rtcp_size.
draft-ietf-avt-rtp-new-07 section 6.3.2 says:

   Upon joining the session, the participant initializes tp to 0, tc to
   0, senders to 0, pmembers to 1, members to 1, we_sent to false,
   rtcp_bw to the specified fraction of the session bandwidth, initial
   to true, and
            -->      avg_rtcp_size to the size of the very first packet
   constructed by the application. <--

   Is it not more reasonable to initialize avg_rtcp_size to a size that
it is probable that the first
RTCP packet will have? I think the current text is to unspecified and
hard to interpret.

The second question is about section 6.3.5 Timing Out an SSRC.

Is there a reason why the time out interval is set to M*Td and not to
the M latest RTCP transmission interval?
You could save the need of a last active timer for each member and
instead use a much smaller status variable counting how many periods
since last active. Is there no problem with premature timing out of
users when you have many users leaving a session, especially if they do
not send BYE message?


Magnus Westerlund

Audio Technology, Ericsson Research
----------------------------------------------------------------------
Ericssson Radio Systems AB | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se





From rem-conf Tue Apr 18 10:36:06 2000 
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Date: Tue, 18 Apr 2000 10:30:00 -0700
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 4/19 Multicast Debugging -- Mark Handley, AT&T Center for
  Internet Research at ICSI (ACIRI)
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Berkeley Multimedia, Graphics and Interfaces Seminar

Multicast Debugging

Wednesday April 19, 2000
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

Mark Handley
AT&T Center for Internet Research at ICSI (ACIRI) 

Network topologies can be complex, and the application and control traffic
can create a 'chemistry' of loads and flows that leads to unanticipated
network congestion and application failure. This session will provide
network engineers valuable information on debugging congestion,
misconfiguration, application failure to help keep networks and content
delivery reliable.
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Thu Apr 20 03:25:50 2000 
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Date: Thu, 20 Apr 2000 03:03:03 -0700
From: Scott Keagy <scott@sknetworks.com>
Organization: SK Networks, Inc.
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Subject: RTP payload types in draft-ietf-avt-profile-new-08.txt
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Steve et al,

Have you considered assigning more RTP payload types to options of G.723
and G.729 with and without VAD support? G.723.1 5.3k vs. 6.3k ? As it
is, vendors must choose to support all or no options of these codecs to
ensure interoperability. Cisco's current SIP implementation, which
appears mostly compliant with the IETF draft, leaves room for
interoperability issues with other vendors because it can support
subsets of these codec types.

(I summarized the details below)

-Scott Keagy

=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================


##################################################################
IOS version:    12.1(1)T      (c3640-p7-mz_121-1_T.bin)
Feature:            SIP/2.0 user agent

Problem Description:

Cisco's interpretation of RTP payload types may not be consistent with
other vendors, which may impact SIP interoperability.

For example, "draft-ietf-avt-profile-new-08.txt" (Jan 14, 2000)
indicates RTP PT=4 corresponds with G.723.1, including the 5.3kbps,
6.3kbps, and Annex A VAD options.

Cisco may indicate a media capability of G.723.1 to other vendors, when
it really only means G.723.1 @ 6.3kbps without Annex A. Other vendors
may justifiably assume that the Cisco router supports all options of
G.723.1, and send RTP packets with G.723.1 @5.3kbps and Annex A VAD.
This scenario would fail if the router is not configured for all of the
options of G.723.1. Cisco is over- advertising media capabilities. But
there is no method for Cisco to only advertise certain options of these
codecs- it's either all or nothing. The RTP payload types should
accomodate this.

A similar problem exists with the G.729 payload type for RTP.


!!The following IOS configuration for codecs........

voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g723ar53
 codec preference 4 g723ar63
 codec preference 5 g723r53
 codec preference 6 g723r63
 codec preference 7 g726r16
 codec preference 8 g726r24
 codec preference 9 g726r32
 codec preference 10 g728
 codec preference 11 g729br8
 codec preference 12 g729r8

!!Results in this capabilities advertisement (from a SIP INVITE)....

m=audio 20864 RTP/AVP 8 0 65535 65535 65535 4 2 65535 65535 15 65535 18

!!Note the 65535 for codecs that Cisco does not deem to be included as
standard RTP PTs.




From rem-conf Thu Apr 20 07:12:09 2000 
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Date: Thu, 20 Apr 2000 10:05:30 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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CC: casner@cisco.com, rem-conf@es.net
Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
References: <38FED5D7.4F182ABF@sknetworks.com>
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I agree that having separate names for the different rates would be a
good idea. Note, however, that RTP no longer assigns fixed numerical
payload type values to codecs but relies on SDP or similar protocols to
map names to numbers. Thus, Cisco's implementation seems to take a
short-cut (to put it politely) in that regard, but Steve might be in a
position to educate his colleagues....

Scott Keagy wrote:
> 
> Steve et al,
> 
> Have you considered assigning more RTP payload types to options of G.723
> and G.729 with and without VAD support? G.723.1 5.3k vs. 6.3k ? As it
> is, vendors must choose to support all or no options of these codecs to
> ensure interoperability. Cisco's current SIP implementation, which
> appears mostly compliant with the IETF draft, leaves room for
> interoperability issues with other vendors because it can support
> subsets of these codec types.
> 
> (I summarized the details below)
> 
> -Scott Keagy
> 
> =======================================================
>  Scott Keagy, CCIE #3985
>  Senior Networking Consultant
>  SK Networks, Inc.         mailto:scott@sknetworks.com
> =======================================================
> 
> ##################################################################
> IOS version:    12.1(1)T      (c3640-p7-mz_121-1_T.bin)
> Feature:            SIP/2.0 user agent
> 
> Problem Description:
> 
> Cisco's interpretation of RTP payload types may not be consistent with
> other vendors, which may impact SIP interoperability.
> 
> For example, "draft-ietf-avt-profile-new-08.txt" (Jan 14, 2000)
> indicates RTP PT=4 corresponds with G.723.1, including the 5.3kbps,
> 6.3kbps, and Annex A VAD options.
> 
> Cisco may indicate a media capability of G.723.1 to other vendors, when
> it really only means G.723.1 @ 6.3kbps without Annex A. Other vendors
> may justifiably assume that the Cisco router supports all options of
> G.723.1, and send RTP packets with G.723.1 @5.3kbps and Annex A VAD.
> This scenario would fail if the router is not configured for all of the
> options of G.723.1. Cisco is over- advertising media capabilities. But
> there is no method for Cisco to only advertise certain options of these
> codecs- it's either all or nothing. The RTP payload types should
> accomodate this.
> 
> A similar problem exists with the G.729 payload type for RTP.
> 
> !!The following IOS configuration for codecs........
> 
> voice class codec 1
>  codec preference 1 g711alaw
>  codec preference 2 g711ulaw
>  codec preference 3 g723ar53
>  codec preference 4 g723ar63
>  codec preference 5 g723r53
>  codec preference 6 g723r63
>  codec preference 7 g726r16
>  codec preference 8 g726r24
>  codec preference 9 g726r32
>  codec preference 10 g728
>  codec preference 11 g729br8
>  codec preference 12 g729r8
> 
> !!Results in this capabilities advertisement (from a SIP INVITE)....
> 
> m=audio 20864 RTP/AVP 8 0 65535 65535 65535 4 2 65535 65535 15 65535 18
> 
> !!Note the 65535 for codecs that Cisco does not deem to be included as
> standard RTP PTs.

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



From rem-conf Thu Apr 20 08:01:52 2000 
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: Scott Keagy <scott@sknetworks.com>, casner@cisco.com, rem-conf@es.net
Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
In-Reply-To: Message from Henning Schulzrinne <schulzrinne@cs.columbia.edu> of "Thu, 20 Apr 2000 10:05:30 EDT." <38FF0EAA.EF2249C@cs.columbia.edu>
Date: Thu, 20 Apr 2000 16:00:51 +0100
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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The codecs in question have static payload type assignments in the revised
a/v profile. It may be worth specifying SDP parameters, such that these
codecs - with the revised parameters - can be mapped to a dynamic payload
type.

Colin

--> Henning Schulzrinne writes:
>I agree that having separate names for the different rates would be a
>good idea. Note, however, that RTP no longer assigns fixed numerical
>payload type values to codecs but relies on SDP or similar protocols to
>map names to numbers. Thus, Cisco's implementation seems to take a
>short-cut (to put it politely) in that regard, but Steve might be in a
>position to educate his colleagues....
>
>Scott Keagy wrote:
>> 
>> Steve et al,
>> 
>> Have you considered assigning more RTP payload types to options of G.723
>> and G.729 with and without VAD support? G.723.1 5.3k vs. 6.3k ? As it
>> is, vendors must choose to support all or no options of these codecs to
>> ensure interoperability. Cisco's current SIP implementation, which
>> appears mostly compliant with the IETF draft, leaves room for
>> interoperability issues with other vendors because it can support
>> subsets of these codec types.
>> 
>> (I summarized the details below)
>> 
>> -Scott Keagy
>> 
>> =======================================================
>>  Scott Keagy, CCIE #3985
>>  Senior Networking Consultant
>>  SK Networks, Inc.         mailto:scott@sknetworks.com
>> =======================================================
>> 
>> ##################################################################
>> IOS version:    12.1(1)T      (c3640-p7-mz_121-1_T.bin)
>> Feature:            SIP/2.0 user agent
>> 
>> Problem Description:
>> 
>> Cisco's interpretation of RTP payload types may not be consistent with
>> other vendors, which may impact SIP interoperability.
>> 
>> For example, "draft-ietf-avt-profile-new-08.txt" (Jan 14, 2000)
>> indicates RTP PT=4 corresponds with G.723.1, including the 5.3kbps,
>> 6.3kbps, and Annex A VAD options.
>> 
>> Cisco may indicate a media capability of G.723.1 to other vendors, when
>> it really only means G.723.1 @ 6.3kbps without Annex A. Other vendors
>> may justifiably assume that the Cisco router supports all options of
>> G.723.1, and send RTP packets with G.723.1 @5.3kbps and Annex A VAD.
>> This scenario would fail if the router is not configured for all of the
>> options of G.723.1. Cisco is over- advertising media capabilities. But
>> there is no method for Cisco to only advertise certain options of these
>> codecs- it's either all or nothing. The RTP payload types should
>> accomodate this.
>> 
>> A similar problem exists with the G.729 payload type for RTP.
>> 
>> !!The following IOS configuration for codecs........
>> 
>> voice class codec 1
>>  codec preference 1 g711alaw
>>  codec preference 2 g711ulaw
>>  codec preference 3 g723ar53
>>  codec preference 4 g723ar63
>>  codec preference 5 g723r53
>>  codec preference 6 g723r63
>>  codec preference 7 g726r16
>>  codec preference 8 g726r24
>>  codec preference 9 g726r32
>>  codec preference 10 g728
>>  codec preference 11 g729br8
>>  codec preference 12 g729r8
>> 
>> !!Results in this capabilities advertisement (from a SIP INVITE)....
>> 
>> m=audio 20864 RTP/AVP 8 0 65535 65535 65535 4 2 65535 65535 15 65535 18
>> 
>> !!Note the 65535 for codecs that Cisco does not deem to be included as
>> standard RTP PTs.
>
>-- 
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Thu Apr 20 09:40:40 2000 
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To: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--Multipart_Thu_Apr_20_12:36:41_2000-1
Content-Type: text/plain; charset=US-ASCII

Dear all, 

Please note that for G.723.1, the same payload type is used for both
bitrates and for the silence removal (SID) mode, since the bitrate is
indicated in the first two bits of the bitstream. 

To avoid further confusion, you might consider amending
draft-ietf-avt-profile-new-??.{txt,ps} to include a diagram showing
the bit packaging for the G.723.1 modes. Please find attached such a
description, in ASCII format. I hope you find it useful.

Cheers,
Simao.


--Multipart_Thu_Apr_20_12:36:41_2000-1
Content-Type: application/octet-stream
Content-Disposition: attachment; filename="pack-g723.1.txt"
Content-Transfer-Encoding: base64


--Multipart_Thu_Apr_20_12:36:41_2000-1--



From rem-conf Thu Apr 20 10:53:52 2000 
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From: Scott Keagy <scott@sknetworks.com>
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To: Simao Campos <campos@ctd.comsat.com>
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Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
References: <schulzrinne@cs.columbia.edu>
		<38FF0EAA.EF2249C@cs.columbia.edu>
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Simao,

I agree that it is possible for a receiver to determine which version of
G.723.1 it is receiving, but the RTP payload type leaves ambiguity for
*advertising* which versions of G.723.1 a machine supports.

As an example, I indicated that Cisco routers do not necessarily support all
options of G.723.1  (or all options may not be configured), but they will
advertise in a SIP INVITE that they support G.723.1. The receiver of such an
invite will assume by the IETF draft that all options are supported, which
leads to confusion during the media establishment:

1) Cisco router's SIP INVITE: "I understand G.723.1"
2) Other vendor's 200 OK: "Great, G.723.1 it is"
3) (session established, other router sends G.723.1 @5.3k w/ Annex A)
4) Cisco router complains: "Wait, actually I  meant G.7236.3k w/o Annex A)

It would be better to avoid the ambiguity up front than to deal with this as
an error condition after the media is already established.

-Scott

Simao Campos wrote:

> Dear all,
>
> Please note that for G.723.1, the same payload type is used for both
> bitrates and for the silence removal (SID) mode, since the bitrate is
> indicated in the first two bits of the bitstream.
>
> To avoid further confusion, you might consider amending
> draft-ietf-avt-profile-new-??.{txt,ps} to include a diagram showing
> the bit packaging for the G.723.1 modes. Please find attached such a
> description, in ASCII format. I hope you find it useful.
>
> Cheers,
> Simao.
>
>   ------------------------------------------------------------------------
>                       Name: pack-g723.1.txt
>    pack-g723.1.txt    Type: Plain Text (text/plain)
>                   Encoding: base64

--
=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================





From rem-conf Thu Apr 20 11:07:09 2000 
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Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
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So should an SDP advertisement for all G.723.1 options look like this?

 m=audio 20864 RTP/AVP 4 4 4 4
 a=rtpmap 4 G723/5.3
 a=rtpmap 4 G723/6.3
 a=rtpmap 4 G723/5.3/A
 a=rtpmap 4 G723/6.3/A


Henning Schulzrinne wrote:

> I agree that having separate names for the different rates would be a
> good idea. Note, however, that RTP no longer assigns fixed numerical
> payload type values to codecs but relies on SDP or similar protocols to
> map names to numbers. Thus, Cisco's implementation seems to take a
> short-cut (to put it politely) in that regard, but Steve might be in a
> position to educate his colleagues....
>
> Scott Keagy wrote:
> >
> > Steve et al,
> >
> > Have you considered assigning more RTP payload types to options of G.723
> > and G.729 with and without VAD support? G.723.1 5.3k vs. 6.3k ? As it
> > is, vendors must choose to support all or no options of these codecs to
> > ensure interoperability. Cisco's current SIP implementation, which
> > appears mostly compliant with the IETF draft, leaves room for
> > interoperability issues with other vendors because it can support
> > subsets of these codec types.
> >
> > (I summarized the details below)
> >
> > -Scott Keagy
> >
> > =======================================================
> >  Scott Keagy, CCIE #3985
> >  Senior Networking Consultant
> >  SK Networks, Inc.         mailto:scott@sknetworks.com
> > =======================================================
> >
> > ##################################################################
> > IOS version:    12.1(1)T      (c3640-p7-mz_121-1_T.bin)
> > Feature:            SIP/2.0 user agent
> >
> > Problem Description:
> >
> > Cisco's interpretation of RTP payload types may not be consistent with
> > other vendors, which may impact SIP interoperability.
> >
> > For example, "draft-ietf-avt-profile-new-08.txt" (Jan 14, 2000)
> > indicates RTP PT=4 corresponds with G.723.1, including the 5.3kbps,
> > 6.3kbps, and Annex A VAD options.
> >
> > Cisco may indicate a media capability of G.723.1 to other vendors, when
> > it really only means G.723.1 @ 6.3kbps without Annex A. Other vendors
> > may justifiably assume that the Cisco router supports all options of
> > G.723.1, and send RTP packets with G.723.1 @5.3kbps and Annex A VAD.
> > This scenario would fail if the router is not configured for all of the
> > options of G.723.1. Cisco is over- advertising media capabilities. But
> > there is no method for Cisco to only advertise certain options of these
> > codecs- it's either all or nothing. The RTP payload types should
> > accomodate this.
> >
> > A similar problem exists with the G.729 payload type for RTP.
> >
> > !!The following IOS configuration for codecs........
> >
> > voice class codec 1
> >  codec preference 1 g711alaw
> >  codec preference 2 g711ulaw
> >  codec preference 3 g723ar53
> >  codec preference 4 g723ar63
> >  codec preference 5 g723r53
> >  codec preference 6 g723r63
> >  codec preference 7 g726r16
> >  codec preference 8 g726r24
> >  codec preference 9 g726r32
> >  codec preference 10 g728
> >  codec preference 11 g729br8
> >  codec preference 12 g729r8
> >
> > !!Results in this capabilities advertisement (from a SIP INVITE)....
> >
> > m=audio 20864 RTP/AVP 8 0 65535 65535 65535 4 2 65535 65535 15 65535 18
> >
> > !!Note the 65535 for codecs that Cisco does not deem to be included as
> > standard RTP PTs.
>
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

--
=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================





From rem-conf Thu Apr 20 13:56:50 2000 
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From: "Fox, Mike" <mfox@nuera.com>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, Colin Perkins
	 <c.perkins@cs.ucl.ac.uk>
Cc: rem-conf@es.net
Subject: Question regarding SDP and MIME codec registration...
Date: Thu, 20 Apr 2000 13:49:38 -0700
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	Nuera has a voice codec which encodes 64K PCM into one of 3 data
rates, 7.47, 8.0, and 9.6 Kbps. We would like to register it with IANA in
the Audio/Vendor tree as vnd.nuera.ecelp. Our intention is that the data
rates are explicitly indicated as required parameters. So the MIME
registration description will have the following:

Required parameters : <clock rate> one of 7470, 8000, or 9600.

And that the SDP RFC convention of:

     a=rtpmap:<payload type> <encoding name>/<clock rate>[/<encoding
     parameters>]

would be met with the following SDP.
 
a=rtpmap:100 vnd.nuera.ecelp/7470.

Would this correct? 
Suggestions welcome.

Best regards,
Mike Fox
Nuera Communications.





From rem-conf Thu Apr 20 14:32:24 2000 
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    rem-conf@es.net
Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
In-Reply-To: Message from Scott Keagy <scott@sknetworks.com> of "Thu, 20 Apr 2000 11:04:50 PDT." <38FF46C2.B3AE223F@sknetworks.com>
Date: Thu, 20 Apr 2000 22:21:30 +0100
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--> Scott Keagy writes:
>So should an SDP advertisement for all G.723.1 options look like this?
>
> m=audio 20864 RTP/AVP 4 4 4 4
> a=rtpmap 4 G723/5.3
> a=rtpmap 4 G723/6.3
> a=rtpmap 4 G723/5.3/A
> a=rtpmap 4 G723/6.3/A

If it's acceptable to have implementations which only support some
varients, then a scheme like this makes sense (although the chosen 
payload type number should be different for each).

Colin



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Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
In-reply-to: Your message of "Thu, 20 Apr 2000 03:03:03 PDT." <38FED5D7.4F182ABF@sknetworks.com>
Date: Thu, 20 Apr 2000 22:46:14 +0100
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There is no need to specify support for 5.3 and 6.3 kbit/s on the
grounds of capability in advertisements.  The summary on page 3 of ITU
Recommendation G.723.1 (03/96) states: "Both rates are a mandatory
part of the encoder and decoder.  It is possible to switch between the
two rates at any frame boundary".  Applications that do not support
both modes should simply not advertise G.723.1.  There is a policy
issue regarding bandwidth usage or acceptable audio quality.  A
clarification of preferred formats would be useful; the default
(first) encoding type in SDP probably suffices with the 5.3/6.3
mode explicitly advertised.

Support for G.723.1 Annex A is optional, but the consequences of
mis-advertisement are minor when the contents of the annex are
considered.  Annex A describes a voice activity detection (VAD)
algorithm and a comfort noise generation (CNG) algorithm.  VAD can be
removed from consideration as there is no information transmitted from
the sender to receiver about what type of VAD algorith is employed.
The VAD algorithm is specified because there are useful measures in
the encoding stages for judging whether a period is silence or not.

When CNG is active an additional frame type, 4 bytes small, is sent to
describe the sound of silence at the end of talkspurts and as the
power spectrum of the silence changes above a threshold.  If the
sender supports CNG and the receiver does not, the receiver will
discard these frames when it inspects frame type present in the first
octet of each frame (which it has to do because a G.723.1 encoder is
permitted to change between 5.3 and 6.3 kbit/s at any frame boundary).
A receiver might not support CNG, but know that a CNG frame is 4 bytes
and therefore be able to fragment compound CNG+audio packets.  In
terms of cost CNG data computation is neglible relative to the
encoding process and the amount of wasted bandwith is small under
normal circumstances.

G.729 is different because the frame type is not included within each
frame and several distinct frame types exist between the original
recommendation and it's annexes (audio at 6.4, 8, and 11.8 kbit/s, and
CNG).  Distinct payload identifiers must be advertised and negotiated
for each of the available encodings as receivers cannot decode the
data otherwise.

- Orion.

Orion Hodson =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-= Research Student
Networked Multimedia Research Group      Tel: ++(0)20 7679 3704
Department of Computer Science           Fax: ++(0)20 7387 1397
University College London, Gower Street, London,  WC1E 6BT,  UK     
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-


<38FED5D7.4F182ABF@sknetworks.com>Scott Keagy writes:
> Steve et al,
> 
> Have you considered assigning more RTP payload types to options of G.723
> and G.729 with and without VAD support? G.723.1 5.3k vs. 6.3k ? As it
> is, vendors must choose to support all or no options of these codecs to
> ensure interoperability. Cisco's current SIP implementation, which
> appears mostly compliant with the IETF draft, leaves room for
> interoperability issues with other vendors because it can support
> subsets of these codec types.
> 
> (I summarized the details below)
> 
> -Scott Keagy
> 
> =======================================================
> Scott Keagy, CCIE #3985
> Senior Networking Consultant
> SK Networks, Inc.         mailto:scott@sknetworks.com
> =======================================================
> 
> 
> ##################################################################
> IOS version:    12.1(1)T      (c3640-p7-mz_121-1_T.bin)
> Feature:            SIP/2.0 user agent
> 
> Problem Description:
> 
> Cisco's interpretation of RTP payload types may not be consistent with
> other vendors, which may impact SIP interoperability.
> 
> For example, "draft-ietf-avt-profile-new-08.txt" (Jan 14, 2000)
> indicates RTP PT=4 corresponds with G.723.1, including the 5.3kbps,
> 6.3kbps, and Annex A VAD options.
> 
> Cisco may indicate a media capability of G.723.1 to other vendors, when
> it really only means G.723.1 @ 6.3kbps without Annex A. Other vendors
> may justifiably assume that the Cisco router supports all options of
> G.723.1, and send RTP packets with G.723.1 @5.3kbps and Annex A VAD.
> This scenario would fail if the router is not configured for all of the
> options of G.723.1. Cisco is over- advertising media capabilities. But
> there is no method for Cisco to only advertise certain options of these
> codecs- it's either all or nothing. The RTP payload types should
> accomodate this.
> 
> A similar problem exists with the G.729 payload type for RTP.
> 
> 
> !!The following IOS configuration for codecs........
> 
> voice class codec 1
> codec preference 1 g711alaw
> codec preference 2 g711ulaw
> codec preference 3 g723ar53
> codec preference 4 g723ar63
> codec preference 5 g723r53
> codec preference 6 g723r63
> codec preference 7 g726r16
> codec preference 8 g726r24
> codec preference 9 g726r32
> codec preference 10 g728
> codec preference 11 g729br8
> codec preference 12 g729r8
> 
> !!Results in this capabilities advertisement (from a SIP INVITE)....
> 
> m=audio 20864 RTP/AVP 8 0 65535 65535 65535 4 2 65535 65535 15 65535 18
> 
> !!Note the 65535 for codecs that Cisco does not deem to be included as
> standard RTP PTs.
> 
> 



From rem-conf Thu Apr 20 15:07:18 2000 
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Date: Thu, 20 Apr 2000 18:07:06 -0400
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Subject: Call for Paper - Local Computer Networks (LCN) 2000
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                        CALL FOR PAPERS
                           LCN 2000
    The 25th Annual IEEE Conference on Local Computer Networks

                     November 8 - 10, 2000
              Embassy Suites USF, Tampa, Florida
                    http://www.ieeelcn.org

            Sponsored by the IEEE Computer Society
            with support from GTE Data Services,
            Paradyne Corporation, and the University
            of South Florida College of Engineering.

Important dates:
----------------
   Paper submission: May 19, 2000
   Notification of acceptance: July 14, 2000
   Camera-ready copy due: August 18, 2000

General Information:
--------------------

The IEEE LCN conference is the premier conference on leading edge and
practical computer networking.  The emphasis of this conference is on
practical solutions to important problems in computer networking.  During
the 25 years of this conference, we have moved from problems in the local
network to problems in the global Internet and World Wide Web.  Our unique
approach stimulates a workshop environment and enables an effective
interchange among researchers, users, and product developers.  We encourage
you to submit original papers describing research results or practical
solutions.  Paper topics include, but are not limited to:

- Local Area Networks
- Home Networks
- Wireless Networks
- Storage Area Networks
- Optical Networks
- Realtime Networks
- Active Networks
- ATM
- Gigabit Ethernet
- LAN/WAN Internetworking
- DSL Technologies
- Network Management
- Network Security
- Network Reliability
- Multicasting
- Enabling QoS in High-Speed Networks
- Always On / Always Connected
- Internet / Intranet
- Anything-over-IP
- IP-over-Anything
- Performance Evaluation
- Protocol Design and Validation

Authors are invited to submit full or short papers for presentation at
the conference.  Full papers should present novel perspectives on computer
networking within the general scope of the conference topics listed above
and may be up to 10 camera-ready pages in length.  Short papers are an
opportunity to present preliminary or interim results and are limited to 2
camera-ready pages in length.  All papers will be reviewed by a minimum of
three reviewers.  A best paper award will be presented. Several student
travel scholarships will be available courtesy of the LCN corporate
supporters.   All submitted papers must include title, complete contact
information for all authors, abstract, and keywords on the cover page.
The correspondence author must be clearly identified.

Paper Submission Instructions:
------------------------------

Email one postscript version of your manuscript to the program chair at
christen@csee.usf.edu.  Alternatively, send five hard copies via postal
mail to:

    Dr. Kenneth J. Christensen
    Department of Computer Science and Engineering
    4202 East Fowler Avenue, ENB 118
    University of South Florida
    Tampa, FL  33620

Tutorials:
----------

LCN 2000 will begin with one full day of tutorials.  Individuals interested
in giving a tutorial should contact the Tutorials Chair (Garry Kessler at
kessler@symquest.com).

LCN 2000 Committee:
-------------------

General Chair:
F. Huebner, AT&T (fhuebner@att.com)

Program Chairs:
K. Christensen, USF (christen@csee.usf.edu)
P. Martini, Univ. of Bonn (martini@informatik.uni-bonn.de)

IEEE Liaison:
J. Bumblis, Seagate (Joseph_R_Bumblis@notes.seagate.com)

Finance Chair:
K. Prasad, UMass-Lowell (Kanti_Prasad@uml.edu)

Tutorials Chair:
G. Kessler, SymQuest (kessler@symquest.com)

Panels Chairs:
J. W. Atwood, Concordia (bill@cs.concordia.ca)
T. Strayer, BBN (strayer@bbn.com)

Arrangements Chairs:
K. Christensen, USF (christen@csee.usf.edu)
R. Sankar, USF (sankar@csee.usf.edu)

Webmaster:
G. Kessler, SymQuest  (kessler@symquest.com)

Overseas Advisors:
S. Jha, UNSW, (Australia) (sjha@cse.unsw.edu.au)
P. Martini, Univ of Bonn, (Europe) (martini@informatik.uni-bonn.de)

Standing Committee:
M. McKee, Bowling Green
E. Nolley, Strategic Growth
H. Salwen, OpenRoute

---



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Date: Thu, 20 Apr 2000 15:15:22 -0700
From: Scott Keagy <scott@sknetworks.com>
Organization: SK Networks, Inc.
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To: Colin Perkins <c.perkins@cs.ucl.ac.uk>
CC: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, casner@cisco.com,
        rem-conf@es.net
Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
References: <200004202121.WAA13758@csperkins.demon.co.uk>
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Two questions:

1) Does SDP allow any RTP payload type to be dynamically defined,
   or only those in the "dynamic" range 96-127 ?

2) If SDP can re-redefine the payload types, then isn't this an inefficient

    way of using SDP & RTP for standard codecs ? The RTP payload field is
not
    adding value in this scheme, and extra lines are required in the SDP
description.
    (yeah SDP bandwidth is neglible compared to media, but it's the thought
that counts).

(Also, point noted by Orion re: G.723 & G.729; these questions of RTP
payload
    and SDP usage are still relevant in the case of G.729).


-Scott

Colin Perkins wrote:

> --> Scott Keagy writes:
> >So should an SDP advertisement for all G.723.1 options look like this?
> >
> > m=audio 20864 RTP/AVP 4 4 4 4
> > a=rtpmap 4 G723/5.3
> > a=rtpmap 4 G723/6.3
> > a=rtpmap 4 G723/5.3/A
> > a=rtpmap 4 G723/6.3/A
>
> If it's acceptable to have implementations which only support some
> varients, then a scheme like this makes sense (although the chosen
> payload type number should be different for each).
>
> Colin






From rem-conf Fri Apr 21 07:38:39 2000 
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Subject: Re: Question regarding SDP and MIME codec registration...
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"Fox, Mike" wrote:
> 
>         Nuera has a voice codec which encodes 64K PCM into one of 3 data
> rates, 7.47, 8.0, and 9.6 Kbps. We would like to register it with IANA in
> the Audio/Vendor tree as vnd.nuera.ecelp. Our intention is that the data
> rates are explicitly indicated as required parameters. So the MIME
> registration description will have the following:
> 
> Required parameters : <clock rate> one of 7470, 8000, or 9600.
> 
> And that the SDP RFC convention of:
> 
>      a=rtpmap:<payload type> <encoding name>/<clock rate>[/<encoding
>      parameters>]
> 
> would be met with the following SDP.
> 
> a=rtpmap:100 vnd.nuera.ecelp/7470.
> 
> Would this correct?
> Suggestions welcome.

Sounds about right. Would be useful to include some amount of detail in
the IANA registration (e.g., whether the format itself is described
somewhere - I suspect not...)

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



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From: Simao Campos <campos@ctd.comsat.com>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
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Henning,

I have attached now as plain text. Also re-sending to rem-conf, in
case someone also missed it.

Regards,
Simao.

------------------
> From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
> Sender: hgs@cs.columbia.edu
> To: Simao Campos <campos@ctd.comsat.com>
> Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
> Date: Fri, 21 Apr 2000 10:46:55 -0400
> 
> Your email attachment was empty.
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 

PACKETIZATION FOR G.723.1 AT 6.3 KBIT/S

    0                   1                   2                   3 
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | * |                      LPC                      |ACL0       |
   |   |                                               |           |
   |0 0|0 1 2 3 4 5 6 7 8 9 1 1 1 1 1 1 1 1 1 1 2 2 2 2|0 1 2 3 4 5|
   |   |                    0 1 2 3 4 5 6 7 8 9 0 1 2 3|           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

    3               4                   5                   6       
    2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | |ACL|    ACL2     |ACL|        GAIN0          |     GAIN1     |
   | | 1 |             | 3 |                       |               |
   |6|0 1|0 1 2 3 4 5 6|0 1|0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3 4 5 6 7|
   | |   |             |   |                    0 1|               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

    6           7                   8                   9           
    4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       |   GAIN2               |   GAIN3               | GRID  |
   |       |                       |                       |       |
   |8 9 1 1|0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3|
   |    0 1|                    0 1|                    0 1|       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

            1                   1                   1              
    9       0                   1                   2              
    6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |*|  MSBPOS                 |POS0                           |POS|
   |*|                         |                               | 1 |
   |0|0 1 2 3 4 5 6 7 8 9 1 1 1|0 1 2 3 4 5 6 7 8 9 1 1 1 1 1 1|0 1|
   | |                    0 1 2|                    0 1 2 3 4 5|   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

    1   1                   1                   1                  
    2   3                   4                   5                  
    8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |POS1                   |POS2                           |POS3   |
   |                       |                               |       |
   |2 3 4 5 6 7 8 9 1 1 1 1|0 1 2 3 4 5 6 7 8 9 1 1 1 1 1 1|0 1 2 3|
   |                0 1 2 3|                    0 1 2 3 4 5|       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

    1                   1                   1                   1  
    6                   7                   8                   9  
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                   |PSIG0      |PSIG1    |PSIG2      |PSIG3    |
   |                   |           |         |           |         |
   |4 5 6 7 8 9 1 1 1 1|0 1 2 3 4 5|0 1 2 3 4|0 1 2 3 4 5|0 1 2 3 4|
   |            0 1 2 3|           |         |           |         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

NOTES: 
(*)  Bits 0 and 1 are always "0 0" for the G.723.1 operation at 6.3
     kbit/s.
(**) Bit 96 is set to '0'


PACKETIZATION FOR G.723.1 AT 5.3 KBIT/S

    0                   1                   2                   3 
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | * |                      LPC                      |ACL0       |
   |   |                                               |           |
   |1 0|0 1 2 3 4 5 6 7 8 9 1 1 1 1 1 1 1 1 1 1 2 2 2 2|0 1 2 3 4 5|
   |   |                    0 1 2 3 4 5 6 7 8 9 0 1 2 3|           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

    3               4                   5                   6       
    2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | |ACL|    ACL2     |ACL|        GAIN0          |     GAIN1     |
   | | 1 |             | 3 |                       |               |
   |6|0 1|0 1 2 3 4 5 6|0 1|0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3 4 5 6 7|
   | |   |             |   |                    0 1|               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

    6           7                   8                   9           
    4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       |   GAIN2               |   GAIN3               | GRID  |
   |       |                       |                       |       |
   |8 9 1 1|0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3|
   |    0 1|                    0 1|                    0 1|       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

            1                   1                   1              
    9       0                   1                   2              
    6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |POS0                   |POS1                   |POS2           |
   |                       |                       |               |
   |0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3 4 5 6 7|
   |                    0 1|                    0 1|               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

    1   1                   1                   1                  
    2   3                   4                   5                  
    8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |POS2   |POS3                   |PSIG0  |PSIG1  |PSIG2  |PSIG3  |
   |       |                       |       |       |       |       |
   |8 9 1 1|0 1 2 3 4 5 6 7 8 9 1 1|0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
   |    0 1|2                   0 1|       |       |       |       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

NOTE: 
(*) Bits 0 and 1 are always "1 0" respectively for the G.723.1
    operation at 5.3 kbit/s.


PACKETIZATION FOR G.723.1 SID FRAMES

    0                   1                   2                   3 
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | * |                      LPC                      |GAIN       |
   |   |                                               |           |
   |0 1|0 1 2 3 4 5 6 7 8 9 1 1 1 1 1 1 1 1 1 1 2 2 2 2|0 1 2 3 4 5|
   |   |                    0 1 2 3 4 5 6 7 8 9 0 1 2 3|           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+




From rem-conf Fri Apr 21 08:50:43 2000 
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I should have read Mike's message more carefully - what I said is wrong,
as are the other examples given for SDP selection of codecs provided
earlier by others. Note that the SDP parameter is the *clock* rate, not
the bit rate. This parameter is included to allow receivers that do not
understand the details of the encoding to deal with RTP/RTCP information
measured in clock samples. (For example, an RTP recorder doesn't need to
understand the media encoding, but needs to know what the clock rate is
so that it can play back the recorded data without the network jitter.)

For all G.723 and G.729 varieties, the clock (sampling) rate is 8000 Hz.
Thus, different versions of the codec need to be labeled and cannot be
distinguished by the clock rate parameter. Thus, the rtpmap format would
be something like

a=rtpmap:100 vnd.nuera.ecelp7470/8000

Using the third parameter (as in vnd.nuera.ecelp/8000/7470) for the bit
rate is probably not a good idea since this parameter is reserved for
the number of audio channels. RFC 2327 also says explicitly
"Codec-specific parameters should be added in other attributes."

I would also argue that using the third parameter for the bit rate
complicates implementations, as they now have to look in three places to
create a table entry. Encoding it as part of the name seems easier,
particularly given that the number of rates seems to be small.

Hope this clarifies things.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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In-reply-to: Your message of "Fri, 21 Apr 2000 11:48:27 EDT."
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>For all G.723 and G.729 varieties, the clock (sampling) rate is 8000 Hz.
>Thus, different versions of the codec need to be labeled and cannot be
>distinguished by the clock rate parameter. Thus, the rtpmap format would
>be something like
>
>a=rtpmap:100 vnd.nuera.ecelp7470/8000
>
>Using the third parameter (as in vnd.nuera.ecelp/8000/7470) for the bit
>rate is probably not a good idea since this parameter is reserved for
>the number of audio channels. RFC 2327 also says explicitly
>"Codec-specific parameters should be added in other attributes."

Note that there is also a b= field in SDP specifically for specifying
bandwidths.  

Cheers,
	Mark



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Subject: Re: RTP payload types in draft-ietf-avt-profile-new-08.txt
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Scott Keagy wrote:
> 
> Two questions:
> 
> 1) Does SDP allow any RTP payload type to be dynamically defined,
>    or only those in the "dynamic" range 96-127 ?

This is not an SDP issue so much as an RTP issue. The new rfc1890 says:

> This profile reserves payload type numbers in the range 96-127
>    exclusively for dynamic assignment. Applications SHOULD first use
>    values in this range for dynamic payload types. Those applications
>    which need to define more than 32 dynamic payload types MAY bind
>    codes below 96, in which case it is RECOMMENDED that unassigned
>    payload type numbers be used first. However, the statically assigned
>    payload types are default bindings and MAY be dynamically bound to
>    new encodings if needed. Redefining payload types below 96 may cause
>    incorrect operation if an attempt is made to join a session without
>    obtaining session description information that defines the dynamic
>    payload types.

so, you are allowed to do this, although mostly as a last resort.

> 
> 2) If SDP can re-redefine the payload types, then isn't this an inefficient
> 
>     way of using SDP & RTP for standard codecs ? The RTP payload field is
> not
>     adding value in this scheme, and extra lines are required in the SDP
> description.
>     (yeah SDP bandwidth is neglible compared to media, but it's the thought
> that counts).

Huh? Of course you still need to have this. The payload type is needed
in the RTP to allow for the type to vary on the fly during the session.
This is essential for useful application of comfort noise codecs and of
the avt-tones mechanisms, both of which cause an on the fly change. You
still need the number in SDP so the other side knows what codec is
associated with which RTP payload type number.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From rem-conf Mon Apr 24 08:57:23 2000 
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: Mark Handley <mjh@aciri.org>, rem-conf@es.net
Subject: Re: Question regarding SDP and MIME codec registration...
In-Reply-To: Message from Henning Schulzrinne <schulzrinne@cs.columbia.edu> of "Fri, 21 Apr 2000 13:23:00 EDT." <39008E74.27ECC4B6@cs.columbia.edu>
Date: Mon, 24 Apr 2000 16:41:06 +0100
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[...rem-conf added back to the cc list...]

--> Henning Schulzrinne writes:
>Mark Handley wrote:
>> This wasn't obvious to me from the previous messages in this thread.
>> I agree that b= isn't useful for negotiation.  Yet another case where
>> SDP falls short because it doesn't have a grouping mechanism for
>> parameters of formats.
>
>Yes, we are definitely reaching the limits of the protocol.
>
>> 
>> There seem to be several alternatives
>> 
>>   a=rtpmap:100 vnd.nuera.ecelp7470/8000
>> 
>>   a=rtpmap:100 vnd.nuera.ecelp;bps=7470/8000
>> 
>>   a=rtpmap:100 vnd.nuera.ecelp/8000
>>   a=fmtp:100 bps=7470
>> 
>> The first overloads the MIME subtype.  The second overloads the format
>> name, by encoding a MIME parameter in it when used with SDP (SDP won't
>> care - it's just a token) but the MIME registration is clean.  The
>> last doesn't overload anything, but may not satisfy them that it's a
>> required parameter.
>
>None of them are great, I agree. Among the bunch, the second one is
>probably the least bad, in my view. (2327 doesn't say anything about
>what's legal in an 'encoding name', so that anything except a / would
>work. It's not really a token, in the strict sense, since tokens can't
>contain semicolons.)

The MIME registration for the standard codecs in the a/v profile
(draft-ietf-avt-rtp-mime-02.txt) uses the third option, with a 
statement that some parameters are required. 

Colin





>I don't think the third one would work as it now requires that the
>negotiation code goes fishing elsewhere. You'd end up with something
>like
>
>a=rtpmap:100 vnd.nuera.ecelp/8000
>a=rtpmap:101 vnd.nuera.ecelp/8000
>a=rtpmap:102 vnd.nuera.ecelp/8000
>
>a=fmtp:100 bps=7470
>a=fmtp:101 bps=8000
>a=fmtp:102 bps=9600
>
>which is likely to be very confusing.
>
>
>
>
>
>> 
>> None of these really seem all that satisfactory, but I can't think of
>> any other good alternatives.
>> 
>> Cheers,
>>         Mark
>
>-- 
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Mon Apr 24 13:31:18 2000 
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Thanks for the clarification-

I overlooked  tones & comfort noise.

The only issues I have left are vendor-specific :^)

-Scott

Jonathan Rosenberg wrote:

> Scott Keagy wrote:
> >
> > Two questions:
> >
> > 1) Does SDP allow any RTP payload type to be dynamically defined,
> >    or only those in the "dynamic" range 96-127 ?
>
> This is not an SDP issue so much as an RTP issue. The new rfc1890 says:
>
> > This profile reserves payload type numbers in the range 96-127
> >    exclusively for dynamic assignment. Applications SHOULD first use
> >    values in this range for dynamic payload types. Those applications
> >    which need to define more than 32 dynamic payload types MAY bind
> >    codes below 96, in which case it is RECOMMENDED that unassigned
> >    payload type numbers be used first. However, the statically assigned
> >    payload types are default bindings and MAY be dynamically bound to
> >    new encodings if needed. Redefining payload types below 96 may cause
> >    incorrect operation if an attempt is made to join a session without
> >    obtaining session description information that defines the dynamic
> >    payload types.
>
> so, you are allowed to do this, although mostly as a last resort.
>
> >
> > 2) If SDP can re-redefine the payload types, then isn't this an inefficient
> >
> >     way of using SDP & RTP for standard codecs ? The RTP payload field is
> > not
> >     adding value in this scheme, and extra lines are required in the SDP
> > description.
> >     (yeah SDP bandwidth is neglible compared to media, but it's the thought
> > that counts).
>
> Huh? Of course you still need to have this. The payload type is needed
> in the RTP to allow for the type to vary on the fly during the session.
> This is essential for useful application of comfort noise codecs and of
> the avt-tones mechanisms, both of which cause an on the fly change. You
> still need the number in SDP so the other side knows what codec is
> associated with which RTP payload type number.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

--
=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================





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From rem-conf Mon Apr 24 13:58:24 2000 
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I don't have much hope, but just in case: Is there source or Unix (any
flavor) object code for either G.723.1 or G.729 **other than the
slow-as-molasses ITU code** for internal research use? (I'm aware of the
patent issues.)
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Mon Apr 24 14:07:20 2000 
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Colin Perkins wrote:
> 

> >>   a=rtpmap:100 vnd.nuera.ecelp/8000
> >>   a=fmtp:100 bps=7470
> >>
> >> The first overloads the MIME subtype.  The second overloads the format
> >> name, by encoding a MIME parameter in it when used with SDP (SDP won't
> >> care - it's just a token) but the MIME registration is clean.  The
> >> last doesn't overload anything, but may not satisfy them that it's a
> >> required parameter.
> >
> >None of them are great, I agree. Among the bunch, the second one is
> >probably the least bad, in my view. (2327 doesn't say anything about
> >what's legal in an 'encoding name', so that anything except a / would
> >work. It's not really a token, in the strict sense, since tokens can't
> >contain semicolons.)
> 
> The MIME registration for the standard codecs in the a/v profile
> (draft-ietf-avt-rtp-mime-02.txt) uses the third option, with a
> statement that some parameters are required.
> 

Hmm. I still think this is a really bad idea.
draft-ietf-avt-rtp-mime-02.txt doesn't really mention parameters that
are necessary to do media type and capability negotiation. None of the
registrations included describe this case that a single MIME type
describes two different encodings (L16 does, but the parameter, the
clock rate, is part of the rtpmap parameter). I think this should be
fixed before it's too late. This should only affect a few encodings.
    

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



From rem-conf Mon Apr 24 20:59:48 2000 
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Date: Mon, 24 Apr 2000 20:51:56 -0700
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Colin Perkins <c.perkins@cs.ucl.ac.uk>
From: Jeff Ayars <jeffa@real.com>
Subject: Re: Question regarding SDP and MIME codec registration...
Cc: Mark Handley <mjh@aciri.org>, rem-conf@es.net
In-Reply-To: <3904B76F.650146F6@cs.columbia.edu>
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At 05:06 PM 4/24/00 -0400, Henning Schulzrinne wrote:
>Colin Perkins wrote:
> >
>
> > >>   a=rtpmap:100 vnd.nuera.ecelp/8000
> > >>   a=fmtp:100 bps=7470
> > >>
> >
> > The MIME registration for the standard codecs in the a/v profile
> > (draft-ietf-avt-rtp-mime-02.txt) uses the third option, with a
> > statement that some parameters are required.
> >
>
>Hmm. I still think this is a really bad idea.
>draft-ietf-avt-rtp-mime-02.txt doesn't really mention parameters that
>are necessary to do media type and capability negotiation. None of the
>registrations included describe this case that a single MIME type
>describes two different encodings (L16 does, but the parameter, the
>clock rate, is part of the rtpmap parameter). I think this should be
>fixed before it's too late. This should only affect a few encodings.
>

Using fmtp is how I would have expected it to be done.  I think using the 
rtpmap <optional parameters> part is a interop nightmare.  What about RFC 
2361?  If I want to send Microsoft G.723 from a .wav file to a player that 
played audio using ACM codecs I think

m= audio 0 RTP/AVP 101
a=rtpmap: 101 vnd.wav/22050/1
a=fmtp: 101 codec=42

is the right way to do it.


JEff




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From rem-conf Tue Apr 25 06:44:46 2000 
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Date: Tue, 25 Apr 2000 09:40:26 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Jeff Ayars <jeffa@real.com>
CC: Colin Perkins <c.perkins@cs.ucl.ac.uk>, Mark Handley <mjh@aciri.org>,
        rem-conf@es.net
Subject: Re: Question regarding SDP and MIME codec registration...
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Jeff Ayars wrote:
> 
> At 05:06 PM 4/24/00 -0400, Henning Schulzrinne wrote:
> >Colin Perkins wrote:
> > >
> >
> > > >>   a=rtpmap:100 vnd.nuera.ecelp/8000
> > > >>   a=fmtp:100 bps=7470
> > > >>
> > >
> > > The MIME registration for the standard codecs in the a/v profile
> > > (draft-ietf-avt-rtp-mime-02.txt) uses the third option, with a
> > > statement that some parameters are required.
> > >
> >
> >Hmm. I still think this is a really bad idea.
> >draft-ietf-avt-rtp-mime-02.txt doesn't really mention parameters that
> >are necessary to do media type and capability negotiation. None of the
> >registrations included describe this case that a single MIME type
> >describes two different encodings (L16 does, but the parameter, the
> >clock rate, is part of the rtpmap parameter). I think this should be
> >fixed before it's too late. This should only affect a few encodings.
> >
> 
> Using fmtp is how I would have expected it to be done.  I think using the
> rtpmap <optional parameters> part is a interop nightmare.  What about RFC
> 2361?  If I want to send Microsoft G.723 from a .wav file to a player that
> played audio using ACM codecs I think
> 
> m= audio 0 RTP/AVP 101
> a=rtpmap: 101 vnd.wav/22050/1
> a=fmtp: 101 codec=42
> 
> is the right way to do it.

I disagree. This is an implementation nightmare, since now the
negotiation code (which is often not the media agent) has to be aware of
all the possible codecs and their parameters that influence
interoperability. As long as I have names, I can easily export a table
of capabilities from the media agent, without having to grope through
assorted fmtp's.

If two codecs are different and not distinguishable at the bit-level
(such as G.723.1) and are usable independently (unlike G.723.1, where
the claim is that the ITU protocol police will pull you over if you
support 5.6 kb/s without supporting 6.3 kb/s), they deserve different
names, even though they might be lumped together in the same "marketing"
name. They have different bit patterns, different software for decoding
it - thus, for all technical purposes, they are different (albeit
technically related) codecs. If (to cite a recent example) Nuera had
chosen to name their codecs Alpha, Beta and Gamma, we wouldn't hesitate
a moment to register them under different names, even though they might
share some fundamental principles of audio signal compression.

Also, I don't think that using Wave codec names is the right solution,
as it is heavily OS dependent. Designation of codec 42 is meaningless to
those not running ACM. We have codec names, such as G.729, that are
standardized and widely recognized, with appropriate payload formats.
Note that codec 42 makes no sense since it does not describe the payload
format itself, just a file format. In all but trivial cases (like
mu-law), these are not likely to be the same.

> 
> JEff

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



From rem-conf Tue Apr 25 07:37:37 2000 
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To: rem-conf@es.net
From: position.Offers.USA@Freelance.com
Subject: Freelance.com
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Date: Tue, 25 Apr 2000 16:22:06 +0200
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From rem-conf Tue Apr 25 09:06:31 2000 
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Date: Tue, 25 Apr 2000 09:07:15 -0700
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Ashok Yerneni <ashok@graham.com>
Subject: Information about SAP and SDP protocols...
Cc: Jeff Ayars <jeffa@real.com>, Colin Perkins <c.perkins@cs.ucl.ac.uk>,
        Mark Handley <mjh@aciri.org>, rem-conf@es.net
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Where do I get detailed information about these protocols. Is there a 
public domain
library that implements these protocols?

thanks,

Ashok
Ashok Yerneni				(408) 366 8001,x501
VP of Engineering				(408) 366 8004(fax)
Graham Technology Solutions			ashok@graham.com
20823 Stevens Creek Blvd, 			www.graham.com
Cupertino, CA 95014-2111.




From rem-conf Tue Apr 25 10:04:26 2000 
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Subject: 4/26 Integrating Timing into XML Documents -- Patrick Schmitz,
  Microsoft Bay Area Research Center 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Integrating Timing into XML Documents

Wednesday April 26, 2000, 
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

Patrick Schmitz
Microsoft Bay Area Research Center 

Growing interest in SMIL underscores the importance of a common language to
describe timing and synchronization in documents. SMIL 1 defined syntax and
semantics for a basic multimedia language. This was an important first
step, but does not allow authors to use the timing semantics with other
languages, like HTML and CSS. Authors need to be able to describe timing
for wide range of documents, and to combine the semantics of timing and
sync with other existing tools and semantics that are in common use, such
as CSS.  In addition, they need the flexibility to accommodate different
editing/authoring scenarios and constraints.  

We need to provide a common model for timing so that authors are not forced
to learn and remember different models for different document types or
different authoring scenarios. In addition, where different syntactic
approaches are available (e.g. to address different authoring scenarios),
it is important to allow authors to combine the different approaches in a
straightforward manner. 

This talk will describe three approaches to the integration of timing and
synchronization with XML languages. I will describe the current work in
SMIL Boston to support the integration of SMIL with HTML/CSS, as well as
other applications and future work. I will also describe a proposed
framework for combining the approaches. 
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Tue Apr 25 10:28:35 2000 
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Date: Tue, 25 Apr 2000 13:37:02 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Ashok Yerneni <ashok@graham.com>
CC: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Jeff Ayars <jeffa@real.com>, Colin Perkins <c.perkins@cs.ucl.ac.uk>,
        Mark Handley <mjh@aciri.org>, rem-conf@es.net
Subject: Re: Information about SAP and SDP protocols...
References: <200004241541.QAA23980@csperkins.demon.co.uk>
	 <4.2.0.58.20000424203610.044ccdc0@mail.real.com> <4.3.1.0.20000425090551.00a7cf00@yoyodyne.graham.com>
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SAP:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sap-v2-06.txt

SDP:
rfc2327, http://www.ietf.org/rfc/rfc2327.txt

-Jonathan R.

Ashok Yerneni wrote:
> 
> Where do I get detailed information about these protocols. Is there a
> public domain
> library that implements these protocols?
> 
> thanks,
> 
> Ashok
> Ashok Yerneni                           (408) 366 8001,x501
> VP of Engineering                               (408) 366 8004(fax)
> Graham Technology Solutions                     ashok@graham.com
> 20823 Stevens Creek Blvd,                       www.graham.com
> Cupertino, CA 95014-2111.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From rem-conf Tue Apr 25 12:23:50 2000 
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Jeff Ayars <jeffa@real.com>
Subject: Re: Question regarding SDP and MIME codec registration...
Cc: Colin Perkins <c.perkins@cs.ucl.ac.uk>, Mark Handley <mjh@aciri.org>,
        rem-conf@es.net
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Comments inline...

At 09:40 AM 4/25/00 -0400, Henning Schulzrinne wrote:
>Jeff Ayars wrote:
> >
> > At 05:06 PM 4/24/00 -0400, Henning Schulzrinne wrote:
> > >Colin Perkins wrote:
> > > >
> > >
> > > > >>   a=rtpmap:100 vnd.nuera.ecelp/8000
> > > > >>   a=fmtp:100 bps=7470
> > > > >>
> > > >
> > > > The MIME registration for the standard codecs in the a/v profile
> > > > (draft-ietf-avt-rtp-mime-02.txt) uses the third option, with a
> > > > statement that some parameters are required.
> > > >
> > >
> > >Hmm. I still think this is a really bad idea.
> > >draft-ietf-avt-rtp-mime-02.txt doesn't really mention parameters that
> > >are necessary to do media type and capability negotiation. None of the
> > >registrations included describe this case that a single MIME type
> > >describes two different encodings (L16 does, but the parameter, the
> > >clock rate, is part of the rtpmap parameter). I think this should be
> > >fixed before it's too late. This should only affect a few encodings.
> > >
> >
> > Using fmtp is how I would have expected it to be done.  I think using the
> > rtpmap <optional parameters> part is a interop nightmare.  What about RFC
> > 2361?  If I want to send Microsoft G.723 from a .wav file to a player that
> > played audio using ACM codecs I think
> >
> > m= audio 0 RTP/AVP 101
> > a=rtpmap: 101 vnd.wav/22050/1
> > a=fmtp: 101 codec=42
> >
> > is the right way to do it.
>
>I disagree. This is an implementation nightmare, since now the
>negotiation code (which is often not the media agent) has to be aware of
>all the possible codecs and their parameters that influence
>interoperability. As long as I have names, I can easily export a table
>of capabilities from the media agent, without having to grope through
>assorted fmtp's.

So you are saying that it's your wish that a=fmtp in SDP be explicitly 
excluded from consideration during any format negotiation phase?  I can see 
that makes it easier.

>If two codecs are different and not distinguishable at the bit-level
>(such as G.723.1) and are usable independently (unlike G.723.1, where
>the claim is that the ITU protocol police will pull you over if you
>support 5.6 kb/s without supporting 6.3 kb/s), they deserve different
>names, even though they might be lumped together in the same "marketing"
>name. They have different bit patterns, different software for decoding
>it - thus, for all technical purposes, they are different (albeit
>technically related) codecs. If (to cite a recent example) Nuera had
>chosen to name their codecs Alpha, Beta and Gamma, we wouldn't hesitate
>a moment to register them under different names, even though they might
>share some fundamental principles of audio signal compression.

This is why in draft-ietf-avt-profile-new-08.txt the G.726 
bitrates/bitdepths are nudged in the direction of G726-16, G726-24, 
G726-32, and G726-40?

>Also, I don't think that using Wave codec names is the right solution,
>as it is heavily OS dependent. Designation of codec 42 is meaningless to
>those not running ACM. We have codec names, such as G.729, that are
>standardized and widely recognized, with appropriate payload formats.
>Note that codec 42 makes no sense since it does not describe the payload
>format itself, just a file format. In all but trivial cases (like
>mu-law), these are not likely to be the same.

I don't know that Wave codec names necessarily implies any OS 
dependency.  Sure the codec name is defined for a particular OS but the 
codec may exist on multiple platforms - Indio by Intel, Cenepac by 
Supermac.  There is a standard that registers a mime type for these codecs 
- RFC 2361.  It uses parameters to the general mime type to identify the 
encoding type.

Are you suggesting that SDP not support announcing sessions that use codecs 
defined in RFC 2361?

 From RFC 2361:

"The purpose of this paper is to establish a mechanism by which codecs 
registered within Microsoft's WAVE and AVI Registries may be referenced 
within the IANA Namespace by Internet applications."

So RFC 2361 isn't tied to a file format, but to the codec - the data type.

Or are you saying that every data type needs an explicit mime type 
registered for it's RTP payload format so if I wanted to send Cenepac via 
RTP I would need to register a new mime type, different from video/vnd.avi; 
codec=CVID, for to send it via RTP and that as above, I should include the 
codec in the mime name instead of as a parameter?

JEff





From rem-conf Tue Apr 25 18:43:05 2000 
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> 
> So you are saying that it's your wish that a=fmtp in SDP be explicitly
> excluded from consideration during any format negotiation phase?  I can see
> that makes it easier.

Yes, fmtp should be used to *configure* a codec, but not to decide
whether it's compatible. In my implementation model, the SIP agent (say)
would simply pass down the fmtp content to the media agent, completely
oblivious what it means. It simply gathers the 
advertised receive capabilities = codecs A, B, C, D (from the m and
rtpmap lines) and, for example, pops up a GUI panel of the sort after
subtracting the codecs that it doesn't know anything about: "The caller
can receive audio A, B, C (suitably non-techie descriptions). Which ones
do you want to use?" and then configures the media agent. At that point,
the GUI part can easily handle codecs that it has never heard about, as
long as it is given a simple list of names describing the media agent
capabilities.


> This is why in draft-ietf-avt-profile-new-08.txt the G.726
> bitrates/bitdepths are nudged in the direction of G726-16, G726-24,
> G726-32, and G726-40?

Yes, although I have to admit that the recent discussion has emphasized
the need for this even more.

> 
> >Also, I don't think that using Wave codec names is the right solution,
> >as it is heavily OS dependent. Designation of codec 42 is meaningless to
> >those not running ACM. We have codec names, such as G.729, that are
> >standardized and widely recognized, with appropriate payload formats.
> >Note that codec 42 makes no sense since it does not describe the payload
> >format itself, just a file format. In all but trivial cases (like
> >mu-law), these are not likely to be the same.
> 
> I don't know that Wave codec names necessarily implies any OS
> dependency.  Sure the codec name is defined for a particular OS but the
> codec may exist on multiple platforms - Indio by Intel, Cenepac by
> Supermac.  There is a standard that registers a mime type for these codecs
> - RFC 2361.  It uses parameters to the general mime type to identify the
> encoding type.
> 
> Are you suggesting that SDP not support announcing sessions that use codecs
> defined in RFC 2361?
> 
>  From RFC 2361:
> 
> "The purpose of this paper is to establish a mechanism by which codecs
> registered within Microsoft's WAVE and AVI Registries may be referenced
> within the IANA Namespace by Internet applications."
> 
> So RFC 2361 isn't tied to a file format, but to the codec - the data type.
> 
> Or are you saying that every data type needs an explicit mime type
> registered for it's RTP payload format so if I wanted to send Cenepac via
> RTP I would need to register a new mime type, different from video/vnd.avi;
> codec=CVID, for to send it via RTP and that as above, I should include the
> codec in the mime name instead of as a parameter?

Yes, as discussed in the RTP MIME registration document, just having a
file format description is not sufficient for RTP transport. We can now
do one of two things:

- make sure that there's an interoperable RTP packetization for each of
the video/vnd.avi codecs (probably a useful exercise)

- map these to the 'explicit' RTP codec names where possible to avoid
the former.


> 
> JEff



From rem-conf Wed Apr 26 04:43:18 2000 
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Subject: New Proto
Date: Wed, 26 Apr 2000 13:36:11 +0200
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This is a multi-part message in MIME format.

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Hello, I'm new to this mailing list.
My name is Paolo Possanzini and I'm from Italy.
I'm working to a new protocol to see Video On InterNet. I works with =
UDP.=20
I need someone to help me to talk about it. I'm interested to your =
opinion on my Idea.
I'm sorry for my bad english.

Bye

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<DIV><FONT face=3DArial size=3D2>Hello, I'm new to this mailing =
list.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>My name is Paolo Possanzini and I'm =
from=20
Italy.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I'm working to a new protocol to see =
Video On=20
InterNet. I works with UDP. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I need someone to help me to talk about =
it. I'm=20
interested to your opinion on my Idea.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I'm sorry for my bad =
english.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Bye</FONT></DIV></BODY></HTML>

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From rem-conf Wed Apr 26 07:33:36 2000 
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Date: Wed, 26 Apr 2000 10:28:58 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Art Yerkes wrote:
> 
> For G.723, http://www.cwi.nl/ftp/audio/ccitt-adpcm.tar.gz.  These
> sources come from sun.  Not sure if they're any faster, though.

Thanks for all the responses. A brief summary:

- The ITU floating point code for G.729 is significantly faster than the
fixed-point code. Decoding takes about 8% of a SPARC Ultra CPU, encoding
about a third.

- The term G.723 is the subject of endless confusion. The source cited
above is the "old" G.723. The RTP profile explains it thusly:

"Note: In 1990, ITU-T Recommendation G.721 was merged with
Recommendation
G.723 into ITU-T Recommendation G.726. Thus, G726-32 designates the same
algorithm as G721 in RFC 1890."

Most people (myself included) seem to use the term G.723 now for what's
more precisely called G.723.1, the 5.3/6.3 kb/s codec.

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



From rem-conf Wed Apr 26 07:39:37 2000 
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From: jhall@UU.NET (Jeremy Hall)
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Subject: audio coverage of space shuttle mission STS-101
To: mbone@isi.edu
Date: Wed, 26 Apr 2000 10:36:57 -0400 (EDT)
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Hi,

Out of curiosity, is anybody interested in mp3 streams of shuttle audio
during space missions?

_J



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To: Art Yerkes <ayerkes@genxnet.com>
Cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
    rem-conf@es.net
Subject: Re: G723
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Art,

Yes, G.723 is a "vacant" recommendation number now. For a short period
of time, experts referred to the 6.3/5.3 kb/s ACELP/MP-LPC ITU codec
as G.723, but this was quickly changed to G.723.1, to avoid confusion
with the 24, 40 kb/s ADPCM G.723 spec. So any "modern" reference to
G.723 has been to actually designate G.723.1. Do you feel we need more
clear text on that regard in RTP profile?

BTW, the code from Sun in ccitt-adpcm.tar.gz does not process the ITU
test vectors correctly, hence it cannot be claimed to comply to the
ITU Standard (Note for newbies: the ITU-T was formerly known as
CCITT). The code in the ITU Software Tool Library does, and runs very
fast on today's CPUs.

Simao.

>>>>> "Henning" == Henning Schulzrinne <schulzrinne@cs.columbia.edu> writes:

    Henning> Art Yerkes wrote:
    >> 
    >> For G.723, http://www.cwi.nl/ftp/audio/ccitt-adpcm.tar.gz.  These
    >> sources come from sun.  Not sure if they're any faster, though.

    Henning> Thanks for all the responses. A brief summary:

    Henning> - The ITU floating point code for G.729 is significantly faster than the
    Henning> fixed-point code. Decoding takes about 8% of a SPARC Ultra CPU, encoding
    Henning> about a third.

    Henning> - The term G.723 is the subject of endless confusion. The source cited
    Henning> above is the "old" G.723. The RTP profile explains it thusly:

    Henning> "Note: In 1990, ITU-T Recommendation G.721 was merged with
    Henning> Recommendation
    Henning> G.723 into ITU-T Recommendation G.726. Thus, G726-32 designates the same
    Henning> algorithm as G721 in RFC 1890."

    Henning> Most people (myself included) seem to use the term G.723 now for what's
    Henning> more precisely called G.723.1, the 5.3/6.3 kb/s codec.

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




From rem-conf Wed Apr 26 09:24:04 2000 
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Subject: 4/26 SMIL/XML ... -- Patrick Schmitz, Microsoft Bay Area
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Berkeley Multimedia, Graphics and Interfaces Seminar

SMIL/XML ...

Wednesday April 26, 2000 
1:10-2:30 p.m. PST
Fujitsu Seminar Room (405 Soda Hall) 

Patrick Schmitz
Microsoft Bay Area Research Center 

Growing interest in SMIL underscores the importance of a common language to
describe timing and synchronization in documents. SMIL 1 defined syntax and
semantics for a basic multimedia language. This was an important first
step, but does not allow authors to use the timing semantics with other
languages, like HTML and CSS. Authors need to be able to describe timing
for wide range of documents, and to combine the semantics of timing and
sync with other existing tools and semantics that are in common use, such
as CSS.  In addition, they need the flexibility to accommodate different
editing/authoring scenarios and constraints.  

We need to provide a common model for timing so that authors are not forced
to learn and remember different models for different document types or
different authoring scenarios. In addition, where different syntactic
approaches are available (e.g. to address different authoring scenarios),
it is important to allow authors to combine the different approaches in a
straightforward manner. 

This talk will describe three approaches to the integration of timing and
synchronization with XML languages. I will describe the current work in
SMIL Boston to support the integration of SMIL with HTML/CSS, as well as
other applications and future work. I will also describe a proposed
framework for combining the approaches. 
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Wed Apr 26 10:43:44 2000 
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Subject: Re: audio coverage of space shuttle mission STS-101
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Jeremy Hall wrote:
> 
> Hi,
> 
> Out of curiosity, is anybody interested in mp3 streams of shuttle audio
> during space missions?
> 
Would this be accompanied with MPEG-1,MPEG-2 or MPEG-4 video?

Pete



From rem-conf Wed Apr 26 11:19:39 2000 
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From: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
To: <rem-conf@es.net>
Subject: Echo Cancellers 
Date: Wed, 26 Apr 2000 11:09:47 -0700
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Hi,
  I am interested in knowing what kind of echo cancellers do we require in
typical VoIP applications. Is there a requirement for delays greater than
64msecs?

Regards
Hari





From rem-conf Wed Apr 26 12:10:00 2000 
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From: "Michael F. Speer" <speer@valathar.eng.sun.com>
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Below is 2 URL's announcing a RTSP/RTP streaming solution from Sun.  One
URL is the press release and the other is a pointer to where one can
find a try and buy URL.

http://www.sun.com/smi/Press/sunflash/2000-04/sunflash.20000410.2.html

http://www.sun.com/storage/

Regards,
Michael




From rem-conf Wed Apr 26 12:45:12 2000 
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In-Reply-To: <39072988.44677606@kpnqwest.fi> from Petri Helenius at "Apr 26,
 2000 08:38:16 pm"
To: Petri Helenius <pete@kpnqwest.fi>
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not by me. :)

In the new year, Petri Helenius wrote:
> Jeremy Hall wrote:
> > 
> > Hi,
> > 
> > Out of curiosity, is anybody interested in mp3 streams of shuttle audio
> > during space missions?
> > 
> Would this be accompanied with MPEG-1,MPEG-2 or MPEG-4 video?
> 
> Pete
> 




From rem-conf Wed Apr 26 12:51:11 2000 
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From: Scott Keagy <scott@sknetworks.com>
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I think there is a good application for high-delay echo cancellers.
Consider the following scenarios:

	<user A>----<VoIP network>----<PSTN>----<mobile network>---<userB>
or
	<VoIP network>----<PSTN>-------<VoIP network>
or
	<VoIP network>----<any high delay hop-off to PSTN>

Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
routers) only provide 32 msec of coverage. If a hop-off tail circuit has
a round-trip delay that exceeds the coverage of the echo canceller (on
the outbound edge of the VoIP network), then the additional delay of the
VoIP network (100 msec or more usually) exacerbates the echo problem.

This is an intractable problem without the use of echo cancellers that
have longer coverage ability.


-Scott


Hari Krishna Vuppaladhadiam wrote:
> 
> Hi,
>   I am interested in knowing what kind of echo cancellers do we require in
> typical VoIP applications. Is there a requirement for delays greater than
> 64msecs?
> 
> Regards
> Hari

-- 
=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant  
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================



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Scott:

Thanks for you input to the group on this subject.  There seems to be a
considerable amount of misunderstanding about echo control with VoIP networks.
I repeat a label your networks below to make some points of discussion:

Case 1:
     <user A>----<VoIP network>----<PSTN>----<mobile network>---<userB>

Case 2:


     <VoIP network>----<PSTN>-------<VoIP network>

Case 3:

     <VoIP network>----<any high delay hop-off to PSTN>

In case 1, user A has a four-wire connection to the network and user B also has
a four-wire connection to the network.  That is - unless somewhere in the mobile
network or in the PSTN there is some two wire circuits.  Unless user A is using
a poor quality speakerphone, there should be no need for any echo cancellation
at all.

for case 2, there should also be no need for any echo cancellers unless, again,
the PSTN has been poorly engineered and has two-wire connections somewhere.

For case 3, it the right side of the network has a Media Gateway.  At some point
to the right of that gateway there will be a two-four wire hybrid which will be
the source of an echo.  The echo would be user a "talker echo".  Therefore, the
media gateway has ther responsibility of echo cancelling this echo.  This is
typicaly short - as in certainly less than 20 msec.  It is true - that the
longer the VoIP network delay, the better the echo canceller in the network must
work.  This is why it is important for VoIP networks to manage their delay
downwards as much as possible.  To do otherwise causes a) the media gateway echo
canceller to "work harder" than it should have to and b) cause conversational
double-talking that is subjectively annoying even when the echo control is
perfect.

The basic misimpression that I see all too often is that people believe that
"station side" or subscriber equipment echo cancellers need to be deployed at
the subscriber or user's equipment.  This is, for the most part, untrue.  If it
turns out to be true, someone architected their network poorly.  Echo cancellers
with long tail coverage are an expensive commitment to poor network design.

If I recall correctly, even with normal characteristic impedances in cable (free
space is faster) a tail circuit across the country will not exceed 20 msecs.

Kevin





Scott Keagy <scott@sknetworks.com> on 04/26/2000 03:48:08 PM

To:   Hari Krishna Vuppaladhadiam <hariv@chiplogic.com>
cc:   rem-conf@es.net (bcc: Kevin Bruemmer/Natural MicroSystems/US)
Subject:  Re: Echo Cancellers




I think there is a good application for high-delay echo cancellers.
Consider the following scenarios:

     <user A>----<VoIP network>----<PSTN>----<mobile network>---<userB>
or
     <VoIP network>----<PSTN>-------<VoIP network>
or
     <VoIP network>----<any high delay hop-off to PSTN>

Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
routers) only provide 32 msec of coverage. If a hop-off tail circuit has
a round-trip delay that exceeds the coverage of the echo canceller (on
the outbound edge of the VoIP network), then the additional delay of the
VoIP network (100 msec or more usually) exacerbates the echo problem.

This is an intractable problem without the use of echo cancellers that
have longer coverage ability.


-Scott


Hari Krishna Vuppaladhadiam wrote:
>
> Hi,
>   I am interested in knowing what kind of echo cancellers do we require in
> typical VoIP applications. Is there a requirement for delays greater than
> 64msecs?
>
> Regards
> Hari

--
=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================








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Kevin et al,

feel free to correct me if I'm wrong in the details below....

I added more details to the diagrams to clarify what we're talking
about.
<see diagrams at bottom>


ITU Echo Canceller Requirements
-------------------------------

ITU-T Recommendation G.131 "Control of Talker Echo" (section 4.1) states
the following:
"In general, it is recommended that the active echo control devices
shall be deployed on all connections which exceed the total one-way
talker echo transmission path time of 25 ms."

This implies that public phone networks may not provide echo
cancellation for tail-end hop-off circuits that have up to 50 msec
round-trip latency.


Where echo originates
---------------------
Impedance mismatches at 2-wire/4-wire hybrids (transitions from 2-wire
to 4-wire circuits) can cause signal reflections. The likely spots where
these hybrids can cause problems for VoIP hop-off scenarios are
illustrated below. Carrier equipment may not be optimally tuned, or
customer equipment may not be optimally tuned. In addition to these
spots, acoustic echoes may occur from the destination speaker/mic
equipment (i.e. speakerphone or cheap telephone handset).

I submit as proof that not all networks are ideally designed, that I
frequently call residential locations in India from the U.S. I here my
own echo on at least half of the conversations, which implies that some
telco central offices in India have poor hybrid terminations.


What a company can do about it
-------------------------------
Given the ITU recommendations that support a carrier's right to not
employ echo cancellers in medium-delay networks (i.e. up to 50 msec
round-trip), a company employing a VoIP solution has no recourse to
block the echo other than to use echo cancellers with high delay
coverage, as illustrated below. A given company can't fix the network
problems for different calling destinations (i.e. adjust the impedance
coupling at the hybrid to increase the return loss), but it can take
personal action to defend the voice quality experienced by its
employees.

Note that this is an enterprise-centric approach to the problem. A
carrier-centric approach might be to negotiate with the remote telco to
fix the problem, so that the local carrier does not get stuck with the
echo-canceller bill. So high delay echo cancellers may be useful for
low-end and enterprise products, but not for carrier products. Also note
that up to 64 msec is sufficient given that all networks are designed
according to ITU spec.


==== : 4-wire circuit
---- : 2-wire circuit

**************************************************************************
                                                  ||=====<digital user
C>
                                                  ||
                                                  ||
<user A>=====<VoIP cloud>===<long delay PSTN>====<PBX>-----<analog user
B>
                        ^                             ^
                        |                             |
                        |                             |
                        |                 (signal from A reflected here)
                        |
                   (High delay echo canceller needed here)

**************************************************************************

<user A>=====<VoIP cloud>=====<long delay PSTN>-------<user B>
                        ^                     ^
                        |                     |
                        |                     |
                        |            (signal from A reflected here)
                        |
                   (High delay echo canceller needed here)

***************************************************************************

-Scott

Kevin_Bruemmer@nmss.com wrote:
> 
> Scott:
> 
> Thanks for you input to the group on this subject.  There seems to be a
> considerable amount of misunderstanding about echo control with VoIP networks.
> I repeat a label your networks below to make some points of discussion:
> 
> Case 1:
>      <user A>----<VoIP network>----<PSTN>----<mobile network>---<userB>
> 
> Case 2:
> 
>      <VoIP network>----<PSTN>-------<VoIP network>
> 
> Case 3:
> 
>      <VoIP network>----<any high delay hop-off to PSTN>
> 
> In case 1, user A has a four-wire connection to the network and user B also has
> a four-wire connection to the network.  That is - unless somewhere in the mobile
> network or in the PSTN there is some two wire circuits.  Unless user A is using
> a poor quality speakerphone, there should be no need for any echo cancellation
> at all.
> 
> for case 2, there should also be no need for any echo cancellers unless, again,
> the PSTN has been poorly engineered and has two-wire connections somewhere.
> 
> For case 3, it the right side of the network has a Media Gateway.  At some point
> to the right of that gateway there will be a two-four wire hybrid which will be
> the source of an echo.  The echo would be user a "talker echo".  Therefore, the
> media gateway has ther responsibility of echo cancelling this echo.  This is
> typicaly short - as in certainly less than 20 msec.  It is true - that the
> longer the VoIP network delay, the better the echo canceller in the network must
> work.  This is why it is important for VoIP networks to manage their delay
> downwards as much as possible.  To do otherwise causes a) the media gateway echo
> canceller to "work harder" than it should have to and b) cause conversational
> double-talking that is subjectively annoying even when the echo control is
> perfect.
> 
> The basic misimpression that I see all too often is that people believe that
> "station side" or subscriber equipment echo cancellers need to be deployed at
> the subscriber or user's equipment.  This is, for the most part, untrue.  If it
> turns out to be true, someone architected their network poorly.  Echo cancellers
> with long tail coverage are an expensive commitment to poor network design.
> 
> If I recall correctly, even with normal characteristic impedances in cable (free
> space is faster) a tail circuit across the country will not exceed 20 msecs.
> 
> Kevin
> 
> Scott Keagy <scott@sknetworks.com> on 04/26/2000 03:48:08 PM
> 
> To:   Hari Krishna Vuppaladhadiam <hariv@chiplogic.com>
> cc:   rem-conf@es.net (bcc: Kevin Bruemmer/Natural MicroSystems/US)
> Subject:  Re: Echo Cancellers
> 
> I think there is a good application for high-delay echo cancellers.
> Consider the following scenarios:
> 
>      <user A>----<VoIP network>----<PSTN>----<mobile network>---<userB>
> or
>      <VoIP network>----<PSTN>-------<VoIP network>
> or
>      <VoIP network>----<any high delay hop-off to PSTN>
> 
> Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
> routers) only provide 32 msec of coverage. If a hop-off tail circuit has
> a round-trip delay that exceeds the coverage of the echo canceller (on
> the outbound edge of the VoIP network), then the additional delay of the
> VoIP network (100 msec or more usually) exacerbates the echo problem.
> 
> This is an intractable problem without the use of echo cancellers that
> have longer coverage ability.
> 
> -Scott
> 
> Hari Krishna Vuppaladhadiam wrote:
> >
> > Hi,
> >   I am interested in knowing what kind of echo cancellers do we require in
> > typical VoIP applications. Is there a requirement for delays greater than
> > 64msecs?
> >
> > Regards
> > Hari
> 
> --
> =======================================================
>  Scott Keagy, CCIE #3985
>  Senior Networking Consultant
>  SK Networks, Inc.         mailto:scott@sknetworks.com
> =======================================================

-- 
=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant  
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================



From rem-conf Wed Apr 26 15:24:49 2000 
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Reply-To: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
From: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
To: "Scott Keagy" <scott@sknetworks.com>, <Kevin_Bruemmer@nmss.com>
Cc: <rem-conf@es.net>
Subject: Re: Echo Cancellers
Date: Wed, 26 Apr 2000 15:12:19 -0700
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Hi,
  I see in many of the company websites of VoIP specifying Echo cancellers
(ITU-G.165/G.168) standards. When it comes to delay it goes upto 128 msecs.
Now I understand a case where Mobile networks is involved, one can find
larger delays.
It takes lot of MIPS to run Echo cancellers for larger delays. Where do I
get information regarding the DELAY?
And also the necessity of an Echo canceller for say a residential gateway or
an enterprise gateway?

Hari

-----Original Message-----
From: Scott Keagy <scott@sknetworks.com>
To: Kevin_Bruemmer@nmss.com <Kevin_Bruemmer@nmss.com>
Cc: Hari Krishna Vuppaladhadiam <hariv@chiplogic.com>; rem-conf@es.net
<rem-conf@es.net>
Date: Wednesday, April 26, 2000 2:58 PM
Subject: Re: Echo Cancellers


>Kevin et al,
>
>feel free to correct me if I'm wrong in the details below....
>
>I added more details to the diagrams to clarify what we're talking
>about.
><see diagrams at bottom>
>
>
>ITU Echo Canceller Requirements
>-------------------------------
>
>ITU-T Recommendation G.131 "Control of Talker Echo" (section 4.1) states
>the following:
>"In general, it is recommended that the active echo control devices
>shall be deployed on all connections which exceed the total one-way
>talker echo transmission path time of 25 ms."
>
>This implies that public phone networks may not provide echo
>cancellation for tail-end hop-off circuits that have up to 50 msec
>round-trip latency.
>
>
>Where echo originates
>---------------------
>Impedance mismatches at 2-wire/4-wire hybrids (transitions from 2-wire
>to 4-wire circuits) can cause signal reflections. The likely spots where
>these hybrids can cause problems for VoIP hop-off scenarios are
>illustrated below. Carrier equipment may not be optimally tuned, or
>customer equipment may not be optimally tuned. In addition to these
>spots, acoustic echoes may occur from the destination speaker/mic
>equipment (i.e. speakerphone or cheap telephone handset).
>
>I submit as proof that not all networks are ideally designed, that I
>frequently call residential locations in India from the U.S. I here my
>own echo on at least half of the conversations, which implies that some
>telco central offices in India have poor hybrid terminations.
>
>
>What a company can do about it
>-------------------------------
>Given the ITU recommendations that support a carrier's right to not
>employ echo cancellers in medium-delay networks (i.e. up to 50 msec
>round-trip), a company employing a VoIP solution has no recourse to
>block the echo other than to use echo cancellers with high delay
>coverage, as illustrated below. A given company can't fix the network
>problems for different calling destinations (i.e. adjust the impedance
>coupling at the hybrid to increase the return loss), but it can take
>personal action to defend the voice quality experienced by its
>employees.
>
>Note that this is an enterprise-centric approach to the problem. A
>carrier-centric approach might be to negotiate with the remote telco to
>fix the problem, so that the local carrier does not get stuck with the
>echo-canceller bill. So high delay echo cancellers may be useful for
>low-end and enterprise products, but not for carrier products. Also note
>that up to 64 msec is sufficient given that all networks are designed
>according to ITU spec.
>
>
>==== : 4-wire circuit
>---- : 2-wire circuit
>
>**************************************************************************
>                                                  ||=====<digital user
>C>
>                                                  ||
>                                                  ||
><user A>=====<VoIP cloud>===<long delay PSTN>====<PBX>-----<analog user
>B>
>                        ^                             ^
>                        |                             |
>                        |                             |
>                        |                 (signal from A reflected here)
>                        |
>                   (High delay echo canceller needed here)
>
>**************************************************************************
>
><user A>=====<VoIP cloud>=====<long delay PSTN>-------<user B>
>                        ^                     ^
>                        |                     |
>                        |                     |
>                        |            (signal from A reflected here)
>                        |
>                   (High delay echo canceller needed here)
>
>***************************************************************************
>
>-Scott
>
>Kevin_Bruemmer@nmss.com wrote:
>>
>> Scott:
>>
>> Thanks for you input to the group on this subject.  There seems to be a
>> considerable amount of misunderstanding about echo control with VoIP
networks.
>> I repeat a label your networks below to make some points of discussion:
>>
>> Case 1:
>>      <user A>----<VoIP network>----<PSTN>----<mobile network>---<userB>
>>
>> Case 2:
>>
>>      <VoIP network>----<PSTN>-------<VoIP network>
>>
>> Case 3:
>>
>>      <VoIP network>----<any high delay hop-off to PSTN>
>>
>> In case 1, user A has a four-wire connection to the network and user B
also has
>> a four-wire connection to the network.  That is - unless somewhere in the
mobile
>> network or in the PSTN there is some two wire circuits.  Unless user A is
using
>> a poor quality speakerphone, there should be no need for any echo
cancellation
>> at all.
>>
>> for case 2, there should also be no need for any echo cancellers unless,
again,
>> the PSTN has been poorly engineered and has two-wire connections
somewhere.
>>
>> For case 3, it the right side of the network has a Media Gateway.  At
some point
>> to the right of that gateway there will be a two-four wire hybrid which
will be
>> the source of an echo.  The echo would be user a "talker echo".
Therefore, the
>> media gateway has ther responsibility of echo cancelling this echo.  This
is
>> typicaly short - as in certainly less than 20 msec.  It is true - that
the
>> longer the VoIP network delay, the better the echo canceller in the
network must
>> work.  This is why it is important for VoIP networks to manage their
delay
>> downwards as much as possible.  To do otherwise causes a) the media
gateway echo
>> canceller to "work harder" than it should have to and b) cause
conversational
>> double-talking that is subjectively annoying even when the echo control
is
>> perfect.
>>
>> The basic misimpression that I see all too often is that people believe
that
>> "station side" or subscriber equipment echo cancellers need to be
deployed at
>> the subscriber or user's equipment.  This is, for the most part, untrue.
If it
>> turns out to be true, someone architected their network poorly.  Echo
cancellers
>> with long tail coverage are an expensive commitment to poor network
design.
>>
>> If I recall correctly, even with normal characteristic impedances in
cable (free
>> space is faster) a tail circuit across the country will not exceed 20
msecs.
>>
>> Kevin
>>
>> Scott Keagy <scott@sknetworks.com> on 04/26/2000 03:48:08 PM
>>
>> To:   Hari Krishna Vuppaladhadiam <hariv@chiplogic.com>
>> cc:   rem-conf@es.net (bcc: Kevin Bruemmer/Natural MicroSystems/US)
>> Subject:  Re: Echo Cancellers
>>
>> I think there is a good application for high-delay echo cancellers.
>> Consider the following scenarios:
>>
>>      <user A>----<VoIP network>----<PSTN>----<mobile network>---<userB>
>> or
>>      <VoIP network>----<PSTN>-------<VoIP network>
>> or
>>      <VoIP network>----<any high delay hop-off to PSTN>
>>
>> Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
>> routers) only provide 32 msec of coverage. If a hop-off tail circuit has
>> a round-trip delay that exceeds the coverage of the echo canceller (on
>> the outbound edge of the VoIP network), then the additional delay of the
>> VoIP network (100 msec or more usually) exacerbates the echo problem.
>>
>> This is an intractable problem without the use of echo cancellers that
>> have longer coverage ability.
>>
>> -Scott
>>
>> Hari Krishna Vuppaladhadiam wrote:
>> >
>> > Hi,
>> >   I am interested in knowing what kind of echo cancellers do we require
in
>> > typical VoIP applications. Is there a requirement for delays greater
than
>> > 64msecs?
>> >
>> > Regards
>> > Hari
>>
>> --
>> =======================================================
>>  Scott Keagy, CCIE #3985
>>  Senior Networking Consultant
>>  SK Networks, Inc.         mailto:scott@sknetworks.com
>> =======================================================
>
>--
>=======================================================
> Scott Keagy, CCIE #3985
> Senior Networking Consultant
> SK Networks, Inc.         mailto:scott@sknetworks.com
>=======================================================




From rem-conf Thu Apr 27 00:29:08 2000 
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To: Scott Keagy <scott@sknetworks.com>
cc: Hari Krishna Vuppaladhadiam <hariv@chiplogic.com>, rem-conf@es.net
Subject: Re: Echo Cancellers
In-reply-to: Your message of "Wed, 26 Apr 2000 12:48:08 PDT." <390747F8.BA10F60B@sknetworks.com>
Date: Thu, 27 Apr 2000 08:20:19 +0100
Message-ID: <1246.956820019@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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In message <390747F8.BA10F60B@sknetworks.com>, Scott Keagy typed:

 >>Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
 >>routers) only provide 32 msec of coverage. If a hop-off tail circuit has
 >>a round-trip delay that exceeds the coverage of the echo canceller (on
 >>the outbound edge of the VoIP network), then the additional delay of the
 >>VoIP network (100 msec or more usually) exacerbates the echo problem.
 
 >>This is an intractable problem without the use of echo cancellers that
 >>have longer coverage ability.

 Scott


well a well engineered path from London to San Francisco consistently
has a delay of around 150ms.... (see below) - i dont see why someone
cant think a bit about echo cancellors for this.....references would
be interesting on why it is "intractable" - is it only if the delay is
also highly variable (the path below has low delay variance - probably
lower than a lot of paths with lower e2e delays by about an order of
magnitude)

j.
1  cisco (128.16.6.150)  1.290 ms  0.822 ms  0.774 ms
 2  cisco-pb (128.40.14.245)  1.037 ms  1.005 ms  0.931 ms
 3  10.0.121.5 (10.0.121.5)  1.091 ms  1.116 ms  0.943 ms
 4  128.40.255.153 (128.40.255.153)  1.011 ms  0.978 ms  0.961 ms
 5  atmr-ulcc.lmn.net.uk (194.83.100.62)  2.442 ms  2.072 ms  2.861 ms
6  146.97.255.65 (146.97.255.65)  2.888 ms  3.083 ms  3.552 ms
 7  us-gw2.ja.net (128.86.1.242)  2.644 ms  4.118 ms  4.282 ms
 8  193.62.157.10 (193.62.157.10)  77.606 ms  78.305 ms  76.007 ms
 9  ny-pop.i2.ja.net (193.62.157.2)  76.505 ms  77.377 ms  77.356 ms
10  clev-nycm.abilene.ucaid.edu (198.32.8.29)  91.581 ms  91.423 ms
91.912 ms
11  ipls-clev.abilene.ucaid.edu (198.32.8.25)  97.521 ms  97.082 ms
97.209 ms
12  kscy-ipls.abilene.ucaid.edu (198.32.8.5)  106.739 ms  106.723 ms
106.687 ms
13  denv-kscy.abilene.ucaid.edu (198.32.8.13)  117.296 ms  117.093 ms
117.063 m
s
14  scrm-denv.abilene.ucaid.edu (198.32.8.1)  139.592 ms  140.627 ms
138.953 ms
15  BERK--abilene.POS.calren2.net (198.32.249.41)  142.507 ms  142.512
ms  142.5
65 ms
16  pos1-0.inr-000-eva.Berkeley.EDU (128.32.0.89)  142.498 ms  142.269
ms  142.5
63 ms
17  pos5-0-0.inr-002-eva.Berkeley.EDU (128.32.0.74)  139.891 ms
139.231 ms  139
.991 ms
18  fast9-0-0.inr-010-cory.Berkeley.EDU (128.32.0.42)  140.512 ms
141.192 ms  1
40.178 ms
19  f1-0.inr-181-soda.Berkeley.EDU (128.32.120.181)  144.808 ms
143.865 ms  144
.345 ms
20  router1.ICSI.Berkeley.EDU (192.150.186.2)  146.302 ms  145.192 ms
146.140 m
s
21  www.aciri.org (192.150.187.11)  146.306 ms  153.327 ms  146.852 ms




From rem-conf Thu Apr 27 07:08:14 2000 
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Date: Thu, 27 Apr 2000 15:52:26 +0200
From: Bart Van Doorselaer <Bart.Van_Doorselaer@alcatel.be>
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CC: Scott Keagy <scott@sknetworks.com>,
        Hari Krishna Vuppaladhadiam <hariv@chiplogic.com>, rem-conf@es.net
Subject: Re: Echo Cancellers
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Let us not confuse near-end and far-end echo canceling.

Assume a PSTN-GW-IP-GW-PSTN scenario.

If you only provide near-end echo cancelling you only have to take into account
low delays (i.e. mainly the transmission delay in the PSTN between the hybrid
and the gateway - which is about 5 microseconds per km).
In that case, it is of course a prerequisite to provide near-end echo
cancelling on both gateways (and how do you guarantee that?)

If you want to provide far-end echo cancelling as well, the more-than-100-ms
assumption will be realistic indeed.

Bart


Jon Crowcroft wrote:

> In message <390747F8.BA10F60B@sknetworks.com>, Scott Keagy typed:
>
>  >>Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
>  >>routers) only provide 32 msec of coverage. If a hop-off tail circuit has
>  >>a round-trip delay that exceeds the coverage of the echo canceller (on
>  >>the outbound edge of the VoIP network), then the additional delay of the
>  >>VoIP network (100 msec or more usually) exacerbates the echo problem.
>
>  >>This is an intractable problem without the use of echo cancellers that
>  >>have longer coverage ability.
>
>  Scott
>
> well a well engineered path from London to San Francisco consistently
> has a delay of around 150ms.... (see below) - i dont see why someone
> cant think a bit about echo cancellors for this.....references would
> be interesting on why it is "intractable" - is it only if the delay is
> also highly variable (the path below has low delay variance - probably
> lower than a lot of paths with lower e2e delays by about an order of
> magnitude)
>
> j.

--
                                              ____________________
Bart Van Doorselaer                           \                  /
_______________________________________________\                /____
                                                \              /
Research Engineer                                \  ALCATEL   /
DN2                                          Network Strategy Group
Tel   : +32 3 240 86 41                      Francis Wellesplein  1
Fax   : +32 3 240 99 32                     B-2018 Antwerpen Belgium
_____________________________________________________\    /__________
mailto:Bart.Van_Doorselaer@alcatel.be                 \  /
                                                       \/





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Jon:

Unless I misunderstand what you are saying below, you should confuse the end-end
delay of an entire connection with what coverage the echo canceller must have.
The network echo canceller (G.165/G.168 definition) should cancel the echo as
reflected on the far end.  In other words, the echo canceller's responsibility
is to cancel the talker echo at or near the two-four wire hybrid at the
listener's end of the connection.

Kevin




Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk> on 04/27/2000 03:20:19 AM

To:   Scott Keagy <scott@sknetworks.com>
cc:   Hari Krishna Vuppaladhadiam <hariv@chiplogic.com>, rem-conf@es.net (bcc:
      Kevin Bruemmer/Natural MicroSystems/US)
Subject:  Re: Echo Cancellers




In message <390747F8.BA10F60B@sknetworks.com>, Scott Keagy typed:

 >>Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
 >>routers) only provide 32 msec of coverage. If a hop-off tail circuit has
 >>a round-trip delay that exceeds the coverage of the echo canceller (on
 >>the outbound edge of the VoIP network), then the additional delay of the
 >>VoIP network (100 msec or more usually) exacerbates the echo problem.

 >>This is an intractable problem without the use of echo cancellers that
 >>have longer coverage ability.

 Scott


well a well engineered path from London to San Francisco consistently
has a delay of around 150ms.... (see below) - i dont see why someone
cant think a bit about echo cancellors for this.....references would
be interesting on why it is "intractable" - is it only if the delay is
also highly variable (the path below has low delay variance - probably
lower than a lot of paths with lower e2e delays by about an order of
magnitude)

j.
1  cisco (128.16.6.150)  1.290 ms  0.822 ms  0.774 ms
 2  cisco-pb (128.40.14.245)  1.037 ms  1.005 ms  0.931 ms
 3  10.0.121.5 (10.0.121.5)  1.091 ms  1.116 ms  0.943 ms
 4  128.40.255.153 (128.40.255.153)  1.011 ms  0.978 ms  0.961 ms
 5  atmr-ulcc.lmn.net.uk (194.83.100.62)  2.442 ms  2.072 ms  2.861 ms
6  146.97.255.65 (146.97.255.65)  2.888 ms  3.083 ms  3.552 ms
 7  us-gw2.ja.net (128.86.1.242)  2.644 ms  4.118 ms  4.282 ms
 8  193.62.157.10 (193.62.157.10)  77.606 ms  78.305 ms  76.007 ms
 9  ny-pop.i2.ja.net (193.62.157.2)  76.505 ms  77.377 ms  77.356 ms
10  clev-nycm.abilene.ucaid.edu (198.32.8.29)  91.581 ms  91.423 ms
91.912 ms
11  ipls-clev.abilene.ucaid.edu (198.32.8.25)  97.521 ms  97.082 ms
97.209 ms
12  kscy-ipls.abilene.ucaid.edu (198.32.8.5)  106.739 ms  106.723 ms
106.687 ms
13  denv-kscy.abilene.ucaid.edu (198.32.8.13)  117.296 ms  117.093 ms
117.063 m
s
14  scrm-denv.abilene.ucaid.edu (198.32.8.1)  139.592 ms  140.627 ms
138.953 ms
15  BERK--abilene.POS.calren2.net (198.32.249.41)  142.507 ms  142.512
ms  142.5
65 ms
16  pos1-0.inr-000-eva.Berkeley.EDU (128.32.0.89)  142.498 ms  142.269
ms  142.5
63 ms
17  pos5-0-0.inr-002-eva.Berkeley.EDU (128.32.0.74)  139.891 ms
139.231 ms  139
.991 ms
18  fast9-0-0.inr-010-cory.Berkeley.EDU (128.32.0.42)  140.512 ms
141.192 ms  1
40.178 ms
19  f1-0.inr-181-soda.Berkeley.EDU (128.32.120.181)  144.808 ms
143.865 ms  144
.345 ms
20  router1.ICSI.Berkeley.EDU (192.150.186.2)  146.302 ms  145.192 ms
146.140 m
s
21  www.aciri.org (192.150.187.11)  146.306 ms  153.327 ms  146.852 ms









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From: Kevin_Bruemmer@nmss.com
To: Bart Van Doorselaer <Bart.Van_Doorselaer@alcatel.be>
cc: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>,
	Scott Keagy <scott@sknetworks.com>,
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Bart:

As far as I know, ETSI TIPHON, VoIP Forum ( IMTC) and the new ITU G.799 require
the use of echo canceller on gateways.  All the gateway vendors that I am aware
of have network echo cancellers in them.  If you sell a gateway without it,
nobody in their right mind should buy it.

In general - I believe that if an application needs a 20 msec echo canceller
they either have a wireless application or something else is wrong.  For
wireless applications, the longest that I have heard as a requirement is 60
msecs.  People are over-spec'ing echo canceller out of ignorance, not out of
real need.

Do you agree?

Kevin,




Bart Van Doorselaer <Bart.Van_Doorselaer@alcatel.be> on 04/27/2000 09:52:26 AM

To:   Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
cc:   Scott Keagy <scott@sknetworks.com>, Hari Krishna Vuppaladhadiam
      <hariv@chiplogic.com>, rem-conf@es.net (bcc: Kevin Bruemmer/Natural
      MicroSystems/US)
Subject:  Re: Echo Cancellers



Let us not confuse near-end and far-end echo canceling.

Assume a PSTN-GW-IP-GW-PSTN scenario.

If you only provide near-end echo cancelling you only have to take into account
low delays (i.e. mainly the transmission delay in the PSTN between the hybrid
and the gateway - which is about 5 microseconds per km).
In that case, it is of course a prerequisite to provide near-end echo
cancelling on both gateways (and how do you guarantee that?)

If you want to provide far-end echo cancelling as well, the more-than-100-ms
assumption will be realistic indeed.

Bart


Jon Crowcroft wrote:

> In message <390747F8.BA10F60B@sknetworks.com>, Scott Keagy typed:
>
>  >>Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
>  >>routers) only provide 32 msec of coverage. If a hop-off tail circuit has
>  >>a round-trip delay that exceeds the coverage of the echo canceller (on
>  >>the outbound edge of the VoIP network), then the additional delay of the
>  >>VoIP network (100 msec or more usually) exacerbates the echo problem.
>
>  >>This is an intractable problem without the use of echo cancellers that
>  >>have longer coverage ability.
>
>  Scott
>
> well a well engineered path from London to San Francisco consistently
> has a delay of around 150ms.... (see below) - i dont see why someone
> cant think a bit about echo cancellors for this.....references would
> be interesting on why it is "intractable" - is it only if the delay is
> also highly variable (the path below has low delay variance - probably
> lower than a lot of paths with lower e2e delays by about an order of
> magnitude)
>
> j.

--
                                              ____________________
Bart Van Doorselaer                           \                  /
_______________________________________________\                /____
                                                \              /
Research Engineer                                \  ALCATEL   /
DN2                                          Network Strategy Group
Tel   : +32 3 240 86 41                      Francis Wellesplein  1
Fax   : +32 3 240 99 32                     B-2018 Antwerpen Belgium
_____________________________________________________\    /__________
mailto:Bart.Van_Doorselaer@alcatel.be                 \  /
                                                       \/










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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 5/3 Design and Evaluations of Multimedia Proxy Caching
  Mechanisms for the Internet -- Reza Rejaie, AT&T Labs - Research 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Design and Evaluations of Multimedia Proxy Caching Mechanisms for the Internet

Wednesday May 3, 2000 
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

Reza Rejaie
AT&T Labs - Research 

Most of the existing Internet multimedia streaming applications are based
on the client-server architecture. This architecture has two major
limitations: 1) The quality of delivered streams is limited to the
bottleneck bandwidth between the server and the client. Thus a client with
a high bandwidth local connectivity may receive low quality streams due to
a remote bottleneck. 2) Its scalability is limited, in that it is difficult
to support a large number of concurrent high quality sessions due to
network and server load. Multimedia proxy caching (MCaching) is a natural
extension of the client-server architecture that addresses both limitations
simultaneously. The idea is to cache popular streams with appropriate
quality at a proxy close to interested clients. Thus the proxy can
effectively enhance the delivered quality and improve scalability by
significantly reducing the load on the network and the server. 

In this talk, we address the design and performance evaluations of MCaching
mechanisms. First the design of an efficient MCaching mechanism for
layered-encoded streams is outlined. We observe that in addition to cache
efficiency (measured by hit ratio), MCaching introduces the notion of
``quality'' for delivered streams as a new dimension to Web caching. Thus
performance of MCaching mechanisms should be collectively evaluated along
both dimensions caching efficiency and delivered quality. We identify a
fundamental tradeoff between these two dimensions. Our simulation results
show that our proposed MCaching design can adaptively exploit this tradeoff
to improve overall performance whereas simple extension of current Web
caching schemes can only enhance the performance along one dimension.
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Thu Apr 27 13:22:28 2000 
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Date: Thu, 27 Apr 2000 13:13:21 -0700
From: Scott Keagy <scott@sknetworks.com>
Organization: SK Networks, Inc.
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        Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>,
        Hari Krishna Vuppaladhadiam <hariv@chiplogic.com>, rem-conf@es.net
Subject: Re: Echo Cancellers
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Distance between New York, NY and :
San Diego, CA = 3914 km          ====> 39.14 msec round trip
San Francisco, CA = 4156 km      ====> 41.56 msec   "     "
Seattle, WA = 3884 km            ====> 38.84 msec   "     "

These times assume fiber spread "as the crow flies"
and no switching/regeneration delays.

An international company with a single U.S. office in one of these
cities (or any city on either coast) will have problems with VoIP
tail-end hop-off to other major population centers in the U.S.

The engineering solution says, "why don't they just centrally locate
their office in Chicago?" Echo cancellers seems like a funny reason to
relocate a business though.

-Scott

Kevin_Bruemmer@nmss.com wrote:
> 
> Bart:
> 
> As far as I know, ETSI TIPHON, VoIP Forum ( IMTC) and the new ITU G.799 require
> the use of echo canceller on gateways.  All the gateway vendors that I am aware
> of have network echo cancellers in them.  If you sell a gateway without it,
> nobody in their right mind should buy it.
> 
> In general - I believe that if an application needs a 20 msec echo canceller
> they either have a wireless application or something else is wrong.  For
> wireless applications, the longest that I have heard as a requirement is 60
> msecs.  People are over-spec'ing echo canceller out of ignorance, not out of
> real need.
> 
> Do you agree?
> 
> Kevin,
> 
> Bart Van Doorselaer <Bart.Van_Doorselaer@alcatel.be> on 04/27/2000 09:52:26 AM
> 
> To:   Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
> cc:   Scott Keagy <scott@sknetworks.com>, Hari Krishna Vuppaladhadiam
>       <hariv@chiplogic.com>, rem-conf@es.net (bcc: Kevin Bruemmer/Natural
>       MicroSystems/US)
> Subject:  Re: Echo Cancellers
> 
> Let us not confuse near-end and far-end echo canceling.
> 
> Assume a PSTN-GW-IP-GW-PSTN scenario.
> 
> If you only provide near-end echo cancelling you only have to take into account
> low delays (i.e. mainly the transmission delay in the PSTN between the hybrid
> and the gateway - which is about 5 microseconds per km).
> In that case, it is of course a prerequisite to provide near-end echo
> cancelling on both gateways (and how do you guarantee that?)
> 
> If you want to provide far-end echo cancelling as well, the more-than-100-ms
> assumption will be realistic indeed.
> 
> Bart
> 
> Jon Crowcroft wrote:
> 
> > In message <390747F8.BA10F60B@sknetworks.com>, Scott Keagy typed:
> >
> >  >>Typical echo cancellers at the edge of the VoIP network (e.g. Cisco
> >  >>routers) only provide 32 msec of coverage. If a hop-off tail circuit has
> >  >>a round-trip delay that exceeds the coverage of the echo canceller (on
> >  >>the outbound edge of the VoIP network), then the additional delay of the
> >  >>VoIP network (100 msec or more usually) exacerbates the echo problem.
> >
> >  >>This is an intractable problem without the use of echo cancellers that
> >  >>have longer coverage ability.
> >
> >  Scott
> >
> > well a well engineered path from London to San Francisco consistently
> > has a delay of around 150ms.... (see below) - i dont see why someone
> > cant think a bit about echo cancellors for this.....references would
> > be interesting on why it is "intractable" - is it only if the delay is
> > also highly variable (the path below has low delay variance - probably
> > lower than a lot of paths with lower e2e delays by about an order of
> > magnitude)
> >
> > j.
> 
> --
>                                               ____________________
> Bart Van Doorselaer                           \                  /
> _______________________________________________\                /____
>                                                 \              /
> Research Engineer                                \  ALCATEL   /
> DN2                                          Network Strategy Group
> Tel   : +32 3 240 86 41                      Francis Wellesplein  1
> Fax   : +32 3 240 99 32                     B-2018 Antwerpen Belgium
> _____________________________________________________\    /__________
> mailto:Bart.Van_Doorselaer@alcatel.be                 \  /
>                                                        \/

-- 
=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant  
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================



From rem-conf Thu Apr 27 13:37:03 2000 
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I believe that you are assuming that the company has only one gateway.
Generally, each location will have a gateway.
Do you agree?





From rem-conf Thu Apr 27 13:42:10 2000 
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---------------------- Forwarded by Bruce Levens/Natural MicroSystems/US on
04/27/2000 04:46 PM ---------------------------


Bruce Levens
04/27/2000 04:40 PM

To:   Scott Keagy <scott@sknetworks.com>
cc:
Subject:  Re: Echo Cancellers  (Document link: Bruce Levens' Mail)

Or perhaps you are implying that the company is located on the west coast and
calls a gateway (via PSTN) in NY to reach its international offices, instead of
having a gateway on the west coast.






From rem-conf Thu Apr 27 14:31:44 2000 
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From: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
To: <rem-conf@es.net>
Subject: Re: Echo Cancellers
Date: Thu, 27 Apr 2000 14:23:41 -0700
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A typical chain like this:

Telephone ==> Central Office  ==> Gateway ==> Internet ==> Gateway ==>
Central office ==> Telephone

Can we consider a telephone itself as a Hybrid device? If so does it require
an EC?
The central office where Hybrid conversion takes place need to have an EC!
Does the Gateway require an EC if the central office already has one?
I somehow did not see companies seriously discussing Acoustic Echo
canceller!

Say if Residential Gateway is designed then is there a necessity of an EC?
(if not why?)

May I request someone to clear my doubts.

Regards
Hari








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Hi everybody !

Just to say that JRTPLIB v2.3 is out. This is an object-oriented library
which makes the use of RTP a bit easier. It is written in C++ and seems to
work just fine on several kinds of unix-like platforms and also on a windows
platform.

Changes from version 2.2 include:
 - some bugfixes
 - multicasting support
 - a configure script which should make the compilation a bit easier
 - a small tutorial and some small sample applications

For a more complete list of changes, you should take a look at the ChangeLog
in the archive.

In case you're interested, you can download the library from this url:
http://lumumba.luc.ac.be/jori/rtp/rtp.html

Bye !
Jori




From rem-conf Thu Apr 27 19:15:09 2000 
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To: rem-conf@es.net
Subject: Re: Echo Cancellers 
In-Reply-To: Message from "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com> 
   of "Thu, 27 Apr 2000 14:23:41 MST." <01aa01bfb08e$fc854420$03000004@hariv.domain> 
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From: George Michaelson <ggm@dstc.edu.au>
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We just (re)deployed an ATM/AAL5 layered videowall with subb 100ms end-to-end
delay in the network (IP measure, other payload same medium) and also some ISDN
(>100ms) and IP (variable)

Back about 5 years ago, decent audio/hw with echo cancelling built in was
bloody expensive. I recall Van discussing this in terms of the complexity
of the models which have to try and deal with an 'average' size room and
the kinds of echo you can get, and how badly cheap phone 'handsfree' echo
cancellation works (If you've been on a multiway call recently you will
know this I am sure. it has a long baseline mute/volume behaviour which
stinks, and one person typing on a keyboard which is not voice data, in
a very specific part of the sound spectrum, and damn loud can wipe out any
echo canceller I've used).

We looked again just a few weeks ago. a unit designed to plug into a VideoConf
unit is still around $700-$1000 $AU. same for a PC plugin unit.

a non-echo cancelling senheisser mike is still around $1000. Use something
for $25 from Tandy, and it can work. But its really horses for courses.

Decent audio gear comes at a premium. I doubt if the chipcount in these
devices is very high, or in real compute terms the complexity is very
high given FPGAs or even a PC-on-a-card to deal with it.

The market expectation is that you sell few of them, and charge a lot for it.
  
If somebody wants to invest a lifetime, writing some code to model this stuff
so I can dial into VAT or some other app "My room is 25M**2 and I appear to
have a wall delay echo of 5ms, and a keyboard near to the mike and want it
to not unmute mike but do want speech to turn it on, with minimal clipping"
would be sweet. Right now, our $750 unit sends my cohort tapping his foot
against the table as a deep throbbing drum in the middle of conversations and
the speaking is so 'bright' it almost rings.

cheers
	-George
--
George Michaelson         |  DSTC Pty Ltd
Email: ggm@dstc.edu.au    |  University of Qld 4072
Phone: +61 7 3365 4310    |  Australia
  Fax: +61 7 3365 4311    |  http://www.dstc.edu.au





From rem-conf Fri Apr 28 00:01:51 2000 
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Date: Thu, 27 Apr 2000 23:51:25 -0700
From: Scott Keagy <scott@sknetworks.com>
Organization: SK Networks, Inc.
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I am indeed talking about a company with a single office in the U.S.,
which happens to be located on either the east or west coast. Consider
for example, an international company w/ offices in Amsterdam & New
York. The Amsterdam office wants to call Los Angeles. They call via
their private VoIP network to New York, then do tail-end hop-off via the
PSTN to Los Angeles. If there is a hybrid problem in Los Angeles (which
causes signal reflection back toward New York), then the New York router
doesn't have long enough echo-canceler coverage to filter out the echo.
The echo keeps propogating back to Amerstam, only the delay is much
larger now. Even a quiet echo is annoying if it comes 500 msec later.


-Scott

Bruce_Levens@nmss.com wrote:
> 
> ---------------------- Forwarded by Bruce Levens/Natural MicroSystems/US on
> 04/27/2000 04:46 PM ---------------------------
> 
> Bruce Levens
> 04/27/2000 04:40 PM
> 
> To:   Scott Keagy <scott@sknetworks.com>
> cc:
> Subject:  Re: Echo Cancellers  (Document link: Bruce Levens' Mail)
> 
> Or perhaps you are implying that the company is located on the west coast and
> calls a gateway (via PSTN) in NY to reach its international offices, instead of
> having a gateway on the west coast.

-- 
=======================================================
 Scott Keagy, CCIE #3985
 Senior Networking Consultant  
 SK Networks, Inc.         mailto:scott@sknetworks.com
=======================================================



From rem-conf Fri Apr 28 11:44:05 2000 
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From rem-conf Fri Apr 28 16:02:02 2000 
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Thanks for the detailed reply.  That scenario indeed looks like it calls for a
(moderately) long echo canceller.





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