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Message-ID: <375464E5.62550239@nps.navy.mil>
Date: Tue, 01 Jun 1999 15:55:34 -0700
From: Don Brutzman <brutzman@nps.navy.mil>
Organization: Naval Postgraduate School
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To: "Baugher, Mark" <mbaugher@intel.com>
CC: AVT working group <rem-conf@es.net>,
        Francisco Afonso <afonso@cs.nps.navy.mil>
Subject: Re: draft-ietf-avt-mib-05.txt question:  MIB in XML?
References: <F70F37F77E9FD211AC3F00A0C96B78DA7CE4A4@orsmsx47.jf.intel.com>
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"Baugher, Mark" wrote:
> 
> Don
>   I don't think this is within the scope of the RTP MIB.
> There is a list for general discussion of MIB-related
> issues at mibs-request@ops.ietf.org - though I'm not
> sure it is an SMI or SNMP issue, properly considered
> 
> Mark

Thanks for the redirect, very helpful.  Though not sure if we
have time to go there yet... Someday.

If anyone is interested, read on, otherwise hit D now.  :)

To provide at least a clue about what we're looking at, here
are Francisco's document type definition (.dtd) and example
recording files, including one marked up in XML that complies
with the dtd.  So it is a verbose but rigorous data-recording
format that has lots of benefits regarding correctness, tools etc.

Caveat:  this is probably pretty buggy stuff, though it does
pass XML well-formed & validity checks.  So far, manually created.

More XML info available at http://www.w3.org/XML/
Caveat 2:  be careful diving in, we haven't found the bottom yet!

> > -----Original Message-----
> > From: Don Brutzman [mailto:brutzman@nps.navy.mil]
> > Sent: Tuesday, May 18, 1999 3:00 PM
> > To: AVT working group
> > Subject: draft-ietf-avt-mib-05.txt question: MIB in XML?
> >
> >
> > Apologies if this question is out of scope for the group,
> > redirects welcome.
> >
> > We are beginning to record RTP statistics and are considering XML tags
> > for the data format.  Does anyone know of work being done
> > that might map
> > MIB entities to an XML document type definition (DTD)?
> >
> > Looking downstream:  should there be an XML DTD as part of the AVT MIB
> > so that such tags are well specified and easily available?
> >
> > all the best, Don
> > --
> > Don Brutzman  Naval Postgraduate School, Code UW/Br Root 200
> > work 831.656.2149
> >               Monterey California 93943-5000 USA
> > fax  831.656.3679
> > Virtual worlds/underwater robots/Internet
> > http://web.nps.navy.mil/~brutzman
> >

-- 
all the best, Don
-- 
Don Brutzman  Naval Postgraduate School, Code UW/Br Root 200  work 831.656.2149
              Monterey California 93943-5000 USA              fax  831.656.3679
Virtual worlds/underwater robots/Internet     http://web.nps.navy.mil/~brutzman
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--------------69346C5FC5979537A7FF3EE3--




From rem-conf Wed Jun 02 11:38:53 1999 
From rem-conf-request@es.net Wed Jun 02 11:38:51 1999
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Date: Wed, 02 Jun 1999 14:29:20 -0400
From: Kamalanand Dasu <kdasu@fore.com>
Organization: Wipro Infotech
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Hi,

    I am running vic v2.7b4 and vat v4.0b2 on NT 4.0.
I am having a problem in changing the ttl parameter. No matter waht ttl
I specify the multicast packet goes out with the ttl=1.
Is this a problem related to this version or am I missing something. I
need to multicast accross subets.

-Kamal




From rem-conf Thu Jun 03 01:03:59 1999 
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To: "A. Tawfik" <atawfik@sun6.math.upei.ca>
Cc: rem-conf@es.net
Subject: Re: RTCP usability 
In-reply-to: atawfik's message of Wed, 02 Jun 1999 23:59:31 -0300.	     <199906030259.XAA22219@sun6.math.upei.ca.> 
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From: Thierry Turletti <Thierry.Turletti@sophia.inria.fr>
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> I have a question regarding the MBone IVS application. IVS uses RTP and RTCP. I 
> have read the article about scalable feedback in IVS by Bolot et al. The article 
> does not mention RTCP but rather mentions that the salability is done via 
> probing and the sender elicits feedback from the receivers. These things are not 
> done by RTCP, so how does IVS use RTCP?

IVS is a videoconferencing tool developed a long time ago (first version 
released in 1992). Firsts versions of IVS used "regular" version 1 of RTCP
packets to report back loss rate from the receivers and adapt the coder rate
to the network state (see "Videoconferencing on the internet" appeared in
ToN, June 1996). The paper you mention (I suppose appeared on Sigcomm'94), 
describes a "new" experimental feedback control mechanism based on the
Ian Wakeman's probing mechanism and implemented in the final version of IVS
released around 1995. This scheme use RTCPv1 options to transmit the 
non-standard information required to estimate for example the number of 
receivers or the state of the network.

> 
> The second question is related to sender based adaptive application. Which of 
> the MBone tools: VAT, IVS, NS, NeVoT, RAT use sender based rate adaptive schemes 
> and if they use the sender based adaptive schemes, do they use RTCP in that?
> 

I don't think that any of VAT, NV, VIC, NeVoT or RAT audio/video conferencing
tools use adaptive rate control schemes. IVS uses a sender-based congestion
control scheme but the point with such source-based schemes is that they don't 
handle receiver/network heterogeneity, i.e they can only adapt to the worst 
receiver or a percentage of congested receivers. To optimize the transmission
and satisfy all receivers, hierarchical coding and receiver-based schemes 
could be efficient (such as RLM developed by Steve McCanne) but unluckily
we are still waiting for the corresponding VIC version... Note that putting 
some intelligence in some routers could also represent an alternative to 
sender/receiver -based congestion control schemes.


Thierry



From rem-conf Thu Jun 03 02:03:58 1999 
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Date: Thu, 03 Jun 1999 02:06:14 -0700
From: Jean Bolot <bolot@ensim.com>
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"A. Tawfik" wrote:

> I have a question regarding the MBone IVS application. IVS uses RTP and RTCP. I
> have read the article about scalable feedback in IVS by Bolot et al. The article
> does not mention RTCP but rather mentions that the salability is done via
> probing and the sender elicits feedback from the receivers. These things are not
> done by RTCP, so how does IVS use RTCP?

Work on IVS stopped about 3 years ago, and the work you mention took
place during the transition of what were then RTPv1 and RTPv2, when one
question was whether to use unicast or multicast feedback. So IVS is not
the best MBone tool to get a good feel for recent uses of RTCP.... Refer
instead to vic or to RendezVous, the successor of IVS at Inria.

> The second question is related to sender based adaptive application. Which of
> the MBone tools: VAT, IVS, NS, NeVoT, RAT use sender based rate adaptive schemes
> and if they use the sender based adaptive schemes, do they use RTCP in that?

Of those, I know RAT uses it (as well as Inria's FreePhone tool). We use a
TCP-friendly scheme in FreePhone, actually a joint rate/error control scheme.
Refer for our recent paper at Infocom for more details
ftp://ftp-sop.inria.fr/rodeo/bolot/99.Audio_fec.ps.gz

-Jean

-------------------------------------------------------------
 Jean Bolot, Ensim Corp.
 Ensim - enabling shared outsourcing of network services



From rem-conf Thu Jun 03 03:52:19 1999 
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Subject: CERN MBone Announcement: Next ATLAS Plenary Sessions
To: rem-conf@es.net (List Mbone-WW)
Date: Thu, 3 Jun 1999 12:48:27 +0200 (MET DST)
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  CERN is pleased to announce the MBONE broadcast of the

	     ATLAS Physics Plenary Sessions
	     ==============================
		     8-11 June 1999


Place: CERN Council Chamber and Main Auditorium

*** Times are UTC+2 ***

Tuesday Afternoon
=================
8 June 1999

14h00 - 17h30  ATLAS Physics Plenary      Salle de Conseil

Wednesday Whole Day
===================
9 June 1999

09h00 - 17h30  ATLAS Plenary              Main Auditorium

Thursday Whole Day
==================
10 June 1999

09h00 - 17h30  ATLAS Plenary              Main Auditorium

Friday Morning
==============
11 June 1999

09h00 - 12h30  ATLAS Plenary              Main Auditorium


The MBONE Broadcast is Announced via 'sdr' as "CERN ATLAS"
The 'vat'&'vic' Applications will be Used with a 'ttl=127'

The Sessions will also be Recorded Using the 'wrtp' Application. They can
be Downloaded from:  "http://sunmed2.cern.ch/cgi-bin/nph-MBone-sessions/"

In Case of Questions or Problems Please Contact:  "multicast@noc.cern.ch"





From rem-conf Thu Jun 03 06:55:00 1999 
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Subject: Re: CERN MBone Announcement: Next ATLAS Plenary Sessions
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On Thu, 3 Jun 1999, Daniel Davids CERN/CN wrote:

>   CERN is pleased to announce the MBONE broadcast of the
> 
> 	     ATLAS Physics Plenary Sessions
> 	     ==============================
> 		     8-11 June 1999

Friends, Once again it is clear that the sdr packets announcing this
session are going missing.  I've reported this to CERN on a number of
recent occasions and they, and SWITCH, have clearly had a good look,
but as yet it seems the problem is not resolved.

This means that although the recent sessions themselves have been very
good as received here in Glasgow, I have had to start the apps
manually by picking up the session parameters from UCSB:

http://imj.ucsb.edu/sdr-monitor/

where we currently see that although advertised with ttl 127, only
five of the contributing sites have in fact received the announcement.

I hope CERN, and Simon at SWITCH, won't resent me going "public" on
this, but if anyone can suggest what's going wrong here, I for one
would be very grateful.

On this occasion I've been running mtrace for a while, and the
problematical part of the path is this:

192.65.185.100  mbone-cern.switch.ch
     |     ^      ttl   51        51 pps        1/1    = --    0 pps
     v     |      hop    4 ms     51 pps        1/1    = --    0 pps
212.1.193.130   ch-ws.ten-155.net
     |     ^      ttl   52        51 pps        0/0    = --    0 pps
     v     |      hop    9 ms     51 pps        0/0    = --    0 pps
212.1.193.233
212.1.193.35    ws1.de.ten-155.net
     |     ^      ttl   53       526 pps        0/0    = --    0 pps

On recent studies, almost all of the packets go missing on the leg
>from mbone-cern.switch.ch; today I can see that one packet managed to
get past ch-ws.ten-155.net, but then got dropped at the next leg.
None of them have reached me.  

I gather that before I started looking, one of them must have reached
dap@aber.ac.uk, who is on the UCSB list.

best regards




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From: Francisco Afonso <afonso@cs.nps.navy.mil>
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Subject: RTP MIB implementation with JMF
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We are sending a comparison between RTP MIB fields, as proposed in the
Internet Draft, and the set of RTP Statistics provided by Sun's Java
Media Framework (JMF).

It would be great if JMF could provide all necessary fields for creating
an RTP MIB application.

On the other side, some JMF statistics are good candidates for joining
the RTP MIB draft.

Comments will be welcome.








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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
   <meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-=
8859-1">
   <meta name=3D"Generator" content=3D"Microsoft Word 97">
   <meta name=3D"GENERATOR" content=3D"Mozilla/4.51 [en] (Win98; I) [Nets=
cape]">
   <meta name=3D"Author" content=3D"Francisco Carlos Afonso">
   <title>RTP MIB vs JMF</title>
</head>
<body link=3D"#0000FF" vlink=3D"#800080">

<center><b><font size=3D+1>RTP MIB implementation using Java Media Framew=
ork(JMF
2.0)</font></b></center>

<div align=3Dright>2 Jun 99</div>

<center>Francisco Afonso (<a href=3D"mailto:afonso@nps.navy.mil">afonso@n=
ps.navy.mil</a>)
&amp; Don Brutzman (<a href=3D"mailto:brutzman@nps.navy.mil">brutzman@nps=
.navy.mil</a>)
<br>Naval Postgraduate School
<br>Monterey, California USA</center>

<h4>
1. Overview and References</h4>
This document compares the attributes of the Management Information Base
(MIB), defined for managing Real-Time Transport Protocol (RTP) systems,
with the existing Java Media Framework set of RTP statistics used by the
RTPMonitor.
<p>The RTP MIB is defined in the document "Real-Time Transport Protocol
Management Information Base", dated April 12<sup>th</sup> 1999. It can
be found in following URL:
<p><b><a href=3D"http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-m=
ib-05.txt">http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mib-05.=
txt</a></b>
<p>Java Media Framework (JMF) is a product of SUN and other companies lik=
e
IBM. The current version is 1.1, but version 2.0 has already been release=
d
in its early version. Version 1.1 allows only media playback and RTP rece=
ption,
while version 2.0 provides media capture and RTP transmission. It can be
downloaded from the following URL:
<p><b><a href=3D"http://www.javasoft.com/products/java-media/jmf/index.ht=
ml">http://www.javasoft.com/products/java-media/jmf/index.html</a></b>
<p>RTPMonitor is a Java application developed by Naval Postgraduate Schoo=
l
as part of the <a href=3D"http://www.web3d.org/WorkingGroups/vrtp/dis-jav=
a-vrml/">DIS-Java-VRML</a>
and <a href=3D"http://www.web3d.org/WorkingGroups/vrtp/">VRTP</a> project=
s.
It was implemented using Java Media Framework (JMF) version 2.0, and it
has the following capabilities:
<dir>- presents global session statistics, individual stream statistics
and feedback from all participants;
<br>- records the statistics in text files, continuously, in intervals
defined by the user;
<br>- plays the received streams (audio and video); and
<br>- participates in the session by sending RTCP packets, if desired.</d=
ir>
<b>2. Comparison RTP MIB versus JMF 2.0-based RTPMonitor</b>
<p>RTP MIB was designed for being used with SNMP. The RTP agents running
this MIB can be either RTP hosts (end systems) or RTP Monitors. The objec=
tive
is to collect statistical data about RTP sessions and its streams, for
diagnostics and management purposes. Each agent maintains a MIB that can
be queried by a Manager. Only the last updated statistic is stored in the
MIB.
<p>RTP MIB has basically three tables:
<ul>
<ul>
<li>
a Session Table, that stores general information about each session being
monitored;</li>

<li>
a Sender Table, that stores information about each stream being sent;</li=
>

<li>
a Receiver Table, that stores feedback information from receivers about
senders.</li>
</ul>
</ul>
RTPMonitor is an application that collects RTP statistics and stores them
on files for later analysis. It is supposed to store data periodically
over the whole duration of the session. The default interval between samp=
les
is 30 seconds.
<p>RTPMonitor collects three types of data:
<ul>
<ul>
<li>
global statistics: about the whole session (most from class GlobalRecepti=
onStats
in JMF )</li>

<li>
stream statistics: about each stream ( most from class RTPReceptionStats
in JMF)</li>

<li>
feedback statistics: about each RR packet ( most from class RTCPFeedback
in JMF).</li>
</ul>
</ul>
The following tables compare the fields in RTP MIB with the ones in RTPMo=
nitor.
A comment is provided for each RTP MIB field not implemented in RTPMonito=
r.
Some MIB fields can not be implemented by RTPMonitor because JMF does not
provide such capability.
<br>&nbsp;
<p><b>SessionTable</b>
<br>&nbsp;
<center><table BORDER COLS=3D3 WIDTH=3D"100%" >
<tr>
<td>
<center><b><font size=3D-1>RTP MIB</font></b></center>
</td>

<td>
<center><b><font size=3D-1>JMF-based RTP Monitor Application</font></b></=
center>
</td>

<td>
<center><b><font size=3D-1>Comments</font></b></center>
</td>
</tr>

<tr>
<td><font size=3D-1>SessionIndex ( Integer32) =96 an index of the concept=
ual
row which is for SNMP purposes only and has no relation with any protocol
value.</font></td>

<td>
<center><font size=3D-1>Not applicable.</font></center>
</td>

<td>
<center>-</center>
</td>
</tr>

<tr>
<td><font size=3D-1>SessionDomain ( TDomain ) =96 the transport layer pro=
tocol
used for sending or receiving the stream of RTP packets in this session.<=
/font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It can be added.&nbsp;</font>
<br><font size=3D-1>JMF uses UDP as the transport protocol. Extensibility
to other protocols is possible, but not provided.&nbsp;</font></td>
</tr>

<tr>
<td><font size=3D-1>SessionRemAddr (Taddress ) =96 the remote destination=
 transport
address on which the RTP data packet is sent and/or received.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It can be added.&nbsp;</font>
<br><font size=3D-1>In RTP Monitor the session transport address names th=
e
directory where the files are created and updated.</font>
<br><font size=3D-1>e.g. session224.2.2.2port44444</font></td>
</tr>

<tr>
<td><font size=3D-1>SessionLocAddr (Taddress) =96 the local destination t=
ransport
address on which the stream of data packet is being sent and/or received.=
</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It can be added.&nbsp;</font></td>
</tr>

<tr>
<td><font size=3D-1>SessionIfIndex ( InterfaceIndex) =96 this value is se=
t
to the corresponding value from the Internet Standard MIB. This is the
interface that the RTPStream is being sent to or received from.</font></t=
d>

<td>
<center><font size=3D-1>Not applicable.</font></center>
</td>

<td>
<center>-</center>
</td>
</tr>

<tr>
<td><font size=3D-1>SessionSenderJoins (Counter32) =96 the number of send=
ers
that have been observed joined the session since SessionStartTime ( see
below).</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>Not part of native JMF statistics. It can be derived
in an JMF application by monitoring NewRecvStreamEvents.</font></td>
</tr>

<tr>
<td><font size=3D-1>SessionReceiverJoins(Counter32) - the number of recei=
vers
that have been observed joined the session since SessionStartTime&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;
(see below).</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>Not part of native JMF statistics. It can be derived
in an JMF application by searching for new participants in the list of
participants managed by RTPSessionManager.</font></td>
</tr>

<tr>
<td><font size=3D-1>SessionByes (Counter32) =96 a count of RTCP BYE.</fon=
t></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>Not part of native JMF statistics. It can be derived
in an JMF application by monitoring ByeEvents.</font></td>
</tr>

<tr>
<td><font size=3D-1>SessionStartTime (TimeStamp) =96 the value of SysUpTi=
me
at the time that this row is created.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>That is the time when the Session Monitoring starts.
It can be added to RTPMonitor.</font></td>
</tr>

<tr>
<td><font size=3D-1>SessionMonitor (ThuthValue) =96 set to true if sender=
 or
receivers in addition to the local RTP System are to be monitored.</font>=
</td>

<td>
<center><font size=3D-1>Not applicable.</font></center>
</td>

<td><font size=3D-1>In RTPMonitor it is always true.</font></td>
</tr>

<tr>
<td><font size=3D-1>SessionRowStatus (RowStatus) =96 active when RTP/RTCP=
 Messages
are being sent or received by an RTPSystem. If this row is "notInService"
it may be removed after 5 minutes.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It can be implemented in RTP Monitor, but without rem=
ovals.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>Time ( hh:mm:ss) =96 time of the report.</font></td>

<td><font size=3D-1>Needed. Implicit in SNMP?</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>TotalParticipants (int) =96 the total number of parti=
cipants
attending the session.&nbsp;</font></td>

<td><font size=3D-1>Needed. Equals active + passive participants and also
remote + local.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>RemoteParticipants(int) =96 the number of remote part=
icipants
attending the session.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>ActiveParticipants (int) =96 the number of active par=
ticipants
(senders) attending the session.</font></td>

<td><font size=3D-1>Needed.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>TotalBytesRecd (int) =96 the number of bytes received=
 in
the session, before any validation.</font></td>

<td><font size=3D-1>Needed for bandwidth calculations.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>TotalPacketsRecd (int) =96 the total number of RTP an=
d
RTCP packets received in the session before any packet validation.&nbsp;<=
/font></td>

<td><font size=3D-1>&nbsp;Needed for comparison of bandwidth versus packe=
ts
per second performance.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>RTCPPacketsRecd (int ) =96 the total number of RTCP p=
ackets
received in the session before any header validation.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>SRPacketsRecd (int) =96 the total number of sender re=
ports
received in the session.&nbsp;</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>BadRTPPackets (int) =96 the total number of RTP data =
packets
that failed the RTP header validation check.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>BadRTCPPackets (int) =96 the total number of RTCP pac=
kets
that failed the RTCP header validation check.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>MalformedSR (int) =96 the total number of invalid sen=
der
reports due to length inconsistency.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>MalformedRR (int) =96 the total number of invalid rec=
eiver
reports due to length inconsistency.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>MalformedSDES (int) =96 the total number of invalid S=
DES
packets due to length inconsistency.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>MalformedBYE (int) =96 the total number of invalid BY=
E
packets due to length inconsistency.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>LocalCollisions (int) =96 the total number of local c=
ollisions
(SSRC collisions).&nbsp;</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>RemoteCollisions (int) =96 the total number of remote=
 collisions
(SSRC collisions).</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>PacketsLooped (int) =96 the total number of packets l=
ooped.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>FailedTransmission (int) =96 the number of packets th=
at
failed to get transmitted.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>UnknownRTCPType (int) =96 the number of individual RT=
CP
packets types that were not implemented or not recognized.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>
</table></center>

<h4>
SenderTable</h4>

<table BORDER COLS=3D3 WIDTH=3D"100%" >
<tr>
<td>
<center><b><font size=3D-1>RTP MIB</font></b></center>
</td>

<td>
<center><b><font size=3D-1>JMF-based RTP Monitor Application</font></b></=
center>
</td>

<td>
<center><b><font size=3D-1>Comments</font></b></center>
</td>
</tr>

<tr>
<td><font size=3D-1>SenderSSRC (Unsigned32) =96 the sender synchronizatio=
n
source identifier.</font></td>

<td><font size=3D-1>SenderSSRC (long)</font></td>

<td><font size=3D-1>Slight type mismatch, but workable.</font></td>
</tr>

<tr>
<td><font size=3D-1>SenderCNAME (DisplayString) =96 the canonical name of=
 the
sender.</font></td>

<td><font size=3D-1>SenderCNAME (String )</font></td>

<td>
<center>-</center>
</td>
</tr>

<tr>
<td><font size=3D-1>SenderAddress ( Taddress) =96 the unicast transport s=
ource
address of the sender.</font></td>

<td>
<center><font size=3D-1>Not provide by JMF.</font></center>
</td>

<td>
<center>-</center>
</td>
</tr>

<tr>
<td><font size=3D-1>SenderPackets ( Counter64) =96 count of RTP packets s=
ent
by this sender, or observed by an RTP Monitor, since SenderStartTime.</fo=
nt></td>

<td><font size=3D-1>SenderPackets (int)</font></td>

<td><font size=3D-1>Called ProcessedPDU by JMF =96 number of valid packet=
s
received from the selected source. Possible type issue. It can overflow.<=
/font></td>
</tr>

<tr>
<td><font size=3D-1>SenderOctets (Counter64) =96 count of RTP octets sent=
 by
this sender or observed by an RTP Monitor, since SenderStartTime.</font><=
/td>

<td>
<center><font size=3D-1>Not provided by JMF.</font></center>
</td>

<td><font size=3D-1>JMF only has a corresponding session count.</font></t=
d>
</tr>

<tr>
<td><font size=3D-1>SenderTool (DisplayString) =96 Name of the applicatio=
n
program source of the stream.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It can be read from JMF RTPSourceDescription object.<=
/font></td>
</tr>

<tr>
<td><font size=3D-1>SRs (Counter32) =96 a counter of the number of RTCP S=
ender
Reports that have been sent from this sender or observed if the RTP entit=
y
is a monitor, since SenderStartTime.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>Not part of native JMF statistics. It can be derived
in an JMF application by monitoring RecvSenderReportEvents.</font></td>
</tr>

<tr>
<td><font size=3D-1>SenderSRTime (TimeStamp) =96 the value of SysUpTime a=
t
the time that the last SR was received from this sender, in the case of
a monitor or receiving host, or sent by this sender, in case of a sending
host.</font></td>

<td>
<center><font size=3D-1>Not provided by JMF.</font></center>
</td>

<td>
<center><font size=3D-1>-</font></center>
</td>
</tr>

<tr>
<td><font size=3D-1>SenderPT ( integer 0..127) =96 static or dynamic payl=
oad
type from the RTP Header.</font></td>

<td>
<center><font size=3D-1>Not provided by JMF.</font></center>
</td>

<td>
<center>-</center>
</td>
</tr>

<tr>
<td><font size=3D-1>SenderStartTime (TimeStamp) =96 the value of SysUpTim=
e
at the time this row was created.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It is the time when the Monitor detected this source.
It can be derived by the JMF application.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>LostPDU (int) =96 the difference between the number o=
f
packets expected as determined by the RTP sequence number range and the
count of packets actually received and validated.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>MisorderedPDU (int) =96 the total number of data pack=
ets
that came in out of order as per the RTP sequence number.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>InvalidPDU (int) =96 the total number of RTP data pac=
kets
that have failed to be within an acceptable sequence number range for an
established SSRC id.</font></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>

<tr>
<td>
<center><font size=3D-1>Not provided by MIB.</font></center>
</td>

<td><font size=3D-1>DuplicatePDU (int) =96 the total number of RTP data p=
ackets
that match the sequence number of another already in the input queue.</fo=
nt></td>

<td><font size=3D-1>Desirable.</font></td>
</tr>
</table>

<p><b>RcvrTable</b>
<br>&nbsp;
<table BORDER COLS=3D3 WIDTH=3D"100%" >
<tr>
<td>
<center><b><font size=3D-1>RTP MIB</font></b></center>
</td>

<td>
<center><b><font size=3D-1>JMF-based RTP Monitor Application</font></b></=
center>
</td>

<td>
<center><b><font size=3D-1>Comments</font></b></center>
</td>
</tr>

<tr>
<td><font size=3D-1>RcvrSRCSSRC (Unsigned32) =96 the SSRC of the sender.<=
/font></td>

<td><font size=3D-1>RcvrSRCSSRC (long)&nbsp;</font></td>

<td><font size=3D-1>Type issue.</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvrSSRC (Unsigned32) =96 the SSRC of the receiver.</=
font></td>

<td><font size=3D-1>RcvrSSRC (long )</font></td>

<td><font size=3D-1>Type issue.</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvrCNAME (DisplayString) =96 the canonical name of t=
he
receiver.&nbsp;</font></td>

<td><font size=3D-1>RcvrCNAME (String)</font></td>

<td>
<center>-</center>
</td>
</tr>

<tr>
<td><font size=3D-1>RcvrAddr (Taddress) =96 the unicast transport address=
 of
the receiver.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It can be added.</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvrRTT (Gauge32) =96 the round trip time measurement=
 taken
by the source of the RTP stream.</font></td>

<td>
<center><font size=3D-1>Not provided by JMF.</font></center>
</td>

<td><font size=3D-1>This value can only be calculated by senders after re=
ceiving
RTP feedback about their streams. JMF does not provide this statistic.
This capability was requested was requested in the jmf-interest mailing
list</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvrLostPackets (Counter64) =96 a count of RTP packet=
s
lost as observed by this receiver.</font></td>

<td><font size=3D-1>RcvrLostPackets (long)</font></td>

<td>
<center>-</center>
</td>
</tr>

<tr>
<td><font size=3D-1>RcvrJitter (Gauge32) =96 an estimate of delay variati=
on
as observed by this receiver.</font></td>

<td><font size=3D-1>RcvrJitter (long )</font></td>

<td><font size=3D-1>Type issue.</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvrTool (DisplayString) =96 the name of the applicat=
ion
program source of the stream.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It can be read from JMF RTPSourceDescription object.<=
/font></td>
</tr>

<tr>
<td><font size=3D-1>RRs (Counter32) =96 a count of the number of RTCP Rec=
eiver
Reports that have been sent from this receiver since RcvrStartTime.</font=
></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>Not part of native JMF statistics. It can be derived
in an JMF application by monitoring RecvSenderReportEvents.</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvrRRTime (TimeStamp) =96 the value of SysUpTime at =
the
last RTCP Receiver Report was received or sent (in case of the sender).</=
font></td>

<td>
<center><font size=3D-1>Not provided by JMF.</font></center>
</td>

<td><font size=3D-1>JMF does not provide the time a feedback is received
or sent. This capability was requested was requested in the jmf-interest
mailing list.</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvrPT (Integer) =96 static or dynamic payload type f=
rom
the RTP header.</font></td>

<td>
<center><font size=3D-1>Not provided by JMF.</font></center>
</td>

<td>
<center>-</center>
</td>
</tr>

<tr>
<td><font size=3D-1>RcvrPackets (Counter64) =96 count of RTP packets rece=
ived
by this RTP host since RcvrStartTime.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>Called ProcessedPDU by JMF =96 number of valid packet=
s
received from the selected source. Same as SenderPackets in the SenderTab=
le.</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvOctets (Counter64) =96 count of RTP octets receive=
d
since RcvrStartTime.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>Not provided by JMF. . Same as SenderOctets in the Se=
nderTable.</font></td>
</tr>

<tr>
<td><font size=3D-1>RcvStartTime (TimeStamp) =96 the value of SysUpTime a=
t
the time that this row is created.</font></td>

<td>
<center><font size=3D-1>Not implemented.</font></center>
</td>

<td><font size=3D-1>It can be added.</font></td>
</tr>
</table>

<p>General comment: variable names used in JMF frequently do not match
names proposed by the RTP MIB. A name correspondence table would be helpf=
ul
in JMF.
<br>&nbsp;
<h4>
3. Conclusion</h4>

<dl>
<dt>
Perhaps all these respective design decisions in the draft RTP MIB and
the new JMF 2.0 were made for conscious, deliberate reasons. Further work
is needed on the MIB and the JMF to resolve these differences.</dt>

<dt>
&nbsp;</dt>
</dl>

</body>
</html>

--------------3066EA5CF36C6A1836E68F74--




From rem-conf Thu Jun 03 11:27:32 1999 
From rem-conf-request@es.net Thu Jun 03 11:27:31 1999
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From: "Strahm, Bill" <bill.strahm@intel.com>
To: "'Francisco Afonso'" <afonso@cs.nps.navy.mil>,
        rem-conf
	 <rem-conf@es.net>, jmf <jmf-interest@sun.com>
Cc: brutzman <brutzman@nps.navy.mil>
Subject: RE: RTP MIB implementation with JMF
Date: Thu, 3 Jun 1999 11:24:13 -0700 
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>From your session table 
1)Not provided by MIB. Time ( hh:mm:ss) - time of the report. Needed.
Implicit in SNMP? 
	Time of the reports are in the sender/receiver report tables...

2)Not provided by MIB. TotalParticipants (int) - the total number of
participants attending the session.  	Needed. Equals active + passive
participants and also remote + local. 
We argued about whether we should track Sender/Receiver Joins/Byes, as a
high watermark, a total count, or a current count.  We decided on the
counter.  Depending on how you want to track "TotalParticipants" it very
well may be rtpSessionSenderJoins+rtpSessionReceiverJoins.

3) Not provided by MIB. RemoteParticipants(int) - the number of remote
participants attending the session. 	Desirable. 
Can simply be the count of receivers in the receiver table
4)Not provided by MIB. ActiveParticipants (int) - the number of active
participants (senders) attending the 	session. Needed. 
See above

5) Not provided by MIB. TotalBytesRecd (int) - the number of bytes received
in the session, before any 	validation. Needed for bandwidth
calculations. 
rtpRcvrOctets in the receiver table.

6) Not provided by MIB. TotalPacketsRecd (int) - the total number of RTP and
RTCP packets received in the 	session before any packet validation.
Needed for comparison of bandwidth versus packets per second 	performance.

rtpRcvrPackets in the receiver table

7) Not provided by MIB. RTCPPacketsRecd (int ) - the total number of RTCP
packets received in the session before 	any header validation. Desirable. 
Combination of SR/RR counts out of the sender/receiver table.

8)Not provided by MIB. SRPacketsRecd (int) - the total number of sender
reports received in the session.  	Desirable. 
rtpSenderSR from the sender table.

9) Not provided by MIB. BadRTPPackets (int) - the total number of RTP data
packets that failed the RTP header 	validation check. Desirable. 
Not available in the MIB

10) Not provided by MIB. BadRTCPPackets (int) - the total number of RTCP
packets that failed the RTCP header 	validation check. Desirable. 
Not available in the MIB

11) Not provided by MIB. MalformedSR (int) - the total number of invalid
sender reports due to length 	inconsistency. Desirable. 
Not available in the MIB

12) Not provided by MIB. MalformedRR (int) - the total number of invalid
receiver reports due to length 	inconsistency. Desirable. 
Not available in the MIB

13) Not provided by MIB. MalformedSDES (int) - the total number of invalid
SDES packets due to length 	inconsistency. Desirable. 
14) Not provided by MIB. MalformedBYE (int) - the total number of invalid
BYE packets due to length 	inconsistency. Desirable. 
15) Not provided by MIB. LocalCollisions (int) - the total number of local
collisions (SSRC collisions).  	Desirable. 
16) Not provided by MIB. RemoteCollisions (int) - the total number of remote
collisions (SSRC collisions). 	Desirable. 
17) Not provided by MIB. PacketsLooped (int) - the total number of packets
looped. Desirable. 
18 ) Not provided by MIB. FailedTransmission (int) - the number of packets
that failed to get transmitted. 	Desirable. 
19)Not provided by MIB. UnknownRTCPType (int) - the number of individual
RTCP packets types that were not 	implemented or not recognized.
Desirable. 
Not available in the MIB

>From the Sender Table

20) Not provided by MIB. LostPDU (int) - the difference between the number
of packets expected as determined by 	the RTP sequence number range and
the count of packets actually received and validated. Desirable. 
RcvrLostPackets in the Receiver Table
21) Not provided by MIB. MisorderedPDU (int) - the total number of data
packets that came in out of order as per 	the RTP sequence number.
Desirable. 
Not availbe in the MIB... it would have to be in the rcvrTable

22) Not provided by MIB. InvalidPDU (int) - the total number of RTP data
packets that have failed to be within 	an acceptable sequence number range
for an established SSRC id. Desirable. 
Not available in the MIB... It would have to be in the RcvrTable

23) Not provided by MIB. DuplicatePDU (int) - the total number of RTP data
packets that match the sequence 	number of another already in the
input queue. Desirable. 
Not available in the MIB... it would have to be in the Rcvr Table 

>From these issues it looks like we handle many of them, not necissarily the
same way that JMF does.  Many of the things that are not included in the MIB
are in the catagory of Protocol errors that we decided not to include
because they add overhead to the management framework to track and are not
necissarily useful in a fully compliant session.

I think we can meet all of your "Needed" and many of your "Desired" requests
with the current MIB.

Bill
-----Original Message-----
From: Francisco Afonso [mailto:afonso@cs.nps.navy.mil]
Sent: Thursday, June 03, 1999 10:35 AM
To: rem-conf; jmf
Cc: brutzman
Subject: RTP MIB implementation with JMF


We are sending a comparison between RTP MIB fields, as proposed in the
Internet Draft, and the set of RTP Statistics provided by Sun's Java
Media Framework (JMF).

It would be great if JMF could provide all necessary fields for creating
an RTP MIB application.

On the other side, some JMF statistics are good candidates for joining
the RTP MIB draft.

Comments will be welcome.










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Date: Wed, 02 Jun 1999 12:44:33 -0500
From: Bezalel Gavish <gavishb@ctrvax.Vanderbilt.Edu>
Subject: CFPs ICTEC 99
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Enclosed is the Call for Papers for the Second International Conference on
Telecommunications and Electronic Commerce. I look forward to receive your
contribution

Sincerely yours,
Bezalel Gavish

PS You may receive multiple copies of this message. The reason for it is
that you appear on multiple mailing lists.



--=====================_928363473==_
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In response to several requests, the submission deadline for ICTEC99 has been
extended to June 15, 1999. Please note that this is a hard deadline.


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

CALL FOR PAPERS and PANELS

Second International Conference on Telecommunications and Electronic Commerce 
(ICTEC)
Nashville, USA, October 6-8, 1999

http://munin.utdallas.edu/atsma/ictec

Topics of interest for the conference include (but are not limited to) the 
following:

* Impact of TEC on supply chain, distribution chain management, EDI and IOS
* Secure transaction processing methods for EC
* The impact of mobility on EC mechanisms 
* Internet traffic management and analysis 
* Pricing Internet services 
* TEC market mechanisms and business models 
* TEC intermediary roles and their analysis
* Electronic payment mechanisms
* Economics of TEC
* Design and analysis of intranets and extranets 
* Web Data management
* Managing Web-based information services 
* Effective management of EC transaction servers 
* Workflow Management issues in TEC
* EC developments and challenges in specific industries 
* Globalization and policy issues in TEC (standards, regulation, property 
rights, etc.)

SUBMISSIONS: Authors should submit an extended abstract of at least 2000 words, 
or a complete paper, by May 15, 1999 (June 15, 1999). Proposals for panels are also
invited. Three hard-copies should be submitted, to an appropriate Regional Coordinator. 
Electronic submissions are welcome, but must be augmented with at least one 
hard-copy.  Authors of accepted papers will be asked to submit their complete 
manuscripts to the Conference Manager, Ms. Dru Lundeng,  by August 31, 1999 for 
inclusion in the conference proceedings.

Conference Chair: Bezalel Gavish, Vanderbilt University, Nashville, USA

Program Committee (Regional Coordinators)

EUROPE and AFRICA

Asuman Dogac							
Software R&D 						
Department of Computer Eng.				
Middle East Tech. Univ.						
Ankara, Turkey
asuman@srdc.metu.edu.tr

Eric van Heck
School of Management
Erasmus University
Rotterdam
The Netherlands
e.heck@fac.fbk.eur.nl

Timo Saarinen 
Helsinki School of Economics
Runeberginkatu 14-16
Helsinki FIN-00100  06531							
Finland
saarinen@hkkk.fi  

AUSTRALIA and NEW ZEALAND

Tony Eyers			
Dept. of Computer Engg.
Univ. of Wollongong
Wollongong NSW 2522, Australia
tony@elec.uow.edu.au

Robyn Lindley				
School of IT & Computer Science		
Univ. of Wollongong			
Wollongong NSW 2522, Australia				
robyn_lindley@uow.edu.au   			 

Ananth Srinivasan
Commerce C 512
University of Auckland
Private Bag 92019
Auckland, New Zealand
a.srinivasan@auckland.ac.nz

ASIA

Robert Blanning	
Dept. of Information Systems		
Univ. of Science and Technology
Clear Water Bay, Kowloon, Hong Kong
Blannirw@ust.hk

Norihisa Komoda
Dept. of I.S. Engg. 			
Osaka University 			
Yamadaoka 2-1, Osaka 565-0871, Japan					
komoda@ise.eng.osaka-u.ac.jp    			

James C. Westland	
Dept. of Information Systems 
Univ. of Science and Technology	
Clear Water Bay, Kowloon, Hong Kong
westland@ust.hk

NORTH AMERICA and ALL OTHER REGIONS
						
Amit Basu
Owen Grad. School of Mgt.
Vanderbilt University 		
Nashville, TN 37203, USA 
Amit.Basu@owen.vanderbilt.edu 	

Ramayya Krishnan			
Heinz School of Public Policy & Mgt.
Carnegie Mellon University		
Pittsburgh, PA 15215, USA 
rk2x@cmu.edu

Industry Panels and Plenaries
Industry Canada
Prabir Neogi
300 Slater Street
Ottawa, Ont., Canada, K1A 0C8
Neogi.Prabir@ic.gc.ca 

PROGRAM COMMITTEE MEMBERS

Abdullah Al-Dhelaan, King Saud U., Saudi Arabia
Kemal Altinkemer, Purdue U., USA
Israel Ben-Shaul, Technion, Israel 
Martin Bichler, U. of Vienna, Austria 
Imrich Chlamtac, U. T. Dallas, USA
H. Michael Chung, Cal State U., USA
Anindya Datta, Georgia Tech., USA
Amitava Dutta, George Mason U., USA
Soumitra Dutta, INSEAD, France
Stuart Feldman, IBM Corp., USA 
Erol Gelenbe, Duke U., USA
Sigmund Handelman, IBM Corp., USA
Joakim Kalvenes, U.T. Dallas, USA
Ajit Kambil, New York U., USA
Kalevi Kilkki, Nokia Corp., Finland 
Wolfgang Klas, U. of Ulm, Germany 
Akhil Kumar, U. of Colorado, USA
Ronald Lee, Erasmus U., Netherlands
Ting-Peng Liang, Sun Yat-Sen U., Tiawan
Alberto Mendelzon, U. of Toronto, Canada
Juzar Motiwalla, Nat. U. of Singapore
Rajat Mukherjee, IBM Corp., USA
Richard W. Oliver, Vanderbilt U., USA
Hasan Pirkul, U. T. Dallas, USA
Sara Porat, IBM Corp., Israel
Louiqa Raschid, U. of Md., USA
Aviel Rubin, AT&T Corp., USA
Arie Segev, U.C. Berkeley, USA
Avi Silberschatz, Bell Labs., USA
Jacob Slonim, Dalhousie U., Canada
Sang Son, U. Va., USA
Kar Yan Tam, U.S.T., Hong Kong
Doug Tygar, UC Berkeley, USA
Andrew Whinston, U.T. Austin, USA
Chee Sing Yap, Multimedia Dev. Corp., Malaysia
Leon Zhao, U.S.T., Hong Kong



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----------------------------------------------------------------------------
--------------
Bezalel Gavish
Owen Graduate School of Management
Vanderbilt University
Nashville, TN 37220
USA

Tel: Office (615) 322-3659            FAX (615) 343-7177
         Home (615) 370-0813

Email: Gavishb@ctral1.vanderbilt.edu
--=====================_928363473==_--




From rem-conf Fri Jun 04 13:54:50 1999 
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From: "Strahm, Bill" <bill.strahm@intel.com>
To: "'hmibs@mailbag.intel.com'" <hmibs@mailbag.cps.intel.com>,
        "'rem-conf@es.net'" <rem-conf@es.net>
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	 <tlam@ascend.com>,
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	 <pwang@ascend.com>,
        "'Dave Thaler'" <dthaler@dthaler.microsoft.com>
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Date: Fri, 4 Jun 1999 13:44:09 -0700 
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I am wanting to send out a summary of changes we are planning on making to
the RTP MIB draft and solicit some input on a couple of key design areas.

Thank you David for your review of our draft, all of the typos,
clarifications etc. have been added to the draft in question.

We have three major points to consider
1) The addition of a "Reverse Lookup Table" to convert address/port pairs
into the corresponding rtpSessionIndex
2) A modification to allow more efficient bandwidth aggregation
3) Additional Objects available from the Java Media Framework

My comments on these issues are:
1) We will include this table as an OPTIONAL table.  Currently many of the
H.341 implementers are seeing the need for 1 session for audio 1 session for
video in their environment and do not want the added complexity of this
table.  If the rtpSessionTable is small the overhead of using this table is
not worth the implementation overhead.

2) One of the questions that David Thaler wants answered is what is the
aggregate RTP Bandwidth from the device.  To do this the rtpSenderTable must
be read, leading to inefficiencies.  There are two ways of optimizing this
lookup, one is require hosts to only include bits they are directly sending
or receiving if rtpSessionMonitor is set.  The other is to include an
additional index into the rtpSenderTable and rtpRcvrTable that is a
TruthValue rtpSenderIsLocal (or some name close to that).  The H.341
community is not interested in having to include 2 rtpSessionEntries to be
able to track what the receivers of their bits are receiving (Create one
entry for the required Host Mode, Create one entry for the Monitor Mode so
they can include the receiver table).  

The issue that needs to be resolved is the line between complexity in the
RTP MIB Agent versus the Management application, and its effect on network
management traffic.  My current thinking is that calculating total RTP
bandwidth for a device belongs as a function of the Management application.
I also do not see much of an improvement in the number of roundtrips needed
to get the values needed to do the aggregation, especially in the presence
of GET-BULK in SNMP v2 and beyond.   Additional indexes will add processing
time to table handling within the agent therefore slowing the host down. 

3) I believe that we have met all of Francisco Needed requirements from the
JMF, and no one else has asked for these variables.  I do not think adding
any of these objects would be a good idea, many implementations will not
have access to this information.

Bill


______________________________________________
Bill Strahm        Programming today is a race between
bill.strahm@      software engineers striving to build
intel.com           bigger and better idiot-proof programs,
(503) 264-4632   and the Universe trying to produce
            bigger and better idiots.  So far, the
                        Universe is winning.--Rich Cook
I am not speaking for Intel.  And Intel rarely speaks for me




From rem-conf Fri Jun 04 18:48:06 1999 
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Hi!
    I'm looking for recent statistics on the Mbone, e.g. how many people
are using it, etc.  Can you help?

    John Foote




From rem-conf Sun Jun 06 13:55:00 1999 
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From rem-conf Mon Jun 07 10:51:15 1999 
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From rem-conf Mon Jun 07 14:18:07 1999 
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I'm wondering how the consensus is emerging on DTMF carriage.  Are we
leaning more toward an out of band (TCP) method as in H.245, or an
inband (UDP) method with redundancy  as in ietf-avt-dtmf-01.txt?  Also
wondering if anyone can think of any common applications involving voice
and DTMF intermingled (in the same direction).  Thanks.





From rem-conf Mon Jun 07 14:29:40 1999 
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Bruce_Levens@nmss.com wrote:
> 
> I'm wondering how the consensus is emerging on DTMF carriage.  Are we
> leaning more toward an out of band (TCP) method as in H.245, or an
> inband (UDP) method with redundancy  as in ietf-avt-dtmf-01.txt?  Also
> wondering if anyone can think of any common applications involving voice
> and DTMF intermingled (in the same direction).  Thanks.

The revised spec draft for the latter should appear later today or
tomorrow, depending on the changes Scott Petrack and I still need to
work out.

IVR and any type of information service (banking, phone course
registration, etc.) are obvious examples of mixing the two. Given that
the DTMF detection is usually done in the DSP handling the voice bits,
doing DTMF in-band seems somewhat easier.


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



From rem-conf Tue Jun 08 22:56:32 1999 
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From: Evi Nemeth <evi@anchor.cs.colorado.edu>
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the usenix technical conference will be mboned jun 9-11,
9am to 5:30pm, pacific time (gmt-8).  see www.usenix.org/events/usenix99
for detailed program.  here are the parameters in case you cant see it
in sdr.

vic	224.2.187.101/59298
vat	224.2.192.144/17856
wb	224.2.138.245/43590

-evi



From rem-conf Wed Jun 09 07:09:43 1999 
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I've put the PostScript and PDF versions of the merged DTMF and tones
draft at http://www.cs.columbia.edu/~hgs/rtp/doc/dtmf/tones.pdf and
tones.ps. Scott and I would appreciate your comments. I will turn this
into an (ASCII) I-D shortly, barring "you must be crazy" comments.

Thanks.



From rem-conf Wed Jun 09 08:59:01 1999 
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A couple of quick questions/comments:

  1.  What's the relationship of the line events (section 2.9) and trunk
events (section 3.1) to the various signalling-over-IP proposals that are
floating around (which I'm not very familiar with, thus the question.)?

  2.  In particular, I'm not quite sure how or why things such as
'incoming seizure' would be transmitted via RTP -- seems to me that I'd be
expecting these to be carried 'higher up', via SIP, MGCP, H.245 or
something.

  3.  The chart on P.11 has 3 columns of numbers.  I assume that the
second two columns are cadence (on/off) information.  It appears that the
cadence information did not find its way into the payload format.  (I seem
to recall it being in the previous tones draft).  I generally agree with
its removal, because of the difficulty of figuring out the cadence without
listening to a long sample to figure it out -- as pointed out in the
introduction.

  4.  WRT the comment in the introduction "When calling a foreign country,
the caller generally hears the tones as defined in the country called,
rather than those familiar to him": As I understand ISDN, this is not
generally how things happen in that environment.   For example, if a call
is placed from the US to France, and the called party is busy, an
indication is sent back to the calling switch, which generates the busy
tone, and the overseas call is dropped.  I don't rightly know how other
tones (ringing, for example) are handled.  Not that it matters really --
there isn't any reason why one couldn't implement a system where the tones
are generated remotely.  So, I'd recommend leaving the comment in, but
change the wording so it sounds like a "possible use," recognizing that
different RTP users may decide to go the ISDN-style route -- right now it
reads more like "this is how it should be done."  (even though it doesn't
have the 'SHOULD' keyword.)

  5.  Similarly, I'd suggest that the last paragraph on the first page be
modified to read something like "Gateway implementations which send
signalling events via RTP SHOULD send both named signals ...."

  6.  What was the rationale for removing 'Flash' from the DTMF events and
adding it to the Line events?  It seems to me to logically belong in the
DTMF events (even though it's *not* DTMF), for two reasons:

   -- Apart from 'on hook' and 'off hook', it's the only line event that
goes from a terminal to a gateway.
   -- In the PSTN, it's used much as DTMF is -- as an input device by the
user to indicate that the switch should do something.  I imagine that in a
VoIP environment, the usage would still be similar.

 7.  Speaking of on- and off- hook, wouldn't transmitting these as line
events imply that there was a continual RTP stream being generated from
the terminal?  I guess that if someone wanted to do this, they could, but
I would expect the on- and off- hook messages to generate a message in a
different protocol (again, SIP or H.323 or whatever) which caused the RTP
stream to be instantiated.

 8.  I like breaking up the draft into 5 different payload types -- allows
implementors the ability to pick-and-choose what they need for their
implementation.

 9.  Section 2.3, at the bottom of the page:  "An audio source SHOULD
start transmitting event packets as soon as it recognizes an event and
every 50 ms thereafter."  May I suggest that this should be something like
"at intervals of at most 50 ms thereafter."  I'm not even sure of the 50ms
limit -- seems to me that you'd want the event packets to come out at
about the same rate as the audio codec.  (as alluded to in 2.6.)

 10.  'Duration' for several events is undefined, as they reflect a state
transition, for instance the hook signals, seizure, etc...

  Many thanks!

--
C



Henning Schulzrinne wrote:

> I've put the PostScript and PDF versions of the merged DTMF and tones
> draft at http://www.cs.columbia.edu/~hgs/rtp/doc/dtmf/tones.pdf and
> tones.ps. Scott and I would appreciate your comments. I will turn this
> into an (ASCII) I-D shortly, barring "you must be crazy" comments.
>
> Thanks.




From rem-conf Wed Jun 09 17:39:32 1999 
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Chris Fulmer wrote:
> 
> A couple of quick questions/comments:
> 
>   1.  What's the relationship of the line events (section 2.9) and trunk
> events (section 3.1) to the various signalling-over-IP proposals that are
> floating around (which I'm not very familiar with, thus the question.)?

There are several answers to this. I'll name one, hopefully not too
controversial:

Imagine a "trunk replacement" service, where a T1 carrying voice is to
be replaced by an IP span. The voice is coded as G.723.1, channel by
channel, and carried as an "RTP trunk" (on-going AVT work on muxing).
The signaling (MF, if a digital trunk recovered from the LSB, which
wouldn't survive the G.723.1 encoding) is translated to the trunk
events. Possibly, DTMF is recognized and carried as a separate payload
as well.

Yes, one could use SIGTRANS in this case, but my hunch is that the draft
at hand would be much simpler to implement for in-band signaling.

> 
>   2.  In particular, I'm not quite sure how or why things such as
> 'incoming seizure' would be transmitted via RTP -- seems to me that I'd be
> expecting these to be carried 'higher up', via SIP, MGCP, H.245 or
> something.

If true signaling, yes, I'd using "real" signaling as well, as it avoids
exposing Internet systems to the historical details of MF signaling,
although others might disagree. See previous paragraph.

> 
>   3.  The chart on P.11 has 3 columns of numbers.  I assume that the
> second two columns are cadence (on/off) information.  It appears that the
> cadence information did not find its way into the payload format.  (I seem
> to recall it being in the previous tones draft).  I generally agree with
> its removal, because of the difficulty of figuring out the cadence without
> listening to a long sample to figure it out -- as pointed out in the
> introduction.

Simply missing enumeration (aka as my laziness). To be filled in once we
have rough consensus on the approach.


> 
>   4.  WRT the comment in the introduction "When calling a foreign country,
> the caller generally hears the tones as defined in the country called,
> rather than those familiar to him": As I understand ISDN, this is not
> generally how things happen in that environment.   For example, if a call
> is placed from the US to France, and the called party is busy, an
> indication is sent back to the calling switch, which generates the busy
> tone, and the overseas call is dropped.  I don't rightly know how other
> tones (ringing, for example) are handled.  Not that it matters really --
> there isn't any reason why one couldn't implement a system where the tones
> are generated remotely.  So, I'd recommend leaving the comment in, but
> change the wording so it sounds like a "possible use," recognizing that
> different RTP users may decide to go the ISDN-style route -- right now it
> reads more like "this is how it should be done."  (even though it doesn't
> have the 'SHOULD' keyword.)

Good point; will be added.

> 
>   5.  Similarly, I'd suggest that the last paragraph on the first page be
> modified to read something like "Gateway implementations which send
> signalling events via RTP SHOULD send both named signals ...."

Agreed.

> 
>   6.  What was the rationale for removing 'Flash' from the DTMF events and
> adding it to the Line events?  It seems to me to logically belong in the
> DTMF events (even though it's *not* DTMF), for two reasons:
> 
>    -- Apart from 'on hook' and 'off hook', it's the only line event that
> goes from a terminal to a gateway.
>    -- In the PSTN, it's used much as DTMF is -- as an input device by the
> user to indicate that the switch should do something.  I imagine that in a
> VoIP environment, the usage would still be similar.

I wasn't quite sure where this belonged. Many systems that use DTMF
(say, a gateway), will never send this. I'll move it back.

> 
>  7.  Speaking of on- and off- hook, wouldn't transmitting these as line
> events imply that there was a continual RTP stream being generated from
> the terminal?  I guess that if someone wanted to do this, they could, but
> I would expect the on- and off- hook messages to generate a message in a
> different protocol (again, SIP or H.323 or whatever) which caused the RTP
> stream to be instantiated.

I'm also dubious about on-hook and off-hook as line events. Anybody has
a reason why they'd be useful?

> 
>  8.  I like breaking up the draft into 5 different payload types -- allows
> implementors the ability to pick-and-choose what they need for their
> implementation.
> 
>  9.  Section 2.3, at the bottom of the page:  "An audio source SHOULD
> start transmitting event packets as soon as it recognizes an event and
> every 50 ms thereafter."  May I suggest that this should be something like
> "at intervals of at most 50 ms thereafter."  I'm not even sure of the 50ms
> limit -- seems to me that you'd want the event packets to come out at
> about the same rate as the audio codec.  (as alluded to in 2.6.)

Will be fixed.

> 
>  10.  'Duration' for several events is undefined, as they reflect a state
> transition, for instance the hook signals, seizure, etc...

This would be how long this condition has persisted so far.

> 
>   Many thanks!
> 
> --
> C
> 
> Henning Schulzrinne wrote:
> 
> > I've put the PostScript and PDF versions of the merged DTMF and tones
> > draft at http://www.cs.columbia.edu/~hgs/rtp/doc/dtmf/tones.pdf and
> > tones.ps. Scott and I would appreciate your comments. I will turn this
> > into an (ASCII) I-D shortly, barring "you must be crazy" comments.
> >
> > Thanks.



From rem-conf Thu Jun 10 10:48:52 1999 
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From: rodgers@nlm.nih.gov (R. P. Channing Rodgers, M.D.)
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To: rem-conf@es.net
Subject: Re: usenix technical conference
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Is anyone seeing any of these sessions (sdr. vic, vat, wb)???  I see
nohting here...

Cheerio, Rick Rodgers
-------
   R. P. C. Rodgers, M.D. * rodgers@nlm.nih.gov * (301)496-9305 (voice, fax)
   CSB, LHNCBC, National Library of Medicine, NIH
   8600 Rockville Pike, Bethesda MD 20894 USA
   http://www.nlm.nih.gov/, search personnel roster for "rodgers"


> From rem-conf-request@es.net Wed Jun  9 02:06:17 1999
> Date: Tue, 8 Jun 1999 23:45:59 -0600 (MDT)
> From: Evi Nemeth <evi@anchor.cs.colorado.edu>
> To: rem-conf@es.net
> Subject: usenix technical conference 
> Cc: evi@anchor.cs.colorado.edu
> X-Mailing-List: <rem-conf@es.net> 
> X-Loop: rem-conf@es.net
> Content-Length: 288
> 
> the usenix technical conference will be mboned jun 9-11,
> 9am to 5:30pm, pacific time (gmt-8).  see www.usenix.org/events/usenix99
> for detailed program.  here are the parameters in case you cant see it
> in sdr.
> 
> vic	224.2.187.101/59298
> vat	224.2.192.144/17856
> wb	224.2.138.245/43590
> 
> -evi
> 
> 



From rem-conf Thu Jun 10 11:49:19 1999 
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From: "S.Grant" <grant@psc.edu>
To: "R. P. Channing Rodgers, M.D." <rodgers@nlm.nih.gov>, <rem-conf@es.net>
References: <199906101738.NAA12356@billings.csb>
Subject: Re: usenix technical conference
Date: Thu, 10 Jun 1999 14:45:15 -0400
Organization: NLANR/NCNE Pittsburgh Supercomputing Center
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I'm getting video, but rat's not carrying the audio.

    Shirl Grant                       NCNE Engineering Services
   grant@ncne.org               NLANR / Pittsburgh Supercomputing Center

-


----- Original Message -----
From: R. P. Channing Rodgers, M.D. <rodgers@nlm.nih.gov>
To: <rem-conf@es.net>
Sent: Thursday, June 10, 1999 1:38 PM
Subject: Re: usenix technical conference


>
> Is anyone seeing any of these sessions (sdr. vic, vat, wb)???  I see
> nohting here...
>
> Cheerio, Rick Rodgers
> -------
>    R. P. C. Rodgers, M.D. * rodgers@nlm.nih.gov * (301)496-9305
(voice, fax)
>    CSB, LHNCBC, National Library of Medicine, NIH
>    8600 Rockville Pike, Bethesda MD 20894 USA
>    http://www.nlm.nih.gov/, search personnel roster for "rodgers"
>
>
> > From rem-conf-request@es.net Wed Jun  9 02:06:17 1999
> > Date: Tue, 8 Jun 1999 23:45:59 -0600 (MDT)
> > From: Evi Nemeth <evi@anchor.cs.colorado.edu>
> > To: rem-conf@es.net
> > Subject: usenix technical conference
> > Cc: evi@anchor.cs.colorado.edu
> > X-Mailing-List: <rem-conf@es.net>
> > X-Loop: rem-conf@es.net
> > Content-Length: 288
> >
> > the usenix technical conference will be mboned jun 9-11,
> > 9am to 5:30pm, pacific time (gmt-8).  see
www.usenix.org/events/usenix99
> > for detailed program.  here are the parameters in case you cant see
it
> > in sdr.
> >
> > vic 224.2.187.101/59298
> > vat 224.2.192.144/17856
> > wb 224.2.138.245/43590
> >
> > -evi
> >
> >
>




From rem-conf Thu Jun 10 12:03:07 1999 
From rem-conf-request@es.net Thu Jun 10 12:03:07 1999
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To: "Evi Nemeth" <evi@anchor.cs.colorado.edu>, <rem-conf@es.net>
Cc: <evi@anchor.cs.colorado.edu>
References: <199906090545.XAA16604@anchor.cs.colorado.edu>
Subject: Re: usenix technical conference 
Date: Thu, 10 Jun 1999 14:58:34 -0400
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Evi,

    Can you broadcast such that "rat" can hear the sessions?  If you're
broadcasting with vat, do so with "vat -r".


Thanks,

    Shirl Grant                       NCNE Engineering Services
   grant@ncne.org               NLANR / Pittsburgh Supercomputing Center

-


----- Original Message -----
From: Evi Nemeth <evi@anchor.cs.colorado.edu>
To: <rem-conf@es.net>
Cc: <evi@anchor.cs.colorado.edu>
Sent: Wednesday, June 09, 1999 1:45 AM
Subject: usenix technical conference


> the usenix technical conference will be mboned jun 9-11,
> 9am to 5:30pm, pacific time (gmt-8).  see
www.usenix.org/events/usenix99
> for detailed program.  here are the parameters in case you cant see it
> in sdr.
>
> vic 224.2.187.101/59298
> vat 224.2.192.144/17856
> wb 224.2.138.245/43590
>
> -evi
>




From rem-conf Thu Jun 10 12:04:22 1999 
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From: "Cain, Michael E." <mcain@mediaone.com>
To: <rem-conf@es.net>
Subject: RE: usenix technical conference
Date: Thu, 10 Jun 1999 15:02:55 -0400
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Interesting.  I'm getting audio with rat and an agenda with wb, but have not
seen any vido packets.
Michael Cain
MediaOne Labs
Westminster, CO

> ----------
> From: 	S.Grant[SMTP:grant@psc.edu]
> Sent: 	Thursday, June 10, 1999 12:45 PM
> To: 	R. P. Channing Rodgers, M.D.; rem-conf@es.net
> Subject: 	Re: usenix technical conference
> 
> I'm getting video, but rat's not carrying the audio.
> 
>     Shirl Grant                       NCNE Engineering Services
>    grant@ncne.org               NLANR / Pittsburgh Supercomputing Center
> 
> -
> 
> 
> ----- Original Message -----
> From: R. P. Channing Rodgers, M.D. <rodgers@nlm.nih.gov>
> To: <rem-conf@es.net>
> Sent: Thursday, June 10, 1999 1:38 PM
> Subject: Re: usenix technical conference
> 
> 
> >
> > Is anyone seeing any of these sessions (sdr. vic, vat, wb)???  I see
> > nohting here...
> >
> > Cheerio, Rick Rodgers
> > -------
> >    R. P. C. Rodgers, M.D. * rodgers@nlm.nih.gov * (301)496-9305
> (voice, fax)
> >    CSB, LHNCBC, National Library of Medicine, NIH
> >    8600 Rockville Pike, Bethesda MD 20894 USA
> >    http://www.nlm.nih.gov/, search personnel roster for "rodgers"
> >
> >
> > > From rem-conf-request@es.net Wed Jun  9 02:06:17 1999
> > > Date: Tue, 8 Jun 1999 23:45:59 -0600 (MDT)
> > > From: Evi Nemeth <evi@anchor.cs.colorado.edu>
> > > To: rem-conf@es.net
> > > Subject: usenix technical conference
> > > Cc: evi@anchor.cs.colorado.edu
> > > X-Mailing-List: <rem-conf@es.net>
> > > X-Loop: rem-conf@es.net
> > > Content-Length: 288
> > >
> > > the usenix technical conference will be mboned jun 9-11,
> > > 9am to 5:30pm, pacific time (gmt-8).  see
> www.usenix.org/events/usenix99
> > > for detailed program.  here are the parameters in case you cant see
> it
> > > in sdr.
> > >
> > > vic 224.2.187.101/59298
> > > vat 224.2.192.144/17856
> > > wb 224.2.138.245/43590
> > >
> > > -evi
> > >
> > >
> >
> 
> 




From rem-conf Thu Jun 10 13:03:38 1999 
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Subject: Re: usenix technical conference 
In-Reply-To: Your message of "Thu, 10 Jun 1999 15:02:55 EDT."
             <B147CE8D09F1D111BD7100805F19B1C04D8DA6@coexch01.mediaone.com> 
From: bmah@CA.Sandia.GOV (Bruce A. Mah)
Reply-To: bmah@CA.Sandia.GOV
X-Face: g~c`.{#4q0"(V*b#g[i~rXgm*w;:nMfz%_RZLma)UgGN&=j`5vXoU^@n5<Pi&akO)o^8;[r
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--==_Exmh_-1995191658P
Content-Type: text/plain; charset=us-ascii

If memory serves me right, "Cain, Michael E." wrote:
> Interesting.  I'm getting audio with rat and an agenda with wb, but have not
> seen any vido packets.

Just so someone knows that they *are* alive over there, I'm currently
receiving video with vic 2.8.1 at 48kbps, no loss (of a blank screen,
unfortunately), and I've been listening to the audio for the past day
and a half (vat 4.0pre7).  wb has two pages on it...the first is
miscellaneous graffiti and the second is the second day's schedule.

I have control packets from about eight participants in vat.

Bruce.




--==_Exmh_-1995191658P
Content-Type: application/pgp-signature

-----BEGIN PGP MESSAGE-----
Version: 2.6.2

iQCVAwUBN2AZYqjOOi0j7CY9AQEtXAP+OVNi77quMQPX5/a+9+MoTb/xT9sWfUM6
Rph3bVN8XKJ11UDACLp6qV+rFDUbFOuvHYuyXm5xbanGKRx7R+XicbOoh3dlD3bk
oNKIvpsl5l0RB7Hhpt7tcyqws/4lqb5SPvidolhijgc3F9SHGJ6koVkl8ZJJbRZz
QrFxvBy0+VY=
=pDaH
-----END PGP MESSAGE-----

--==_Exmh_-1995191658P--



From rem-conf Thu Jun 10 16:46:30 1999 
From rem-conf-request@es.net Thu Jun 10 16:46:29 1999
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Date: Thu, 10 Jun 1999 19:43:17 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Graeme Gibbs <gibbs@nortelnetworks.com>
CC: "'rem-conf@es.net'" <rem-conf@es.net>,
        Rade Gvozdanovic <rade@nortelnetworks.com>,
        Rajakulasingam Raviraj <raviraj@nortelnetworks.com>,
        "'Chaoxin.Qiu@mail.sprint.com'" <Chaoxin.Qiu@mail.sprint.com>
Subject: Re: Additional Named Signal Events
References: <391D82D86F58D111B8250000F805C4D0023526EF@bhari886.europe.nortel.com>
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Graeme Gibbs wrote:
> 
> To whoever is editting/championing "RTP Payloads for Telephone Signal
> Events"
> 
> Firstly I have not been following the mailing list so apologies if this
> topic is old & cold.
> 
> Basically I believe this draft would benefit from the addition of 2 extra
> Named Signal Events ie
> 2100Hz tones (ITU G.164)&
> 2100Hz tone with Phase reversals (ITU G.165)

The draft (at http://www.cs.columbia.edu/~hgs/rtp/doc/dtmf/tones.pdf)
currently contains CED in the 'fax' set. CED appears to be a 2,100 Hz
tone, so it covers the G.164 case. Is a special G.164 version of this
tone needed or will this do? Also, I've tentatively added a 2,100 Hz
tone with phase reversals into the fax set. Does this tone have a common
name?


> 
> Rationale:
> These tones, while not used by traditional TDM switches per se, are used by
> echo cancellers within the TDM network to turn off echo cancellation for
> calls carrying modems & faxes.
> Certain carriers may wish to build telephony over IP transit networks,
> where, at least initially, modem/faxes are carried transparently as G.711
> (64kbit/s) packets over IP
> (yes I know this amounts to data over voice over data, but it may be some
> period before all gateways support all modulation techniques, and carriers
> must provide equivalence to current Public Switched Telephone Network from
> Day 1). Modems/faxes sending such tones would expect both the intervening IP
> network and prior/subsequent echo cancellers in PSTN to moderate their
> behaviour accordingly.
> 
> Take  a scenario where the VOIP gateways are running voice calls as
> compressed to G.729, with silence suppression and echo cancellation, and
> such a 2100Hz tone is detected
> The local GW must: Upspeed to 64 kbit/s & turn off silence suppression, and
> echo cancellation
> The remote GW must also: Upspeed to 64 kbit/s & turn off silence
> suppression, and echo cancellation
> The tone must be propagated back into the PSTN by remote GW
> 
> 2100Hz tones will not reliably propagate through G.729, therefore if the
> system  relied on passing  this tone in-band, the near-end modem would have
> to detect tone , decide to upspeed (maybe local CAC decisions), and then
> execute upspeeding before far end GW can begin to detect tones in the stream
> from the IP network- this may take too long. This method also relies on
> codec that can receive one codec type and transmit in another (desirable BUT
> not always the case)
> However if the near-end GW were to send a Named Signal Event once it had
> detected the tone, this would allow CAC and upspeeding to occur in parallel,
> together with a correct tone being enforced to the PSTN
> 
> Any replies, please mail direct to me as I do not subscribe to avt list

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



From rem-conf Thu Jun 10 17:21:04 1999 
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From: "Fox, Mike" <mfox@nuera.com>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, Graeme Gibbs
	 <gibbs@nortelnetworks.com>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Additional Named Signal Events
Date: Thu, 10 Jun 1999 17:17:27 -0700
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Perhaps the ANS signal designation of the 2100 Hz answer tone from V.25
would be a more common signal name. A bar over ANS indicates the phase
reversed signal.

Mike Fox

		-----Original Message-----
		From:	Henning Schulzrinne
[mailto:schulzrinne@cs.columbia.edu]
		Sent:	Thursday, June 10, 1999 4:43 PM
		To:	Graeme Gibbs
		Cc:	'rem-conf@es.net'; Rade Gvozdanovic; Rajakulasingam
Raviraj; 'Chaoxin.Qiu@mail.sprint.com'
		Subject:	Re: Additional Named Signal Events

		Graeme Gibbs wrote:
		> 
		> To whoever is editting/championing "RTP Payloads for
Telephone Signal
		> Events"
		> 
		> Firstly I have not been following the mailing list so
apologies if this
		> topic is old & cold.
		> 
		> Basically I believe this draft would benefit from the
addition of 2 extra
		> Named Signal Events ie
		> 2100Hz tones (ITU G.164)&
		> 2100Hz tone with Phase reversals (ITU G.165)

		The draft (at
http://www.cs.columbia.edu/~hgs/rtp/doc/dtmf/tones.pdf)
		currently contains CED in the 'fax' set. CED appears to be a
2,100 Hz
		tone, so it covers the G.164 case. Is a special G.164
version of this
		tone needed or will this do? Also, I've tentatively added a
2,100 Hz
		tone with phase reversals into the fax set. Does this tone
have a common
		name?


		> 
		> Rationale:
		> These tones, while not used by traditional TDM switches
per se, are used by
		> echo cancellers within the TDM network to turn off echo
cancellation for
		> calls carrying modems & faxes.
		> Certain carriers may wish to build telephony over IP
transit networks,
		> where, at least initially, modem/faxes are carried
transparently as G.711
		> (64kbit/s) packets over IP
		> (yes I know this amounts to data over voice over data, but
it may be some
		> period before all gateways support all modulation
techniques, and carriers
		> must provide equivalence to current Public Switched
Telephone Network from
		> Day 1). Modems/faxes sending such tones would expect both
the intervening IP
		> network and prior/subsequent echo cancellers in PSTN to
moderate their
		> behaviour accordingly.
		> 
		> Take  a scenario where the VOIP gateways are running voice
calls as
		> compressed to G.729, with silence suppression and echo
cancellation, and
		> such a 2100Hz tone is detected
		> The local GW must: Upspeed to 64 kbit/s & turn off silence
suppression, and
		> echo cancellation
		> The remote GW must also: Upspeed to 64 kbit/s & turn off
silence
		> suppression, and echo cancellation
		> The tone must be propagated back into the PSTN by remote
GW
		> 
		> 2100Hz tones will not reliably propagate through G.729,
therefore if the
		> system  relied on passing  this tone in-band, the near-end
modem would have
		> to detect tone , decide to upspeed (maybe local CAC
decisions), and then
		> execute upspeeding before far end GW can begin to detect
tones in the stream
		> from the IP network- this may take too long. This method
also relies on
		> codec that can receive one codec type and transmit in
another (desirable BUT
		> not always the case)
		> However if the near-end GW were to send a Named Signal
Event once it had
		> detected the tone, this would allow CAC and upspeeding to
occur in parallel,
		> together with a correct tone being enforced to the PSTN
		> 
		> Any replies, please mail direct to me as I do not
subscribe to avt list

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



From rem-conf Thu Jun 10 18:56:57 1999 
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From: "Fox, Mike" <mfox@nuera.com>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, Graeme Gibbs
	 <gibbs@nortelnetworks.com>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Additional Named Signal Events
Date: Thu, 10 Jun 1999 18:53:44 -0700
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Hi,
I just looked at T.30 in section 4.1.1 it designates the Called terminal
identification (CED) answer tone, a continuous 2100 Hz  * 15 Hz tone. In
section 4.1.2 it specifies the optional procedures of V.8 using the ANSam
defined in V.8.
In section 6.1 for V.34 Recommendations, the spec uses the ANSam signal from
V.8. 

The V.8 Modified answer tone ANSam is a sinewave signal at 2100 * 1 Hz with
phase reversals at an interval of 450 * 25 ms, amplitude-modulated by a
sinewave at 15 * 0.1 Hz.

In V.25 the signals are called ANS, and ANS(with a bar on top). 

If there was to be a vote, I would pick ANS and ANSam for the signal
indications. When the 2100 is detected the ANS is indicated, and if the
phase reversal is detected a ANSam signal is indicated until the ANSam
signal goes away.

Best regards,
Mike Fox
Nuera Communications



From rem-conf Thu Jun 10 19:41:11 1999 
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Date: Thu, 10 Jun 1999 22:37:53 -0400
To: "Fox, Mike" <mfox@nuera.com>,
        Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Graeme Gibbs <gibbs@nortelnetworks.com>
From: Fatih Erdin <fatih.erdin@ascend.com>
Subject: RE: Additional Named Signal Events
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
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 om>
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At 06:53 PM 6/10/99 -0700, Fox, Mike wrote:
>Hi,
>I just looked at T.30 in section 4.1.1 it designates the Called terminal
>identification (CED) answer tone, a continuous 2100 Hz  * 15 Hz tone. In
>section 4.1.2 it specifies the optional procedures of V.8 using the ANSam
>defined in V.8.
>In section 6.1 for V.34 Recommendations, the spec uses the ANSam signal from
>V.8. 
>
>The V.8 Modified answer tone ANSam is a sinewave signal at 2100 * 1 Hz with
>phase reversals at an interval of 450 * 25 ms, amplitude-modulated by a
>sinewave at 15 * 0.1 Hz.
>
>In V.25 the signals are called ANS, and ANS(with a bar on top). 
>
>If there was to be a vote, I would pick ANS and ANSam for the signal
>indications. When the 2100 is detected the ANS is indicated, and if the
>phase reversal is detected a ANSam signal is indicated until the ANSam
>signal goes away.

But in this case, the calling terminal can not distinguish whether the
called terminal is V.32 (sends /ANS, 2100Hz with phase reversal as answer
tone) or V.34 (ANSam, 2100Hz + PR + AM) and thinks it as V.34, even though
it's V.32 actually. I think they all should be indicated separately: ANS,
/ANS and ANSam (even Cre for V.8bis devices. CRe consists of a dual-tone
signal with tones at 1375 Hz and 2002 Hz for 400ms followed by a single
tone at 400 Hz for 100 ms). 

Regards,
Fatih Erdin
Ascend Communications


>
>Best regards,
>Mike Fox
>Nuera Communications
>
>
>



From rem-conf Fri Jun 11 07:49:52 1999 
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Fatih Erdin wrote:
> 
>
> 
> But in this case, the calling terminal can not distinguish whether the
> called terminal is V.32 (sends /ANS, 2100Hz with phase reversal as answer
> tone) or V.34 (ANSam, 2100Hz + PR + AM) and thinks it as V.34, even though
> it's V.32 actually.

According to V.8 and V.25, ANSam is a sinewave signal at 2100 Hz with
phase reversals at 450 ms, and amplitude-modulated.

V.32 uses ANS (no phase reversals), from what I can tell. I don't see a
separate /ANS signal mentioned in the V.32 spec. Can somebody point me
to the section. (V.32 does mention that it uses the V.25 startup
procedure.)



From rem-conf Fri Jun 11 08:44:17 1999 
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Shirl,

  I think it's a multicast problem rather than an application problem
that's causing you to not get the audio.  The multicast had some trouble
on Wednesday because it was singly-homed to the vBNS and the vBNS was
experiencing MSDP session flaps to the rest of the world due to an apparent
IOS bug.  Dave Meyer kindly added a tunnel to AS10888 at PAIX so we should
have much better connectivity for Thursday and Friday.

  If you're still having trouble, perhaps you could try an mtrace?
mtrace mbone.conference.usenix.org 224.2.192.144 for the audio, or
mtrace mbone.conference.usenix.org 224.2.187.101 for the video.

  Bill



From rem-conf Fri Jun 11 12:50:46 1999 
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From: "Fox, Mike" <mfox@nuera.com>
To: Henning Schulzrinne <hgs@cs.columbia.edu>, Fatih Erdin
	 <fatih.erdin@ascend.com>
Cc: "Fox, Mike" <mfox@nuera.com>, Graeme Gibbs <gibbs@nortelnetworks.com>, 
	"'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Additional Named Signal Events
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I think Fatih is correct ( I would like to change my vote ) in that ANS,
/ANS, and ANSam (even /ANSam) are different signals. If a modem using the
V.25 procedures would like to disable the echo cancellers and suppressors it
will emit ANS /ANS ANS /ANS ( V32bis 6.2)  ... For a modem using V.8 answer
procedures it may emit ANSam, /ANSam, ANSam ... to disable echo cancellers
and suppressors. In V.8, section 7.2 the modem may emit ANSam without phase
reversals if the echo cancellers need not be disabled ( I do not know if any
products do this... ).

So do the signal indication packets look like:
Procedure:		indications (time -->)
1. V.25+V.8	 	ANS, ANS, ANS... 
2. V.25 ( EC disable )	ANS..., /ANS..., ANS..., /ANS...
3. V.8  			ANSam, ANSam,....
3. V.8 (EC disable )	ANSam..., /ANSam..., ANSam..., /ANSam...

Also V.8 section 7.2 notes that the DCE needs to distinguish ANSam from ANS.
This will probably add some delay and complexity to some implementations of
ANS detectors. 

Best regards,
Mike
Nuera Communications


		-----Original Message-----
		From:	Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
		Sent:	Friday, June 11, 1999 7:41 AM
		To:	Fatih Erdin
		Cc:	Fox, Mike; Graeme Gibbs; 'rem-conf@es.net'
		Subject:	Re: Additional Named Signal Events

		Fatih Erdin wrote:
		> 
		>
		> 
		> But in this case, the calling terminal can not distinguish
whether the
		> called terminal is V.32 (sends /ANS, 2100Hz with phase
reversal as answer
		> tone) or V.34 (ANSam, 2100Hz + PR + AM) and thinks it as
V.34, even though
		> it's V.32 actually.

		According to V.8 and V.25, ANSam is a sinewave signal at
2100 Hz with
		phase reversals at 450 ms, and amplitude-modulated.

		V.32 uses ANS (no phase reversals), from what I can tell. I
don't see a
		separate /ANS signal mentioned in the V.32 spec. Can
somebody point me
		to the section. (V.32 does mention that it uses the V.25
startup
		procedure.)



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Date: Fri, 11 Jun 1999 15:42:52 -0400
To: Henning Schulzrinne <hgs@cs.columbia.edu>
From: Fatih Erdin <fatih.erdin@ascend.com>
Subject: Re: Additional Named Signal Events
Cc: "Fox, Mike" <mfox@nuera.com>, Graeme Gibbs <gibbs@nortelnetworks.com>,
        "'rem-conf@es.net'" <rem-conf@es.net>
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At 10:41 AM 6/11/99 -0400, Henning Schulzrinne wrote:
>Fatih Erdin wrote:
>> 
>>
>> 
>> But in this case, the calling terminal can not distinguish whether the
>> called terminal is V.32 (sends /ANS, 2100Hz with phase reversal as answer
>> tone) or V.34 (ANSam, 2100Hz + PR + AM) and thinks it as V.34, even though
>> it's V.32 actually.
>
>According to V.8 and V.25, ANSam is a sinewave signal at 2100 Hz with
>phase reversals at 450 ms, and amplitude-modulated.
>
>V.32 uses ANS (no phase reversals), from what I can tell. I don't see a
>separate /ANS signal mentioned in the V.32 spec. Can somebody point me
>to the section. (V.32 does mention that it uses the V.25 startup
>procedure.)
>

You're right, V.32 spec doesn't clearly mention ANS (with a bar on top)
with phase reversals (what I meant with /ANS was ANS with a bar on top,
2100 Hz with phase reversals). 
Since V.32 is duplex modem, the line (voice) echo cancellers on the path
should be disabled. The echo canceller disabler tone has phase reversals
according to G.165, section 4 (the plain 2100 Hz disables the echo
suppressors only). I've also changed the modulation type to V.32 of my
modem and observed the phase reversals in the answer tone.

Regards,

Fatih Erdin
Ascend Communications


>
>
**************************************************************
Fatih Erdin                     tel   : (732) 578-3113
Senior DSP/Software Engineer    fax   : (732) 578-3131 
Ascend Communications, Inc.     email : fatih.erdin@ascend.com
http://www.ascend.com
**************************************************************




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Subject: Re: Additional Named Signal Events
References: <3.0.3.32.19990610223753.009dcb90@basset.eng.ascend.com>
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I've updated the DTMF/tones/events payload draft according to the
suggestions received. If I missed something, please let me know. One
change made was a change from the name "fax" to "data", since this seems
more appropriate for the current collection of signals.



From rem-conf Fri Jun 11 15:05:11 1999 
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To: Bill Fenner <fenner@research.att.com>
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Subject: Re: usenix technical conference 
In-Reply-To: Your message of "Fri, 11 Jun 1999 08:42:04 PDT."
             <199906111542.LAA16894@windsor.research.att.com> 
From: bmah@CA.Sandia.GOV (Bruce A. Mah)
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If memory serves me right, Bill Fenner wrote:

> The multicast had some trouble
> on Wednesday because it was singly-homed to the vBNS and the vBNS was
> experiencing MSDP session flaps to the rest of the world due to an apparent
> IOS bug.  Dave Meyer kindly added a tunnel to AS10888 at PAIX so we should
> have much better connectivity for Thursday and Friday.
> 
>   If you're still having trouble, perhaps you could try an mtrace?
> mtrace mbone.conference.usenix.org 224.2.192.144 for the audio, or
> mtrace mbone.conference.usenix.org 224.2.187.101 for the video.

Bill, et al--

I was having (for the most part) pretty good reception the last couple 
of days.  Now I'm receiving video but no audio.

I'm starting to suspect an application or user problem because I'm 
continuing to get control packets on the audio session, just no data.

For kicks, I've appended an mtrace on the audio and video multicast
addresses.  Personally, for me, it's not critically important that
"someone" (indefinite third-person) fix this problem, but I'm providing
this info for anyone who can use it.

Bruce.

Mtrace from 209.179.127.5 to 146.246.243.44 via group 224.2.192.144
Querying full reverse path... 
  0  stennis.ca.sandia.gov (146.246.243.44)
 -1  con243.ca.sandia.gov (146.246.243.254)  PIM/Special  thresh^ 0  
 -2  snl-outnet.ca.sandia.gov (146.246.250.254)  PIM/Special  thresh^ 0  
 -3  esnetatm-outnetfddi.llnl.gov (192.188.35.16)  PIM/Special  thresh^ 32  
 -4  lbl-mcp2p-llnl.es.net (198.129.78.125)  PIM/Special  thresh^ 0  
 -5  ames2-mcp2p-lbl.es.net (134.55.33.2)  PIM/BGP4+  thresh^ 64  
 -6  ? (198.9.201.254)  PIM/BGP4+  thresh^ 0  
 -7  pao5.pao4.verio.net (198.9.200.1)  PIM/Special  thresh^ 0  Reached RP/Core
 -8  mbone.conference.usenix.org (209.179.127.5)  DVMRP  thresh^ 64  
 -9  mbone.conference.usenix.org (209.179.127.5)
Round trip time 58 ms; total ttl of 68 required.
 
Waiting to accumulate statistics...Results after 10 seconds:
 
  Source        Response Dest    Overall     Packet Statistics For Traffic From
209.179.127.5   224.0.1.32       Packet      209.179.127.5 To 224.2.192.144
     v       __/  rtt   61 ms     Rate       Lost/Sent = Pct  Rate
209.179.127.5   mbone.conference.usenix.org 
     v     ^      ttl   65        23 pps        0/2    = --    0 pps
129.250.16.158 
198.9.200.1     pao5.pao4.verio.net Reached RP/Core
     v     ^      ttl   65       147 pps        0/2    = --    0 pps
198.9.200.2    
198.9.201.254   ?              
     v     ^      ttl   65       386 pps        0/2    = --    0 pps
198.9.201.13   
134.55.33.2     ames2-mcp2p-lbl.es.net 
     v     ^      ttl   68       404 pps        0/2    = --    0 pps
134.55.33.1    
198.129.78.125  lbl-mcp2p-llnl.es.net 
     v     ^      ttl   68       413 pps        0/2    = --    0 pps
198.129.78.126 
192.188.35.16   esnetatm-outnetfddi.llnl.gov 
     v     ^      ttl   68        56 pps        0/2    = --    0 pps
192.188.35.3   
146.246.250.254 snl-outnet.ca.sandia.gov 
     v     ^      ttl   68        46 pps        0/2    = --    0 pps
146.246.250.252
146.246.243.254 con243.ca.sandia.gov 
     v      \__   ttl   68        26 pps        ?/2            0 pps
146.246.243.44  146.246.243.44
  Receiver      Query Source
 
Mtrace from 209.179.127.5 to 146.246.243.44 via group 224.2.187.101
Querying full reverse path... 
  0  stennis.ca.sandia.gov (146.246.243.44)
 -1  con243.ca.sandia.gov (146.246.243.254)  PIM/Special  thresh^ 0  
 -2  snl-outnet.ca.sandia.gov (146.246.250.254)  PIM/Special  thresh^ 0  
 -3  esnetatm-outnetfddi.llnl.gov (192.188.35.16)  PIM/Special  thresh^ 32  
 -4  lbl-mcp2p-llnl.es.net (198.129.78.125)  PIM/Special  thresh^ 0  
 -5  ames2-mcp2p-lbl.es.net (134.55.33.2)  PIM/BGP4+  thresh^ 64  
 -6  ? (198.9.201.254)  PIM/BGP4+  thresh^ 0  
 -7  pao5.pao4.verio.net (198.9.200.1)  PIM/Special  thresh^ 0  Reached RP/Core
 -8  mbone.conference.usenix.org (209.179.127.5)  DVMRP  thresh^ 64  
 -9  mbone.conference.usenix.org (209.179.127.5)
Round trip time 56 ms; total ttl of 68 required.
 
Waiting to accumulate statistics...Results after 10 seconds:
 
  Source        Response Dest    Overall     Packet Statistics For Traffic From
209.179.127.5   224.0.1.32       Packet      209.179.127.5 To 224.2.187.101
     v       __/  rtt  481 ms     Rate       Lost/Sent = Pct  Rate
209.179.127.5   mbone.conference.usenix.org 
     v     ^      ttl   65        19 pps        0/189  =  0%  18 pps
129.250.16.158 
198.9.200.1     pao5.pao4.verio.net Reached RP/Core
     v     ^      ttl   65       158 pps        0/189  =  0%  18 pps
198.9.200.2    
198.9.201.254   ?              
     v     ^      ttl   65       404 pps        0/189  =  0%  18 pps
198.9.201.13   
134.55.33.2     ames2-mcp2p-lbl.es.net 
     v     ^      ttl   68       387 pps       -1/189  =  0%  18 pps
134.55.33.1    
198.129.78.125  lbl-mcp2p-llnl.es.net 
     v     ^      ttl   68       422 pps        7/190  =  3%  19 pps
198.129.78.126 
192.188.35.16   esnetatm-outnetfddi.llnl.gov 
     v     ^      ttl   68        39 pps        0/183  =  0%  18 pps
192.188.35.3   
146.246.250.254 snl-outnet.ca.sandia.gov 
     v     ^      ttl   68        33 pps        0/183  =  0%  18 pps
146.246.250.252
146.246.243.254 con243.ca.sandia.gov 
     v      \__   ttl   68        21 pps        ?/183         18 pps
146.246.243.44  146.246.243.44
  Receiver      Query Source
 



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From rem-conf Mon Jun 14 08:52:27 1999 
From rem-conf-request@es.net Mon Jun 14 08:52:26 1999
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Message-ID: <61891BA043DED21180920090273F173802E71D@argentina.telogy.com>
From: Roy Spitzer <rspitzer@telogy.com>
To: 'Henning Schulzrinne' <hgs@cs.columbia.edu>, Fatih Erdin
	 <fatih.erdin@ascend.com>
Cc: "Fox, Mike" <mfox@nuera.com>, Graeme Gibbs <gibbs@nortelnetworks.com>, 
	"'rem-conf@es.net'" <rem-conf@es.net>, Zoran Mladenovic
	 <zmladenovic@telogy.com>
Subject: RE: Additional Named Signal Events
Date: Mon, 14 Jun 1999 11:32:36 -0400
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We at Telogy Networks, Inc. believe there is some confusion here.  Our modem
expert offers this explanation.

There is a confusion between V.8 and V.25 with the modulation mode (V.32 or
V.34 or V.90). 

With V.32 modems, one of the major problems was lack of specified automoding
procedure (how you fall back to V.22, V.22bis etc.). People realized the
need to do it up front (during the standardization of V.34), in the very
first phase of handshake. That is how V.8 was created. As a means to
distinguish V.8 capable modem, from non-V.8 modems (or modems that
typically, but not necessarily comply with V.25), ANSam was used. During the
standardization process, special attention was given to the fact that all
installed network devices need to continue functioning properly in the
presence of ANSam (i.e. echo cancellers in particular).
Now, noting prevents modem manufacturers to manufacture V.32 modem, with the
V.8 handshake up front. Most, if not all, of V.32 designs are old designs
and consequently they support V.25 only.

As far as discussion going on, I am not sure exactly what was the point. V.8
does not specifically mention ANSam w/o phase reversals, although in some
cases (fax for example), it might be beneficial to have network echo
cancellers present. In my opinion, V.8 does not exclude ANSam w/o phase
reversals either.

Regards,
Roy

Roy Spitzer				Telogy Networks, Inc.
Phone:	(301) 515-6531			20250 Century Blvd.
Fax:	(301) 515-7954			Germantown, MD 20874
Email:	rspitzer@telogy.com


		-----Original Message-----
		From:	Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
		Sent:	Friday, June 11, 1999 10:41 AM
		To:	Fatih Erdin
		Cc:	Fox, Mike; Graeme Gibbs; 'rem-conf@es.net'
		Subject:	Re: Additional Named Signal Events

		Fatih Erdin wrote:
		> 
		>
		> 
		> But in this case, the calling terminal can not distinguish
whether the
		> called terminal is V.32 (sends /ANS, 2100Hz with phase
reversal as answer
		> tone) or V.34 (ANSam, 2100Hz + PR + AM) and thinks it as
V.34, even though
		> it's V.32 actually.

		According to V.8 and V.25, ANSam is a sinewave signal at
2100 Hz with
		phase reversals at 450 ms, and amplitude-modulated.

		V.32 uses ANS (no phase reversals), from what I can tell. I
don't see a
		separate /ANS signal mentioned in the V.32 spec. Can
somebody point me
		to the section. (V.32 does mention that it uses the V.25
startup
		procedure.)



From rem-conf Mon Jun 14 18:07:01 1999 
From rem-conf-request@es.net Mon Jun 14 18:06:59 1999
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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        Terry G Lyons <tlyons@holmdel.exchange.lucent.com>
Subject: DTMF draft
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Since there seems to be general interest in moving this draft along
quickly, I'd like to make sure that the one technical change made to the
DTMF part receives proper comment at this stage.

Previously, the DTMF digit extended (logically) backward in time. For
example, a timestamp of 1000 and a duration of 50 meant that the digit
started at 950 and continued until 1000.

Unfortunately, this contradicts the definition of timestamps in the RTP
spec, which refer to the "first sample". We don't have samples here, but
it makes more sense to have time "move forward" as for all other media
types, so that if we send both audio and DTMF, the same timestamp refers
to the same position in time. This makes playout algorithms work
naturally.

The only effect of this change seems to be that you get multiple DTMF
packets with the same timestamp (and increasing duration), but that
seems harmless.

Comments are appreciated.

Since the general discussion seems to have subsided a bit, I will submit
the draft as an I-D by mid-week.

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



From rem-conf Mon Jun 14 21:46:04 1999 
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Date: Tue, 15 Jun 1999 14:42:07 +1000
To: mbone@isi.edu, multicast@apan.net, multicast-eng@apan.net
From: Markus Buchhorn <markus@acsys.anu.edu.au>
Subject: Global Graphical Session Directory Monitor
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Hi All

By stomping on the shoulders of giants (Kevin Almeroth and Peter Parnes)
I've finally built something I drew up nearly 5 years ago :-)

http://mbone.au.apan.net/MSDL

MSDL is a worldwide session directory monitor, using input from Almeroth
sites (sdr-plugin) and mSD sites (Parnes Java code) and some mSD-clones
around the globe, together with some bodgy geography and suspect Perl code
to produce a (geo+)graphical log of global (ttl >= 127) session
advertisements on the MBONE over the last 24 hours.

At this stage it shows around 50 sites in 3 regions (Asia Pacific, North
America, Europe), with a graph of the total count of sessions over the last
24 hours at 15 minute intervals (where appropriate). Graphs are displayed
in a semi-geographically-correct fashion, so that you can eyeball regions
with problems. Each graph is itself a link to a high-resolution graph, plus
any others at the same "site" (city). See the top page for details.

It does not yet show connectivity between sites, nor view-ability on a
per-session basis. The latter I am working on right now. The former is
harder...

This is distinctly v1.0 code, and is bound to contain bugs. :-)

I'd appreciate any comments, suggestions, bug reports, etc. At the bottom
of the main page you will see my current ToDo list. 

I'd also appreciate any 'ground-truth' data, i.e. if your site is shown do
you actually see the number of sessions I attribute to you.

If you'd like to see your site involved, I'd suggest either installing mSD
and letting me know, or installing Kevin Almeroth's sdr-plugin and I will
automatically pick you up. The latter also increases Kevin's sample size,
so I would recommend that (even though I personally prefer mSD's approach
:-)). Or do both !

A heartfelt "Thank you" to Kevin and Peter for their great efforts/code,
without which I'd still be at the drawing board.

Cheers,
	Markus

Markus Buchhorn,  Advanced Computational Systems CRC     | Ph: +61 2 62798810
email: markus@acsys.anu.edu.au, snail: ACSys, RSISE Bldg,|Fax: +61 2 62798602
Australian National University, Canberra 0200, Australia |Mobile: 0417 281429  



From rem-conf Tue Jun 15 07:04:10 1999 
From rem-conf-request@es.net Tue Jun 15 07:04:08 1999
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Date: Tue, 15 Jun 1999 07:00:22 -0700 (PDT)
From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
Message-Id: <199906151400.HAA25674@jackson.cs.ucsb.edu>
To: rem-conf@es.net
Subject: Re:  Global Graphical Session Directory Monitor
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>>If you'd like to see your site involved, I'd suggest either installing mSD
>>and letting me know, or installing Kevin Almeroth's sdr-plugin and I will
>>automatically pick you up. The latter also increases Kevin's sample size,
>>so I would recommend that (even though I personally prefer mSD's approach
>>:-)). Or do both !

Let me encourage a few more people to join the sdr-monitor effort.  If
you are running a recent version of sdr on a unix platform the sdr-plugin
takes two seconds to install and the data will help tremendously.  Check
us out at http://imj.ucsb.edu/sdr-monitor/global/

-Kevin



From rem-conf Tue Jun 15 16:44:12 1999 
From rem-conf-request@es.net Tue Jun 15 16:44:11 1999
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From: "Fox, Mike" <mfox@nuera.com>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, 
	Scott.Petrack@metatel.com
Cc: rem-conf@es.net
Subject: RE: DTMF draft
Date: Tue, 15 Jun 1999 16:18:23 -0700
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Hi Henning and Scott,
I was just going over the June 99 issue of Computer Telephony where four
VOIP products were tested in a "Branch Office VoIP Gateway Shootout!". I
think there is a better spec. is better mentality in equipment buyers, and
every vendor would like to have the fewest DTMF failures when tested on a
congested network. It appears that one product implements an error corrected
channel for DTMF, and in doing so achieves a significantly better DTMF
transmission performance than the other products. I think that all of the
other products use some sort of a redundancy scheme.

I am raising the question of if perhaps a more "reliable" channel can be
*optionally* constructed using the RTP tones payload format. 

A possible mechanism would be to optimize the redundancy mechanism by
allowing  full duplex connection of the RTP tones to ACK the receipt of each
RTP tones packet back toward the source. This way the source can then
efficiently remove or maintain the redundant information in the
transmission.

I hate to open this can of worms again, but the basis of much of the
redundancy and reliability discussions in the VOIP Forum and rem-conf did
not anticipate that specific test scenarios would be used to compare
equipment.

Best regards,
Mike Fox



From rem-conf Tue Jun 15 17:06:21 1999 
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Date: Tue, 15 Jun 1999 20:03:31 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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CC: Scott.Petrack@metatel.com, rem-conf@es.net
Subject: Re: DTMF draft
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"Fox, Mike" wrote:
> 
> Hi Henning and Scott,
> I was just going over the June 99 issue of Computer Telephony where four
> VOIP products were tested in a "Branch Office VoIP Gateway Shootout!". I
> think there is a better spec. is better mentality in equipment buyers, and
> every vendor would like to have the fewest DTMF failures when tested on a
> congested network. It appears that one product implements an error corrected
> channel for DTMF, and in doing so achieves a significantly better DTMF
> transmission performance than the other products. I think that all of the
> other products use some sort of a redundancy scheme.
> 
> I am raising the question of if perhaps a more "reliable" channel can be
> *optionally* constructed using the RTP tones payload format.

The current scheme is already rather reliable. In the case of an
auto-dialer dialing a standard xxx xxx-xxxx number, you would end up
sending every digit between three and ten times.

If you want to add true NACK-based reliability, you could fall back to
RFC 2032, Section 5.2.2.

> 
> A possible mechanism would be to optimize the redundancy mechanism by
> allowing  full duplex connection of the RTP tones to ACK the receipt of each
> RTP tones packet back toward the source. This way the source can then
> efficiently remove or maintain the redundant information in the
> transmission.
> 
> I hate to open this can of worms again, but the basis of much of the
> redundancy and reliability discussions in the VOIP Forum and rem-conf did
> not anticipate that specific test scenarios would be used to compare
> equipment.
> 
> Best regards,
> Mike Fox

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



From rem-conf Tue Jun 15 19:20:47 1999 
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From: Scott Petrack <scott.petrack@metatel.com>
To: "Fox, Mike" <mfox@nuera.com>
cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, rem-conf@es.net
Subject: RE: DTMF draft
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The bit rate is so low that you can do a lot of things with this scheme.

Since packets tend to get lost in bursts, delaying some packets just a
little bit helped a lot in certain cases, and so did layering (so that 
the DTMF stream is actually sent over radically different paths in the
Internet between the two endpoints).

There was a time when I was playing with some RTCP receiver reports which
reported about a specific payload format, so that the receiver could
report some specific statistics about the receipt of the DTMF payload.
Although it looked sort of promising, I never finished and closed on
the whole issue of "if I send enough DTMF to get a statistically useful
sample, then the statistics are very similar to the ordinary receiver
reports for the whole stream." 

If any one else has done this sort of stuff in the context of layered
codecs, I'd be sort of curious to know. 

Thinking of the DTMF payload as working together with the audio payload
in a layered codec scheme, I would be very surprised if you would get
DTMF transmission that is worse than the PSTN. 

explicit ACKS and such is a terrible thing to do to a realtime bitstream.

Scott

On Tue, 15 Jun 1999, Fox, Mike wrote:

> Hi Henning and Scott,
> I was just going over the June 99 issue of Computer Telephony where four
> VOIP products were tested in a "Branch Office VoIP Gateway Shootout!". I
> think there is a better spec. is better mentality in equipment buyers, and
> every vendor would like to have the fewest DTMF failures when tested on a
> congested network. It appears that one product implements an error corrected
> channel for DTMF, and in doing so achieves a significantly better DTMF
> transmission performance than the other products. I think that all of the
> other products use some sort of a redundancy scheme.
> 
> I am raising the question of if perhaps a more "reliable" channel can be
> *optionally* constructed using the RTP tones payload format. 
> 
> A possible mechanism would be to optimize the redundancy mechanism by
> allowing  full duplex connection of the RTP tones to ACK the receipt of each
> RTP tones packet back toward the source. This way the source can then
> efficiently remove or maintain the redundant information in the
> transmission.
> 
> I hate to open this can of worms again, but the basis of much of the
> redundancy and reliability discussions in the VOIP Forum and rem-conf did
> not anticipate that specific test scenarios would be used to compare
> equipment.
> 
> Best regards,
> Mike Fox
> 
> 




From rem-conf Wed Jun 16 09:30:01 1999 
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Date: Wed, 16 Jun 1999 12:20:57 -0400
From: Sanjoy Paul <sanjoy@dnrc.bell-labs.com>
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To: ieeetcpc@ccvm.sunysb.edu, ietf@ietf.org, int-serv@ISI.EDU,
        isadsoc@fokus.gmd.de, itc@ieee.org,
        modern-heuristics@uk.ac.mailbase.ucdavis.edu,
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Subject: IEEE Network Magazine Special Issue on Multicasting
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Folks:

  the deadline for IEEE Network Special Issue on Multicasting is
coming up (July 1). You may take an additional week for submission
given that INFOCOM deadline is also July 1.
  You may find the details of the announcement and submission
procedure at 

   http://www.comsoc.org/socstr/techcom/ntwrk/special/cfp-paul.html

  Looking forward to your submissions.

Regards,
   Sanjoy Paul



From rem-conf Wed Jun 16 15:28:40 1999 
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From: Patrick Bihan-Faou <patrick@mindstep.com>
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Hi,

I went back about a year in the archive for the mailing list, and I have
not seen any discussion about the "protection operation" described in
"draft-ietf-avt-fec-05.txt".

Maybe I am missing the point completely, but it seems to me that there
are some major problems with the recommended protection operation. I
have included its description at the end of the message.

The problem I see is that:
- the 2 version bits from the original RTP packet beeing protected are
dropped
- only 3 out of the 4 bits of the CC field of the original RTP packet
are kept.

I can not see any justification for that, but I can find a lot of
arguments against these 2 points. 

First, which 3 out of the 4 bits of the CC field should be kept ?
Why not include the 4 of them ?
Why insisting on discarding the version field ?

The obvious results of these choices to me is that almost the entire
original RTP packet has to be shifted by 3 bits in order to produce
something that complies with the recommended protection operation. It
seems to me that this requires a lot of work for no benefits at all
since the RTP packets have to be padded to 4 bytes boundaries anyway.


I can understand that there is no need to protect the version field
since the value never changes, but it seems that shifting everything is
a lot of trouble just to not transmit 2 bits that don't change (and in
the process add 2 padding bits anyway).




Maybe I am missing the point completely, and if it is the case, please
let me know what I missed!



Thanks for you attention,


Patrick.



 

>From "draft-ietf-avt-fec-05.txt" page 8:


8 Protection Operation

   The protection operation involves taking a variety of fields from the
   various headers, appending the payloads, appending the whole thing
   together, padding zeroes, and then computing the FEC across the
   resulting binary block. The result is then placed into the FEC
   packet.

   The following procedure MUST be followed for the protection opera-
   tion. For each media packet to be protected, a binary array is gen-
   erated by appending the following fields together:

        o Padding Bit (1 bit)
        o Extension Bit (1 bit)
        o CC bits (3 bits)
        o Marker bit (1 bit)
        o Payload Type (7 bits)
        o Timestamp (32 bits)
        o Natural binary representation of the length of the CSRC List
          plus padding plus payload plus extension of the media packet
          (16 bits)
        o CSRC List (if CC is 1), else null (variable)
        o Header Extension (if X is 1), else null (variable)
        o the payload (variable)
        o Padding, if present (variable)

   Note that the padding bit forms the most significant bit of the
   binary array.





>From RFC 1889 page 10



5.1 RTP Fixed Header Fields

      The RTP header has the following format:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|X|  CC   |M|     PT      |       sequence number         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



-- 
et les Shadoks pompaient...




From rem-conf Wed Jun 16 18:44:01 1999 
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Date: Wed, 16 Jun 1999 21:38:12 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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Patrick Bihan-Faou wrote:
> 
> The problem I see is that:
> - the 2 version bits from the original RTP packet beeing protected are
> dropped

This is on purpose. If the version were included in the protection
operation, the value written into the V field in the FEC packet would
not likely be 2. The version is used for many RTP validity checks. As
such, I believe its important to keep it as 2. Since the version does
not change from packet to packet in a session, there is nothing gained
by recovering it. 


> - only 3 out of the 4 bits of the CC field of the original RTP packet
> are kept.

This is an error. Its been corrected. It should, of course, read 4 bits
everywhere. Thanks for pointing it out.

-Jonathan R.

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



From rem-conf Wed Jun 16 21:00:38 1999 
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-------------------------------------------------------------------------
We apologize if you receive this message more than once.
-------------------------------------------------------------------------

        JOURNAL OF APPLIED SYSTEMS STUDIES
Methodologies and Applications for Systems Approaches
		[ JASS ]

	http://www.unipi.gr/jass/


	CALL FOR PAPERS

Topics of interest to JASS include :

=B7 Applications of cybernetics using the viable system model
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There is no time limit for the submission of papers.






From rem-conf Thu Jun 17 05:09:12 1999 
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Date: Thu, 17 Jun 1999 08:02:56 -0400
Subject: Re: Questions about the recommended FEC protection operation
From: "Patrick Bihan-Faou" <patrick@mindstep.com>
To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Hi,


Jonathan Rosenberg wrote:
>
> Patrick Bihan-Faou wrote:
>> 
>> The problem I see is that:
>> - the 2 version bits from the original RTP packet beeing protected are
>> dropped
>
> This is on purpose. If the version were included in the protection
> operation, the value written into the V field in the FEC packet would
> not likely be 2. The version is used for many RTP validity checks. As
> such, I believe its important to keep it as 2. Since the version does
> not change from packet to packet in a session, there is nothing gained
> by recovering it.
>

Of course I am not arguing the fact that there is little gained in
"protecting" a value that never changes, neither do I contest that having
the version field set to 2 is necessary. My point is that by forcing this
field out of the protected data set, forces the ENTIRE original RTP packet
to be shifted by 2 bits. Then since we are sending an integral number of
BYTES, these two bits are going to be added at the end of the data set and
are likely to contain another totally useless value as far as the protection
operation goes.

So my point is that I do not understand why
1. we should shift X bytes by 2 bits
2. add 2 bits of padding
3. compute the protection function
4. transmit the protection RTP packet

then
5. execute the recovery function on the received protection packet
6. shift the resulting byte array by 2 bits
7. add the version field with the value 2 back.


It looks to me that operation 1 and 2 as well as 6 and 7 are unnecessary
considering that we are not even saving on the bandwidth used on the
internet, increase the computational requirements, and maybe the memory
requirement on the end units of the RTP session.

Furthermore I could argue that by leaving the version field in there, you
may enable protection of future versions of RTP with an RTP/RTCP version 2
session. However I do not consider that as a good point...


So the question (in my mind) remains unanswered.


Thanks for your help!


Patrick.

--
Patrick Bihan-Faou,          email: patrick.bihan-faou@mindstep.com
MindStep Corporation         tel:   (613) 526 5062
                             fax:   (613) 526 2417
                             web:   www.mindstep.com




From rem-conf Thu Jun 17 08:24:24 1999 
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From: Roy Spitzer <rspitzer@telogy.com>
To: 'Henning Schulzrinne' <hgs@cs.columbia.edu>, rem-conf@es.net
Cc: Scott Petrack <Scott.Petrack@metatel.com>, 
	MEGACO@standards.nortelnetworks.com, Ed Morgan <emorgan@telogy.com>, 
	Mihai Sirbu <msirbu@telogy.com>, Zoran Mladenovic
	 <zmladenovic@telogy.com>, Frank Fruth <ffruth@telogy.com>
Subject: RE: First cut at DTMF/tones draft
Date: Thu, 17 Jun 1999 11:17:32 -0400
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Henning,

I apologize for my late comments.

I applaud the reuse of the current RTP definitions for DTMF.  The
application of the same technique for supervisory signaling is desirable.

1.	The problem I have is that VoIP and the use of RTP is fast becoming
an international set of procedures.  To be compatible with the differing
encoding of supervisory signaling, the prime emphasis should be towards
creating a standard set of supervisory signal (event) names as discussed in
section 5.  The translation of these names becomes a function of the Media
Gateway.  These name could translate to standard country encodings or custom
audio response messages.  This need not be known by the generator of the
signal (event).
2.	Item 1 is fine when there is a translation of the signal (event).
For the cases when there is no translation of the signal (event) the
associated tones may be used.   But this may cause confusion at the
destination end since that combination of tones may already be assigned to
another signal(event).  The other alternative is to drop those signals on
the floor.  Obviously, there if there is no translation table, all signals
would be dropped on the floor.  The other alternative is to use the local
signal translation if present, used the associated tones when not present,
and drop unknown signals on the floor only when local signal translation is
present.  Please comment on this.
3.	To support V.34 Modems and V.34 FAX phase reversal needs as either a
parameter to CED or as distinguishable event.
4.	I am confused about the meaning of the events 
*	V.21 Channel 1, "0"
*	V.21 Channel 1, "1"
*	V.21 Channel 2, "0"
*	V.21 Channel 2, "1"
5.	Is V.21 flag indication available?

Regards,
Roy

Roy Spitzer				Telogy Networks, Inc.
Phone:	(301) 515-6531			20250 Century Blvd.
Fax:	(301) 515-7954			Germantown, MD 20874
Email:	rspitzer@telogy.com


		-----Original Message-----
		From:	Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
		Sent:	Wednesday, June 09, 1999 10:02 AM
		To:	rem-conf@es.net
		Cc:	Scott Petrack; MEGACO@standards.nortelnetworks.com
		Subject:	First cut at DTMF/tones draft

		I've put the PostScript and PDF versions of the merged DTMF
and tones
		draft at
http://www.cs.columbia.edu/~hgs/rtp/doc/dtmf/tones.pdf and
		tones.ps. Scott and I would appreciate your comments. I will
turn this
		into an (ASCII) I-D shortly, barring "you must be crazy"
comments.

		Thanks.



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Date: Thu, 17 Jun 1999 16:41:29 EDT
Subject: The Education Coalition (TEC) Newsletter
To: carole@oxford.school.nz, Ggrath@msg.pacbell.com, Jesimin@pacbell.com,
        kjoffe@us.pmc-vacc.com, cannings@pepperdine.edu, garnell@phoebus.com,
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The Education Coalition (TEC) Newsletter
An update to keep our business associates, and partners informed about TEC=
=92s=20
activities in educational technology and distributed learning. For further=20
information, visit our website at TECWeb.org or call 949-369-3867.=20
TEC Newsletter  June 1999
Contents:=20
Dr. Lane key speaker at KPMG Peat Marwick Summit Conference
Dr. Carla Lane, Executive Director of TEC, was invited by the internationall=
y=20
renowned  business consulting firm, KPMG Peat Marwick, to be a key presenter=20
at their three-day "21st Century Summit Conference" in Boca Raton, Florida.=20
Considered an expert in the area of educational technologies and learning=20
styles, Dr. Lane addressed 200 top industry leaders and their alliance=20
partners on the topic of  distributed learning and how  the use of learning=20
styles and multiple intelligences increases the impact of learning in=20
mediated courses.

TEC Personal Learning Model BETA Nears Completion
Providing course material in a student=92s preferred learning style offers=20
quicker, better learning with a higher retention rate.  The question becomes=20
how to ascertain learning styles for online courses. To meet this need, TEC,=20
in partnership with Performance Learning Systems of Nevada City, California,=20
is developing the Personal Learning Model (PLM); the PLM is an online=20
learning styles/ multiple intelligences (LS/MI) assessment that will=20
determine a student=92s preferred learning style and secondary learning mode=
s.
The PLM is not a "paper-and-pencil" type test; rather it will be a series of=20
tasks and problems to solve in a virtual environment. The responses will be=20
tracked and built into a LS/MI profile.  The profile is tagged, stored in a=20
database, and passed to an online course so material can be presented in the=20
preferred learning style. PLM adheres to the Educause/Information Management=20
System (IMS) standards, and utilizes object-oriented, reusable learning=20
objects.=20
The PLM, developed in Asymetrix software, is nearing completion and we are=20
pleased to announce that BETA testing on the software will commence shortly.=20

Online Cosmetic Surgery Training Utilizes 3-D Interactive Simulation=20
TEC is currently working with Dr. James Fulton, co-developer of Retin-A, and=20
renown cosmetic/plastic surgeon in Newport Beach, California, to develop=20
web-based training for his bleeding-edge (no pun intended! ) practices. The=20
courses will be offered to physicians throughout the world. The training=20
utilizes a virtual-reality product from Eon Reality, Inc., which allows user=
s=20
to manipulate and practice surgical techniques on a 3-D interactive=20
simulation of a person after watching Dr. Fulton perform a training session=20
online. TEC foresees terrific potential for the Eon software to be used in=20
any training that requires hands-on manipulation of objects, and/or is aimin=
g=20
to reach the kinesthetic learner. =20

TEAMS Project Evaluation: Los Angeles County Office of Education  =20
TEAMS Distance Learning brings exemplary learning opportunities to K-8=20
students, teachers, and parents across the United States through nationally=20
televised satellite broadcasts and the Internet. Learners use instructional=20
technologies to access a combination of the best features of time-dependent=20
(synchronous) video-based instruction along with time-independent=20
(asynchronous) computer access to multimedia and the Internet.=20
TEAMS is now the largest satellite distance learning provider of instruction=20
for the elementary grades in the United States. Over 140,000 students and=20
4,000 teachers from 20 states plus the District of Columbia receive direct=20
student instruction and professional development transmitted via satellite=20
>from the ETN studio facilities at the Los Angeles County Office of Education=
.=20

TEAMS Distance Learning has won eleven national awards for program=20
excellence, and is highly regarded in the educational community for its=20
unique learning design which provides hands-on motivational instruction for=20
students and long term professional development for teachers. TEC conducted=20
the largest research study ever conducted on a distance learning program  an=
d=20
surveyed thousands of TEAMS participants over a four year period. This surve=
y=20
of teachers and students demonstrated conclusively that this distance=20
learning program significantly improves instruction in the elementary and=20
middle school grades. TEC is proud to be named the evaluator for the project=20
for the eighth year.

TEAMS is funded in part by Star Schools legislation through the U.S.=20
Department of Education's Office of Educational Research and Improvement=20
(O.E.R.I.).  Web site http://teams.lacoe.edu/

Mountain Plains Distance Learning Partnership=20
Mountain Plains Distance Learning Partnership focuses on the needs of the=20
Native American population in the vast intermountain region which it=20
services.  This partnership of four states is developing an electronic,=20
virtual campus employing a two-way audio and visual interactive connection=20
using fully scaleable high speed digital ATM microwave technology, electroni=
c=20
transmission and receiving classrooms, and computer-assisted instructional=20
programming centers. Teachers are empowered to design and deliver interactiv=
e=20
multimedia curricula to remote communities for K-14 students and adults usin=
g=20
a comprehensive training model and multimedia curriculum strategies. =20
The project provides services in Utah, Wyoming, Colorado, and Montana.  TEC=20
is moving into the third year as the project=92s evaluator and was the=20
evaluator on the Four Corners Project for three years prior to the next Star=20
Schools funding round.  TEC conducted the on-site evaluation in April.
The project is funded by Star Schools legislation through the U.S. Departmen=
t=20
of Education=92s Office of Educational Research and Improvement (O.E.R.I.). =20
Check out the web site at http://stars-cwc.cwc.whecn.edu/

San Bernardino Student and Teacher Excellence Project (STEP) Evaluation
San Bernardino County is one of the few areas in the US selected to offer it=
s=20
students accelerated science and math programs based on material from NASA.=20
TEC, chosen as the evaluator of this innovative program with its winning use=20
of technology in the classroom, is working closely with the STEP staff to=20
provide  total educational improvement through the integration of staff=20
development, curriculum development, home access, evaluation and technology/=20
infrastructure support. The project is built on the belief that educational=20
improvement is not a function of a single intervention, but a system of=20
interventions.  TEC has completed the research design and survey instruments=20
and will conduct site surveys for four days in June. =20

High quality curriculum development is a project emphasis. Project STEP =20
targets highly focused math and science concepts and explore these concepts=20
in depth. It follows the recommendations of the Third International=20
Mathematics and Science Study (TIMSS) and the National Science Standards=20
that, "more emphasis should be placed on studying a few fundamental science=20
concepts" and limit the number of topics covered at each grade level.

NASA resources are invaluable in helping students see the relevance of math=20
and science in the real world. Each Project STEP unit of study is=20
results-driven with a rigorous application activity, based on NASA project=20
resources, embedded into the curriculum. Learning progresses from=20
understanding to application, and then to application in dynamic,=20
unpredictable situations. Instruction emphasizes "doing" rather than just=20
"knowing".

In collaboration with the Jet Propulsion Laboratory and the Dreyden Flight=20
Center, teachers are involved in wide variety of professional development=20
activities.  Professional development activities are a key feature of the=20
STEP project with a full-time mentor teacher assigned at each project school=
.=20
This commitment to on-going multi-strategy professional development supports=20
networks of teachers at the site, regional, and county levels.
Web site at  http://step.k12.ca.us


Oakland Unified School District Core Values Project Evaluation
The Oakland Unified School District received a California state Challenge=20
grant  1998.  TEC has been named the evaluator and recently completed a=20
weeklong site visit for the evaluation.

The Core Values Projects addresses four identified needs of its 10,694 middl=
e=20
school students: They don=92t think critically or historically. They are not=20
proficient readers, writers, and speakers. They are not presented with=20
opportunities to work within a constructivist approach to learning. They do=20
not systematically use technology tools to support academic progress.=20
However, the adoption of curriculum embedded assessments has allowed Oakland=20
to go beyond measuring failure to design a data-driven program for middle=20
schools that is at the heart of this proposal.=20

The project incorporates a linked progression of content, technology skills=20
and habits of mind that move students from investigating the past to taking=20
an active citizenship role to address current community problems. To develop=20
these skills students have access to networked, multi-media research pods an=
d=20
the instruction based on their usage. Professional development for teachers=20
within the LA/SS/ELD content areas connects technology, curriculum and=20
classroom practice to ensure progress in student achievement, curriculum=20
development and technology integration. Teachers are collaborating with U.C.=20
Berkeley and the Oakland Museum to create on-line resources to address=20
specific needs in the LA/SS/ELD curriculum. Schools will recruit and train=20
staff from community based organizations, technology volunteers and parents=20
to be "community service project partners," increase classroom support, and=20
continue networking classes. Teachers have "just-in-time" on-site technical=20
assistance. The district's 24 hour educational cable channel, Web page,=20
videoconferencing, and School Calling system extend staff development and th=
e=20
instructional day, give access to enriched learning resources, and improve=20
home-school communications.=20

Using Project-Based Learning Strategies, OUSD Core Values students:
*=09Work in Cooperative Groups- students build knowledge collaboratively,=20
encountering multiple perspectives.
*=09Read and Write for Understanding- students analyze and synthesize=20
information; record reactions, thoughts, and feelings; clarify and organize=20
information using a wide variety of writing activities.
*=09Use Inquiry Learning Methods- challenging problem solving and=20
research tasks that require higher-order thinking skills and multiple=20
abilities, and student ownership of work in heterogeneous groupings.
*=09Use Technology as a Tool- for research, communication, and=20
production; content comes first, supported by uses of technology; students=20
are communicators and authors.
*=09Demonstrate Content Knowledge- students are challenged to demonstrate=20
their knowledge and skills in various modalities and media: performance,=20
portfolio, and presentation.

To Support Project-Based Learning strategies, OUSDCore Values Teachers:
1.=09Plan for and use Spiraling- topics and information build outwardly=20
and sequentially from the basic to the complex, and Scaffolding- support=20
students in ways that enable dependent success, moving towards independent=20
success
2.=09Use Standard English Proficiency (SEP) and Sheltered Instructional=20
Strategies- appeal to Diverse Learning Populations, Second Language Learners=
,=20
and Multiple Abilities, as well as instructional strategies which support an=
d=20
strengthen under-prepared students.
3.=09Facilitate Group Work- tap into abilities and strengths of others,=20
develop skills communally while working with a variety of people; requires=20
teacher and student management of time and facilities.
4.=09Receive Professional Development- address best teaching practices,=20
classroom management, lesson design, meeting new teachers' needs and the new=20
needs of experienced teachers.
5.=09Use Curriculum-Embedded and Alternative, Authentic Assessments-=20
portfolio, performance, exhibition.

Check out the web site at http://www.ousd.k12.ca.us/cv/
Seeking New DL Solutions: TeleCon East=20
The TeleCon East (Washington, DC, March) and TeleCon West (Anahaim, CA, Oct =
)=20
conferences are super-charged shows that feature cutting-edge technologies,=20
and the opportunity to network with others focused on using technology to=20
enhance learning.   The TeleCon East conference has been reconfigured to be=20
the world=92s LEADING conference in Distance Learning and, as chairs and=20
producers of the TeleCon shows, TEC is seeking examples of brand-new distanc=
e=20
learning applications and products. Contact us so we can spotlight your=20
solutions !
=09
Dr. Carla Lane is co-chairing the TeleCon East show with Shirley Davis,=20
Director, Learning Innovations, PBS Adult Learning  Service, who is the=20
current president of the United States Distance Learning Association (USDLA)=20
and Dr. Jolly Holden who is USDLA president-elect  and Chief Learning=20
Strategist Spacenet , Inc

TEC Online Courses=20
Too busy to attend a sit-down class after working all day? Join other busy=20
professionals who want to stay current in the latest trends in educational=20
technologies - enroll in a TEC online course. Customized courses are=20
available when five or more people enroll from the same organization. =20
The following courses are currently being offered:
Introduction to Distributed Learning - 15 hours - asynchronous online and=20
audio conferencing - $89
An overview of the online learning environment, products, and learning model=
s=20
used in presenting online education to users in diverse locations. Includes=20
the characteristics of a good candidate for online courses.=20
Using Technology to Address Learning Styles - 15 hours  - asynchronous onlin=
e=20
and audio-conferencing- $89 =20
For all the years since Socrates, most teachers and trainers have presented=20
information to trainees or students in one format=97lecture.  However, studi=
es=20
has shown that presenting information to students in their preferred learnin=
g=20
style provides a learning impact more quickly, with learning easier to=20
retrieve, easier to apply, and more deeply understood. Acquire an overview o=
f=20
how to addresses learning styles in online courses.=20

TEC courses are facilitated by Dr. Lane and Joanne Carle-Accornero,=20
developers and instructors for numerous online programs including UCLA, UC=20
Berkeley, and the University of Phoenix Online.=20
Course sections will have limited enrollments! For more information and to=20
enroll, call 949-369-3867. =20
TECWeb.org              Tel:  949-369-3867     Fax: 949-369-3865




From rem-conf Thu Jun 17 17:28:12 1999 
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Message-ID: <376991AA.4E7FB66C@dnrc.bell-labs.com>
Date: Thu, 17 Jun 1999 20:24:10 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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To: Patrick Bihan-Faou <patrick@mindstep.com>
CC: rem-conf@es.net
Subject: Re: Questions about the recommended FEC protection operation
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Patrick Bihan-Faou wrote:
> 
> > This is on purpose. If the version were included in the protection
> > operation, the value written into the V field in the FEC packet would
> > not likely be 2. The version is used for many RTP validity checks. As
> > such, I believe its important to keep it as 2. Since the version does
> > not change from packet to packet in a session, there is nothing gained
> > by recovering it.
> >
> 
> Of course I am not arguing the fact that there is little gained in
> "protecting" a value that never changes, neither do I contest that having
> the version field set to 2 is necessary. My point is that by forcing this
> field out of the protected data set, forces the ENTIRE original RTP packet
> to be shifted by 2 bits. Then since we are sending an integral number of
> BYTES, these two bits are going to be added at the end of the data set and
> are likely to contain another totally useless value as far as the protection
> operation goes.

You don't send the data generated by the protection operation. It
contains a portion of the header (note the SN is missing too) and the
body. 

In any case, the specification describes the operation. Its
implementation is up to you. You could just XOR the packets together and
then overwrite the fields not protected (like the SN and version) with
the correct values. The spec does say that you MUST follow the procedure
listed. In fact, this should be a MAY, and it should say you can use any
other procedure with the same results. I've changed this for the next
rev.

Thanks,
Jonathan R.

> 
> So my point is that I do not understand why
> 1. we should shift X bytes by 2 bits
> 2. add 2 bits of padding
> 3. compute the protection function
> 4. transmit the protection RTP packet
> 
> then
> 5. execute the recovery function on the received protection packet
> 6. shift the resulting byte array by 2 bits
> 7. add the version field with the value 2 back.
> 
> It looks to me that operation 1 and 2 as well as 6 and 7 are unnecessary
> considering that we are not even saving on the bandwidth used on the
> internet, increase the computational requirements, and maybe the memory
> requirement on the end units of the RTP session.
> 
> Furthermore I could argue that by leaving the version field in there, you
> may enable protection of future versions of RTP with an RTP/RTCP version 2
> session. However I do not consider that as a good point...
> 
> So the question (in my mind) remains unanswered.
> 
> Thanks for your help!
> 
> Patrick.
> 
> --
> Patrick Bihan-Faou,          email: patrick.bihan-faou@mindstep.com
> MindStep Corporation         tel:   (613) 526 5062
>                              fax:   (613) 526 2417
>                              web:   www.mindstep.com

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



From rem-conf Fri Jun 18 07:30:51 1999 
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Hi, has any motion occured on the issue
	of RTP multiplexing?

[\] Robert Wood   

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

mailto:robert_wood@mitel.com http://www.magma.ca/~rgwood/



From rem-conf Fri Jun 18 07:37:03 1999 
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From: "Patrick Bihan-Faou" <patrick@mindstep.com>
To: "Jonathan Rosenberg" <jdrosen@dnrc.bell-labs.com>
Cc: <rem-conf@es.net>
Subject: Re: Questions about the recommended FEC protection operation
Date: Fri, 18 Jun 1999 10:42:16 -0400
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Jonathan Rosenberg wrote:

>Patrick Bihan-Faou wrote:
>>
>> > This is on purpose. If the version were included in the protection
>> > operation, the value written into the V field in the FEC packet would
>> > not likely be 2. The version is used for many RTP validity checks. As
>> > such, I believe its important to keep it as 2. Since the version does
>> > not change from packet to packet in a session, there is nothing gained
>> > by recovering it.
>> >
>>
>> Of course I am not arguing the fact that there is little gained in
>> "protecting" a value that never changes, neither do I contest that having
>> the version field set to 2 is necessary. My point is that by forcing this
>> field out of the protected data set, forces the ENTIRE original RTP
packet
>> to be shifted by 2 bits. Then since we are sending an integral number of
>> BYTES, these two bits are going to be added at the end of the data set
and
>> are likely to contain another totally useless value as far as the
protection
>> operation goes.
>
>You don't send the data generated by the protection operation. It
>contains a portion of the header (note the SN is missing too) and the
>body.


???

I must really be missing something here. What do you mean by "you don't send
the data generated by the protection opertation" ? You do send the resulting
XOR of the selected packets you want to protect right ?


Anyway, once again I am comming back to that issue, and I will until I get
it (i.e. somebody gives me a reasonable reason). I understand that one may
not want to send/protect unecessay data. No problem with that. I just don't
agree to take it to an extreme where NOT doing so (sending unecessary /
redundant data) requires a lot of additional computing power.

So I will reformulate my question:

- the version field which is 2 bits long is excluded from the protection
mechanism, right ?

- the bit field on which the XOR operation is carried is "continuous" (i.e.
not unused bits / padding bits in the middle), right ?

- removing 2 bits from the beginning of the bits field to be protected
implies that ALL the subsequent bits have to be shifted, right ?

If you answered YES to these three questions, then I will maintain that
removing the version field is a questionable design decision. How many
systems deal (address) with memory at the bit level ? If you are going to
eliminate an integral number of bytes (as with the SN field), then I don't
have any problem (even though I may argue that these days saving less than a
multiple of 4 bytes is questionable).

My objection is that the spec implies that you can deal with memory at the
bit level where these days it is usually easier to deal with memory in 1/2/4
byte(s) units.

In other words: what are you saving by eliminating 2 bits ? How much is it
going to cost ? Is it worth it ?

These are really the questions that I am looking answers for to stop ranting
;-).


>In any case, the specification describes the operation. Its
>implementation is up to you. You could just XOR the packets together and
>then overwrite the fields not protected (like the SN and version) with
>the correct values. The spec does say that you MUST follow the procedure
>listed. In fact, this should be a MAY, and it should say you can use any
>other procedure with the same results. I've changed this for the next
>rev.
>

Well I don't have a problem with the MUST. I do not think that it would be a
very good idea to have everybody implement something radically different.
One of the purpose of these documents is to have compatible
implementations...


Thanks,

Patrick (the 2 bits guy).

--
Patrick Bihan-Faou,          email: patrick.bihan-faou@mindstep.com
MindStep Corporation         tel:   (613) 526 5062
                             fax:   (613) 526 2417
                             web:   www.mindstep.com





From rem-conf Fri Jun 18 09:15:00 1999 
From rem-conf-request@es.net Fri Jun 18 09:15:00 1999
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To: rem-conf@es.net
Subject: Working group last call on SSRC sampling draft
Date: Fri, 18 Jun 1999 17:10:32 +0100
Message-ID: <3224.929722232@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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This is to announce a two week working group last call on the Sampling of
the Group Membership in RTP draft (draft-ietf-avt-rtpsample-04.txt). If no
comments requiring changes are received in the next two weeks (by 2nd
July), this draft will be submitted for publication as an experimental RFC.

Colin



From rem-conf Fri Jun 18 11:12:04 1999 
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Date: Fri, 18 Jun 1999 11:07:54 -0700 (PDT)
From: Dan Wing <dwing@cisco.com>
To: Robert Wood <robert_wood@Mitel.COM>
cc: iptel@lists.research.bell-labs.com, rem-conf@es.net
Subject: Re: RTP multiplexing
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On Fri, 18 Jun 1999 10:21 -0400, Robert Wood wrote:

We have a new proposal I have been editing which will be submitted as an
I-D shortly.  I will also publish it on our FTP site as I expect the I-D
publishing delay is getting large as we approach the I-D submission cutoff
date.

-Dan Wing

> Hi, has any motion occured on the issue
> 	of RTP multiplexing?
> 
> [\] Robert Wood   
> 
> The St. Lawrence river - fresh, warm, visible diving.
> 
> mailto:robert_wood@mitel.com http://www.magma.ca/~rgwood/
> 




From rem-conf Fri Jun 18 13:33:56 1999 
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From: "David E. Taylor" <DavidT@GNSconsulting.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: URGENT!!!
Date: Fri, 18 Jun 1999 15:30:47 -0500
MIME-Version: 1.0
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This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

------_=_NextPart_001_01BEB9C9.72CD045A
Content-Type: text/plain

How do I regiester my Internet Multicast?  I have a live show to broadcast
tomarrow. I can cast on my local net, but need to register a time and put it
on MBONE.

Thanks.

David Taylor, MCSE CCA
218-590-3531
218-722-8231

------_=_NextPart_001_01BEB9C9.72CD045A
Content-Type: text/html
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2448.0">
<TITLE>URGENT!!!</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>How do I regiester my Internet Multicast?&nbsp; I =
have a live show to broadcast tomarrow. I can cast on my local net, but =
need to register a time and put it on MBONE.</FONT></P>

<P><FONT SIZE=3D2>Thanks.</FONT>
</P>

<P><FONT SIZE=3D2>David Taylor, MCSE CCA</FONT>
<BR><FONT SIZE=3D2>218-590-3531</FONT>
<BR><FONT SIZE=3D2>218-722-8231</FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01BEB9C9.72CD045A--



From rem-conf Fri Jun 18 13:35:34 1999 
From rem-conf-request@es.net Fri Jun 18 13:35:33 1999
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From:      <bigen@flashmail.com>
To:        <rem-conf@es.net>
Message-Id: <419.436329.55899329bigen@flashmail.com>
Subject: ENTREPRENEURS WANTED!
Mime-Version: 1.0
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Date: Fri, 18 Jun 1999 13:35:05 -0700
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Our Company is rapidly expanding in many
local & worldwide markets and is seeking
several motivated individuals who are
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To be removed from our list,simply click reply, type remove, then send





From rem-conf Fri Jun 18 13:44:12 1999 
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	 <rem-conf@es.net>
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Date: Fri, 18 Jun 1999 13:42:50 -0700
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Is the answer to your question
 
Using SDR, or similar announcement tool, create a session advertisement to
you out to the mbone ?
 
Or is your question
 
How do I connect my local network into the Mbone so other people can get
multicast that I am sending out ?
 
I fear that if it is the second you won't make tomorrow
Bill

-----Original Message-----
From: David E. Taylor [mailto:DavidT@GNSconsulting.com]
Sent: Friday, June 18, 1999 1:31 PM
To: 'rem-conf@es.net'
Subject: URGENT!!!



How do I regiester my Internet Multicast?  I have a live show to broadcast
tomarrow. I can cast on my local net, but need to register a time and put it
on MBONE.

Thanks. 

David Taylor, MCSE CCA 
218-590-3531 
218-722-8231 




From rem-conf Fri Jun 18 22:29:05 1999 
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Message-ID: <376B1FAA.C6C73320@dnrc.bell-labs.com>
Date: Sat, 19 Jun 1999 00:42:18 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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To: Patrick Bihan-Faou <patrick@mindstep.com>
CC: rem-conf@es.net
Subject: Re: Questions about the recommended FEC protection operation
References: <01beb998$c2a7d8a0$0b01a8c0@olivier.sandbox.ca>
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Patrick Bihan-Faou wrote:
> 
> >You don't send the data generated by the protection operation. It
> >contains a portion of the header (note the SN is missing too) and the
> >body.
> 
> ???
> 
> I must really be missing something here. What do you mean by "you don't send
> the data generated by the protection opertation" ? You do send the resulting
> XOR of the selected packets you want to protect right ?

I'm sorry. This sentence really doesn't make sense. Let me try again.
The data in the protection array is not a complete or correctly
formatted RTP packet. Besides the missing version number, the SN is not
covered by the operation. There is also a length value that is
protected, which is not present in the RTP packet but IS covered by the
protection operation. So, the protection array cannot be simply
transmitted, even if we added the version back in. But, see my further
comments below on a way to do this anyway.



> >In any case, the specification describes the operation. Its
> >implementation is up to you. You could just XOR the packets together and
> >then overwrite the fields not protected (like the SN and version) with
> >the correct values. The spec does say that you MUST follow the procedure
> >listed. In fact, this should be a MAY, and it should say you can use any
> >other procedure with the same results. I've changed this for the next
> >rev.
> >
> 
> Well I don't have a problem with the MUST. I do not think that it would be a
> very good idea to have everybody implement something radically different.
> One of the purpose of these documents is to have compatible
> implementations...

No, you miss the point. My statement above was that you can implement
this in any other way, so long as THE RESULT IS THE SAME AS IF YOU HAD
FOLLOWED THE SPECIFIED ALGORITHM. Thus interoperability is guaranteed.
So, given this, let me suggest an implementation which meets exactly
your non-shifting criteria. 

For simplicities sake, consider here a "mini RTP packet" with three
fields, A, B, and C, instead of the ones normally used in RTP. A is 2
bits, B is 3 bits, and C is 3 bits. An FEC packet has B always set to
001, but the A and C fields are protected using the same basic algorithm
described by the protection operation in the FEC spec. This means you
would take A, append C, xor this with the other A+C fields, and then
place the first 2 bits into the A field in the FEC packet, and the next
3 bits into the last three bits of the FEC packet. This has the same
shift problem you describe: the protection is over 5 bits, not 8.

OK - heres an alternate implementation. Take the packets being
protected, and just xor them together as a single byte - fields A,B, and
C. The result is a one byte packet with the A and C values the same as
the resulting FEC packet from the operation in the previous paragraph.
Of course, B is wrong. No problem. Set it to 001. The result is then the
same FEC packet as above. Since the result is the same, we have
interoperability.

So, in the case of real RTP, don't shift. Include the version in the xor
operation (the SN too while you're at it). Then, overwrite these in the
output with the correct v and SN values. You'll need to do the length
computation separately.

-Jonathan R.



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



From rem-conf Sat Jun 19 19:08:10 1999 
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Subject: Re: Questions about the recommended FEC protection operation
From: "Patrick Bihan-Faou" <patrick@mindstep.com>
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All right now I get it!


Thank you (Jonathan Rosenberg) for your patience and explenations. I was
under the impression that the resulting XOR of the protection bit field was
the payload of the RTP packet transmitted. My mistake! BUt again thanks for
making that clear.


Sorry for the bother.

Patrick.


--
Et les Shadoks pompaient...




From rem-conf Sun Jun 20 18:25:07 1999 
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Date: Mon, 21 Jun 1999 09:19:07 +0800 (CST)
From: Jan-Ming Ho <hoho@iis.sinica.edu.tw>
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To: rem-conf@es.net
Cc: Sheng-Hsiung Huang <huangk@gate.sinica.edu.tw>, chinfu@iis.sinica.edu.tw,
        Chia-Hui Wang <chwang@iis.sinica.edu.tw>,
        Lih-Hsiang Chen <chens@mail.ncku.edu.tw>, feng@iis.sinica.edu.tw
Subject: request for Mbone feed
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We are representing Taiwan to seek for Mbone feed to our Academic Network,
Taiwan Academic Network, or TANET for short.  We had long been suffering
a low bandwidth connecting to international internet traffic.  But, we
recently upgrade our connection to the US by a T3 lease line.  We 
appreciate it very much if we could have any institute here to provide
us with permanent MBONE feed.  We will run a permanent mbone root for
the academic family including universities in Taiwan.   We are also 
willing to transit this MBONE traffic to nearby countries based on the
availability of physical connections and bandwidth.

Thank you for your attention with

Best Regards,


Jan-Ming Ho, PhD
Associate Editor, IEEE Transaction on Multimedia
Research Fellow			(O) 886-2-2788-3799 x1803
IIS, Academis Sinica		(F) 886-2-2788-3799 x1606,  886-2-2782-4814
Nankang, Taipei			email:  hoho@iis.sinica.edu.tw
Taiwan				URL:	http://www.iis.sinica.edu.tw/~hoho
					http://www.iis.sinica.edu.tw/CCL
					http://twstudy.sinica.edu.tw

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






From rem-conf Mon Jun 21 08:08:02 1999 
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From: <telegen0@hotmail.com>
To: rem-conf@es.net
Subject: E-Commerce Conference - Wembley 1999
Date: Mon, 21 Jun 1999 15:56:35
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--====================5453515:56====
Content-Type: text/plain; charset="us-ascii"

E-Commerce - the future - how?

Find attached a brochure and application to join the forthcoming e-commerce exhibition being held at Wembley in July 1999.  Don't get left behind come and find out what, how and why e-commerce can help your business create MORE BUSINESS.

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--====================5453515:56====
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--====================5453515:56====--



From rem-conf Mon Jun 21 08:55:27 1999 
From rem-conf-request@es.net Mon Jun 21 08:55:26 1999
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Message-ID: <024b01bebbfe$28b6aba0$a63db680@psc.edu>
From: "S.Grant" <grant@psc.edu>
To: <rem-conf@es.net>
Subject: What can be done?
Date: Mon, 21 Jun 1999 11:52:58 -0400
Organization: NLANR/NCNE Pittsburgh Supercomputing Center
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What can be done about the volume of spam sent to this list?  Can the
list posts be restricted to only those subscribed?  At this point, I'm
ready to unsub due to the volume of spam.

Does anyone else here have any ideas?

Thanks for listening.

    Shirl Grant                       NLANR Engineering Services
   grant@ncne.org                NCNE / Pittsburgh Supercomputing Center

-






From rem-conf Mon Jun 21 09:57:12 1999 
From rem-conf-request@es.net Mon Jun 21 09:57:11 1999
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Date: Mon, 21 Jun 1999 12:53:53 -0400 (EDT)
From: Bob Dixon <rdixon@stargate.acs.ohio-state.edu>
To: "S.Grant" <grant@psc.edu>
cc: rem-conf@es.net
Subject: Re: What can be done?
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I agree completely. This has been discussed here before, but the problem
remains. The amount of spam I get which is related to my subscription to
this list is huge.


On Mon, 21 Jun 1999, S.Grant wrote:

> What can be done about the volume of spam sent to this list?  Can the
> list posts be restricted to only those subscribed?  At this point, I'm
> ready to unsub due to the volume of spam.
> 
> Does anyone else here have any ideas?
> 
> Thanks for listening.
> 
>     Shirl Grant                       NLANR Engineering Services
>    grant@ncne.org                NCNE / Pittsburgh Supercomputing Center
> 
> -
> 
> 
> 
> 

Robert S. Dixon                    Voice    614-292-1638
Chief Research Engineer            Fax      614-292-7081
Office of the CIO                  Lab      614-292-7425
Ohio State University              Email    Bob_Dixon@osu.edu
1971 Neil Ave, room 451            Video    128.146.199.85
Columbus, OH 43210




From rem-conf Mon Jun 21 10:06:35 1999 
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From: Bruce_Levens@nmss.com
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
	Scott.Petrack@metatel.com
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Date: Mon, 21 Jun 1999 13:04:38 -0400
Subject: inband dtmf/tones proposal
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Is it your intention in the new Schulzrinne/Petrack proposal that the
inband dtmf/tone  rtp packets could overlap in time with voice rtp packets?

     A receiver/decoder would then receive a udp packet containing "voice /
encoded tone" as well as a udp packet covering some or all of the same
     time interval with a dtmf/tone message.  This is my reading of the
proposal and I see no problem with it.  It is also in line with H.245's out
of band        method.

More importantly:
 Is it your intention also that the redundancy must be applied to all
packets: voice as well as dtmf/tones?
     It is hard to imagine how it could work otherwise because it would be
difficult to determine that a dtmf/tone packet was missing. This
requirement could   add a lot of overhead however.  Also, the redundancy
adds some latency.  For the dtmf/tones this is not such a huge problem, but
for the voice it         might be, especially as the quality/Bandwidth of
the Internet connections improve.  Hence, if there is some way to operate
the protocol with             redundancy applied only to the dtmf/tone
packets, it could be important.





From rem-conf Mon Jun 21 10:06:36 1999 
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Message-ID: <027b01bebc08$450ea320$a63db680@psc.edu>
From: "S.Grant" <grant@psc.edu>
To: "Bob Dixon" <rdixon@stargate.acs.ohio-state.edu>
Cc: <rem-conf@es.net>
References: <Pine.SOL.4.05.9906211252400.1793-100000@stargate.acs.ohio-state.edu>
Subject: Re: What can be done?
Date: Mon, 21 Jun 1999 13:05:31 -0400
Organization: NLANR/NCNE Pittsburgh Supercomputing Center
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> I agree completely. This has been discussed here before, but the
problem
> remains. The amount of spam I get which is related to my subscription
to
> this list is huge.

I maintin a list that was getting hit alot too.  I changed it over to a
subscriber posting only list.  It was a PITA for a little bit.  I have
another list of 'approved' posting addresses, and this is where I out
people who post that may be subscribed via a local reflector.  After the
first couple of weeks it settled down.  It was worth it.  All spam gets
bounced to me as a reminder of the Good Thing I did.  :^)

I really don't want to unsubscribe to this list, but I cannot deal with
all the spam.

-Shirl




From rem-conf Mon Jun 21 10:12:10 1999 
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If the list owner cannot help, my only suggestion is to adopt the
convention that all messages to this group must have "RTP" in their
subject line. List subscribers could then filter the messages
appropriately. (:-)

Bob Dixon wrote:
> 
> I agree completely. This has been discussed here before, but the problem
> remains. The amount of spam I get which is related to my subscription to
> this list is huge.
> 
> On Mon, 21 Jun 1999, S.Grant wrote:
> 
> > What can be done about the volume of spam sent to this list?  Can the
> > list posts be restricted to only those subscribed?  At this point, I'm
> > ready to unsub due to the volume of spam.
> >
> > Does anyone else here have any ideas?
> >
> > Thanks for listening.
> >
> >     Shirl Grant                       NLANR Engineering Services
> >    grant@ncne.org                NCNE / Pittsburgh Supercomputing Center
> >
> > -
> >
> >
> >
> >
> 
> Robert S. Dixon                    Voice    614-292-1638
> Chief Research Engineer            Fax      614-292-7081
> Office of the CIO                  Lab      614-292-7425
> Ohio State University              Email    Bob_Dixon@osu.edu
> 1971 Neil Ave, room 451            Video    128.146.199.85
> Columbus, OH 43210



From rem-conf Mon Jun 21 10:18:14 1999 
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Date: Mon, 21 Jun 1999 19:21:50 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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CC: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
 	Scott.Petrack@metatel.com, rem-conf@es.net
Subject: Re: inband dtmf/tones proposal
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Bruce_Levens@nmss.com wrote:
> 
> Is it your intention in the new Schulzrinne/Petrack proposal that the
> inband dtmf/tone  rtp packets could overlap in time with voice rtp packets?

Yes, I believe the draft points this out somewhere.

> 
>      A receiver/decoder would then receive a udp packet containing "voice /
> encoded tone" as well as a udp packet covering some or all of the same
>      time interval with a dtmf/tone message.  This is my reading of the
> proposal and I see no problem with it.  It is also in line with H.245's out
> of band        method.
> 
> More importantly:
>  Is it your intention also that the redundancy must be applied to all
> packets: voice as well as dtmf/tones?

No, definitely not. There are other redundancy schemes, including the
one in the draft, but using low bit-rate codecs, not repetition, as well
as FEC, to handle regular audio packets.

>      It is hard to imagine how it could work otherwise because it would be
> difficult to determine that a dtmf/tone packet was missing. This

The RTP sequence number tells you if packets (audio or DTMF) are
missing. To provide some protection against missing the last part of the
last digit, it is transmitted repeatedly (three times). I guess I don't
understand the concern exactly.

> requirement could   add a lot of overhead however.  Also, the redundancy
> adds some latency.  For the dtmf/tones this is not such a huge problem, but
> for the voice it         might be, especially as the quality/Bandwidth of
> the Internet connections improve.  Hence, if there is some way to operate
> the protocol with             redundancy applied only to the dtmf/tone
> packets, it could be important.



From rem-conf Mon Jun 21 11:05:36 1999 
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	rem-conf@es.net
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Subject: Re: inband dtmf/tones proposal
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I take it then that you mean for the rtp sequence numbers for dtmf/tone to
wriggle into the sequence for voice.  For example if the audio stream would
have
  resulted in the following:

     Time:               0    30mS       60mS           90mS

RTP Seq number:          0                1                             2
3


And we detect a tone at around 70 mS which we want to insert into the
stream, we would give it RTP Seq number 3 (wriggled in), and give the 90 mS
voice packet RTP Seq number 4 (bumped up from 3), and then maybe give a
"repeat" of the DTMF/TONE packet  at around time 100 mS sequence number 5,
etc.?

If we do that, then we still don't know whether a missing RTP sequence
number applies to DTMF/Tone or to VOICE.
In my example above, if we miss seq. #3 then we lost a DTMF/Tone packet,
and if we miss seq. #4 then we lost a VOICE packet.

Or have I missed something?





From rem-conf Mon Jun 21 11:15:33 1999 
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I take it then that you mean for the rtp sequence numbers for dtmf/tone to
wriggle into the sequence for voice.  For example if the audio stream would
have
  resulted in the following:

     Time:               0    30mS       60mS           90mS

RTP Seq number:          0                1                             2
3


And we detect a tone at around 70 mS which we want to insert into the
stream, we would give it RTP Seq number 3 (wriggled in), and give the 90 mS
voice packet RTP Seq number 4 (bumped up from 3), and then maybe give a
"repeat" of the DTMF/TONE packet  at around time 100 mS sequence number 5,
etc.?

If we do that, then we still don't know whether a missing RTP sequence
number applies to DTMF/Tone or to VOICE.
In my example above, if we miss seq. #3 then we lost a DTMF/Tone packet,
and if we miss seq. #4 then we lost a VOICE packet.

Or have I missed something?
Perhaps the voice and dtmf/tone information are piggybacked together in the
same rtp packet.  Then it would work but this would contradict your answer
to my first question in the previous email.






From rem-conf Mon Jun 21 13:36:50 1999 
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Date: Mon, 21 Jun 1999 16:34:14 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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Folks,

I've submitted an updated FEC draft to the archives. Until it appears,
you can find a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-avt-fec-06.txt

The only changes were a cleanup of the Figures and a new mechanism for
SDP signaling. The mechanism in the last draft had compatibility issues.
This version uses the fmtp parameter to convey the FEC information, so
its backwards compatible, if a little ugly. I'd rather have working and
ugly than not working.

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



From rem-conf Mon Jun 21 14:53:49 1999 
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Message-ID: <61891BA043DED21180920090273F173802E73D@argentina.telogy.com>
From: Roy Spitzer <rspitzer@telogy.com>
To: 'Henning Schulzrinne' <hgs@cs.columbia.edu>, "'rem-conf@es.net'"
	 <rem-conf@es.net>
Cc: 'Scott Petrack' <Scott.Petrack@metatel.com>, 
	"'MEGACO@standards.nortelnetworks.com'"
	 <MEGACO@standards.nortelnetworks.com>, Ed Morgan <emorgan@telogy.com>, 
	Mihai Sirbu <msirbu@telogy.com>, Zoran Mladenovic
	 <zmladenovic@telogy.com>, Frank Fruth <ffruth@telogy.com>
Subject: RE: First cut at DTMF/tones draft
Date: Mon, 21 Jun 1999 17:51:42 -0400
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Henning,

This is a resend of an earlier message since I received no reply.

Regards,
Roy

Roy Spitzer				Telogy Networks, Inc.
Phone:	(301) 515-6531			20250 Century Blvd.
Fax:	(301) 515-7954			Germantown, MD 20874
Email:	rspitzer@telogy.com


		-----Original Message-----
		From:	Roy Spitzer 
		Sent:	Thursday, June 17, 1999 11:18 AM
		To:	'Henning Schulzrinne'; rem-conf@es.net
		Cc:	Scott Petrack; MEGACO@standards.nortelnetworks.com;
Ed Morgan; Mihai Sirbu; Zoran Mladenovic; Frank Fruth
		Subject:	RE: First cut at DTMF/tones draft

		Henning,

		I apologize for my late comments.

		I applaud the reuse of the current RTP definitions for DTMF.
The application of the same technique for supervisory signaling is
desirable.

1.	The problem I have is that VoIP and the use of RTP is fast becoming
an international set of procedures.  To be compatible with the differing
encoding of supervisory signaling, the prime emphasis should be towards
creating a standard set of supervisory signal (event) names as discussed in
section 5.  The translation of these names becomes a function of the Media
Gateway.  These name could translate to standard country encodings or custom
audio response messages.  This need not be known by the generator of the
signal (event).
2.	Item 1 is fine when there is a translation of the signal (event).
For the cases when there is no translation of the signal (event) the
associated tones may be used.   But this may cause confusion at the
destination end since that combination of tones may already be assigned to
another signal(event).  The other alternative is to drop those signals on
the floor.  Obviously, there if there is no translation table, all signals
would be dropped on the floor.  The other alternative is to use the local
signal translation if present, used the associated tones when not present,
and drop unknown signals on the floor only when local signal translation is
present.  Please comment on this.
3.	To support V.34 Modems and V.34 FAX phase reversal needs as either a
parameter to CED or as distinguishable event.
4.	I am confused about the meaning of the events 
*	V.21 Channel 1, "0"
*	V.21 Channel 1, "1"
*	V.21 Channel 2, "0"
*	V.21 Channel 2, "1"
5.	Is V.21 flag indication available?

		Regards,
		Roy

		Roy Spitzer				Telogy Networks,
Inc.
		Phone:	(301) 515-6531			20250 Century Blvd.
		Fax:	(301) 515-7954			Germantown, MD 20874
		Email:	rspitzer@telogy.com


				-----Original Message-----
				From:	Henning Schulzrinne
[mailto:hgs@cs.columbia.edu]
				Sent:	Wednesday, June 09, 1999 10:02 AM
				To:	rem-conf@es.net
				Cc:	Scott Petrack;
MEGACO@standards.nortelnetworks.com
				Subject:	First cut at DTMF/tones
draft

				I've put the PostScript and PDF versions of
the merged DTMF and tones
				draft at
http://www.cs.columbia.edu/~hgs/rtp/doc/dtmf/tones.pdf and
				tones.ps. Scott and I would appreciate your
comments. I will turn this
				into an (ASCII) I-D shortly, barring "you
must be crazy" comments.

				Thanks.



From rem-conf Mon Jun 21 17:13:50 1999 
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cmF0aGVyIGhhdmUgd29ya2luZyBhbmQNCj4gdWdseSB0aGFuIG5vdCB3b3JraW5nLg0KPiANCj4g
LUpvbmF0aGFuIFIuDQo+IC0tIA0KPiBKb25hdGhhbiBELiBSb3NlbmJlcmcgICAgICAgICAgICAg
ICAgICAgICAgIEx1Y2VudCBUZWNobm9sb2dpZXMNCj4gTWVtYmVyIG9mIFRlY2huaWNhbCBTdGFm
ZiAgICAgICAgICAgICAgICAgICAxMDEgQ3Jhd2ZvcmRzIENvcm5lciBSZC4NCj4gSGlnaCBTcGVl
ZCBOZXR3b3JrcyBSZXNlYXJjaCAgICAgICAgICAgICAgICBIb2xtZGVsLCBOSiAwNzczMw0KPiBG
QVg6ICg3MzIpIDgzNC01Mzc5ICAgICAgICAgICAgICAgICAgICAgICAgIFJtLiA0Qy01MjYNCj4g
RU1BSUw6IGpkcm9zZW5AYmVsbC1sYWJzLmNvbQ0KPiBVUkw6IGh0dHA6Ly93d3cuY3MuY29sdW1i
aWEuZWR1L35qZHJvc2VuDQo+IA0KPiANCg==




From rem-conf Mon Jun 21 20:48:06 1999 
From rem-conf-request@es.net Mon Jun 21 20:48:05 1999
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From: Forrest Blair <fblair@s-vision.com>
To: "S.Grant" <grant@psc.edu>, rem-conf@es.net
Subject: RE: What can be done?
Date: Mon, 21 Jun 1999 21:45:43 -0600
MIME-Version: 1.0
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Every four weeks or so have the list request a reply from each subscriber to
stay on the list.  I suspect most spammers ignore reply mail - especially
those marked with REMOVE in the subject line.  This would probably filter
out a fair share of the spam.

Here is a possible transaction:  The list sends me a message with REMOVE
>FROM LIST in the subject line.  In the body is a note indicating that a
subscription timeout has occurred and I should send a reply with a subject
line of "continue".

Maybe this will spark other ideas.

Forrest Blair
Sorenson Vision, Inc.

	-----Original Message-----
	From:	S.Grant [SMTP:grant@psc.edu]
	Sent:	Monday, June 21, 1999 9:53 AM
	To:	rem-conf@es.net
	Subject:	What can be done?

	What can be done about the volume of spam sent to this list?  Can
the
	list posts be restricted to only those subscribed?  At this point,
I'm
	ready to unsub due to the volume of spam.

	Does anyone else here have any ideas?

	Thanks for listening.

	    Shirl Grant                       NLANR Engineering Services
	   grant@ncne.org                NCNE / Pittsburgh Supercomputing
Center

	-


	



From rem-conf Mon Jun 21 20:48:06 1999 
From rem-conf-request@es.net Mon Jun 21 20:48:05 1999
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Message-ID: <376F0645.208F9244@dnrc.bell-labs.com>
Date: Mon, 21 Jun 1999 23:43:01 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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MIME-Version: 1.0
To: Yung Yi <yiyung@mmlab.snu.ac.kr>
CC: rem-conf@es.net, mrtp@mmlab.snu.ac.kr
Subject: Re: Updated FEC draft
References: <376EA1C6.823AB577@dnrc.bell-labs.com> <001701bebc43$19045e60$260e2e93@apan.snu.ac.kr>
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Right. Thanks.

-Jonathan R.

Yung Yi wrote:
> 
> Hi Jonathan.
> In page 13,
> 
> In Section [10. Example],
> the third line "Packet x uses payload type 11, and packet x uses payload type
>                                                                                       ^^
>  18." should be corrected to "Packet x uses payload type 11, and packet y uses
>                                                                                                               ^^
>  payload type 18.", I think.
> 
> Am I right?
> 
> yiyung@mmlab.snu.ac.kr
> 
> ----- Original Message -----
> From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
> To: <rem-conf@es.net>
> Sent: Tuesday, June 22, 1999 5:34 AM
> Subject: Updated FEC draft
> 
> > Folks,
> >
> > I've submitted an updated FEC draft to the archives. Until it appears,
> > you can find a copy at:
> >
> > http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-avt-fec-06.txt
> >
> > The only changes were a cleanup of the Figures and a new mechanism for
> > SDP signaling. The mechanism in the last draft had compatibility issues.
> > This version uses the fmtp parameter to convey the FEC information, so
> > its backwards compatible, if a little ugly. I'd rather have working and
> > ugly than not working.
> >
> > -Jonathan R.
> > --
> > Jonathan D. Rosenberg                       Lucent Technologies
> > Member of Technical Staff                   101 Crawfords Corner Rd.
> > High Speed Networks Research                Holmdel, NJ 07733
> > FAX: (732) 834-5379                         Rm. 4C-526
> > EMAIL: jdrosen@bell-labs.com
> > URL: http://www.cs.columbia.edu/~jdrosen
> >
> >

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



From rem-conf Mon Jun 21 21:21:30 1999 
From rem-conf-request@es.net Mon Jun 21 21:21:30 1999
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Date: Mon, 21 Jun 1999 21:20:15 -0700
To: Forrest Blair <fblair@s-vision.com>, "S.Grant" <grant@psc.edu>,
        rem-conf@es.net
From: Ari Ollikainen <Ari@OLTECO.com>
Subject: RE: What can be done?
Cc: rem-conf-request@es.net
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At 9:45 PM 6/21/99, Forrest Blair wrote:
>Every four weeks or so have the list request a reply from each subscriber to
>stay on the list.  I suspect most spammers ignore reply mail - especially
>those marked with REMOVE in the subject line.  This would probably filter
>out a fair share of the spam.
>
>Here is a possible transaction:  The list sends me a message with REMOVE
>FROM LIST in the subject line.  In the body is a note indicating that a
>subscription timeout has occurred and I should send a reply with a subject
>line of "continue".
>
>Maybe this will spark other ideas.
>

        Uhhhh...as the list former and former maintainer (:^)) of the
        rem-conf list, I suggest that you direct such suggestions to BOTH

                rem-conf-request@es.net

                        AND

                postmaster@es.net

        perhaps that will either get some useful movement in filtering
        and/or list re-organization...OR get ESnet to look for a new
        listowner/maintainer...

        As to Henning's suggestion of prefixing message subject lines
        with "RTP", I humbly suggest that RTP isn't the ONLY topic of
        discussion on the rem-conf list. [rem-conf] might be a useful
        prefix, however...

        My apologies for adding to the noise.



      OLTECO                    Ari Ollikainen
      P.O. BOX 3688             Networking Architecture and Technology
      Stanford, CA              Ari@OLTECO.com
      94309-3688                415.517.3519                          





From rem-conf Mon Jun 21 23:39:31 1999 
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From: jhall@UU.NET (Jeremy Hall)
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Subject: Re: What can be done?
In-Reply-To: <18A6A966D2B7D211AEDE0090273CE1BA28CB19@el-postino.s-vision.com> from Forrest Blair at "Jun 21, 1999  9:45:43 pm"
To: fblair@s-vision.com (Forrest Blair)
Date: Tue, 22 Jun 1999 02:04:04 -0400 (EDT)
Cc: grant@psc.edu, rem-conf@es.net
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yah right.

Forrest Blair said:
> Every four weeks or so have the list request a reply from each subscriber to
> stay on the list.  I suspect most spammers ignore reply mail - especially
> those marked with REMOVE in the subject line.  This would probably filter
> out a fair share of the spam.
> 
> Here is a possible transaction:  The list sends me a message with REMOVE
> FROM LIST in the subject line.  In the body is a note indicating that a
> subscription timeout has occurred and I should send a reply with a subject
> line of "continue".
> 
> Maybe this will spark other ideas.
> 
> Forrest Blair
> Sorenson Vision, Inc.
> 
> 	-----Original Message-----
> 	From:	S.Grant [SMTP:grant@psc.edu]
> 	Sent:	Monday, June 21, 1999 9:53 AM
> 	To:	rem-conf@es.net
> 	Subject:	What can be done?
> 
> 	What can be done about the volume of spam sent to this list?  Can
> the
> 	list posts be restricted to only those subscribed?  At this point,
> I'm
> 	ready to unsub due to the volume of spam.
> 
> 	Does anyone else here have any ideas?
> 
> 	Thanks for listening.
> 
> 	    Shirl Grant                       NLANR Engineering Services
> 	   grant@ncne.org                NCNE / Pittsburgh Supercomputing
> Center
> 
> 	-
> 
> 
> 	
> 




From rem-conf Mon Jun 21 23:45:28 1999 
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Date: Tue, 22 Jun 1999 08:49:27 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Roy Spitzer <rspitzer@telogy.com>
CC: "'rem-conf@es.net'" <rem-conf@es.net>,
 	'Scott Petrack' <Scott.Petrack@metatel.com>,
 	"'MEGACO@standards.nortelnetworks.com'" 
 <MEGACO@standards.nortelnetworks.com>,
 	Ed Morgan <emorgan@telogy.com>, Mihai Sirbu <msirbu@telogy.com>,
 	Zoran Mladenovic <zmladenovic@telogy.com>,
 	Frank Fruth <ffruth@telogy.com>
Subject: Re: First cut at DTMF/tones draft
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> 1.      The problem I have is that VoIP and the use of RTP is fast becoming
> an international set of procedures.  To be compatible with the differing
> encoding of supervisory signaling, the prime emphasis should be towards
> creating a standard set of supervisory signal (event) names as discussed in
> section 5.  The translation of these names becomes a function of the Media
> Gateway.  These name could translate to standard country encodings or custom
> audio response messages.  This need not be known by the generator of the
> signal (event).

Not sure whether this paragraph is agreeing or disagreeing with the
current approach...

> 2.      Item 1 is fine when there is a translation of the signal (event).
> For the cases when there is no translation of the signal (event) the
> associated tones may be used.   But this may cause confusion at the
> destination end since that combination of tones may already be assigned to
> another signal(event).  The other alternative is to drop those signals on
> the floor.  Obviously, there if there is no translation table, all signals
> would be dropped on the floor.  The other alternative is to use the local
> signal translation if present, used the associated tones when not present,
> and drop unknown signals on the floor only when local signal translation is
> present.  Please comment on this.

As far as I know (and have been able to tell from the ITU collection of
national tones), while the same signal ("busy", "congestion", etc.) has
different tone representations in different countries, I don't think
that the same tone sequence means different things in different
countries. Even if there was this ambiguity, RTP would be no worse than
the current situation in the circuit-switched phone world.

The choice between using tones and signal names depends a bit on who's
listening (tourist at payphone calling home vs. subscriber calling
foreign country). Why shouldn't this be left to the implementor,
depending on what they see is being most useful to their customer base?


> 3.      To support V.34 Modems and V.34 FAX phase reversal needs as either a
> parameter to CED or as distinguishable event.

I'm afraid I can't parse this sentence. I thought we had discussed on
this list the various permutations of ANS: just tone, amplitude
modulated and with phase reversal. Is there another variation?

> 4.      I am confused about the meaning of the events
> *       V.21 Channel 1, "0"
> *       V.21 Channel 1, "1"
> *       V.21 Channel 2, "0"
> *       V.21 Channel 2, "1"
> 5.      Is V.21 flag indication available?

V.21 is a 300 b/s modem protocol, "0" means a zero bit, "1" a phone bit.
The channels are the ones from and to the originator, respectively. I
don't know what you mean by "flag indication".

Henning



From rem-conf Tue Jun 22 10:50:47 1999 
From rem-conf-request@es.net Tue Jun 22 10:50:46 1999
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--NextPart

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

	Title		: An RTP Payload Format for Generic Forward Error 
                          Correction
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-fec-06.txt
	Pages		: 22
	Date		: 21-Jun-99
	
This document specifies a payload format for generic forward error
correction of media encapsulated in RTP. It is engineered for FEC
algorithms based on the exclusive-or (parity) operation. The payload
format allows end systems to transmit using arbitrary block lengths
and parity schemes. It also allows for the recovery of both the pay-
load and critical RTP header fields. Since FEC is sent as a separate
stream, it is backwards compatible with non-FEC capable hosts, so
that receivers which do not wish to implement FEC can just ignore the
extensions.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-fec-06.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-fec-06.txt".

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


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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In the body type:
	"FILE /internet-drafts/draft-ietf-avt-fec-06.txt".
	
NOTE:	The mail server at ietf.org can return the document in
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Content-Type: text/plain
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--NextPart--





From rem-conf Tue Jun 22 12:26:16 1999 
From rem-conf-request@es.net Tue Jun 22 12:26:16 1999
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From: Deepak Srinivasan <Deepak.Srinivasan@ShareWave.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Cc: Greg Parks <Greg.Parks@sharewave.com>, Herman Chao
	 <herman.chao@sharewave.com>
Subject: RTP - Payload Differentiation
Date: Tue, 22 Jun 1999 12:16:03 -0700
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We are seeking more information on RTP Payload Type classification and
valid Payload Types. 

Please advise us on where we can get more information on this issue.

Thanks

Deepak



From rem-conf Tue Jun 22 14:52:48 1999 
From rem-conf-request@es.net Tue Jun 22 14:52:47 1999
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Message-ID: <61891BA043DED21180920090273F173802E741@argentina.telogy.com>
From: Roy Spitzer <rspitzer@telogy.com>
To: 'Henning Schulzrinne' <hgs@cs.columbia.edu>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>, 'Scott Petrack'
	 <Scott.Petrack@metatel.com>, "'MEGACO@standards.nortelnetworks.com'"
	 <MEGACO@standards.nortelnetworks.com>, Ed Morgan <emorgan@telogy.com>, 
	Mihai Sirbu <msirbu@telogy.com>, Zoran Mladenovic
	 <zmladenovic@telogy.com>, Frank Fruth <ffruth@telogy.com>
Subject: RE: First cut at DTMF/tones draft
Date: Tue, 22 Jun 1999 17:49:05 -0400
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Henning,

Comments below.

Regards,
Roy

Roy Spitzer				Telogy Networks, Inc.
Phone:	(301) 515-6531			20250 Century Blvd.
Fax:	(301) 515-7954			Germantown, MD 20874
Email:	rspitzer@telogy.com


		-----Original Message-----
		From:	Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
		Sent:	Tuesday, June 22, 1999 8:49 AM
		To:	Roy Spitzer
		Cc:	'rem-conf@es.net'; 'Scott Petrack';
'MEGACO@standards.nortelnetworks.com'; Ed Morgan; Mihai Sirbu; Zoran
Mladenovic; Frank Fruth
		Subject:	Re: First cut at DTMF/tones draft

		> 1.      The problem I have is that VoIP and the use of RTP
is fast becoming
		> an international set of procedures.  To be compatible with
the differing
		> encoding of supervisory signaling, the prime emphasis
should be towards
		> creating a standard set of supervisory signal (event)
names as discussed in
		> section 5.  The translation of these names becomes a
function of the Media
		> Gateway.  These name could translate to standard country
encodings or custom
		> audio response messages.  This need not be known by the
generator of the
		> signal (event).

		Not sure whether this paragraph is agreeing or disagreeing
with the
		current approach...

		Roy>	I am agreeing with the approach but I want the
document to explicitly state that both the event name and the associated
tone map are mandatory in each tone event message.

		> 2.      Item 1 is fine when there is a translation of the
signal (event).
		> For the cases when there is no translation of the signal
(event) the
		> associated tones may be used.   But this may cause
confusion at the
		> destination end since that combination of tones may
already be assigned to
		> another signal(event).  The other alternative is to drop
those signals on
		> the floor.  Obviously, there if there is no translation
table, all signals
		> would be dropped on the floor.  The other alternative is
to use the local
		> signal translation if present, used the associated tones
when not present,
		> and drop unknown signals on the floor only when local
signal translation is
		> present.  Please comment on this.

		As far as I know (and have been able to tell from the ITU
collection of
		national tones), while the same signal ("busy",
"congestion", etc.) has
		different tone representations in different countries, I
don't think
		that the same tone sequence means different things in
different
		countries. Even if there was this ambiguity, RTP would be no
worse than
		the current situation in the circuit-switched phone world.

		The choice between using tones and signal names depends a
bit on who's
		listening (tourist at payphone calling home vs. subscriber
calling
		foreign country). Why shouldn't this be left to the
implementor,
		depending on what they see is being most useful to their
customer base?

		Roy>	I have a two part response to question.
1.	In order that this feature to be available, the local MG must
somehow determine whether tones should or should not go through the
translation table.  For fixed subscriber to phone definition, this can
easily be done with a configuration of the local MG.  For a pay phone
environment, there must be some MG/MGC dialogues to determine which option
should be chosen.  The associated events have not yet been defined in
MEGACO.
2.	Vendor cannot choose to implement this feature unless the interface
is either standardized or there is a bilateral agreement between the vendor
of the MG and the vendor or the MGC.


		> 3.      To support V.34 Modems and V.34 FAX phase reversal
needs as either a
		> parameter to CED or as distinguishable event.

		I'm afraid I can't parse this sentence. I thought we had
discussed on
		this list the various permutations of ANS: just tone,
amplitude
		modulated and with phase reversal. Is there another
variation?

		Roy>	Non-V.34 FAX is tone, amplitude modulated, without
phase reversal
			V.34 FAX is tone, amplitude modulated, with phase
reversal
		I thought the intention was to support both in the future.

		> 4.      I am confused about the meaning of the events
		> *       V.21 Channel 1, "0"
		> *       V.21 Channel 1, "1"
		> *       V.21 Channel 2, "0"
		> *       V.21 Channel 2, "1"
		> 5.      Is V.21 flag indication available?

		V.21 is a 300 b/s modem protocol, "0" means a zero bit, "1"
a phone bit.
		The channels are the ones from and to the originator,
respectively.

		Roy>	FAX is only transmitted only on V.21 channel 2.  It
is a half duplex protocol.  I do not understand the use of the events
described in item 4.  Please give an example of their use.

		 I
		don't know what you mean by "flag indication".

		Roy>	V.21 flags are a sequence of "7E" bytes.  This is
similar the HDLC flag detection.

		Henning



From rem-conf Tue Jun 22 21:29:25 1999 
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Date: Tue, 22 Jun 1999 22:24:22 -0600 (MDT)
From: Evi Nemeth <evi@anchor.cs.colorado.edu>
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the plenary sessions from the INET99 conference in san jose, california
this week will be broadcast on the mbone.  there are 3 plenary sessions,
wednesday and thursday 8:30-10:30 and friday 10:30-12:30, june 23-25.
all times are pacific daylight time.  the detailed contents of each
day are listed in the white board.  session is advertised thru sdr, 
but just in case:
	audio	224.2.196.221/30158
	video	224.2.132.98/65332
	wb	224.2.135.162/37788

session for wednesday includes the dedication of inet'99 and the jon
postel award, by vint cerf, chairman of the board of the internet society.
the keynote address on wednesday is by irving wladawsky-berger, ibm.

-evi

evi nemeth, assoc prof
computer science dept
university of colorado
boulder, co 80309-0430



From rem-conf Wed Jun 23 05:47:54 1999 
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Message-ID: <03f801bebd75$83866980$a63db680@psc.edu>
From: "S.Grant" <grant@psc.edu>
To: <rem-conf@es.net>
Subject: About all the spam
Date: Wed, 23 Jun 1999 08:40:02 -0400
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I received this reply:

> We are aware of the spam problem and will be looking at
> ways of addressing it in the next few months.
> 


So I will be unsubscribing until some (distant) future date.

    Shirl Grant                       NLANR Engineering Services
   grant@ncne.org               NCNE / Pittsburgh Supercomputing Center

-






From rem-conf Wed Jun 23 10:10:28 1999 
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Date: Wed, 23 Jun 1999 10:10:55 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: "S.Grant" <grant@psc.edu>
cc: rem-conf@es.net
Subject: Re: About all the spam
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On Wed, 23 Jun 1999, S.Grant wrote:

> I received this reply:
> 
> > We are aware of the spam problem and will be looking at
> > ways of addressing it in the next few months.
> > 
> 
> 
> So I will be unsubscribing until some (distant) future date.

Sorry to hear that.  I have reached the conclusion that to control
spam one must install a filter at the receiving endpoint because much
of the spam is individually addressed.  Then the same filters can be
applied to the messages sent to mailing lists as well.  Over time I
have been building up a set of procmail recipies that do a pretty good
job of separating out the spam.  Some still leaks through.

Granted, there are some systematic steps that the list operator can
take, and I presume that is what the reply above is about.  However,
some of the spam filtering has to be a judgement call, and I prefer to
have those messages go to another mailbox where I can scan the headers
when I have time to see if any mail I care about was incorrectly
diverted.  The list operator can't handle that function without human
intervention (or more AI than I trust).
							-- Steve




From rem-conf Wed Jun 23 13:28:45 1999 
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Subject: any pointer to MGCP docs?
To: rem-conf@es.net
Date: Wed, 23 Jun 1999 16:21:11 -0400 (EDT)
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Could someone kindly give me a pointer
to the MGCP (Media Gateway Control Protocol)
documents?

Thanks

Hassan Naser
Univ. of Toronto



From rem-conf Thu Jun 24 09:09:29 1999 
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Date: Thu, 24 Jun 1999 12:05:07 -0400 (EDT)
From: Scott Petrack <scott.petrack@metatel.com>
To: Roy Spitzer <rspitzer@telogy.com>
cc: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'rem-conf@es.net'" <rem-conf@es.net>,
        "'MEGACO@standards.nortelnetworks.com'" <MEGACO@standards.nortelnetworks.com>,
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        Zoran Mladenovic <zmladenovic@telogy.com>,
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Subject: RE: First cut at DTMF/tones draft
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I'd like to stress that the draft simply gives a set of RTP payload
formats for transporting certain kinds of audio and signals. There are
many applications which need this facility, not just MEGACO. If MEGACO
decides that its application requires that redundancy be done a certain
way (for example that both the event payload type and the tones payload
type be used always), well, that is MEGACO's prerogative, but this 
shoud not affect the AVT draft. 

In the "vanilla RTP" application, two payloads have been given for
transmitting a certain kind of information in real-time, and they are
related in the sense that it is possible to use one as a redundant payload
for the other. The information in the two payloads are not identical, but
this is no different from the way in which the information in hifi encoded
audio is not really the same as the information in the "same" audio
encoded via some low-bitrate codec.  

If some particular application (like Megaco) wants to use these RTP
payload types in a particular way, great. But this doesn't need to
constrain other applications.

Scott



On Tue, 22 Jun 1999, Roy Spitzer wrote:

> Henning,
> 
> Comments below.
> 
> Regards,
> Roy
> 
> Roy Spitzer				Telogy Networks, Inc.
> Phone:	(301) 515-6531			20250 Century Blvd.
> Fax:	(301) 515-7954			Germantown, MD 20874
> Email:	rspitzer@telogy.com
> 
> 
> 		-----Original Message-----
> 		From:	Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> 		Sent:	Tuesday, June 22, 1999 8:49 AM
> 		To:	Roy Spitzer
> 		Cc:	'rem-conf@es.net'; 'Scott Petrack';
> 'MEGACO@standards.nortelnetworks.com'; Ed Morgan; Mihai Sirbu; Zoran
> Mladenovic; Frank Fruth
> 		Subject:	Re: First cut at DTMF/tones draft
> 
> 		> 1.      The problem I have is that VoIP and the use of RTP
> is fast becoming
> 		> an international set of procedures.  To be compatible with
> the differing
> 		> encoding of supervisory signaling, the prime emphasis
> should be towards
> 		> creating a standard set of supervisory signal (event)
> names as discussed in
> 		> section 5.  The translation of these names becomes a
> function of the Media
> 		> Gateway.  These name could translate to standard country
> encodings or custom
> 		> audio response messages.  This need not be known by the
> generator of the
> 		> signal (event).
> 
> 		Not sure whether this paragraph is agreeing or disagreeing
> with the
> 		current approach...
> 
> 		Roy>	I am agreeing with the approach but I want the
> document to explicitly state that both the event name and the associated
> tone map are mandatory in each tone event message.
> 
> 		> 2.      Item 1 is fine when there is a translation of the
> signal (event).
> 		> For the cases when there is no translation of the signal
> (event) the
> 		> associated tones may be used.   But this may cause
> confusion at the
> 		> destination end since that combination of tones may
> already be assigned to
> 		> another signal(event).  The other alternative is to drop
> those signals on
> 		> the floor.  Obviously, there if there is no translation
> table, all signals
> 		> would be dropped on the floor.  The other alternative is
> to use the local
> 		> signal translation if present, used the associated tones
> when not present,
> 		> and drop unknown signals on the floor only when local
> signal translation is
> 		> present.  Please comment on this.
> 
> 		As far as I know (and have been able to tell from the ITU
> collection of
> 		national tones), while the same signal ("busy",
> "congestion", etc.) has
> 		different tone representations in different countries, I
> don't think
> 		that the same tone sequence means different things in
> different
> 		countries. Even if there was this ambiguity, RTP would be no
> worse than
> 		the current situation in the circuit-switched phone world.
> 
> 		The choice between using tones and signal names depends a
> bit on who's
> 		listening (tourist at payphone calling home vs. subscriber
> calling
> 		foreign country). Why shouldn't this be left to the
> implementor,
> 		depending on what they see is being most useful to their
> customer base?
> 
> 		Roy>	I have a two part response to question.
> 1.	In order that this feature to be available, the local MG must
> somehow determine whether tones should or should not go through the
> translation table.  For fixed subscriber to phone definition, this can
> easily be done with a configuration of the local MG.  For a pay phone
> environment, there must be some MG/MGC dialogues to determine which option
> should be chosen.  The associated events have not yet been defined in
> MEGACO.
> 2.	Vendor cannot choose to implement this feature unless the interface
> is either standardized or there is a bilateral agreement between the vendor
> of the MG and the vendor or the MGC.
> 
> 
> 		> 3.      To support V.34 Modems and V.34 FAX phase reversal
> needs as either a
> 		> parameter to CED or as distinguishable event.
> 
> 		I'm afraid I can't parse this sentence. I thought we had
> discussed on
> 		this list the various permutations of ANS: just tone,
> amplitude
> 		modulated and with phase reversal. Is there another
> variation?
> 
> 		Roy>	Non-V.34 FAX is tone, amplitude modulated, without
> phase reversal
> 			V.34 FAX is tone, amplitude modulated, with phase
> reversal
> 		I thought the intention was to support both in the future.
> 
> 		> 4.      I am confused about the meaning of the events
> 		> *       V.21 Channel 1, "0"
> 		> *       V.21 Channel 1, "1"
> 		> *       V.21 Channel 2, "0"
> 		> *       V.21 Channel 2, "1"
> 		> 5.      Is V.21 flag indication available?
> 
> 		V.21 is a 300 b/s modem protocol, "0" means a zero bit, "1"
> a phone bit.
> 		The channels are the ones from and to the originator,
> respectively.
> 
> 		Roy>	FAX is only transmitted only on V.21 channel 2.  It
> is a half duplex protocol.  I do not understand the use of the events
> described in item 4.  Please give an example of their use.
> 
> 		 I
> 		don't know what you mean by "flag indication".
> 
> 		Roy>	V.21 flags are a sequence of "7E" bytes.  This is
> similar the HDLC flag detection.
> 
> 		Henning
> 





From rem-conf Thu Jun 24 20:14:27 1999 
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From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Folks,

Some colleagues and I have put together an Internet draft that talks
about the problem of feedback for multicast. Its basically a taxonomy of
the various mechanisms that can be applied for sending feedback on
multicast reception. We believe this taxonomy is useful for both rmt
(reliable multicast transport) and avt. For rmt, it outlines the options
for a feedback building block, and for avt, options for an alternate
profile for RTCP (which was discussed at the last meeting). 

The draft is an individual submission, and will appear in the archives
shortly. Until then, you can pick up a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-hnrs-rmt-avt-feedback-00.txt

Comments welcome.

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



From rem-conf Thu Jun 24 23:27:50 1999 
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Folks,

I've just submitted an updated draft on RTCP conformance testing. This
draft includes several more tests:

* collision detection
* random and uniform distribution of SSRC
* reducing the minimum interval for senders in high bandwidth sessions
* timing out participants

The first three were agreed upon at the last meeting. I decided to add
the fourth since I think its useful as well.

The draft should appear in the archives shortly. Until then, you can
pick up a copy from:

http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-avt-rtcptest-01.txt

Comments welcome as always.

Thanks,
Jonathan R.

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



From rem-conf Fri Jun 25 07:25:53 1999 
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Date: Fri, 25 Jun 1999 10:11:05 -0400
To: rem-conf@es.net
From: Bruce Davie <bdavie@cisco.com>
Subject: Draft of possible interest
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I have revised the draft on Integrated Services in the Presence of
Compressible Flows that I presented at the last IETF meeting and submitted
it to the IETF. For the impatient, there is a copy at 

ftp://ftpeng.cisco.com/bdavie/draft-ietf-intserv-compress-00.txt 

This has some relevance to RTP header compression. We hope to move it ahead
on the standards track after the Oslo meeting. Please send any comments to
me and the int-serv list (int-serv@isi.edu).

Bruce Davie



From rem-conf Fri Jun 25 07:50:58 1999 
From rem-conf-request@es.net Fri Jun 25 07:50:58 1999
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To: rem-conf@es.net
Subject: AVT agenda
Date: Fri, 25 Jun 1999 15:40:11 +0100
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Hi,

A reminder that those wishing to present in the AVT sessions in Oslo should
contact me and Steve as soon as possible. We've currently had requests from
the following:

	RTP Payload for AAC audio (Civanlar)
	RTP Payload for real-time pointers (Civanlar)
	RTP Payload for MPEG-4 with Scaleable & Flexible Error Resiliency (Wesner)
	Header compression for RTP/UDP/IP over cellular links (Degermark)
	Multiplexing header-compressed RTP over PPP (Pazhyannur)    
	Tunneled Compressed RTP (Koren)
	RTP payload for virtual worlds (Stewart)

Cheers,
Colin



From rem-conf Fri Jun 25 08:07:53 1999 
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From: <telegen0@hotmail.com>
To: rem-conf@es.net
Subject: E-Commerce Conference - Wembley 1999
Date: Fri, 25 Jun 1999 15:55:33
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--====================5453515:55====
Content-Type: text/plain; charset="us-ascii"

e-commerce the future - how ?

Find attached a brochure and application to join the forthcoming e-commerce exhibition being held at Wembley in July 1999.  Don't get left behind come and find out, how and why e-commerce can help your business create more business.

If this message has reached you by mistake then please accept our apologies - you will not receive another email as this is a one send campaign.

Thank you for your time, please enjoy the attached information.
--====================5453515:55====
Content-Type: application/octet-stream; name="telezip.ZIP"
Content-Transfer-Encoding: base64
Content-Disposition: attachment; filename="telezip.ZIP"

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WIa4x8Hh/r6Djz/V3xe2+ukXp/AqOLU2top3CIe73cvTQS5vUjd1UDub66A8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--====================5453515:55====--



From rem-conf Fri Jun 25 08:54:29 1999 
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What's the DTMF-over-RTP status?

--
C


Colin Perkins wrote:

> Hi,
>
> A reminder that those wishing to present in the AVT sessions in Oslo should
> contact me and Steve as soon as possible. We've currently had requests from
> the following:
>
>         RTP Payload for AAC audio (Civanlar)
>         RTP Payload for real-time pointers (Civanlar)
>         RTP Payload for MPEG-4 with Scaleable & Flexible Error Resiliency (Wesner)
>         Header compression for RTP/UDP/IP over cellular links (Degermark)
>         Multiplexing header-compressed RTP over PPP (Pazhyannur)
>         Tunneled Compressed RTP (Koren)
>         RTP payload for virtual worlds (Stewart)
>
> Cheers,
> Colin




From rem-conf Fri Jun 25 09:26:21 1999 
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From: "Stefan Wesner" <wesner@rus.uni-stuttgart.de>
To: "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>, <rem-conf@es.net>
Cc: "Jerome. Vieron" <Jerome.Vieron@irisa.fr>,
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        "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
Subject: Updated draft on RTP Payload Format for MPEG-4 with Scaleable and Flexible Error Resiliency
Date: Fri, 25 Jun 1999 18:21:22 +0200
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I've just submitted an updated draft on
RTP Payload Format for MPEG-4 with Scaleable and Flexible Error Resiliency
draft-guillemot-genrtp-01.txt

The document contains beside a new general section on MPEG-4 an improved
section on how to handle forward error correction within the proposed
format.

Additionally a section on how to handle SL-PDUs rather than plain MPEG-4 AUs
and some modifications in the payload format in order to reduce the overhead
has been added.

Comments welcome !

Kind regards,
Stefan Wesner.

The draft should appear in the archives shortly. Until then, you can
pick up a copy from:
http://www-ks.rus.uni-stuttgart.de/PKB/Publications/draft-guillemot-genrtp-0
1.txt






From rem-conf Fri Jun 25 09:50:00 1999 
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To: rem-conf@es.net
Subject: RTP interoperability statement and testing strategy
Date: Fri, 25 Jun 1999 17:48:18 +0100
Message-ID: <5870.930329298@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Folks,

I've just submitted two new internet drafts: an RTP interoperability
statement and a testing strategy.

The interoperability statement is a first attempt to enumerate the things
we need to demonstrate in order to move RTP to draft standard. The testing
strategy draft provides a set of tests which one might use whilst performing 
that demonstration, and is the counterpart to Jonathan's RTCP testing draft.

Until they appear in the archives you can retrieve these drafts from
http://www.cs.ucl.ac.uk/staff/c.perkins/internet-drafts/draft-ietf-avt-rtp-interop-00.txt
http://www.cs.ucl.ac.uk/staff/c.perkins/internet-drafts/draft-ietf-avt-rtptest-00.txt

Cheers,
Colin



From rem-conf Fri Jun 25 12:53:55 1999 
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To: <rem-conf@es.net>
Cc: "Glenn L. Cash" <glenn@research.att.com>
Subject: draft on RTP payload format for pointer
Date: Fri, 25 Jun 1999 15:50:07 -0400
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I have just posted a draft describing an RTP payload format
for
transmitting real-time pointers. If you want to take a look
at it
before it becomes available on the regular web site, it is
available
at:

http://www.research.att.com/~mrc/draft-civanlar-pointer.txt
or
http://www.research.att.com/~mrc/draft-civanlar-pointer.ps

As always, comments are welcome.

Best regards.

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





From rem-conf Fri Jun 25 13:34:59 1999 
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Date: Fri, 25 Jun 1999 16:30:35 -0400
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I've just submitted a first draft on
RTP Payload Format for MPEG-2 AAC Streams
draft-ietf-avt-rtp-mpeg2aac-01.txt

Comments are welcome!

Regards,

Mathias Kretschmer



From rem-conf Fri Jun 25 13:38:42 1999 
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From: "Jerry Scharf" <scharf@vayu.net>
To: "M. Reha Civanlar" <civanlar@research.att.com>,
	<rem-conf@es.net>
Cc: "Glenn L. Cash" <glenn@research.att.com>
Subject: RE: draft on RTP payload format for pointer
Date: Fri, 25 Jun 1999 13:38:00 -0700
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I have several comments about this draft.

First, there are pointing devices with more than two buttons, even if
Windows does not believe it. The mouse indicator should be able to indicate
multiple buttons and button chords.

Second, the assumption that the transmitted media that the pointer is to
combined with is integer pixel based is not necessarily a good one. In
particular, were you to have 3D vector system such as GL or PHIGS, there is
no direct relation from viewbox on the sender to pixels on the receiver. It
is nominal in such systems to represent pointer position as a fraction from
0 to 1 in both x and y directions. Since this will work for presentations as
well, I would recommend positive fixed point numbers between 0 and 1, with
the same right and down semantics of the pixel model. The receiver than
takes the received numbers and multiplies it by the number of pixels of
height and width the display has been scaled to, and the desired mapping is
maintained within one pixel. I would think that 12 bits per direction would
be sufficient. This would also allow a few more bits for representing mouse
buttons.

Also, if mouse button presses are important to send, then you have to make
some attempt to protect against packet drops. Perhaps the logic in the DTMF
draft would be overkill and perhaps not, but including the mouse
functionality without any attempt to protect it seems bad.

jerry




From rem-conf Fri Jun 25 13:44:43 1999 
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Date: Fri, 25 Jun 1999 13:43:04 -0700 (PDT)
From: Dan Wing <dwing@cisco.com>
To: rem-conf@es.net
cc: Stephen Casner <casner@cisco.com>, c.perkins@cs.ucl.ac.uk
Subject: I-D: Tunneled multiplexed Compressed RTP ("TCRTP")
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I just submitted "Tunneled multiplexed Compressed RTP ("TCRTP")",
draft-wing-avt-tcrtp-00.txt, to the Internet-Drafts editor.

  Abstract

     This document describes a mechanism which improves the end-to-end
     bandwidth utilization of RTP streams over an IP network by
     compressing the UDP and RTP headers and allowing the packets of
     multiple RTP streams to be multiplexed into one IP packet.

     This new protocol is useful in low-bandwidth environments, such as
     dialup modems, ISDN, cellular, and G.Lite which wish to reduce the
     encapsulation overhead of normal RTP packets through the entire
     network.  The protocol provides mechanisms to ensure synchronization
     at certain points to reduce the requirement for the receiver to
     communicate synchronization loss to the sender.

As the I-D publishing process is usually quite backlogged near the
cutoff date, I have also made it available on our FTP server if you
would like to review it before it is published as an Internet Draft:

  ftp://ftp-eng.cisco.com/dwing/tcrtp.txt


We have requested a time slot to present this document to the Audio/Video
Transport WG in Oslo.


Comments on the document are appreciated.

Thank you,
-Dan Wing
 dwing@cisco.com
 +1 408 525 5314





From rem-conf Sat Jun 26 16:11:42 1999 
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Date: Sat, 26 Jun 1999 19:04:09 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Roy Spitzer <rspitzer@telogy.com>
CC: "'rem-conf@es.net'" <rem-conf@es.net>,
        "'Scott Petrack'" <Scott.Petrack@metatel.com>,
        "'MEGACO@standards.nortelnetworks.com'" 
 <MEGACO@standards.nortelnetworks.com>,
        Ed Morgan <emorgan@telogy.com>, Mihai Sirbu <msirbu@telogy.com>,
        Zoran Mladenovic <zmladenovic@telogy.com>,
        Frank Fruth <ffruth@telogy.com>
Subject: Re: First cut at DTMF/tones draft
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Roy Spitzer wrote:
> 

> 
>                 Roy>    I am agreeing with the approach but I want the
> document to explicitly state that both the event name and the associated
> tone map are mandatory in each tone event message.

I don't see why the specification should mandate this. It makes sense in
many applications, but maybe not all.


> 
>                 Roy>    I have a two part response to question.
> 1.      In order that this feature to be available, the local MG must
> somehow determine whether tones should or should not go through the
> translation table.  For fixed subscriber to phone definition, this can
> easily be done with a configuration of the local MG.  For a pay phone
> environment, there must be some MG/MGC dialogues to determine which option
> should be chosen.  The associated events have not yet been defined in
> MEGACO.

This is outside the scope of an RTP/AVT draft, which is ignorant of
Megaco.

> 2.      Vendor cannot choose to implement this feature unless the interface
> is either standardized or there is a bilateral agreement between the vendor
> of the MG and the vendor or the MGC.

Again, a Megaco issue.


> 
>                 Roy>    Non-V.34 FAX is tone, amplitude modulated, without
> phase reversal
>                         V.34 FAX is tone, amplitude modulated, with phase
> reversal
>                 I thought the intention was to support both in the future.

I had forgotten to list the events in the table, even though they are
described in the preceding text. There's now /ANS, /ANSam, ANS and
ANSam, which should cover all bases.


>                 Roy>    FAX is only transmitted only on V.21 channel 2.  It
> is a half duplex protocol.  I do not understand the use of the events
> described in item 4.  Please give an example of their use.

Other modem/negotiation standards seem to use V.21 in both directions,
so there seems to be no harm in including it.

> 
>                  I
>                 don't know what you mean by "flag indication".
> 
>                 Roy>    V.21 flags are a sequence of "7E" bytes.  This is
> similar the HDLC flag detection.

Do we really want to include all possible modem bit patterns here? Just
asking...

> 
>                 Henning

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



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        osimcast@bbn.com, performance@haven.epm.ornl.gov, prs@masi.ibp.fr,
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        s.low@ee.mu.oz.au, sb.all@ieee.org, sc6wg4@ntd.comsat.com,
        sig-dce@opengroup.org
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Prelimary Program of Broadband Access Conference (BAC'99) is available on web page:

		http://bac99.kt.agh.edu.pl


You may register now!



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

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

The conference programme is organised in five streamed sessions:

	* Digital Subscriber Line

	* Wireless Local Loop

	* Services for Home & Office

	* Architectures, Traffic & Cost Modelling
	
	* IP Migration towards the User



From rem-conf Sun Jun 27 23:16:15 1999 
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Date: Mon, 28 Jun 1999 16:05:29 +1000
To: Jan-Ming Ho <hoho@iis.sinica.edu.tw>, rem-conf@es.net, jj@startap.net
From: Markus Buchhorn <markus@acsys.anu.edu.au>
Subject: Re: request for Mbone feed
Cc: Sheng-Hsiung Huang <huangk@gate.sinica.edu.tw>, chinfu@iis.sinica.edu.tw,
        Chia-Hui Wang <chwang@iis.sinica.edu.tw>,
        Lih-Hsiang Chen <chens@mail.ncku.edu.tw>, feng@iis.sinica.edu.tw
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At 09:19 21/06/99 +0800, Jan-Ming Ho wrote:
>
>We are representing Taiwan to seek for Mbone feed to our Academic Network,
>Taiwan Academic Network, or TANET for short.  We had long been suffering
>a low bandwidth connecting to international internet traffic.  But, we
>recently upgrade our connection to the US by a T3 lease line.  We 
>appreciate it very much if we could have any institute here to provide
>us with permanent MBONE feed.  We will run a permanent mbone root for
>the academic family including universities in Taiwan.   We are also 
>willing to transit this MBONE traffic to nearby countries based on the
>availability of physical connections and bandwidth.

Since Taiwan/TANet now has a direct connection to the STARTAP in Chicago
I'd suggest getting an MBONE feed from there - contact John Jamison
(jj@startap.net), who may or may not be the right person, (sorry John! :-))
but should be able to point you to the right person. APAN and SingaREN also
get their MBONE feeds off the vBNS through STARTAP. 

Cheers,
	Markus

>Jan-Ming Ho, PhD
>Associate Editor, IEEE Transaction on Multimedia
>Research Fellow			(O) 886-2-2788-3799 x1803
>IIS, Academis Sinica		(F) 886-2-2788-3799 x1606,  886-2-2782-4814
>Nankang, Taipei			email:  hoho@iis.sinica.edu.tw
>Taiwan				URL:	http://www.iis.sinica.edu.tw/~hoho
>					http://www.iis.sinica.edu.tw/CCL
>					http://twstudy.sinica.edu.tw
>
>===========================================================================
===
>
>
>
>
>
Markus Buchhorn,  Advanced Computational Systems CRC     | Ph: +61 2 62798810
email: markus@acsys.anu.edu.au, snail: ACSys, RSISE Bldg,|Fax: +61 2 62798602
Australian National University, Canberra 0200, Australia |Mobile: 0417 281429  



From rem-conf Mon Jun 28 01:44:14 1999 
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Date: Mon, 28 Jun 1999 16:39:33 +0800
From: Kenny Huang <huangk@sinica.edu>
Organization: Academia Sinica
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To: Markus Buchhorn <markus@acsys.anu.edu.au>
CC: Jan-Ming Ho <hoho@iis.sinica.edu.tw>, rem-conf@es.net, jj@startap.net,
        Sheng-Hsiung Huang <huangk@gate.sinica.edu.tw>,
        chinfu@iis.sinica.edu.tw, Chia-Hui Wang <chwang@iis.sinica.edu.tw>,
        Lih-Hsiang Chen <chens@mail.ncku.edu.tw>, feng@iis.sinica.edu.tw,
        jj@startap.net, "ªL®Ú½å" <a00khl00@nchc.gov.tw>
Subject: Re: request for Mbone feed
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Dear ALL and JJ,

I met JJ at the STARTAP Advisory Committee and briefed the mbone
issue to him already. Ken Lin at NCHC (a00khl00@nchc.gov.tw) will
be the primary contact in Taiwan. Please contact Ken Lin for=20
mbone implementation, and keep me in the communication loop.

Regards

--=20
     Kenny Huang =B6=C0=B3=D3=B6=AF                      Tel 886-2-2789-9=
492
     Head, Communication & Network Services  Fax 886-2-2783-6444
     Academia Sinica                         huangk@sinica.edu.tw
     http://www.sinica.edu.tw



Markus Buchhorn wrote:
>=20
> At 09:19 21/06/99 +0800, Jan-Ming Ho wrote:
> >
> >We are representing Taiwan to seek for Mbone feed to our Academic Netw=
ork,
> >Taiwan Academic Network, or TANET for short.  We had long been sufferi=
ng
> >a low bandwidth connecting to international internet traffic.  But, we
> >recently upgrade our connection to the US by a T3 lease line.  We
> >appreciate it very much if we could have any institute here to provide
> >us with permanent MBONE feed.  We will run a permanent mbone root for
> >the academic family including universities in Taiwan.   We are also
> >willing to transit this MBONE traffic to nearby countries based on the
> >availability of physical connections and bandwidth.
>=20
> Since Taiwan/TANet now has a direct connection to the STARTAP in Chicag=
o
> I'd suggest getting an MBONE feed from there - contact John Jamison
> (jj@startap.net), who may or may not be the right person, (sorry John! =
:-))
> but should be able to point you to the right person. APAN and SingaREN =
also
> get their MBONE feeds off the vBNS through STARTAP.
>=20
> Cheers,
>         Markus
>=20
> >Jan-Ming Ho, PhD
> >Associate Editor, IEEE Transaction on Multimedia
> >Research Fellow                        (O) 886-2-2788-3799 x1803
> >IIS, Academis Sinica           (F) 886-2-2788-3799 x1606,  886-2-2782-=
4814
> >Nankang, Taipei                        email:  hoho@iis.sinica.edu.tw
> >Taiwan                         URL:    http://www.iis.sinica.edu.tw/~h=
oho
> >                                       http://www.iis.sinica.edu.tw/CC=
L
> >                                       http://twstudy.sinica.edu.tw
> >
> >=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D
> =3D=3D=3D
> >
> >
> >
> >
> >
> Markus Buchhorn,  Advanced Computational Systems CRC     | Ph: +61 2 62=
798810
> email: markus@acsys.anu.edu.au, snail: ACSys, RSISE Bldg,|Fax: +61 2 62=
798602
> Australian National University, Canberra 0200, Australia |Mobile: 0417 =
281429



From rem-conf Mon Jun 28 05:43:31 1999 
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Date: Mon, 28 Jun 1999 08:33:37 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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The second SIP bake-off will take place at pulver.com (Melville, NY, on
Long Island near NYC) on August 5th and 6th. Details can be found at
http://www.pulver.com/sip/ or through the SIP page at
http://www.cs.columbia.edu/sip.

The emphasis of this second bake-off is on more advanced SIP
functionality, for example, proxying, security and CANCEL.

We will also likely target basic RTP interoperability for testing at the
event.

Details on available hardware and software will be made available later,
but you might want to send a note to Jeff Pulver if you anticipate
needing more than just Ethernet jacks.

Thanks to Jeff Pulver for hosting the event.

Henning



From rem-conf Mon Jun 28 06:19:13 1999 
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Date: Mon, 28 Jun 1999 09:17:51 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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Currently, the new profile draft has the comfort noise RTP payload type
at 19. However, the VoIP Forum Interoperability agreement put it at 13,
since the profile had listed it as unassigned. Previously, VSC (a
proprietary Vocaltec codec) was at 13. Thus, the question:

- Is anybody who *has* to use static payload types still using either 13
for VSC or 19 for CN?

If there are no objections, we would move CN to 13, since a more widely
published document has had it there for a while.

This might also be a good opportunity to gather evidence as to which
static payload types and, in general, RTP payload formats people
actually use. This will be a necessary step as we move towards Draft
Standard status for RTP and the profile. Thus, please send me this
information. I will then include it on the RTP implementations page
(unless it is secret for some reason).

This provides another illustration why static payload types are a bad
idea. End of soapbox.

Thanks.

Henning



From rem-conf Tue Jun 29 04:35:29 1999 
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--NextPart

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

	Title		: RTP Payload Format for DV Format Video
	Author(s)	: K. Kobayashi, A. Ogawa, S. Casner, C. Borman
	Filename	: draft-ietf-avt-dv-video-00.txt
	Pages		: 13
	Date		: 28-Jun-99
	
This document specifies the packetization scheme for encapsulating
the digital video data streams defined by the HD Digital VCR
Conference, commonly known as 'DV', into a payload format for the
Real-Time Transport Protocol (RTP).  The RTP payload format specified
in this document supports three quality levels of digital video
identified as SD-VCR, HD-VCR and SDL-VCR.

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--NextPart

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

	Title		: Conformance Tests for RTP Scalability Algorithms
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-rtcptest-01.txt
	Pages		: 14
	Date		: 28-Jun-99
	
This document defines a set of tests that can be run on an RTP imple-
mentation to determine the level of conformance with the various sca-
lability algorithms defined in the draft version of RTP. These tests
do not require access to internal implementations, and use black box
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--NextPart

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

	Title		: RTP Interoperability Statement
	Author(s)	: C. Perkins
	Filename	: draft-ietf-avt-rtp-interop-00.txt
	Pages		: 5
	Date		: 28-Jun-99
	
It is required to demonstrate interoperability of RTP implementations
in order to move the RTP specification to draft standard.  This memo
outlines those features to be tested, as the first stage of an
interoperability statement.

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DIRECCIONES INFORMATIZADAS PARA MARKETING
FICHEROS INTERNACIONALES ACTUALIZADOS 98/99

SOLICITE PRUEBA GRATUITA

936365223- fax. 936646051
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From rem-conf Wed Jun 30 04:40:48 1999 
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Testing Strategies
	Author(s)	: C. Perkins
	Filename	: draft-ietf-avt-rtptest-00.txt
	Pages		: 9
	Date		: 29-Jun-99
	
This memo describes a possible testing strategy for RTP
implementations and is intended as an aid to implementors.
It does not specify a standard of any kind.

A URL for this Internet-Draft is:
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Subject: 33rd Annual Simulation Symposium
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* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *
*                                                                         *   
*                            Call for Papers                              *
*                                                                         *   
*                    33rd Annual Simulation Symposium                     *
*                         Sheraton City Hotel                             *
*                             Washington D.C.                             *
*                                                                         *
*                           April 16-20, 2000                             *
*                                                                         *   
*                              Sponsored by:                              * 
*                                                                         *
*               The Society for Computer Simulation (SCS)                 *
*                                                                         *
*                        In cooperation with:                             *
*                                                                         *
*                     The IEEE Computer Society                           *
*                                                                         *
*               ACM (Association for Computing Machinery)                 *
*                                                                         *
*                                                                         *
* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *

       The Annual Simulation Symposium is a forum for the exchange of ideas,  
    techniques, and applications among practitioners of simulation in industry,
    government, and  academia. This international Symposium is the oldest 
    continuously operating conference/symposium dedicated to simulation.  
       The paper sessions are designed to promote discussion of concepts, 
    methodology,  and  results between authors and the audience. The structure 
    of the Symposium also provides for a degree of collegiality and continuity 
    in the discussions of the various topics presented during the week.

       In recent years, the Symposium has provided a forum for traditional  
    simulation  topics  in discrete-event, continuous, digital, and analog 
    simulation.  Along with the traditional  subjects,  this year's Symposium 
    is seeking papers in the following topical areas:

		o Simulation of Multiprocessor Architectures        
		o VLSI Circuit Simulators
		o Simulation Languages, Tools, and Environments     
		o Object-Oriented Simulation
		o Simulation of Distributed Systems and Databases   
                o Simulation of Parallel Processing Systems
		o Simulation based Performance Analysis
                o Animations/Virtual Reality
		o Advances in Simulation Methodology and Practices  
		o Network Modeling and Simulation
		o Simulation of Multimedia Applications and Systems 
  		o Parallel and Distributed Simulation
		o Artificial Intelligence in Simulation    
		o Neural Network Models and Simulation
		o Web-based Modeling and Simulation


    General Chair                         Program Committee Chair
    Prof. Taieb F. Znati                  Prof. Helen Karatza 
    Computer Science Department           Department of Informatics
    University of Pittsburgh              Aristotle University of Thessaloniki
    Pittsburgh, PA 15260                  54006 Thessaloniki, Greece
    Tel. : (412) 624-8417                 Tel. : (31) 997974 
    Fax. : (412) 624-8854                 Fax. : (31) 996360
    Email: znati@cs.pitt.edu              Email: karatza@csd.auth.gr


  PAPER SUBMISSION:

   Please send five (5) copies of a complete paper by September 24, 1999 to:

   Professor Helen Karatza                Important Dates
   Department of Informatics              Paper Submission Deadline :  9/24/99
   Aristotle University of Thessaloniki   Acceptance Notification   : 11/15/99
   54006 Thessaloniki, Greece             Camera-Ready Paper        :  1/8/2000

    Papers can also be submitted electronically to karatza@csd.auth.gr 
    or via anonymous ftp to ftp.auth.gr/incoming/ssymp33 by October 1st, 1999. 
    Submitted files must be either in dvi or PostScript format  with  all  
    figures  and  tables included, as created on Unix systems by:

          cat paper.ps | compress | uuencode paper.ps.Z > paper.uue

    or in most other environments by the `print to file'  command  with
    the PostScript output format option.

       All submissions will be fully refereed for accuracy, technical
    content,  and relevance.  Papers should be no longer than 20 
    double-spaced pages. Please  include  full  name,  affiliation,  
    address, phone number, and email address of each author and designate 
    the author who is to be contacted. Authors will receive the reviewers' 
    comments and will be given a chance to respond to these reviews. Authors' 
    replies will be taken into consideration in the final paper selection  
    decision.  Accepted  papers  will  appear  in  the Symposium's  
    Proceedings  which  will be published by the IEEE Computer Society.

       The Ira M. Kay Memorial Prize of $500 will be awarded  to  the
    best  paper presented at the Symposium.  Authors of top papers will
    also be encouraged to submit a follow-on paper for publication
    in the SCS Journals.

       Only papers  which  have  not  been  previously  published  or
    presented  should  be  submitted.   Authors  must  obtain employer,
    client, or government releases prior  to  submitting  the  final
    manuscript.


    PROGRAM COMMITTEE

Taieb Znati, General Chair, University of Pittsburg (znati@cs.pitt.edu)
Helen Karatza, Program Chair, Aristotle University of Thessaloniki (karatza@csd.auth.gr)
Dimetri Avresky, Boston University (avresky@bu.edu)
Simonetta Balsamo, University of Udine (balsamo@dimi.uniud.it, balsamo@di.unipi.it)
Sujata Banerjee, University of Pittsburgh (sujata@icarus.lis.pitt.edu)
Felix Breitenecker, Technical University of Vienna (fbreiten@email.tuwien.ac.at)
Wai Chen, Telcordia Technologies (wchen@research.telcordia.com)
Patrick W. Dowd, University of Maryland at College Park (dowd@eng.umd.edu)
Alois Ferscha, Institut Fuer Angewandte Informatik (ferscha@ani.univie.ac.at)
Paul A. Fishwick, University of Florida (fishwick@cise.ufl.edu)
Richard Fujimoto, Georgia Institute Of Technology (fujimoto@cc.gatech.edu)
Michael Green, Univ. of California at Irvine (green@ece.uci.edu)
Mohsen Guizani, University of West Florida (guizanim@umkc.edu)
Omar Hammami, The University of Aizu, Japan (hammami@u-aizu.ac.jp)
David Kaeli, Northeastern University (kaeli@ece.neu.edu)
Enrqiue V. Kortright, Nicholls State University (ek@reality.nich.edu)
Karen Panetta Lentz, Tufts University (karen@eecs.tufts.edu)
Jason Yi-Bing Lin, Natl. Chiao Tung University, Taiwan, ROC (liny@csie.nctu.edu.tw)
Wayne M. Loucks, University of Waterloo (Wayne.Loucks@uwaterloo.ca)
Robert Macredie, Brunel University, UK (Robert.Macredie@brunel.ac.uk)
John A. Miller, University of Georgia (jam@pollux.cs.uga.edu)
Philippe Mussi, Inria Sophia Antipolis (mussi@sophia.inria.fr)
Tuncer Oren, University of Ottawa and Tubitak-Marmara Research Center, Informatics Institute (oren@csi.uottawa.ca, tuncer@mam.gov.tr)
Vernon Rego, Purdue University (rego@cs.purdue.edu)
William H. Sanders, University of Illinois (whs@crhc.uiuc.edu)
Robert Simon, George Mason University (simon@cs.gmu.edu)
Gary S. H. Tan, National University of Singapore (gtan@comp.nus.edu.sg)
TEO Yong Meng, National University of Singapore (teoym@comp.nus.edu.sg)
Stephen J. Turner, University of Exeter (S.J.Turner@exeter.ac.uk)
Philip A. Wilsey, University of Cincinnati (philip.wilsey@uc.edu)
Shambhu J. Upadhyaya, State University of New York at Buffalo (shambhu@eng.buffalo.edu)
Bernard Zeigler, University of Arizona (zeigler@ece.arizona.edu)
George Zobrist, University of Misouri-Rolla (zobrist@umr.edu)


    FURTHER INFORMATION:

    For   additional   information, please contact or send mail to either Prof.
    Karatza (karatza@csd.auth.gr) or Prof. Taieb Znati (znati@cs.pitt.edu). 
    Updated conference announcement and information are accessible
    at the symposium home page at http://www.csd.auth.gr/ssymp33.


--------------A8D62C01DB93C8DC44881D3B--




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        adtf <adtf@omg.org>, mof <mof@omg.org>, tc <tc@omg.org>,
        ptc <ptc@omg.org>, dtc <dtc@omg.org>, ab <ab@omg.org>,
        mof-rtf <mof-rtf@omg.org>, odp <odp@dstc.edu.au>,
        dbworld <dbworld@cs.wisc.edu>, JavaCORBA <JavaCORBA@luke.org>,
        ISWORLD <ISWORLD@Danann.hea.ie>,
        comsoc-chapters <comsoc-chapters@IEEE.ORG>,
        comsoc-gicb <comsoc-gicb@IEEE.ORG>,
        "comsoc.tac" <comsoc.tac@tab.ieee.org>,
        ctc-members <ctc-members@redbank.tinac.com>,
        end2end-interest <end2end-interest@isi.edu>,
        f-troup <f-troup@codex.cis.upenn.edu>,
        fokus-user <fokus-user@fokus.gmd.de>, g-troup <g-troup@ccrc.wustl.edu>,
        giga <giga@tele.pitt.edu>, hipparch <hipparch@sophia.inria.fr>,
        ieeetcpc <ieeetcpc@ccvm.sunysb.edu>, itc <itc@fokus.gmd.de>,
        rem-conf <rem-conf@es.net>, "sb.all" <sb.all@IEEE.ORG>,
        tccc <tccc@IEEE.ORG>, xtp-relay <xtp-relay@cs.concordia.ca>,
        karatza@csd.auth.gr
Subject: 33rd Annual Simulation Symposium
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* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *
*                                                                         *   
*                            Call for Papers                              *
*                                                                         *   
*                    33rd Annual Simulation Symposium                     *
*                         Sheraton City Hotel                             *
*                             Washington D.C.                             *
*                                                                         *
*                           April 16-20, 2000                             *
*                                                                         *   
*                              Sponsored by:                              * 
*                                                                         *
*               The Society for Computer Simulation (SCS)                 *
*                                                                         *
*                        In cooperation with:                             *
*                                                                         *
*                     The IEEE Computer Society                           *
*                                                                         *
*               ACM (Association for Computing Machinery)                 *
*                                                                         *
*                                                                         *
* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *

       The Annual Simulation Symposium is a forum for the exchange of ideas,  
    techniques, and applications among practitioners of simulation in industry,
    government, and  academia. This international Symposium is the oldest 
    continuously operating conference/symposium dedicated to simulation.  
       The paper sessions are designed to promote discussion of concepts, 
    methodology,  and  results between authors and the audience. The structure 
    of the Symposium also provides for a degree of collegiality and continuity 
    in the discussions of the various topics presented during the week.

       In recent years, the Symposium has provided a forum for traditional  
    simulation  topics  in discrete-event, continuous, digital, and analog 
    simulation.  Along with the traditional  subjects,  this year's Symposium 
    is seeking papers in the following topical areas:

		o Simulation of Multiprocessor Architectures        
		o VLSI Circuit Simulators
		o Simulation Languages, Tools, and Environments     
		o Object-Oriented Simulation
		o Simulation of Distributed Systems and Databases   
                o Simulation of Parallel Processing Systems
		o Simulation based Performance Analysis
                o Animations/Virtual Reality
		o Advances in Simulation Methodology and Practices  
		o Network Modeling and Simulation
		o Simulation of Multimedia Applications and Systems 
  		o Parallel and Distributed Simulation
		o Artificial Intelligence in Simulation    
		o Neural Network Models and Simulation
		o Web-based Modeling and Simulation


    General Chair                         Program Committee Chair
    Prof. Taieb F. Znati                  Prof. Helen Karatza 
    Computer Science Department           Department of Informatics
    University of Pittsburgh              Aristotle University of Thessaloniki
    Pittsburgh, PA 15260                  54006 Thessaloniki, Greece
    Tel. : (412) 624-8417                 Tel. : (31) 997974 
    Fax. : (412) 624-8854                 Fax. : (31) 996360
    Email: znati@cs.pitt.edu              Email: karatza@csd.auth.gr


  PAPER SUBMISSION:

   Please send five (5) copies of a complete paper by September 24, 1999 to:

   Professor Helen Karatza                Important Dates
   Department of Informatics              Paper Submission Deadline :  9/24/99
   Aristotle University of Thessaloniki   Acceptance Notification   : 11/15/99
   54006 Thessaloniki, Greece             Camera-Ready Paper        :  1/8/2000

    Papers can also be submitted electronically to karatza@csd.auth.gr 
    or via anonymous ftp to ftp.auth.gr/incoming/ssymp33 by October 1st, 1999. 
    Submitted files must be either in dvi or PostScript format  with  all  
    figures  and  tables included, as created on Unix systems by:

          cat paper.ps | compress | uuencode paper.ps.Z > paper.uue

    or in most other environments by the `print to file'  command  with
    the PostScript output format option.

       All submissions will be fully refereed for accuracy, technical
    content,  and relevance.  Papers should be no longer than 20 
    double-spaced pages. Please  include  full  name,  affiliation,  
    address, phone number, and email address of each author and designate 
    the author who is to be contacted. Authors will receive the reviewers' 
    comments and will be given a chance to respond to these reviews. Authors' 
    replies will be taken into consideration in the final paper selection  
    decision.  Accepted  papers  will  appear  in  the Symposium's  
    Proceedings  which  will be published by the IEEE Computer Society.

       The Ira M. Kay Memorial Prize of $500 will be awarded  to  the
    best  paper presented at the Symposium.  Authors of top papers will
    also be encouraged to submit a follow-on paper for publication
    in the SCS Journals.

       Only papers  which  have  not  been  previously  published  or
    presented  should  be  submitted.   Authors  must  obtain employer,
    client, or government releases prior  to  submitting  the  final
    manuscript.


    PROGRAM COMMITTEE

Taieb Znati, General Chair, University of Pittsburg (znati@cs.pitt.edu)
Helen Karatza, Program Chair, Aristotle University of Thessaloniki (karatza@csd.auth.gr)
Dimetri Avresky, Boston University (avresky@bu.edu)
Simonetta Balsamo, University of Udine (balsamo@dimi.uniud.it, balsamo@di.unipi.it)
Sujata Banerjee, University of Pittsburgh (sujata@icarus.lis.pitt.edu)
Felix Breitenecker, Technical University of Vienna (fbreiten@email.tuwien.ac.at)
Wai Chen, Telcordia Technologies (wchen@research.telcordia.com)
Patrick W. Dowd, University of Maryland at College Park (dowd@eng.umd.edu)
Alois Ferscha, Institut Fuer Angewandte Informatik (ferscha@ani.univie.ac.at)
Paul A. Fishwick, University of Florida (fishwick@cise.ufl.edu)
Richard Fujimoto, Georgia Institute Of Technology (fujimoto@cc.gatech.edu)
Michael Green, Univ. of California at Irvine (green@ece.uci.edu)
Mohsen Guizani, University of West Florida (guizanim@umkc.edu)
Omar Hammami, The University of Aizu, Japan (hammami@u-aizu.ac.jp)
David Kaeli, Northeastern University (kaeli@ece.neu.edu)
Enrqiue V. Kortright, Nicholls State University (ek@reality.nich.edu)
Karen Panetta Lentz, Tufts University (karen@eecs.tufts.edu)
Jason Yi-Bing Lin, Natl. Chiao Tung University, Taiwan, ROC (liny@csie.nctu.edu.tw)
Wayne M. Loucks, University of Waterloo (Wayne.Loucks@uwaterloo.ca)
Robert Macredie, Brunel University, UK (Robert.Macredie@brunel.ac.uk)
John A. Miller, University of Georgia (jam@pollux.cs.uga.edu)
Philippe Mussi, Inria Sophia Antipolis (mussi@sophia.inria.fr)
Tuncer Oren, University of Ottawa and Tubitak-Marmara Research Center, Informatics Institute (oren@csi.uottawa.ca, tuncer@mam.gov.tr)
Vernon Rego, Purdue University (rego@cs.purdue.edu)
William H. Sanders, University of Illinois (whs@crhc.uiuc.edu)
Robert Simon, George Mason University (simon@cs.gmu.edu)
Gary S. H. Tan, National University of Singapore (gtan@comp.nus.edu.sg)
TEO Yong Meng, National University of Singapore (teoym@comp.nus.edu.sg)
Stephen J. Turner, University of Exeter (S.J.Turner@exeter.ac.uk)
Philip A. Wilsey, University of Cincinnati (philip.wilsey@uc.edu)
Shambhu J. Upadhyaya, State University of New York at Buffalo (shambhu@eng.buffalo.edu)
Bernard Zeigler, University of Arizona (zeigler@ece.arizona.edu)
George Zobrist, University of Misouri-Rolla (zobrist@umr.edu)


    FURTHER INFORMATION:

    For   additional   information, please contact or send mail to either Prof.
    Karatza (karatza@csd.auth.gr) or Prof. Taieb Znati (znati@cs.pitt.edu). 
    Updated conference announcement and information are accessible
    at the symposium home page at http://www.csd.auth.gr/ssymp33.


--------------4002EE02CFE5625AC5E6BBA7--




From rem-conf Wed Jun 30 06:35:42 1999 
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        sc6wg4 <sc6wg4@ntd.comsat.com>, comswtc <comswtc@gmu.edu>,
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        sigmedia <sigmedia@bellcore.com>, commsoft <commsoft@cc.bellcore.com>,
        modern-heuristics <modern-heuristics@uk.ac.mailbase.ucdavis.edu>,
        ckobryn <ckobryn@shl.com>, bomsig <bomsig@omg.org>,
        adtf <adtf@omg.org>, mof <mof@omg.org>, tc <tc@omg.org>,
        ptc <ptc@omg.org>, dtc <dtc@omg.org>, ab <ab@omg.org>,
        mof-rtf <mof-rtf@omg.org>, odp <odp@dstc.edu.au>,
        dbworld <dbworld@cs.wisc.edu>, JavaCORBA <JavaCORBA@luke.org>,
        ISWORLD <ISWORLD@Danann.hea.ie>,
        comsoc-chapters <comsoc-chapters@IEEE.ORG>,
        comsoc-gicb <comsoc-gicb@IEEE.ORG>,
        "comsoc.tac" <comsoc.tac@tab.ieee.org>,
        ctc-members <ctc-members@redbank.tinac.com>,
        end2end-interest <end2end-interest@isi.edu>,
        f-troup <f-troup@codex.cis.upenn.edu>,
        fokus-user <fokus-user@fokus.gmd.de>, g-troup <g-troup@ccrc.wustl.edu>,
        giga <giga@tele.pitt.edu>, hipparch <hipparch@sophia.inria.fr>,
        ieeetcpc <ieeetcpc@ccvm.sunysb.edu>, itc <itc@fokus.gmd.de>,
        rem-conf <rem-conf@es.net>, "sb.all" <sb.all@IEEE.ORG>,
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Subject: 33rd Annual Simulation Symposium
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* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *
*                                                                         *   
*                            Call for Papers                              *
*                                                                         *   
*                    33rd Annual Simulation Symposium                     *
*                         Sheraton City Hotel                             *
*                             Washington D.C.                             *
*                                                                         *
*                           April 16-20, 2000                             *
*                                                                         *   
*                              Sponsored by:                              * 
*                                                                         *
*               The Society for Computer Simulation (SCS)                 *
*                                                                         *
*                        In cooperation with:                             *
*                                                                         *
*                     The IEEE Computer Society                           *
*                                                                         *
*               ACM (Association for Computing Machinery)                 *
*                                                                         *
*                                                                         *
* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *

       The Annual Simulation Symposium is a forum for the exchange of ideas,  
    techniques, and applications among practitioners of simulation in industry,
    government, and  academia. This international Symposium is the oldest 
    continuously operating conference/symposium dedicated to simulation.  
       The paper sessions are designed to promote discussion of concepts, 
    methodology,  and  results between authors and the audience. The structure 
    of the Symposium also provides for a degree of collegiality and continuity 
    in the discussions of the various topics presented during the week.

       In recent years, the Symposium has provided a forum for traditional  
    simulation  topics  in discrete-event, continuous, digital, and analog 
    simulation.  Along with the traditional  subjects,  this year's Symposium 
    is seeking papers in the following topical areas:

		o Simulation of Multiprocessor Architectures        
		o VLSI Circuit Simulators
		o Simulation Languages, Tools, and Environments     
		o Object-Oriented Simulation
		o Simulation of Distributed Systems and Databases   
                o Simulation of Parallel Processing Systems
		o Simulation based Performance Analysis
                o Animations/Virtual Reality
		o Advances in Simulation Methodology and Practices  
		o Network Modeling and Simulation
		o Simulation of Multimedia Applications and Systems 
  		o Parallel and Distributed Simulation
		o Artificial Intelligence in Simulation    
		o Neural Network Models and Simulation
		o Web-based Modeling and Simulation


    General Chair                         Program Committee Chair
    Prof. Taieb F. Znati                  Prof. Helen Karatza 
    Computer Science Department           Department of Informatics
    University of Pittsburgh              Aristotle University of Thessaloniki
    Pittsburgh, PA 15260                  54006 Thessaloniki, Greece
    Tel. : (412) 624-8417                 Tel. : (31) 997974 
    Fax. : (412) 624-8854                 Fax. : (31) 996360
    Email: znati@cs.pitt.edu              Email: karatza@csd.auth.gr


  PAPER SUBMISSION:

   Please send five (5) copies of a complete paper by September 24, 1999 to:

   Professor Helen Karatza                Important Dates
   Department of Informatics              Paper Submission Deadline :  9/24/99
   Aristotle University of Thessaloniki   Acceptance Notification   : 11/15/99
   54006 Thessaloniki, Greece             Camera-Ready Paper        :  1/8/2000

    Papers can also be submitted electronically to karatza@csd.auth.gr 
    or via anonymous ftp to ftp.auth.gr/incoming/ssymp33 by October 1st, 1999. 
    Submitted files must be either in dvi or PostScript format  with  all  
    figures  and  tables included, as created on Unix systems by:

          cat paper.ps | compress | uuencode paper.ps.Z > paper.uue

    or in most other environments by the `print to file'  command  with
    the PostScript output format option.

       All submissions will be fully refereed for accuracy, technical
    content,  and relevance.  Papers should be no longer than 20 
    double-spaced pages. Please  include  full  name,  affiliation,  
    address, phone number, and email address of each author and designate 
    the author who is to be contacted. Authors will receive the reviewers' 
    comments and will be given a chance to respond to these reviews. Authors' 
    replies will be taken into consideration in the final paper selection  
    decision.  Accepted  papers  will  appear  in  the Symposium's  
    Proceedings  which  will be published by the IEEE Computer Society.

       The Ira M. Kay Memorial Prize of $500 will be awarded  to  the
    best  paper presented at the Symposium.  Authors of top papers will
    also be encouraged to submit a follow-on paper for publication
    in the SCS Journals.

       Only papers  which  have  not  been  previously  published  or
    presented  should  be  submitted.   Authors  must  obtain employer,
    client, or government releases prior  to  submitting  the  final
    manuscript.


    PROGRAM COMMITTEE

Taieb Znati, General Chair, University of Pittsburg (znati@cs.pitt.edu)
Helen Karatza, Program Chair, Aristotle University of Thessaloniki (karatza@csd.auth.gr)
Dimetri Avresky, Boston University (avresky@bu.edu)
Simonetta Balsamo, University of Udine (balsamo@dimi.uniud.it, balsamo@di.unipi.it)
Sujata Banerjee, University of Pittsburgh (sujata@icarus.lis.pitt.edu)
Felix Breitenecker, Technical University of Vienna (fbreiten@email.tuwien.ac.at)
Wai Chen, Telcordia Technologies (wchen@research.telcordia.com)
Patrick W. Dowd, University of Maryland at College Park (dowd@eng.umd.edu)
Alois Ferscha, Institut Fuer Angewandte Informatik (ferscha@ani.univie.ac.at)
Paul A. Fishwick, University of Florida (fishwick@cise.ufl.edu)
Richard Fujimoto, Georgia Institute Of Technology (fujimoto@cc.gatech.edu)
Michael Green, Univ. of California at Irvine (green@ece.uci.edu)
Mohsen Guizani, University of West Florida (guizanim@umkc.edu)
Omar Hammami, The University of Aizu, Japan (hammami@u-aizu.ac.jp)
David Kaeli, Northeastern University (kaeli@ece.neu.edu)
Enrqiue V. Kortright, Nicholls State University (ek@reality.nich.edu)
Karen Panetta Lentz, Tufts University (karen@eecs.tufts.edu)
Jason Yi-Bing Lin, Natl. Chiao Tung University, Taiwan, ROC (liny@csie.nctu.edu.tw)
Wayne M. Loucks, University of Waterloo (Wayne.Loucks@uwaterloo.ca)
Robert Macredie, Brunel University, UK (Robert.Macredie@brunel.ac.uk)
John A. Miller, University of Georgia (jam@pollux.cs.uga.edu)
Philippe Mussi, Inria Sophia Antipolis (mussi@sophia.inria.fr)
Tuncer Oren, University of Ottawa and Tubitak-Marmara Research Center, Informatics Institute (oren@csi.uottawa.ca, tuncer@mam.gov.tr)
Vernon Rego, Purdue University (rego@cs.purdue.edu)
William H. Sanders, University of Illinois (whs@crhc.uiuc.edu)
Robert Simon, George Mason University (simon@cs.gmu.edu)
Gary S. H. Tan, National University of Singapore (gtan@comp.nus.edu.sg)
TEO Yong Meng, National University of Singapore (teoym@comp.nus.edu.sg)
Stephen J. Turner, University of Exeter (S.J.Turner@exeter.ac.uk)
Philip A. Wilsey, University of Cincinnati (philip.wilsey@uc.edu)
Shambhu J. Upadhyaya, State University of New York at Buffalo (shambhu@eng.buffalo.edu)
Bernard Zeigler, University of Arizona (zeigler@ece.arizona.edu)
George Zobrist, University of Misouri-Rolla (zobrist@umr.edu)


    FURTHER INFORMATION:

    For   additional   information, please contact or send mail to either Prof.
    Karatza (karatza@csd.auth.gr) or Prof. Taieb Znati (znati@cs.pitt.edu). 
    Updated conference announcement and information are accessible
    at the symposium home page at http://www.csd.auth.gr/ssymp33.


--------------70CDAE2AF9F503EFE45FA229--




From rem-conf Wed Jun 30 10:17:55 1999 
From rem-conf-request@es.net Wed Jun 30 10:17:55 1999
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Date: Wed, 30 Jun 1999 19:11:57 +0200 (MET DST)
From: =?ISO-8859-1?Q?Tobias_=D6brink?= <nv91-tob@nada.kth.se>
To: rem-conf@es.net
Subject: Re: I-D ACTION:draft-ietf-avt-dv-video-00.txt
In-Reply-To: <199906291120.HAA08367@ietf.org>
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I read through draft-ietf-avt-dv-video-00.txt and have some comments
and questions that might be worthwhile discussing.

In section 4.1 RTP header usage:

   Timestamp: 32-bit 90 kHz timestamp representing the time at which
   the first data in the frame was sampled. =20

This statement may lead to the conclusion that the sampling time for
the first frame (maybe obtained from the CIP header as
described in appendix A) should be used as the initial value of the
timestamp field. To avoid unneccessary confusion I would like to add
the following paragraph from the new RTP draft:

   The initial value of the timestamp SHOULD be random, as for the
   sequence number.


In section 4.2 DV data encapsulation into RTP payload:

   Therefore, if VAUX/AAUX information is not
   transmitted in the stream, the equivalent parameters essential to
   playout MUST be provided by some out of band means beyond the
   scope of this document.
   :
   :<snip>
   :
   Therefore, if the RTP receiver
   is feeding the DV stream to a device that requires AAUX
   information and VAUX DIF blocks, the receiver MUST be able to
   generate AAUX within audio DIF blocks and VAUX DIF blocks for the
   device using the parameters provided out of band.

I interpret this as, to comply with the spec, the receiver
implementation MUST retrieve, by some unknown out-of-band-mechanism,
the information needed to recreate the VAUX and AAUX DIF blocks.=20
If that's the case won't we end up with non-interoperable
implementations using proprietary out-of-band-mechanisms?=20
Maybe we should at least state a default out-of-band-mechanism as a
minimal requirement that have to be used by all implementations. The
dynamic payload type assignment using SDP in section 5 comes to mind.


In section 5.2 SDP description for bundled stream:=20

The audio format parameter seems quite detailed. How to find out the=20
values for those parameters if you're about to announce a session in
sdr or writing a SDP-file for a program that haven't yet started?
Aren't there any standardized combinations of those parameters?=20
At least <quantization>, <sampling rate> and the number of channels=20
only can occur in a few different combinations.
Maybe the same applies to some of the other parameters as well?


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





